From b995d76da070440c1e5687c0c13c8fbe51222ea2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Oct 2007 10:41:06 +0200 Subject: [ALSA] hda-codec - Add missing eeepc-p701 model for ALC662 Added the missing model string 'eeepc-p701' for ALC662 codec. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4b48c2e..7b064f2 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -843,6 +843,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch-dig 3-stack (6-channel) with SPDIF 6stack-dig 6-stack with SPDIF lenovo-101e Lenovo laptop + eeepc-p701 ASUS Eeepc auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1c50278..b78f6fc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12057,6 +12057,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_3ST_6ch] = "3stack-6ch", [ALC662_5ST_DIG] = "6stack-dig", [ALC662_LENOVO_101E] = "lenovo-101e", + [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_AUTO] = "auto", }; -- cgit v0.10.2 From 7f16859a8335449c8bf75ce4edd8040a57e2b678 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 18 Oct 2007 17:38:17 +0200 Subject: [ALSA] hda-codec - Add STAC9228 DMIC support Added the missing STAC9228 DMIC support. Also added a new vendor id tag for IDT. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8cbe3bf..dacabe5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -55,6 +55,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x10ec, "Realtek" }, { 0x1057, "Motorola" }, { 0x1106, "VIA" }, + { 0x111d, "IDT" }, { 0x11d4, "Analog Devices" }, { 0x13f6, "C-Media" }, { 0x14f1, "Conexant" }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0401223..86cd3f6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -204,6 +204,11 @@ static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 }; +#define STAC927X_NUM_DMICS 2 +static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { + 0x13, 0x14, 0 +}; + static hda_nid_t stac9205_adc_nids[2] = { 0x12, 0x13 }; @@ -2688,7 +2693,6 @@ static int patch_stac927x(struct hda_codec *codec) spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; @@ -2697,7 +2701,6 @@ static int patch_stac927x(struct hda_codec *codec) spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; @@ -2706,11 +2709,20 @@ static int patch_stac927x(struct hda_codec *codec) spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = stac927x_core_init; spec->mixer = stac927x_mixer; } + switch (codec->subsystem_id) { + case 0x1028020A: /* STAC 9228 */ + case 0x10280209: /* STAC 9228 */ + spec->dmic_nids = stac927x_dmic_nids; + spec->num_dmics = STAC927X_NUM_DMICS; + break; + default: + spec->num_dmics = 0; + } + spec->multiout.dac_nids = spec->dac_nids; /* GPIO0 High = Enable EAPD */ spec->gpio_mask = spec->gpio_data = 0x00000001; -- cgit v0.10.2 From 47744f638a6ee6a9e47cf47cdc6f215d096cb4fc Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 19 Oct 2007 08:19:56 +0200 Subject: [ALSA] hda: Add dmux to STAC 9228 Added a dmux to the STAC9228 cards with DMIC support. And added a STAC_DIGITAL_INPUT_SOURCE macro for repeating mixer code. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 86cd3f6..820208d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -393,6 +393,16 @@ static struct hda_verb stac9205_core_init[] = { {} }; +#define STAC_DIGITAL_INPUT_SOURCE(cnt) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Digital Input Source", \ + .count = cnt, \ + .info = stac92xx_dmux_enum_info, \ + .get = stac92xx_dmux_enum_get, \ + .put = stac92xx_dmux_enum_put,\ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -433,14 +443,7 @@ static struct snd_kcontrol_new stac925x_mixer[] = { }; static struct snd_kcontrol_new stac9205_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Source", - .count = 1, - .info = stac92xx_dmux_enum_info, - .get = stac92xx_dmux_enum_get, - .put = stac92xx_dmux_enum_put, - }, + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), @@ -470,6 +473,7 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), @@ -2718,6 +2722,7 @@ static int patch_stac927x(struct hda_codec *codec) case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; + spec->dmux_nid = 0x1c; break; default: spec->num_dmics = 0; -- cgit v0.10.2 From 83eef75bfeee7bdaf034e30e8f3737b393264b1e Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 22 Oct 2007 12:27:10 +0200 Subject: [ALSA] hda: STAC9228 updated DMUX nid Changed the dmux for STAC9228 from ADC1MUX to ADC0MUX to avoid confusion. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 820208d..876e1d4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2722,7 +2722,7 @@ static int patch_stac927x(struct hda_codec *codec) case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1c; + spec->dmux_nid = 0x1b; break; default: spec->num_dmics = 0; -- cgit v0.10.2 From 61bae0934eddeb51b2559186dfd79a7a6ed3a90a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Oct 2007 17:05:35 +0200 Subject: [ALSA] hda-codec - Add model for Fujitsu V5505 Added model=laptop for Fujitsu V5505 with Cxt5405 codec. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6aa0739..beda297 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -765,6 +765,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), {} }; -- cgit v0.10.2 From 18fe4ac25811ff4006844b6e575acdc9103cbaba Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Mon, 22 Oct 2007 17:41:08 +0200 Subject: [ALSA] soc - Add 'Mono Playback Switch' to WM9712 codec driver The following patch adds 'Mono Playback Switch' control to WM9712 codec SoC driver. Also, it fixes Treble, Bass and Mono playback volume inversion bits. Signed-off-by: Mike Rapoport Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 986b5d59..427cb61 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -102,7 +102,8 @@ SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0), SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), @@ -145,8 +146,8 @@ SOC_ENUM("Bass Control", wm9712_enum[5]), SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), -SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0), -SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0), +SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1), +SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), -- cgit v0.10.2 From f31639b8c5916f58441b529c9c364715921b29af Mon Sep 17 00:00:00 2001 From: Shin-ya Okada Date: Tue, 23 Oct 2007 15:08:18 +0200 Subject: [ALSA] ice1724 - Add support of Onkyo SE-90PCI and SE-200PCI Added the support for Onkyo SE-90PCI and SE-200PCI boards. Signed-off-by: Shin-ya Okada Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7b064f2..69ab91c 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1157,11 +1157,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Chaintech 9CJS * Chaintech AV-710 * Shuttle SN25P + * Onkyo SE-90PCI + * Onkyo SE-200PCI model - Use the given board model, one of the following: revo51, revo71, amp2000, prodigy71, prodigy71lt, prodigy192, aureon51, aureon71, universe, ap192, - k8x800, phase22, phase28, ms300, av710 + k8x800, phase22, phase28, ms300, av710, se200pci, + se90pci This module supports multiple cards and autoprobe. diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile index 65ce66a..ee86a1d 100644 --- a/sound/pci/ice1712/Makefile +++ b/sound/pci/ice1712/Makefile @@ -5,7 +5,7 @@ snd-ice17xx-ak4xxx-objs := ak4xxx.o snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o -snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o juli.o phase.o wtm.o +snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o juli.o phase.o wtm.o se.o # Toplevel Module Dependency obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 58640af..3c3cac3 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -400,6 +400,12 @@ struct snd_ice1712 { struct { struct ak4114 *ak4114; } prodigy192; + struct { + struct { + unsigned char ch1, ch2; + } vol[8]; + } se; + } spec; }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 0b0bbb0..357bdbe 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -51,6 +51,7 @@ #include "juli.h" #include "phase.h" #include "wtm.h" +#include "se.h" MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)"); @@ -65,6 +66,7 @@ MODULE_SUPPORTED_DEVICE("{" JULI_DEVICE_DESC PHASE_DEVICE_DESC WTM_DEVICE_DESC + SE_DEVICE_DESC "{VIA,VT1720}," "{VIA,VT1724}," "{ICEnsemble,Generic ICE1724}," @@ -1933,6 +1935,7 @@ static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_juli_cards, snd_vt1724_phase_cards, snd_vt1724_wtm_cards, + snd_vt1724_se_cards, NULL, }; diff --git a/sound/pci/ice1712/se.c b/sound/pci/ice1712/se.c new file mode 100644 index 0000000..ebfe245 --- /dev/null +++ b/sound/pci/ice1712/se.c @@ -0,0 +1,700 @@ +/* + * ALSA driver for ICEnsemble VT1724 (Envy24HT) + * + * Lowlevel functions for ONKYO WAVIO SE-90PCI and SE-200PCI + * + * Copyright (c) 2007 Shin-ya Okada sh_okada(at)d4.dion.ne.jp + * (at) -> @ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ice1712.h" +#include "envy24ht.h" +#include "se.h" + + +/****************************************************************************/ +/* ONKYO WAVIO SE-200PCI */ +/****************************************************************************/ +/* + * system configuration ICE_EEP2_SYSCONF=0x4b + * XIN1 49.152MHz + * not have UART + * one stereo ADC and a S/PDIF receiver connected + * four stereo DACs connected + * + * AC-Link configuration ICE_EEP2_ACLINK=0x80 + * use I2C, not use AC97 + * + * I2S converters feature ICE_EEP2_I2S=0x78 + * I2S codec has no volume/mute control feature + * I2S codec supports 96KHz and 192KHz + * I2S codec 24bits + * + * S/PDIF configuration ICE_EEP2_SPDIF=0xc3 + * Enable integrated S/PDIF transmitter + * internal S/PDIF out implemented + * S/PDIF is stereo + * External S/PDIF out implemented + * + * + * ** connected chips ** + * + * WM8740 + * A 2ch-DAC of main outputs. + * It setuped as I2S mode by wire, so no way to setup from software. + * The sample-rate are automatically changed. + * ML/I2S (28pin) --------+ + * MC/DM1 (27pin) -- 5V | + * MD/DM0 (26pin) -- GND | + * MUTEB (25pin) -- NC | + * MODE (24pin) -- GND | + * CSBIW (23pin) --------+ + * | + * RSTB (22pin) --R(1K)-+ + * Probably it reduce the noise from the control line. + * + * WM8766 + * A 6ch-DAC for surrounds. + * It's control wire was connected to GPIOxx (3-wire serial interface) + * ML/I2S (11pin) -- GPIO18 + * MC/IWL (12pin) -- GPIO17 + * MD/DM (13pin) -- GPIO16 + * MUTE (14pin) -- GPIO01 + * + * WM8776 + * A 2ch-ADC(with 10ch-selector) plus 2ch-DAC. + * It's control wire was connected to SDA/SCLK (2-wire serial interface) + * MODE (16pin) -- R(1K) -- GND + * CE (17pin) -- R(1K) -- GND 2-wire mode (address=0x34) + * DI (18pin) -- SDA + * CL (19pin) -- SCLK + * + * + * ** output pins and device names ** + * + * 7.1ch name -- output connector color -- device (-D option) + * + * FRONT 2ch -- green -- plughw:0,0 + * CENTER(Lch) SUBWOOFER(Rch) -- black -- plughw:0,2,0 + * SURROUND 2ch -- orange -- plughw:0,2,1 + * SURROUND BACK 2ch -- white -- plughw:0,2,2 + * + */ + + +/****************************************************************************/ +/* WM8740 interface */ +/****************************************************************************/ + +static void __devinit se200pci_WM8740_init(struct snd_ice1712 *ice) +{ + /* nothing to do */ +} + + +static void se200pci_WM8740_set_pro_rate(struct snd_ice1712 *ice, + unsigned int rate) +{ + /* nothing to do */ +} + + +/****************************************************************************/ +/* WM8766 interface */ +/****************************************************************************/ + +static void se200pci_WM8766_write(struct snd_ice1712 *ice, + unsigned int addr, unsigned int data) +{ + unsigned int st; + unsigned int bits; + int i; + const unsigned int DATA = 0x010000; + const unsigned int CLOCK = 0x020000; + const unsigned int LOAD = 0x040000; + const unsigned int ALL_MASK = (DATA | CLOCK | LOAD); + + snd_ice1712_save_gpio_status(ice); + + st = ((addr & 0x7f) << 9) | (data & 0x1ff); + snd_ice1712_gpio_set_dir(ice, ice->gpio.direction | ALL_MASK); + snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask & ~ALL_MASK); + bits = snd_ice1712_gpio_read(ice) & ~ALL_MASK; + + snd_ice1712_gpio_write(ice, bits); + for (i = 0; i < 16; i++) { + udelay(1); + bits &= ~CLOCK; + st = (st << 1); + if (st & 0x10000) + bits |= DATA; + else + bits &= ~DATA; + + snd_ice1712_gpio_write(ice, bits); + + udelay(1); + bits |= CLOCK; + snd_ice1712_gpio_write(ice, bits); + } + + udelay(1); + bits |= LOAD; + snd_ice1712_gpio_write(ice, bits); + + udelay(1); + bits |= (DATA | CLOCK); + snd_ice1712_gpio_write(ice, bits); + + snd_ice1712_restore_gpio_status(ice); +} + +static void se200pci_WM8766_set_volume(struct snd_ice1712 *ice, int ch, + unsigned int vol1, unsigned int vol2) +{ + switch (ch) { + case 0: + se200pci_WM8766_write(ice, 0x000, vol1); + se200pci_WM8766_write(ice, 0x001, vol2 | 0x100); + break; + case 1: + se200pci_WM8766_write(ice, 0x004, vol1); + se200pci_WM8766_write(ice, 0x005, vol2 | 0x100); + break; + case 2: + se200pci_WM8766_write(ice, 0x006, vol1); + se200pci_WM8766_write(ice, 0x007, vol2 | 0x100); + break; + } +} + +static void __devinit se200pci_WM8766_init(struct snd_ice1712 *ice) +{ + se200pci_WM8766_write(ice, 0x1f, 0x000); /* RESET ALL */ + udelay(10); + + se200pci_WM8766_set_volume(ice, 0, 0, 0); /* volume L=0 R=0 */ + se200pci_WM8766_set_volume(ice, 1, 0, 0); /* volume L=0 R=0 */ + se200pci_WM8766_set_volume(ice, 2, 0, 0); /* volume L=0 R=0 */ + + se200pci_WM8766_write(ice, 0x03, 0x022); /* serial mode I2S-24bits */ + se200pci_WM8766_write(ice, 0x0a, 0x080); /* MCLK=256fs */ + se200pci_WM8766_write(ice, 0x12, 0x000); /* MDP=0 */ + se200pci_WM8766_write(ice, 0x15, 0x000); /* MDP=0 */ + se200pci_WM8766_write(ice, 0x09, 0x000); /* demp=off mute=off */ + + se200pci_WM8766_write(ice, 0x02, 0x124); /* ch-assign L=L R=R RESET */ + se200pci_WM8766_write(ice, 0x02, 0x120); /* ch-assign L=L R=R */ +} + +static void se200pci_WM8766_set_pro_rate(struct snd_ice1712 *ice, + unsigned int rate) +{ + if (rate > 96000) + se200pci_WM8766_write(ice, 0x0a, 0x000); /* MCLK=128fs */ + else + se200pci_WM8766_write(ice, 0x0a, 0x080); /* MCLK=256fs */ +} + + +/****************************************************************************/ +/* WM8776 interface */ +/****************************************************************************/ + +static void se200pci_WM8776_write(struct snd_ice1712 *ice, + unsigned int addr, unsigned int data) +{ + unsigned int val; + + val = (addr << 9) | data; + snd_vt1724_write_i2c(ice, 0x34, val >> 8, val & 0xff); +} + + +static void se200pci_WM8776_set_output_volume(struct snd_ice1712 *ice, + unsigned int vol1, unsigned int vol2) +{ + se200pci_WM8776_write(ice, 0x03, vol1); + se200pci_WM8776_write(ice, 0x04, vol2 | 0x100); +} + +static void se200pci_WM8776_set_input_volume(struct snd_ice1712 *ice, + unsigned int vol1, unsigned int vol2) +{ + se200pci_WM8776_write(ice, 0x0e, vol1); + se200pci_WM8776_write(ice, 0x0f, vol2 | 0x100); +} + +static const char *se200pci_sel[] = { + "LINE-IN", "CD-IN", "MIC-IN", "ALL-MIX", NULL +}; + +static void se200pci_WM8776_set_input_selector(struct snd_ice1712 *ice, + unsigned int sel) +{ + static unsigned char vals[] = { + /* LINE, CD, MIC, ALL, GND */ + 0x10, 0x04, 0x08, 0x1c, 0x03 + }; + if (sel > 4) + sel = 4; + se200pci_WM8776_write(ice, 0x15, vals[sel]); +} + +static void se200pci_WM8776_set_afl(struct snd_ice1712 *ice, unsigned int afl) +{ + /* AFL -- After Fader Listening */ + if (afl) + se200pci_WM8776_write(ice, 0x16, 0x005); + else + se200pci_WM8776_write(ice, 0x16, 0x001); +} + +static const char *se200pci_agc[] = { + "Off", "LimiterMode", "ALCMode", NULL +}; + +static void se200pci_WM8776_set_agc(struct snd_ice1712 *ice, unsigned int agc) +{ + /* AGC -- Auto Gain Control of the input */ + switch (agc) { + case 0: + se200pci_WM8776_write(ice, 0x11, 0x000); /* Off */ + break; + case 1: + se200pci_WM8776_write(ice, 0x10, 0x07b); + se200pci_WM8776_write(ice, 0x11, 0x100); /* LimiterMode */ + break; + case 2: + se200pci_WM8776_write(ice, 0x10, 0x1fb); + se200pci_WM8776_write(ice, 0x11, 0x100); /* ALCMode */ + break; + } +} + +static void __devinit se200pci_WM8776_init(struct snd_ice1712 *ice) +{ + int i; + static unsigned short __devinitdata default_values[] = { + 0x100, 0x100, 0x100, + 0x100, 0x100, 0x100, + 0x000, 0x090, 0x000, 0x000, + 0x022, 0x022, 0x022, + 0x008, 0x0cf, 0x0cf, 0x07b, 0x000, + 0x032, 0x000, 0x0a6, 0x001, 0x001 + }; + + se200pci_WM8776_write(ice, 0x17, 0x000); /* reset all */ + /* ADC and DAC interface is I2S 24bits mode */ + /* The sample-rate are automatically changed */ + udelay(10); + /* BUT my board can not do reset all, so I load all by manually. */ + for (i = 0; i < ARRAY_SIZE(default_values); i++) + se200pci_WM8776_write(ice, i, default_values[i]); + + se200pci_WM8776_set_input_selector(ice, 0); + se200pci_WM8776_set_afl(ice, 0); + se200pci_WM8776_set_agc(ice, 0); + se200pci_WM8776_set_input_volume(ice, 0, 0); + se200pci_WM8776_set_output_volume(ice, 0, 0); + + /* head phone mute and power down */ + se200pci_WM8776_write(ice, 0x00, 0); + se200pci_WM8776_write(ice, 0x01, 0); + se200pci_WM8776_write(ice, 0x02, 0x100); + se200pci_WM8776_write(ice, 0x0d, 0x080); +} + +static void se200pci_WM8776_set_pro_rate(struct snd_ice1712 *ice, + unsigned int rate) +{ + /* nothing to do */ +} + + +/****************************************************************************/ +/* runtime interface */ +/****************************************************************************/ + +static void se200pci_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + se200pci_WM8740_set_pro_rate(ice, rate); + se200pci_WM8766_set_pro_rate(ice, rate); + se200pci_WM8776_set_pro_rate(ice, rate); +} + +struct se200pci_control { + char *name; + enum { + WM8766, + WM8776in, + WM8776out, + WM8776sel, + WM8776agc, + WM8776afl + } target; + enum { VOLUME1, VOLUME2, BOOLEAN, ENUM } type; + int ch; + const char **member; + const char *comment; +}; + +static const struct se200pci_control se200pci_cont[] = { + { + .name = "Front Playback Volume", + .target = WM8776out, + .type = VOLUME1, + .comment = "Front(green)" + }, + { + .name = "Side Playback Volume", + .target = WM8766, + .type = VOLUME1, + .ch = 1, + .comment = "Surround(orange)" + }, + { + .name = "Surround Playback Volume", + .target = WM8766, + .type = VOLUME1, + .ch = 2, + .comment = "SurroundBack(white)" + }, + { + .name = "CLFE Playback Volume", + .target = WM8766, + .type = VOLUME1, + .ch = 0, + .comment = "Center(Lch)&SubWoofer(Rch)(black)" + }, + { + .name = "Capture Volume", + .target = WM8776in, + .type = VOLUME2 + }, + { + .name = "Capture Select", + .target = WM8776sel, + .type = ENUM, + .member = se200pci_sel + }, + { + .name = "AGC Capture Mode", + .target = WM8776agc, + .type = ENUM, + .member = se200pci_agc + }, + { + .name = "AFL Bypass Playback Switch", + .target = WM8776afl, + .type = BOOLEAN + } +}; + +static int se200pci_cont_info(struct snd_kcontrol *kc, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_ice1712 *ice; + int n; + int c; + const char **member; + + ice = snd_kcontrol_chip(kc); + n = kc->private_value; + + if (se200pci_cont[n].type == VOLUME1 || + se200pci_cont[n].type == VOLUME2) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; /* mute */ + uinfo->value.integer.max = 0xff; /* 0dB */ + + } else if (se200pci_cont[n].type == BOOLEAN) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + } else if (se200pci_cont[n].type == ENUM) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + member = se200pci_cont[n].member; + if (member == NULL) + return -EINVAL; + for (c = 0; member[c]; c++) + ; + + uinfo->count = 1; + uinfo->value.enumerated.items = c; + if (uinfo->value.enumerated.item >= c) + uinfo->value.enumerated.item = c - 1; + strcpy(uinfo->value.enumerated.name, + member[uinfo->value.enumerated.item]); + } + + return 0; +} + +static int se200pci_cont_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice; + int n; + + ice = snd_kcontrol_chip(kc); + n = kc->private_value; + if (se200pci_cont[n].type == VOLUME1 || + se200pci_cont[n].type == VOLUME2) { + uc->value.integer.value[0] = ice->spec.se.vol[n].ch1; + uc->value.integer.value[1] = ice->spec.se.vol[n].ch2; + + } else if (se200pci_cont[n].type == BOOLEAN || + se200pci_cont[n].type == ENUM) + uc->value.integer.value[0] = ice->spec.se.vol[n].ch1; + + return 0; +} + +static int se200pci_cont_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice; + int n; + unsigned int vol1, vol2; + int changed; + + ice = snd_kcontrol_chip(kc); + n = kc->private_value; + + changed = 0; + vol1 = 0; vol2 = 0; + if (se200pci_cont[n].type == VOLUME1 || + se200pci_cont[n].type == VOLUME2) { + vol1 = uc->value.integer.value[0]; + vol2 = uc->value.integer.value[1]; + if (ice->spec.se.vol[n].ch1 != vol1) + changed = 1; + if (ice->spec.se.vol[n].ch2 != vol2) + changed = 1; + ice->spec.se.vol[n].ch1 = vol1; + ice->spec.se.vol[n].ch2 = vol2; + + } else if (se200pci_cont[n].type == BOOLEAN || + se200pci_cont[n].type == ENUM) { + vol1 = uc->value.integer.value[0]; + if (ice->spec.se.vol[n].ch1 != vol1) + changed = 1; + ice->spec.se.vol[n].ch1 = vol1; + } + + switch (se200pci_cont[n].target) { + case WM8766: + se200pci_WM8766_set_volume(ice, + se200pci_cont[n].ch, vol1, vol2); + break; + + case WM8776in: + se200pci_WM8776_set_input_volume(ice, vol1, vol2); + break; + + case WM8776out: + se200pci_WM8776_set_output_volume(ice, vol1, vol2); + break; + + case WM8776sel: + se200pci_WM8776_set_input_selector(ice, vol1); + break; + + case WM8776agc: + se200pci_WM8776_set_agc(ice, vol1); + break; + + case WM8776afl: + se200pci_WM8776_set_afl(ice, vol1); + break; + + default: + break; + } + + return changed; +} + +static const DECLARE_TLV_DB_SCALE(db_scale_gain1, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(db_scale_gain2, -10350, 50, 1); + +static int __devinit se200pci_add_controls(struct snd_ice1712 *ice) +{ + int i; + struct snd_kcontrol_new cont; + int err; + + memset(&cont, 0, sizeof(cont)); + cont.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cont.info = se200pci_cont_info; + cont.get = se200pci_cont_get; + cont.put = se200pci_cont_put; + for (i = 0; i < ARRAY_SIZE(se200pci_cont); i++) { + cont.private_value = i; + cont.name = se200pci_cont[i].name; + cont.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + cont.tlv.p = NULL; + if (se200pci_cont[i].type == VOLUME1 || + se200pci_cont[i].type == VOLUME2) { + + cont.access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + + if (se200pci_cont[i].type == VOLUME1) + cont.tlv.p = db_scale_gain1; + else + cont.tlv.p = db_scale_gain2; + } + err = snd_ctl_add(ice->card, snd_ctl_new1(&cont, ice)); + if (err < 0) + return err; + } + + return 0; +} + + +/****************************************************************************/ +/* ONKYO WAVIO SE-90PCI */ +/****************************************************************************/ +/* + * system configuration ICE_EEP2_SYSCONF=0x4b + * AC-Link configuration ICE_EEP2_ACLINK=0x80 + * I2S converters feature ICE_EEP2_I2S=0x78 + * S/PDIF configuration ICE_EEP2_SPDIF=0xc3 + * + * ** connected chip ** + * + * WM8716 + * A 2ch-DAC of main outputs. + * It setuped as I2S mode by wire, so no way to setup from software. + * ML/I2S (28pin) -- +5V + * MC/DM1 (27pin) -- GND + * MC/DM0 (26pin) -- GND + * MUTEB (25pin) -- open (internal pull-up) + * MODE (24pin) -- GND + * CSBIWO (23pin) -- +5V + * + */ + + /* Nothing to do for this chip. */ + + +/****************************************************************************/ +/* probe/initialize/setup */ +/****************************************************************************/ + +static int __devinit se_init(struct snd_ice1712 *ice) +{ + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_SE90PCI) { + ice->num_total_dacs = 2; + ice->num_total_adcs = 0; + ice->vt1720 = 1; + return 0; + + } else if (ice->eeprom.subvendor == VT1724_SUBDEVICE_SE200PCI) { + ice->num_total_dacs = 8; + ice->num_total_adcs = 2; + se200pci_WM8740_init(ice); + se200pci_WM8766_init(ice); + se200pci_WM8776_init(ice); + ice->gpio.set_pro_rate = se200pci_set_pro_rate; + return 0; + } + + return -ENOENT; +} + +static int __devinit se_add_controls(struct snd_ice1712 *ice) +{ + int err; + + err = 0; + /* nothing to do for VT1724_SUBDEVICE_SE90PCI */ + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_SE200PCI) + err = se200pci_add_controls(ice); + + return err; +} + + +/****************************************************************************/ +/* entry point */ +/****************************************************************************/ + +static unsigned char se200pci_eeprom[] __devinitdata = { + [ICE_EEP2_SYSCONF] = 0x4b, /* 49.152Hz, spdif-in/ADC, 4DACs */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0x78, /* 96k-ok, 24bit, 192k-ok */ + [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */ + + [ICE_EEP2_GPIO_DIR] = 0x02, /* WM8766 mute 1=output */ + [ICE_EEP2_GPIO_DIR1] = 0x00, /* not used */ + [ICE_EEP2_GPIO_DIR2] = 0x07, /* WM8766 ML/MC/MD 1=output */ + + [ICE_EEP2_GPIO_MASK] = 0x00, /* 0=writable */ + [ICE_EEP2_GPIO_MASK1] = 0x00, /* 0=writable */ + [ICE_EEP2_GPIO_MASK2] = 0x00, /* 0=writable */ + + [ICE_EEP2_GPIO_STATE] = 0x00, /* WM8766 mute=0 */ + [ICE_EEP2_GPIO_STATE1] = 0x00, /* not used */ + [ICE_EEP2_GPIO_STATE2] = 0x07, /* WM8766 ML/MC/MD */ +}; + +static unsigned char se90pci_eeprom[] __devinitdata = { + [ICE_EEP2_SYSCONF] = 0x4b, /* 49.152Hz, spdif-in/ADC, 4DACs */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0x78, /* 96k-ok, 24bit, 192k-ok */ + [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */ + + /* ALL GPIO bits are in input mode */ +}; + +struct snd_ice1712_card_info snd_vt1724_se_cards[] __devinitdata = { + { + .subvendor = VT1724_SUBDEVICE_SE200PCI, + .name = "ONKYO SE200PCI", + .model = "se200pci", + .chip_init = se_init, + .build_controls = se_add_controls, + .eeprom_size = sizeof(se200pci_eeprom), + .eeprom_data = se200pci_eeprom, + }, + { + .subvendor = VT1724_SUBDEVICE_SE90PCI, + .name = "ONKYO SE90PCI", + .model = "se90pci", + .chip_init = se_init, + .build_controls = se_add_controls, + .eeprom_size = sizeof(se90pci_eeprom), + .eeprom_data = se90pci_eeprom, + }, + {} /*terminator*/ +}; diff --git a/sound/pci/ice1712/se.h b/sound/pci/ice1712/se.h new file mode 100644 index 0000000..0b0a9da --- /dev/null +++ b/sound/pci/ice1712/se.h @@ -0,0 +1,15 @@ +#ifndef __SOUND_SE_H +#define __SOUND_SE_H + +/* ID */ +#define SE_DEVICE_DESC \ + "{ONKYO INC,SE-90PCI},"\ + "{ONKYO INC,SE-200PCI}," + +#define VT1724_SUBDEVICE_SE90PCI 0xb161000 +#define VT1724_SUBDEVICE_SE200PCI 0xb160100 + +/* entry struct */ +extern struct snd_ice1712_card_info snd_vt1724_se_cards[]; + +#endif /* __SOUND_SE_H */ -- cgit v0.10.2 From 3f03f7c586471f376eab497ab33394ab086f5301 Mon Sep 17 00:00:00 2001 From: Michael Opdenacker Date: Wed, 24 Oct 2007 10:59:44 +0200 Subject: [ALSA] writing-an-alsa-driver.tmpl: English style improvements This patch brings some English style improvements throughout the document, as well as 1 or 2 extra technical details. Signed-off-by: Michael Opdenacker Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 2c3fc3c..48e4053 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -18,7 +18,7 @@ - September 10, 2007 + Oct 15, 2007 0.3.7 @@ -67,7 +67,7 @@ This document describes how to write an ALSA (Advanced Linux Sound Architecture) - driver. The document focuses mainly on the PCI soundcard. + driver. The document focuses mainly on PCI soundcards. In the case of other device types, the API might be different, too. However, at least the ALSA kernel API is consistent, and therefore it would be still a bit help for @@ -75,23 +75,23 @@ - The target of this document is ones who already have enough - skill of C language and have the basic knowledge of linux - kernel programming. This document doesn't explain the general - topics of linux kernel codes and doesn't cover the detail of - implementation of each low-level driver. It describes only how is + This document targets people who already have enough + C language skills and have basic linux kernel programming + knowledge. This document doesn't explain the general + topic of linux kernel coding and doesn't cover low-level + driver implementation details. It only describes the standard way to write a PCI sound driver on ALSA. - If you are already familiar with the older ALSA ver.0.5.x, you - can check the drivers such as es1938.c or - maestro3.c which have also almost the same + If you are already familiar with the older ALSA ver.0.5.x API, you + can check the drivers such as sound/pci/es1938.c or + sound/pci/maestro3.c which have also almost the same code-base in the ALSA 0.5.x tree, so you can compare the differences. - This document is still a draft version. Any feedbacks and + This document is still a draft version. Any feedback and corrections, please!! @@ -106,7 +106,7 @@
General - The ALSA drivers are provided in the two ways. + The ALSA drivers are provided in two ways. @@ -114,15 +114,14 @@ ALSA's ftp site, and another is the 2.6 (or later) Linux kernel tree. To synchronize both, the ALSA driver tree is split into two different trees: alsa-kernel and alsa-driver. The former - contains purely the source codes for the Linux 2.6 (or later) + contains purely the source code for the Linux 2.6 (or later) tree. This tree is designed only for compilation on 2.6 or later environment. The latter, alsa-driver, contains many subtle - files for compiling the ALSA driver on the outside of Linux - kernel like configure script, the wrapper functions for older, - 2.2 and 2.4 kernels, to adapt the latest kernel API, + files for compiling ALSA drivers outside of the Linux kernel tree, + wrapper functions for older 2.2 and 2.4 kernels, to adapt the latest kernel API, and additional drivers which are still in development or in tests. The drivers in alsa-driver tree will be moved to - alsa-kernel (eventually 2.6 kernel tree) once when they are + alsa-kernel (and eventually to the 2.6 kernel tree) when they are finished and confirmed to work fine. @@ -168,7 +167,7 @@
core directory - This directory contains the middle layer, that is, the heart + This directory contains the middle layer which is the heart of ALSA drivers. In this directory, the native ALSA modules are stored. The sub-directories contain different modules and are dependent upon the kernel config. @@ -181,7 +180,7 @@ The codes for PCM and mixer OSS emulation modules are stored in this directory. The rawmidi OSS emulation is included in the ALSA rawmidi code since it's quite small. The sequencer - code is stored in core/seq/oss directory (see + code is stored in core/seq/oss directory (see below). @@ -200,7 +199,7 @@
core/seq - This and its sub-directories are for the ALSA + This directory and its sub-directories are for the ALSA sequencer. This directory contains the sequencer core and primary sequencer modules such like snd-seq-midi, snd-seq-virmidi, etc. They are compiled only when @@ -229,22 +228,22 @@ include directory This is the place for the public header files of ALSA drivers, - which are to be exported to the user-space, or included by + which are to be exported to user-space, or included by several files at different directories. Basically, the private header files should not be placed in this directory, but you may - still find files there, due to historical reason :) + still find files there, due to historical reasons :)
drivers directory - This directory contains the codes shared among different drivers - on the different architectures. They are hence supposed not to be + This directory contains code shared among different drivers + on different architectures. They are hence supposed not to be architecture-specific. For example, the dummy pcm driver and the serial MIDI driver are found in this directory. In the sub-directories, - there are the codes for components which are independent from + there is code for components which are independent from bus and cpu architectures. @@ -271,7 +270,7 @@ Although there is a standard i2c layer on Linux, ALSA has its - own i2c codes for some cards, because the soundcard needs only a + own i2c code for some cards, because the soundcard needs only a simple operation and the standard i2c API is too complicated for such a purpose. @@ -292,28 +291,28 @@ So far, there is only Emu8000/Emu10k1 synth driver under - synth/emux sub-directory. + the synth/emux sub-directory.
pci directory - This and its sub-directories hold the top-level card modules - for PCI soundcards and the codes specific to the PCI BUS. + This directory and its sub-directories hold the top-level card modules + for PCI soundcards and the code specific to the PCI BUS. - The drivers compiled from a single file is stored directly on - pci directory, while the drivers with several source files are - stored on its own sub-directory (e.g. emu10k1, ice1712). + The drivers compiled from a single file are stored directly + in the pci directory, while the drivers with several source files are + stored on their own sub-directory (e.g. emu10k1, ice1712).
isa directory - This and its sub-directories hold the top-level card modules + This directory and its sub-directories hold the top-level card modules for ISA soundcards.
@@ -321,16 +320,16 @@
arm, ppc, and sparc directories - These are for the top-level card modules which are - specific to each given architecture. + They are used for top-level card modules which are + specific to one of these architectures.
usb directory - This contains the USB-audio driver. On the latest version, the - USB MIDI driver is integrated together with usb-audio driver. + This directory contains the USB-audio driver. In the latest version, the + USB MIDI driver is integrated in the usb-audio driver.
@@ -338,16 +337,17 @@ pcmcia directory The PCMCIA, especially PCCard drivers will go here. CardBus - drivers will be on pci directory, because its API is identical - with the standard PCI cards. + drivers will be in the pci directory, because their API is identical + to that of standard PCI cards.
oss directory - The OSS/Lite source files are stored here on Linux 2.6 (or - later) tree. (In the ALSA driver tarball, it's empty, of course :) + The OSS/Lite source files are stored here in Linux 2.6 (or + later) tree. In the ALSA driver tarball, this directory is empty, + of course :)
@@ -362,7 +362,7 @@
Outline - The minimum flow of PCI soundcard is like the following: + The minimum flow for PCI soundcards is as follows: define the PCI ID table (see the section @@ -370,9 +370,13 @@ ). create probe() callback. create remove() callback. - create pci_driver table which contains the three pointers above. - create init() function just calling pci_register_driver() to register the pci_driver table defined above. - create exit() function to call pci_unregister_driver() function. + create a pci_driver structure + containing the three pointers above. + create an init() function just calling + the pci_register_driver() to register the pci_driver table + defined above. + create an exit() function to call + the pci_unregister_driver() function.
@@ -382,12 +386,12 @@ The code example is shown below. Some parts are kept unimplemented at this moment but will be filled in the - succeeding sections. The numbers in comment lines of - snd_mychip_probe() function are the - markers. + next sections. The numbers in the comment lines of the + snd_mychip_probe() function + refer to details explained in the following section. - Basic Flow for PCI Drivers Example + Basic Flow for PCI Drivers - Example @@ -398,6 +402,7 @@ #include /* module parameters (see "Module Parameters") */ + /* SNDRV_CARDS: maximum number of cards supported by this module */ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; @@ -405,13 +410,13 @@ /* definition of the chip-specific record */ struct mychip { struct snd_card *card; - /* rest of implementation will be in the section - * "PCI Resource Managements" + /* the rest of the implementation will be in section + * "PCI Resource Management" */ }; /* chip-specific destructor - * (see "PCI Resource Managements") + * (see "PCI Resource Management") */ static int snd_mychip_free(struct mychip *chip) { @@ -442,7 +447,7 @@ *rchip = NULL; /* check PCI availability here - * (see "PCI Resource Managements") + * (see "PCI Resource Management") */ .... @@ -454,7 +459,7 @@ chip->card = card; /* rest of initialization here; will be implemented - * later, see "PCI Resource Managements" + * later, see "PCI Resource Management" */ .... @@ -521,7 +526,7 @@ return 0; } - /* destructor -- see "Destructor" sub-section */ + /* destructor -- see the "Destructor" sub-section */ static void __devexit snd_mychip_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); @@ -536,16 +541,16 @@
Constructor - The real constructor of PCI drivers is probe callback. The - probe callback and other component-constructors which are called - from probe callback should be defined with - __devinit prefix. You - cannot use __init prefix for them, + The real constructor of PCI drivers is the probe callback. + The probe callback and other component-constructors which are called + from the probe callback should be defined with + the __devinit prefix. You + cannot use the __init prefix for them, because any PCI device could be a hotplug device. - In the probe callback, the following scheme is often used. + In the probe callback, the following scheme is often used.
@@ -570,7 +575,7 @@ - At each time probe callback is called, check the + Each time the probe callback is called, check the availability of the device. If not available, simply increment the device index and returns. dev will be incremented also later ( - The detail will be explained in the section + The details will be explained in the section Management of Cards and Components. @@ -619,9 +624,9 @@ - The detail will be explained in the section PCI Resource - Managements. + Management.
@@ -640,7 +645,7 @@ The driver field holds the minimal ID string of the - chip. This is referred by alsa-lib's configurator, so keep it + chip. This is used by alsa-lib's configurator, so keep it simple but unique. Even the same driver can have different driver IDs to distinguish the functionality of each chip type. @@ -648,7 +653,7 @@ The shortname field is a string shown as more verbose - name. The longname field contains the information which is + name. The longname field contains the information shown in /proc/asound/cards.
@@ -703,7 +708,7 @@ In the above, the card record is stored. This pointer is - referred in the remove callback and power-management + used in the remove callback and power-management callbacks, too.
@@ -757,22 +762,22 @@ where the last one is necessary only when module options are - defined in the source file. If the codes are split to several - files, the file without module options don't need them. + defined in the source file. If the code is split into several + files, the files without module options don't need them. - In addition to them, you'll need - <linux/interrupt.h> for the interrupt - handling, and <asm/io.h> for the i/o - access. If you use mdelay() or + In addition to these headers, you'll need + <linux/interrupt.h> for interrupt + handling, and <asm/io.h> for I/O + access. If you use the mdelay() or udelay() functions, you'll need to include - <linux/delay.h>, too. + <linux/delay.h> too. - The ALSA interfaces like PCM or control API are defined in other - header files as <sound/xxx.h>. + The ALSA interfaces like the PCM and control APIs are defined in other + <sound/xxx.h> header files. They have to be included after <sound/core.h>. @@ -795,12 +800,12 @@ A card record is the headquarters of the soundcard. It manages - the list of whole devices (components) on the soundcard, such as + the whole list of devices (components) on the soundcard, such as PCM, mixers, MIDI, synthesizer, and so on. Also, the card record holds the ID and the name strings of the card, manages the root of proc files, and controls the power-management states and hotplug disconnections. The component list on the card - record is used to manage the proper releases of resources at + record is used to manage the correct release of resources at destruction. @@ -824,9 +829,8 @@ THIS_MODULE), and the size of extra-data space. The last argument is used to allocate card->private_data for the - chip-specific data. Note that this data - is allocated by - snd_card_new(). + chip-specific data. Note that these data + are allocated by snd_card_new(). @@ -834,10 +838,10 @@ Components After the card is created, you can attach the components - (devices) to the card instance. On ALSA driver, a component is + (devices) to the card instance. In an ALSA driver, a component is represented as a struct snd_device object. A component can be a PCM instance, a control interface, a raw - MIDI interface, etc. Each of such instances has one component + MIDI interface, etc. Each such instance has one component entry. @@ -859,7 +863,7 @@ (SNDRV_DEV_XXX), the data pointer, and the callback pointers (&ops). The device-level defines the type of components and the order of - registration and de-registration. For most of components, the + registration and de-registration. For most components, the device-level is already defined. For a user-defined component, you can use SNDRV_DEV_LOWLEVEL. @@ -867,13 +871,13 @@ This function itself doesn't allocate the data space. The data must be allocated manually beforehand, and its pointer is passed - as the argument. This pointer is used as the identifier - (chip in the above example) for the - instance. + as the argument. This pointer is used as the + (chip identifier in the above example) + for the instance. - Each ALSA pre-defined component such as ac97 or pcm calls + Each pre-defined ALSA component such as ac97 and pcm calls snd_device_new() inside its constructor. The destructor for each component is defined in the callback pointers. Hence, you don't need to take care of @@ -881,19 +885,19 @@ - If you would like to create your own component, you need to - set the destructor function to dev_free callback in - ops, so that it can be released - automatically via snd_card_free(). The - example will be shown later as an implementation of a - chip-specific data. + If you wish to create your own component, you need to + set the destructor function to the dev_free callback in + the ops, so that it can be released + automatically via snd_card_free(). + The next example will show an implementation of chip-specific + data.
Chip-Specific Data - The chip-specific information, e.g. the i/o port address, its + Chip-specific information, e.g. the I/O port address, its resource pointer, or the irq number, is stored in the chip-specific record. @@ -909,13 +913,14 @@ - In general, there are two ways to allocate the chip record. + In general, there are two ways of allocating the chip record.
1. Allocating via <function>snd_card_new()</function>. - As mentioned above, you can pass the extra-data-length to the 4th argument of snd_card_new(), i.e. + As mentioned above, you can pass the extra-data-length + to the 4th argument of snd_card_new(), i.e. @@ -925,7 +930,7 @@ - whether struct mychip is the type of the chip record. + struct mychip is the type of the chip record. @@ -1037,8 +1042,8 @@ Registration and Release After all components are assigned, register the card instance - by calling snd_card_register(). The access - to the device files are enabled at this point. That is, before + by calling snd_card_register(). Access + to the device files is enabled at this point. That is, before snd_card_register() is called, the components are safely inaccessible from external side. If this call fails, exit the probe function after releasing the card via @@ -1047,7 +1052,7 @@ For releasing the card instance, you can call simply - snd_card_free(). As already mentioned, all + snd_card_free(). As mentioned earlier, all components are released automatically by this call. @@ -1055,7 +1060,7 @@ As further notes, the destructors (both snd_mychip_dev_free and snd_mychip_free) cannot be defined with - __devexit prefix, because they may be + the __devexit prefix, because they may be called from the constructor, too, at the false path. @@ -1071,20 +1076,20 @@ - + - PCI Resource Managements + PCI Resource Management
Full Code Example - In this section, we'll finish the chip-specific constructor, - destructor and PCI entries. The example code is shown first, + In this section, we'll complete the chip-specific constructor, + destructor and PCI entries. Example code is shown first, below. - PCI Resource Managements Example + PCI Resource Management Example irq >= 0) free_irq(chip->irq, chip); - /* release the i/o ports & memory */ + /* release the I/O ports & memory */ pci_release_regions(chip->pci); /* disable the PCI entry */ pci_disable_device(chip->pci); @@ -1196,13 +1201,13 @@ .remove = __devexit_p(snd_mychip_remove), }; - /* initialization of the module */ + /* module initialization */ static int __init alsa_card_mychip_init(void) { return pci_register_driver(&driver); } - /* clean up the module */ + /* module clean up */ static void __exit alsa_card_mychip_exit(void) { pci_unregister_driver(&driver); @@ -1228,10 +1233,10 @@ - In the case of PCI devices, you have to call at first - pci_enable_device() function before + In the case of PCI devices, you first have to call + the pci_enable_device() function before allocating resources. Also, you need to set the proper PCI DMA - mask to limit the accessed i/o range. In some cases, you might + mask to limit the accessed I/O range. In some cases, you might need to call pci_set_master() function, too. @@ -1261,15 +1266,15 @@
Resource Allocation - The allocation of I/O ports and irqs are done via standard kernel + The allocation of I/O ports and irqs is done via standard kernel functions. Unlike ALSA ver.0.5.x., there are no helpers for that. And these resources must be released in the destructor function (see below). Also, on ALSA 0.9.x, you don't need to - allocate (pseudo-)DMA for PCI like ALSA 0.5.x. + allocate (pseudo-)DMA for PCI like in ALSA 0.5.x. - Now assume that this PCI device has an I/O port with 8 bytes + Now assume that the PCI device has an I/O port with 8 bytes and an interrupt. Then struct mychip will have the following fields: @@ -1288,7 +1293,7 @@ - For an i/o port (and also a memory region), you need to have + For an I/O port (and also a memory region), you need to have the resource pointer for the standard resource management. For an irq, you have to keep only the irq number (integer). But you need to initialize this number as -1 before actual allocation, @@ -1299,7 +1304,7 @@ - The allocation of an i/o port is done like this: + The allocation of an I/O port is done like this: @@ -1318,12 +1323,12 @@ - It will reserve the i/o port region of 8 bytes of the given + It will reserve the I/O port region of 8 bytes of the given PCI device. The returned value, chip->res_port, is allocated via kmalloc() by request_region(). The pointer must be - released via kfree(), but there is some - problem regarding this. This issue will be explained more below. + released via kfree(), but there is a + problem with this. This issue will be explained later. @@ -1351,8 +1356,8 @@ - On the PCI bus, the interrupts can be shared. Thus, - IRQF_SHARED is given as the interrupt flag of + On the PCI bus, interrupts can be shared. Thus, + IRQF_SHARED is used as the interrupt flag of request_irq(). @@ -1364,7 +1369,7 @@ - I won't define the detail of the interrupt handler at this + I won't give details about the interrupt handler at this point, but at least its appearance can be explained now. The interrupt handler looks usually like the following: @@ -1386,11 +1391,11 @@ Now let's write the corresponding destructor for the resources above. The role of destructor is simple: disable the hardware (if already activated) and release the resources. So far, we - have no hardware part, so the disabling is not written here. + have no hardware part, so the disabling code is not written here. - For releasing the resources, check-and-release + To release the resources, the check-and-release method is a safer way. For the interrupt, do like this: @@ -1410,7 +1415,7 @@ When you requested I/O ports or memory regions via pci_request_region() or - pci_request_regions() like this example, + pci_request_regions() like in this example, release the resource(s) using the corresponding function, pci_release_region() or pci_release_regions(). @@ -1429,7 +1434,7 @@ or request_mem_region, you can release it via release_resource(). Suppose that you keep the resource pointer returned from request_region() - in chip->res_port, the release procedure looks like below: + in chip->res_port, the release procedure looks like: @@ -1442,7 +1447,7 @@ Don't forget to call pci_disable_device() - before all finished. + before the end. @@ -1459,14 +1464,14 @@ Again, remember that you cannot - set __devexit prefix for this destructor. + use the __devexit prefix for this destructor. - We didn't implement the hardware-disabling part in the above. + We didn't implement the hardware disabling part in the above. If you need to do this, please note that the destructor may be called even before the initialization of the chip is completed. - It would be better to have a flag to skip the hardware-disabling + It would be better to have a flag to skip hardware disabling if the hardware was not initialized yet. @@ -1475,14 +1480,14 @@ snd_device_new() with SNDRV_DEV_LOWLELVEL , its destructor is called at the last. That is, it is assured that all other - components like PCMs and controls have been already released. - You don't have to call stopping PCMs, etc. explicitly, but just - stop the hardware in the low-level. + components like PCMs and controls have already been released. + You don't have to stop PCMs, etc. explicitly, but just + call low-level hardware stopping. The management of a memory-mapped region is almost as same as - the management of an i/o port. You'll need three fields like + the management of an I/O port. You'll need three fields like the following: @@ -1561,8 +1566,8 @@
PCI Entries - So far, so good. Let's finish the rest of missing PCI - stuffs. At first, we need a + So far, so good. Let's finish the missing PCI + stuff. At first, we need a pci_device_id table for this chipset. It's a table of PCI vendor/device ID number, and some masks. @@ -1588,13 +1593,13 @@ The first and second fields of - pci_device_id struct are the vendor and - device IDs. If you have nothing special to filter the matching - devices, you can use the rest of fields like above. The last - field of pci_device_id struct is a + the pci_device_id structure are the vendor and + device IDs. If you have no reason to filter the matching + devices, you can leave the remaining fields as above. The last + field of the pci_device_id struct contains private data for this entry. You can specify any value here, for - example, to tell the type of different operations per each - device IDs. Such an example is found in intel8x0 driver. + example, to define specific operations for supported device IDs. + Such an example is found in the intel8x0 driver. @@ -1621,10 +1626,10 @@ The probe and - remove functions are what we already - defined in - the previous sections. The remove should - be defined with + remove functions have already + been defined in the previous sections. + The remove function should + be defined with the __devexit_p() macro, so that it's not defined for built-in (and non-hot-pluggable) case. The name @@ -1665,8 +1670,7 @@ Oh, one thing was forgotten. If you have no exported symbols, - you need to declare it on 2.2 or 2.4 kernels (on 2.6 kernels - it's not necessary, though). + you need to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). @@ -1698,7 +1702,7 @@ For accessing to the PCM layer, you need to include - <sound/pcm.h> above all. In addition, + <sound/pcm.h> first. In addition, <sound/pcm_params.h> might be needed if you access to some functions related with hw_param. @@ -1707,21 +1711,21 @@ Each card device can have up to four pcm instances. A pcm instance corresponds to a pcm device file. The limitation of number of instances comes only from the available bit size of - the linux's device number. Once when 64bit device number is - used, we'll have more available pcm instances. + the Linux's device numbers. Once when 64bit device number is + used, we'll have more pcm instances available. A pcm instance consists of pcm playback and capture streams, and each pcm stream consists of one or more pcm substreams. Some - soundcard supports the multiple-playback function. For example, + soundcards support multiple playback functions. For example, emu10k1 has a PCM playback of 32 stereo substreams. In this case, at each open, a free substream is (usually) automatically chosen and opened. Meanwhile, when only one substream exists and it was - already opened, the succeeding open will result in the blocking - or the error with EAGAIN according to the - file open mode. But you don't have to know the detail in your - driver. The PCM middle layer will take all such jobs. + already opened, the successful open will either block + or error with EAGAIN according to the + file open mode. But you don't have to care about such details in your + driver. The PCM middle layer will take care of such work.
@@ -1944,7 +1948,7 @@
Constructor - A pcm instance is allocated by snd_pcm_new() + A pcm instance is allocated by the snd_pcm_new() function. It would be better to create a constructor for pcm, namely, @@ -1971,23 +1975,23 @@ - The snd_pcm_new() function takes the four + The snd_pcm_new() function takes four arguments. The first argument is the card pointer to which this pcm is assigned, and the second is the ID string. The third argument (index, 0 in the - above) is the index of this new pcm. It begins from zero. When - you will create more than one pcm instances, specify the + above) is the index of this new pcm. It begins from zero. If + you create more than one pcm instances, specify the different numbers in this argument. For example, index = 1 for the second PCM device. The fourth and fifth arguments are the number of substreams - for playback and capture, respectively. Here both 1 are given in - the above example. When no playback or no capture is available, + for playback and capture, respectively. Here 1 is used for + both arguments. When no playback or capture substreams are available, pass 0 to the corresponding argument. @@ -2045,13 +2049,13 @@ - Each of callbacks is explained in the subsection + All the callbacks are described in the - Operators. + Operators subsection. - After setting the operators, most likely you'd like to + After setting the operators, you probably will want to pre-allocate the buffer. For the pre-allocation, simply call the following: @@ -2065,8 +2069,8 @@ - It will allocate up to 64kB buffer as default. The details of - buffer management will be described in the later section Buffer and Memory Management. @@ -2095,13 +2099,13 @@ The destructor for a pcm instance is not always necessary. Since the pcm device will be released by the middle - layer code automatically, you don't have to call destructor + layer code automatically, you don't have to call the destructor explicitly. - The destructor would be necessary when you created some - special records internally and need to release them. In such a + The destructor would be necessary if you created + special records internally and needed to release them. In such a case, set the destructor function to pcm->private_free: @@ -2141,16 +2145,15 @@ When the PCM substream is opened, a PCM runtime instance is allocated and assigned to the substream. This pointer is accessible via substream->runtime. - This runtime pointer holds the various information; it holds - the copy of hw_params and sw_params configurations, the buffer - pointers, mmap records, spinlocks, etc. Almost everything you - need for controlling the PCM can be found there. + This runtime pointer holds most information you need + to control the PCM: the copy of hw_params and sw_params configurations, the buffer + pointers, mmap records, spinlocks, etc. The definition of runtime instance is found in - <sound/pcm.h>. Here is the - copy from the file. + <sound/pcm.h>. Here are + the contents of this file: For the operators (callbacks) of each sound driver, most of these records are supposed to be read-only. Only the PCM - middle-layer changes / updates these info. The exceptions are + middle-layer changes / updates them. The exceptions are the hardware description (hw), interrupt callbacks (transfer_ack_xxx), DMA buffer information, and the private data. Besides, if you use the standard buffer allocation @@ -2285,7 +2288,7 @@ struct _snd_pcm_runtime { - Typically, you'll have a hardware descriptor like below: + Typically, you'll have a hardware descriptor as below: SNDRV_PCM_INFO_XXX. Here, at least, you have to specify whether the mmap is supported and which interleaved format is supported. - When the mmap is supported, add + When the is supported, add the SNDRV_PCM_INFO_MMAP flag here. When the hardware supports the interleaved or the non-interleaved - format, SNDRV_PCM_INFO_INTERLEAVED or + formats, SNDRV_PCM_INFO_INTERLEAVED or SNDRV_PCM_INFO_NONINTERLEAVED flag must be set, respectively. If both are supported, you can set both, too. @@ -2331,7 +2334,7 @@ struct _snd_pcm_runtime { In the above example, MMAP_VALID and - BLOCK_TRANSFER are specified for OSS mmap + BLOCK_TRANSFER are specified for the OSS mmap mode. Usually both are set. Of course, MMAP_VALID is set only if the mmap is really supported. @@ -2345,11 +2348,11 @@ struct _snd_pcm_runtime { pause operation, while the RESUME bit means that the pcm supports the full suspend/resume operation. - If PAUSE flag is set, + If the PAUSE flag is set, the trigger callback below must handle the corresponding (pause push/release) commands. The suspend/resume trigger commands can be defined even without - RESUME flag. See RESUME flag. See Power Management section for details. @@ -2382,7 +2385,7 @@ struct _snd_pcm_runtime { CONTINUOUS bit additionally. The pre-defined rate bits are provided only for typical rates. If your chip supports unconventional rates, you need to add - KNOT bit and set up the hardware + the KNOT bit and set up the hardware constraint manually (explained later). @@ -2390,8 +2393,8 @@ struct _snd_pcm_runtime { rate_min and - rate_max define the minimal and - maximal sample rate. This should correspond somehow to + rate_max define the minimum and + maximum sample rate. This should correspond somehow to rates bits. @@ -2400,7 +2403,7 @@ struct _snd_pcm_runtime { channel_min and channel_max - define, as you might already expected, the minimal and maximal + define, as you might already expected, the minimum and maximum number of channels. @@ -2408,21 +2411,21 @@ struct _snd_pcm_runtime { buffer_bytes_max defines the - maximal buffer size in bytes. There is no + maximum buffer size in bytes. There is no buffer_bytes_min field, since - it can be calculated from the minimal period size and the - minimal number of periods. + it can be calculated from the minimum period size and the + minimum number of periods. Meanwhile, period_bytes_min and - define the minimal and maximal size of the period in bytes. + define the minimum and maximum size of the period in bytes. periods_max and - periods_min define the maximal and - minimal number of periods in the buffer. + periods_min define the maximum and + minimum number of periods in the buffer. - The period is a term, that corresponds to - fragment in the OSS world. The period defines the size at - which the PCM interrupt is generated. This size strongly + The period is a term that corresponds to + a fragment in the OSS world. The period defines the size at + which a PCM interrupt is generated. This size strongly depends on the hardware. Generally, the smaller period size will give you more interrupts, that is, more controls. @@ -2435,8 +2438,8 @@ struct _snd_pcm_runtime { There is also a field fifo_size. - This specifies the size of the hardware FIFO, but it's not - used currently in the driver nor in the alsa-lib. So, you + This specifies the size of the hardware FIFO, but currently it + is neither used in the driver nor in the alsa-lib. So, you can ignore this field. @@ -2450,7 +2453,7 @@ struct _snd_pcm_runtime { Ok, let's go back again to the PCM runtime records. The most frequently referred records in the runtime instance are the PCM configurations. - The PCM configurations are stored on runtime instance + The PCM configurations are stored in the runtime instance after the application sends hw_params data via alsa-lib. There are many fields copied from hw_params and sw_params structs. For example, @@ -2461,11 +2464,11 @@ struct _snd_pcm_runtime { One thing to be noted is that the configured buffer and period - sizes are stored in frames in the runtime + sizes are stored in frames in the runtime. In the ALSA world, 1 frame = channels * samples-size. For conversion between frames and bytes, you can use the - helper functions, frames_to_bytes() and - bytes_to_frames(). + frames_to_bytes() and + bytes_to_frames() helper functions. dma_area is necessary when the buffer is mmapped. If your driver doesn't support mmap, this field is not necessary. dma_addr - is also not mandatory. You can use + is also optional. You can use dma_private as you like, too.
@@ -2524,14 +2527,14 @@ struct _snd_pcm_runtime { Running Status The running status can be referred via runtime->status. - This is the pointer to struct snd_pcm_mmap_status + This is the pointer to the struct snd_pcm_mmap_status record. For example, you can get the current DMA hardware pointer via runtime->status->hw_ptr. The DMA application pointer can be referred via - runtime->control, which points + runtime->control, which points to the struct snd_pcm_mmap_control record. However, accessing directly to this value is not recommended. @@ -2542,14 +2545,14 @@ struct _snd_pcm_runtime { You can allocate a record for the substream and store it in runtime->private_data. Usually, this - done in + is done in the open callback. Don't mix this with pcm->private_data. - The pcm->private_data usually points the + The pcm->private_data usually points to the chip instance assigned statically at the creation of PCM, while the - runtime->private_data points a dynamic - data created at the PCM open callback. + runtime->private_data points to a dynamic + data structure created at the PCM open callback. @@ -2579,7 +2582,7 @@ struct _snd_pcm_runtime { The field transfer_ack_begin and transfer_ack_end are called at - the beginning and the end of + the beginning and at the end of snd_pcm_period_elapsed(), respectively.
@@ -2589,17 +2592,18 @@ struct _snd_pcm_runtime {
Operators - OK, now let me explain the detail of each pcm callback + OK, now let me give details about each pcm callback (ops). In general, every callback must - return 0 if successful, or a negative number with the error - number such as -EINVAL at any - error. + return 0 if successful, or a negative error number + such as -EINVAL. To choose an appropriate + error number, it is advised to check what value other parts of + the kernel return when the same kind of request fails. The callback function takes at least the argument with - snd_pcm_substream pointer. For retrieving the - chip record from the given substream instance, you can use the + snd_pcm_substream pointer. To retrieve + the chip record from the given substream instance, you can use the following macro. @@ -2616,7 +2620,7 @@ struct _snd_pcm_runtime { The macro reads substream->private_data, which is a copy of pcm->private_data. You can override the former if you need to assign different data - records per PCM substream. For example, cmi8330 driver assigns + records per PCM substream. For example, the cmi8330 driver assigns different private_data for playback and capture directions, because it uses two different codecs (SB- and AD-compatible) for different directions. @@ -2709,7 +2713,7 @@ struct _snd_pcm_runtime {
ioctl callback - This is used for any special action to pcm ioctls. But + This is used for any special call to pcm ioctls. But usually you can pass a generic ioctl callback, snd_pcm_lib_ioctl. @@ -2726,9 +2730,6 @@ struct _snd_pcm_runtime { ]]> - - This and hw_free callbacks exist - only on ALSA 0.9.x. @@ -2740,13 +2741,13 @@ struct _snd_pcm_runtime { - Many hardware set-up should be done in this callback, + Many hardware setups should be done in this callback, including the allocation of buffers. Parameters to be initialized are retrieved by - params_xxx() macros. For allocating a + params_xxx() macros. To allocate buffer, you can call a helper function, @@ -2772,8 +2773,8 @@ struct _snd_pcm_runtime { - Thus, you need to take care not to allocate the same buffers - many times, which will lead to memory leak! Calling the + Thus, you need to be careful not to allocate the same buffers + many times, which will lead to memory leaks! Calling the helper function above many times is OK. It will release the previous buffer automatically when it was already allocated. @@ -2782,7 +2783,7 @@ struct _snd_pcm_runtime { Another note is that this callback is non-atomic (schedulable). This is important, because the trigger callback - is atomic (non-schedulable). That is, mutex or any + is atomic (non-schedulable). That is, mutexes or any schedule-related functions are not available in trigger callback. Please see the subsection @@ -2843,15 +2844,15 @@ struct _snd_pcm_runtime { prepared. You can set the format type, sample rate, etc. here. The difference from hw_params is that the - prepare callback will be called at each + prepare callback will be called each time snd_pcm_prepare() is called, i.e. when - recovered after underruns, etc. + recovering after underruns, etc. - Note that this callback became non-atomic since the recent version. - You can use schedule-related functions safely in this callback now. + Note that this callback is now non-atomic. + You can use schedule-related functions safely in this callback. @@ -2871,7 +2872,7 @@ struct _snd_pcm_runtime { Be careful that this callback will be called many times at - each set up, too. + each setup, too.
@@ -2893,7 +2894,7 @@ struct _snd_pcm_runtime { Which action is specified in the second argument, SNDRV_PCM_TRIGGER_XXX in <sound/pcm.h>. At least, - START and STOP + the START and STOP commands must be defined in this callback. @@ -2915,8 +2916,8 @@ struct _snd_pcm_runtime { - When the pcm supports the pause operation (given in info - field of the hardware table), PAUSE_PUSE + When the pcm supports the pause operation (given in the info + field of the hardware table), the PAUSE_PUSE and PAUSE_RELEASE commands must be handled here, too. The former is the command to pause the pcm, and the latter to restart the pcm again. @@ -2925,21 +2926,21 @@ struct _snd_pcm_runtime { When the pcm supports the suspend/resume operation, regardless of full or partial suspend/resume support, - SUSPEND and RESUME + the SUSPEND and RESUME commands must be handled, too. These commands are issued when the power-management status is changed. Obviously, the SUSPEND and - RESUME - do suspend and resume of the pcm substream, and usually, they - are identical with STOP and + RESUME commands + suspend and resume the pcm substream, and usually, they + are identical to the STOP and START commands, respectively. - See + See the Power Management section for details. As mentioned, this callback is atomic. You cannot call - the function going to sleep. + functions which may sleep. The trigger callback should be as minimal as possible, just really triggering the DMA. The other stuff should be initialized hw_params and prepare callbacks properly @@ -2960,8 +2961,8 @@ struct _snd_pcm_runtime { This callback is called when the PCM middle layer inquires the current hardware position on the buffer. The position must - be returned in frames (which was in bytes on ALSA 0.5.x), - ranged from 0 to buffer_size - 1. + be returned in frames, + ranging from 0 to buffer_size - 1. @@ -2983,7 +2984,7 @@ struct _snd_pcm_runtime { These callbacks are not mandatory, and can be omitted in most cases. These callbacks are used when the hardware buffer - cannot be on the normal memory space. Some chips have their + cannot be in the normal memory space. Some chips have their own buffer on the hardware which is not mappable. In such a case, you have to transfer the data manually from the memory buffer to the hardware buffer. Or, if the buffer is @@ -3018,8 +3019,8 @@ struct _snd_pcm_runtime { page callback - This callback is also not mandatory. This callback is used - mainly for the non-contiguous buffer. The mmap calls this + This callback is optional too. This callback is used + mainly for non-contiguous buffers. The mmap calls this callback to get the page address. Some examples will be explained in the later section Buffer and Memory @@ -3035,7 +3036,7 @@ struct _snd_pcm_runtime { role of PCM interrupt handler in the sound driver is to update the buffer position and to tell the PCM middle layer when the buffer position goes across the prescribed period size. To - inform this, call snd_pcm_period_elapsed() + inform this, call the snd_pcm_period_elapsed() function. @@ -3072,7 +3073,7 @@ struct _snd_pcm_runtime { - A typical coding would be like: + Typical code would be like: Interrupt Handler Case #1 @@ -3101,21 +3102,21 @@ struct _snd_pcm_runtime {
- High-frequent timer interrupts + High frequency timer interrupts - This is the case when the hardware doesn't generate interrupts - at the period boundary but do timer-interrupts at the fixed + This happense when the hardware doesn't generate interrupts + at the period boundary but issues timer interrupts at a fixed timer rate (e.g. es1968 or ymfpci drivers). In this case, you need to check the current hardware - position and accumulates the processed sample length at each - interrupt. When the accumulated size overcomes the period + position and accumulate the processed sample length at each + interrupt. When the accumulated size exceeds the period size, call snd_pcm_period_elapsed() and reset the accumulator. - A typical coding would be like the following. + Typical code would be like the following. Interrupt Handler Case #2 @@ -3178,32 +3179,33 @@ struct _snd_pcm_runtime {
Atomicity - One of the most important (and thus difficult to debug) problem - on the kernel programming is the race condition. - On linux kernel, usually it's solved via spin-locks or - semaphores. In general, if the race condition may - happen in the interrupt handler, it's handled as atomic, and you - have to use spinlock for protecting the critical session. If it - never happens in the interrupt and it may take relatively long - time, you should use semaphore. + One of the most important (and thus difficult to debug) problems + in kernel programming are race conditions. + In the Linux kernel, they are usually avoided via spin-locks, mutexes + or semaphores. In general, if a race condition can happen + in an interrupt handler, it has to be managed atomically, and you + have to use a spinlock to protect the critical session. If the + critical section is not in interrupt handler code and + if taking a relatively long time to execute is acceptable, you + should use mutexes or semaphores instead. As already seen, some pcm callbacks are atomic and some are - not. For example, hw_params callback is + not. For example, the hw_params callback is non-atomic, while trigger callback is atomic. This means, the latter is called already in a spinlock held by the PCM middle layer. Please take this atomicity into - account when you use a spinlock or a semaphore in the callbacks. + account when you choose a locking scheme in the callbacks. In the atomic callbacks, you cannot use functions which may call schedule or go to - sleep. The semaphore and mutex do sleep, + sleep. Semaphores and mutexes can sleep, and hence they cannot be used inside the atomic callbacks (e.g. trigger callback). - For taking a certain delay in such a callback, please use + To implement some delay in such a callback, please use udelay() or mdelay(). @@ -3257,7 +3259,7 @@ struct _snd_pcm_runtime { There are many different constraints. - Look in sound/pcm.h for a complete list. + Look at sound/pcm.h for a complete list. You can even define your own constraint rules. For example, let's suppose my_chip can manage a substream of 1 channel if and only if the format is S16_LE, otherwise it supports any format @@ -3346,7 +3348,7 @@ struct _snd_pcm_runtime { - I won't explain more details here, rather I + I won't give more details here, rather I would like to say, Luke, use the source.
@@ -3364,10 +3366,9 @@ struct _snd_pcm_runtime { General The control interface is used widely for many switches, - sliders, etc. which are accessed from the user-space. Its most - important use is the mixer interface. In other words, on ALSA - 0.9.x, all the mixer stuff is implemented on the control kernel - API (while there was an independent mixer kernel API on 0.5.x). + sliders, etc. which are accessed from user-space. Its most + important use is the mixer interface. In other words, since ALSA + 0.9.x, all the mixer stuff is implemented on the control kernel API. @@ -3379,14 +3380,15 @@ struct _snd_pcm_runtime { The control API is defined in <sound/control.h>. - Include this file if you add your own controls. + Include this file if you want to add your own controls.
Definition of Controls - For creating a new control, you need to define the three + To create a new control, you need to define the + following three callbacks: info, get and put. Then, define a @@ -3414,13 +3416,13 @@ struct _snd_pcm_runtime { Most likely the control is created via snd_ctl_new1(), and in such a case, you can - add __devinitdata prefix to the - definition like above. + add the __devinitdata prefix to the + definition as above. - The iface field specifies the type of - the control, SNDRV_CTL_ELEM_IFACE_XXX, which + The iface field specifies the control + type, SNDRV_CTL_ELEM_IFACE_XXX, which is usually MIXER. Use CARD for global controls that are not logically part of the mixer. @@ -3435,12 +3437,11 @@ struct _snd_pcm_runtime { The name is the name identifier - string. On ALSA 0.9.x, the control name is very important, + string. Since ALSA 0.9.x, the control name is very important, because its role is classified from its name. There are pre-defined standard control names. The details are described in - the subsection - - Control Names. + the + Control Names subsection. @@ -3456,15 +3457,15 @@ struct _snd_pcm_runtime { The access field contains the access type of this control. Give the combination of bit masks, SNDRV_CTL_ELEM_ACCESS_XXX, there. - The detailed will be explained in the subsection - - Access Flags. + The details will be explained in + the + Access Flags subsection. The private_value field contains an arbitrary long integer value for this record. When using - generic info, + the generic info, get and put callbacks, you can pass a value through this field. If several small numbers are necessary, you can @@ -3489,7 +3490,7 @@ struct _snd_pcm_runtime {
Control Names - There are some standards for defining the control names. A + There are some standards to define the control names. A control is usually defined from the three parts as SOURCE DIRECTION FUNCTION. @@ -3497,7 +3498,7 @@ struct _snd_pcm_runtime { The first, SOURCE, specifies the source of the control, and is a string such as Master, - PCM, CD or + PCM, CD and Line. There are many pre-defined sources. @@ -3575,22 +3576,22 @@ struct _snd_pcm_runtime { Access Flags - The access flag is the bit-flags which specifies the access type + The access flag is the bitmask which specifies the access type of the given control. The default access type is SNDRV_CTL_ELEM_ACCESS_READWRITE, which means both read and write are allowed to this control. When the access flag is omitted (i.e. = 0), it is - regarded as READWRITE access as default. + considered as READWRITE access as default. When the control is read-only, pass SNDRV_CTL_ELEM_ACCESS_READ instead. In this case, you don't have to define - put callback. + the put callback. Similarly, when the control is write-only (although it's a rare - case), you can use WRITE flag instead, and - you don't need get callback. + case), you can use the WRITE flag instead, and + you don't need the get callback. @@ -3598,15 +3599,15 @@ struct _snd_pcm_runtime { VOLATILE flag should be given. This means that the control may be changed without - notification. Applications should poll such + notification. Applications should poll such a control constantly. When the control is inactive, set - INACTIVE flag, too. + the INACTIVE flag, too. There are LOCK and - OWNER flags for changing the write + OWNER flags to change the write permissions. @@ -3619,10 +3620,10 @@ struct _snd_pcm_runtime { info callback The info callback is used to get - the detailed information of this control. This must store the + detailed information on this control. This must store the values of the given struct snd_ctl_elem_info object. For example, for a boolean control with a single - element will be: + element: Example of info callback @@ -3653,7 +3654,7 @@ struct _snd_pcm_runtime { volume would have count = 2. The value field is a union, and the values stored are depending on the type. The boolean and - integer are identical. + integer types are identical. @@ -3684,7 +3685,7 @@ struct _snd_pcm_runtime { - Some common info callbacks are prepared for easy use: + Some common info callbacks are available for your convenience: snd_ctl_boolean_mono_info() and snd_ctl_boolean_stereo_info(). Obviously, the former is an info callback for a mono channel @@ -3699,7 +3700,7 @@ struct _snd_pcm_runtime { This callback is used to read the current value of the - control and to return to the user-space. + control and to return to user-space. @@ -3722,11 +3723,11 @@ struct _snd_pcm_runtime { - The value field is depending on - the type of control as well as on info callback. For example, + The value field depends on + the type of control as well as on the info callback. For example, the sb driver uses this field to store the register offset, the bit-shift and the bit-mask. The - private_value is set like + private_value field is set as follows: - In get callback, you have to fill all the elements if the + In the get callback, + you have to fill all the elements if the control has more than one elements, i.e. count > 1. In the example above, we filled only one element @@ -3765,7 +3767,7 @@ struct _snd_pcm_runtime { put callback - This callback is used to write a value from the user-space. + This callback is used to write a value from user-space. @@ -3799,7 +3801,7 @@ struct _snd_pcm_runtime { - Like get callback, + As in the get callback, when the control has more than one elements, all elements must be evaluated in this callback, too. @@ -3817,7 +3819,7 @@ struct _snd_pcm_runtime { Constructor When everything is ready, finally we can create a new - control. For creating a control, there are two functions to be + control. To create a control, there are two functions to be called, snd_ctl_new1() and snd_ctl_add(). @@ -3839,14 +3841,14 @@ struct _snd_pcm_runtime { struct snd_kcontrol_new object defined above, and chip is the object pointer to be passed to kcontrol->private_data - which can be referred in callbacks. + which can be referred to in callbacks. snd_ctl_new1() allocates a new snd_kcontrol instance (that's why the definition of my_control can be with - __devinitdata + the __devinitdata prefix), and snd_ctl_add assigns the given control component to the card. @@ -3941,7 +3943,7 @@ struct _snd_pcm_runtime { General The ALSA AC97 codec layer is a well-defined one, and you don't - have to write many codes to control it. Only low-level control + have to write much code to control it. Only low-level control routines are necessary. The AC97 codec API is defined in <sound/ac97_codec.h>. @@ -4004,7 +4006,7 @@ struct _snd_pcm_runtime {
Constructor - For creating an ac97 instance, first call snd_ac97_bus + To create an ac97 instance, first call snd_ac97_bus with an ac97_bus_ops_t record with callback functions. @@ -4042,12 +4044,12 @@ struct _snd_pcm_runtime { - where chip->ac97 is the pointer of a newly created + where chip->ac97 is a pointer to a newly created ac97_t instance. In this case, the chip pointer is set as the private data, so that the read/write callback functions can refer to this chip instance. This instance is not necessarily stored in the chip - record. When you need to change the register values from the + record. If you need to change the register values from the driver, or need the suspend/resume of ac97 codecs, keep this pointer to pass to the corresponding functions. @@ -4098,7 +4100,7 @@ struct _snd_pcm_runtime { - These callbacks are non-atomic like the callbacks of control API. + These callbacks are non-atomic like the control API callbacks. @@ -4110,14 +4112,14 @@ struct _snd_pcm_runtime { The reset callback is used to reset - the codec. If the chip requires a special way of reset, you can + the codec. If the chip requires a special kind of reset, you can define this callback. - The wait callback is used for a - certain wait at the standard initialization of the codec. If the - chip requires the extra wait-time, define this callback. + The wait callback is used to + add some waiting time in the standard initialization of the codec. If the + chip requires the extra waiting time, define this callback. @@ -4172,7 +4174,7 @@ struct _snd_pcm_runtime { snd_ac97_update_bits() is used to update - some bits of the given register. + some bits in the given register. @@ -4185,7 +4187,7 @@ struct _snd_pcm_runtime { Also, there is a function to change the sample rate (of a - certain register such as + given register such as AC97_PCM_FRONT_DAC_RATE) when VRA or DRA is supported by the codec: snd_ac97_set_rate(). @@ -4200,11 +4202,11 @@ struct _snd_pcm_runtime { - The following registers are available for setting the rate: + The following registers are available to set the rate: AC97_PCM_MIC_ADC_RATE, AC97_PCM_FRONT_DAC_RATE, AC97_PCM_LR_ADC_RATE, - AC97_SPDIF. When the + AC97_SPDIF. When AC97_SPDIF is specified, the register is not really changed but the corresponding IEC958 status bits will be updated. @@ -4214,12 +4216,11 @@ struct _snd_pcm_runtime {
Clock Adjustment - On some chip, the clock of the codec isn't 48000 but using a + In some chips, the clock of the codec isn't 48000 but using a PCI clock (to save a quartz!). In this case, change the field bus->clock to the corresponding value. For example, intel8x0 - and es1968 drivers have the auto-measurement function of the - clock. + and es1968 drivers have their own function to read from the clock.
@@ -4239,15 +4240,13 @@ struct _snd_pcm_runtime { When there are several codecs on the same card, you need to call snd_ac97_mixer() multiple times with ac97.num=1 or greater. The num field - specifies the codec - number. + specifies the codec number. - If you have set up multiple codecs, you need to either write + If you set up multiple codecs, you either need to write different callbacks for each codec or check - ac97->num in the - callback routines. + ac97->num in the callback routines.
@@ -4271,7 +4270,7 @@ struct _snd_pcm_runtime { - Some soundchips have similar but a little bit different + Some soundchips have a similar but slightly different implementation of mpu401 stuff. For example, emu10k1 has its own mpu401 routines. @@ -4280,7 +4279,7 @@ struct _snd_pcm_runtime {
Constructor - For creating a rawmidi object, call + To create a rawmidi object, call snd_mpu401_uart_new(). @@ -4307,25 +4306,24 @@ struct _snd_pcm_runtime { - The 4th argument is the i/o port address. Many - backward-compatible MPU401 has an i/o port such as 0x330. Or, it - might be a part of its own PCI i/o region. It depends on the + The 4th argument is the I/O port address. Many + backward-compatible MPU401 have an I/O port such as 0x330. Or, it + might be a part of its own PCI I/O region. It depends on the chip design. - The 5th argument is bitflags for additional information. - When the i/o port address above is a part of the PCI i/o - region, the MPU401 i/o port might have been already allocated + The 5th argument is a bitflag for additional information. + When the I/O port address above is part of the PCI I/O + region, the MPU401 I/O port might have been already allocated (reserved) by the driver itself. In such a case, pass a bit flag MPU401_INFO_INTEGRATED, - and - the mpu401-uart layer will allocate the i/o ports by itself. + and the mpu401-uart layer will allocate the I/O ports by itself. When the controller supports only the input or output MIDI stream, - pass MPU401_INFO_INPUT or + pass the MPU401_INFO_INPUT or MPU401_INFO_OUTPUT bitflag, respectively. Then the rawmidi instance is created as a single stream. @@ -4333,7 +4331,7 @@ struct _snd_pcm_runtime { MPU401_INFO_MMIO bitflag is used to change the access method to MMIO (via readb and writeb) instead of - iob and outb. In this case, you have to pass the iomapped address + iob and outb. In this case, you have to pass the iomapped address to snd_mpu401_uart_new(). @@ -4341,7 +4339,7 @@ struct _snd_pcm_runtime { When MPU401_INFO_TX_IRQ is set, the output stream isn't checked in the default interrupt handler. The driver needs to call snd_mpu401_uart_interrupt_tx() - by itself to start processing the output stream in irq handler. + by itself to start processing the output stream in the irq handler. @@ -4381,7 +4379,7 @@ struct _snd_pcm_runtime { (irq_flags). Otherwise, pass the flags for irq allocation (SA_XXX bits) to it, and the irq will be - reserved by the mpu401-uart layer. If the card doesn't generates + reserved by the mpu401-uart layer. If the card doesn't generate UART interrupts, pass -1 as the irq number. Then a timer interrupt will be invoked for polling. @@ -4392,8 +4390,8 @@ struct _snd_pcm_runtime { When the interrupt is allocated in snd_mpu401_uart_new(), the private - interrupt handler is used, hence you don't have to do nothing - else than creating the mpu401 stuff. Otherwise, you have to call + interrupt handler is used, hence you don't have anything else to do + than creating the mpu401 stuff. Otherwise, you have to call snd_mpu401_uart_interrupt() explicitly when a UART interrupt is invoked and checked in your own interrupt handler. @@ -4480,8 +4478,8 @@ struct _snd_pcm_runtime { The fourth and fifth arguments are the number of output and - input substreams, respectively, of this device. (A substream is - the equivalent of a MIDI port.) + input substreams, respectively, of this device (a substream is + the equivalent of a MIDI port). @@ -4498,7 +4496,7 @@ struct _snd_pcm_runtime { After the rawmidi device is created, you need to set the operators (callbacks) for each substream. There are helper - functions to set the operators for all substream of a device: + functions to set the operators for all the substreams of a device: - If there is more than one substream, you should give each one a - unique name: + If there are more than one substream, you should give a + unique name to each of them: Callbacks - In all callbacks, the private data that you've set for the + In all the callbacks, the private data that you've set for the rawmidi device can be accessed as substream->rmidi->private_data. @@ -4583,8 +4581,8 @@ struct _snd_pcm_runtime { This is called when a substream is opened. - You can initialize the hardware here, but you should not yet - start transmitting/receiving data. + You can initialize the hardware here, but you shouldn't + start transmitting/receiving data yet.
@@ -4632,9 +4630,9 @@ struct _snd_pcm_runtime { To read data from the buffer, call snd_rawmidi_transmit_peek. It will return the number of bytes that have been read; this will be - less than the number of bytes requested when there is no more + less than the number of bytes requested when there are no more data in the buffer. - After the data has been transmitted successfully, call + After the data have been transmitted successfully, call snd_rawmidi_transmit_ack to remove the data from the substream buffer: @@ -4655,7 +4653,7 @@ struct _snd_pcm_runtime { If you know beforehand that the hardware will accept data, you can use the snd_rawmidi_transmit function - which reads some data and removes it from the buffer at once: + which reads some data and removes them from the buffer at once: This is only used with output substreams. This function should wait - until all data read from the substream buffer has been transmitted. + until all data read from the substream buffer have been transmitted. This ensures that the device can be closed and the driver unloaded without losing data. - This callback is optional. If you do not set + This callback is optional. If you do not set drain in the struct snd_rawmidi_ops structure, ALSA will simply wait for 50 milliseconds instead. @@ -4775,24 +4773,24 @@ struct _snd_pcm_runtime {
FM OPL3 - The FM OPL3 is still used on many chips (mainly for backward + The FM OPL3 is still used in many chips (mainly for backward compatibility). ALSA has a nice OPL3 FM control layer, too. The OPL3 API is defined in <sound/opl3.h>. - FM registers can be directly accessed through direct-FM API, + FM registers can be directly accessed through the direct-FM API, defined in <sound/asound_fm.h>. In ALSA native mode, FM registers are accessed through - Hardware-Dependant Device direct-FM extension API, whereas in - OSS compatible mode, FM registers can be accessed with OSS - direct-FM compatible API on /dev/dmfmX device. + the Hardware-Dependant Device direct-FM extension API, whereas in + OSS compatible mode, FM registers can be accessed with the OSS + direct-FM compatible API in /dev/dmfmX device. - For creating the OPL3 component, you have two functions to - call. The first one is a constructor for opl3_t + To create the OPL3 component, you have two functions to + call. The first one is a constructor for the opl3_t instance. @@ -4819,12 +4817,12 @@ struct _snd_pcm_runtime { When the left and right ports have been already allocated by the card driver, pass non-zero to the fifth argument - (integrated). Otherwise, opl3 module will + (integrated). Otherwise, the opl3 module will allocate the specified ports by itself. - When the accessing to the hardware requires special method + When the accessing the hardware requires special method instead of the standard I/O access, you can create opl3 instance separately with snd_opl3_new(). @@ -4845,13 +4843,13 @@ struct _snd_pcm_runtime { access function, the private data and the destructor. The l_port and r_port are not necessarily set. Only the command must be set properly. You can retrieve the data - from opl3->private_data field. + from the opl3->private_data field. After creating the opl3 instance via snd_opl3_new(), call snd_opl3_init() to initialize the chip to the - proper state. Note that snd_opl3_create() always + proper state. Note that snd_opl3_create() always calls it internally. @@ -4884,7 +4882,7 @@ struct _snd_pcm_runtime {
Hardware-Dependent Devices - Some chips need the access from the user-space for special + Some chips need user-space access for special controls or for loading the micro code. In such a case, you can create a hwdep (hardware-dependent) device. The hwdep API is defined in <sound/hwdep.h>. You can @@ -4893,7 +4891,7 @@ struct _snd_pcm_runtime { - Creation of the hwdep instance is done via + The creation of the hwdep instance is done via snd_hwdep_new(). @@ -4912,8 +4910,8 @@ struct _snd_pcm_runtime { You can then pass any pointer value to the private_data. If you assign a private data, you should define the - destructor, too. The destructor function is set to - private_free field. + destructor, too. The destructor function is set in + the private_free field. @@ -4925,7 +4923,7 @@ struct _snd_pcm_runtime { - and the implementation of destructor would be: + and the implementation of the destructor would be: @@ -4943,7 +4941,7 @@ struct _snd_pcm_runtime { The arbitrary file operations can be defined for this instance. The file operators are defined in - ops table. For example, assume that + the ops table. For example, assume that this chip needs an ioctl. @@ -4964,7 +4962,7 @@ struct _snd_pcm_runtime { IEC958 (S/PDIF) Usually the controls for IEC958 devices are implemented via - control interface. There is a macro to compose a name string for + the control interface. There is a macro to compose a name string for IEC958 controls, SNDRV_CTL_NAME_IEC958() defined in <include/asound.h>. @@ -4973,7 +4971,7 @@ struct _snd_pcm_runtime { There are some standard controls for IEC958 status bits. These controls use the type SNDRV_CTL_ELEM_TYPE_IEC958, and the size of element is fixed as 4 bytes array - (value.iec958.status[x]). For info + (value.iec958.status[x]). For the info callback, you don't specify the value field for this type (the count field must be set, though). @@ -5001,7 +4999,7 @@ struct _snd_pcm_runtime { enable/disable or to set the raw bit mode. The implementation will depend on the chip, but the control should be named as IEC958 xxx, preferably using - SNDRV_CTL_NAME_IEC958() macro. + the SNDRV_CTL_NAME_IEC958() macro. @@ -5036,12 +5034,12 @@ struct _snd_pcm_runtime { The allocation of pages with fallback is snd_malloc_xxx_pages_fallback(). This function tries to allocate the specified pages but if the pages - are not available, it tries to reduce the page sizes until the + are not available, it tries to reduce the page sizes until enough space is found. - For releasing the space, call + The release the pages, call snd_free_xxx_pages() function. @@ -5050,8 +5048,8 @@ struct _snd_pcm_runtime { a large contiguous physical space at the time the module is loaded for the later use. This is called pre-allocation. - As already written, you can call the following function at the - construction of pcm instance (in the case of PCI bus). + As already written, you can call the following function at + pcm instance construction time (in the case of PCI bus). @@ -5063,34 +5061,34 @@ struct _snd_pcm_runtime { where size is the byte size to be - pre-allocated and the max is the maximal - size to be changed via prealloc proc file. - The allocator will try to get as large area as possible + pre-allocated and the max is the maximum + size to be changed via the prealloc proc file. + The allocator will try to get an area as large as possible within the given size. The second argument (type) and the third argument (device pointer) are dependent on the bus. - In the case of ISA bus, pass snd_dma_isa_data() + In the case of the ISA bus, pass snd_dma_isa_data() as the third argument with SNDRV_DMA_TYPE_DEV type. For the continuous buffer unrelated to the bus can be pre-allocated with SNDRV_DMA_TYPE_CONTINUOUS type and the snd_dma_continuous_data(GFP_KERNEL) device pointer, - whereh GFP_KERNEL is the kernel allocation flag to + where GFP_KERNEL is the kernel allocation flag to use. For the SBUS, SNDRV_DMA_TYPE_SBUS and snd_dma_sbus_data(sbus_dev) are used instead. For the PCI scatter-gather buffers, use SNDRV_DMA_TYPE_DEV_SG with snd_dma_pci_data(pci) - (see the section + (see the Non-Contiguous Buffers - ). + section). - Once when the buffer is pre-allocated, you can use the - allocator in the hw_params callback + Once the buffer is pre-allocated, you can use the + allocator in the hw_params callback: @@ -5116,8 +5114,8 @@ struct _snd_pcm_runtime { - The first case works fine if the external hardware buffer is enough - large. This method doesn't need any extra buffers and thus is + The first case works fine if the external hardware buffer is large + enough. This method doesn't need any extra buffers and thus is more effective. You need to define the copy and silence callbacks for @@ -5127,25 +5125,25 @@ struct _snd_pcm_runtime { - The second case allows the mmap of the buffer, although you have - to handle an interrupt or a tasklet for transferring the data + The second case allows for mmap on the buffer, although you have + to handle an interrupt or a tasklet to transfer the data from the intermediate buffer to the hardware buffer. You can find an - example in vxpocket driver. + example in the vxpocket driver. - Another case is that the chip uses a PCI memory-map + Another case is when the chip uses a PCI memory-map region for the buffer instead of the host memory. In this case, - mmap is available only on certain architectures like intel. In - non-mmap mode, the data cannot be transferred as the normal - way. Thus you need to define copy and - silence callbacks as well + mmap is available only on certain architectures like the Intel one. + In non-mmap mode, the data cannot be transferred as in the normal + way. Thus you need to define the copy and + silence callbacks as well, as in the cases above. The examples are found in rme32.c and rme96.c. - The implementation of copy and + The implementation of the copy and silence callbacks depends upon whether the hardware supports interleaved or non-interleaved samples. The copy callback is @@ -5184,8 +5182,8 @@ struct _snd_pcm_runtime { What you have to do in this callback is again different - between playback and capture directions. In the case of - playback, you do: copy the given amount of data + between playback and capture directions. In the + playback case, you copy the given amount of data (count) at the specified pointer (src) to the specified offset (pos) on the hardware buffer. When @@ -5202,7 +5200,7 @@ struct _snd_pcm_runtime { - For the capture direction, you do: copy the given amount of + For the capture direction, you copy the given amount of data (count) at the specified offset (pos) on the hardware buffer to the specified pointer (dst). @@ -5216,7 +5214,7 @@ struct _snd_pcm_runtime { - Note that both of the position and the data amount are given + Note that both the position and the amount of data are given in frames. @@ -5247,7 +5245,7 @@ struct _snd_pcm_runtime { - The meanings of arguments are identical with the + The meanings of arguments are the same as in the copy callback, although there is no src/dst argument. In the case of interleaved samples, the channel @@ -5284,8 +5282,8 @@ struct _snd_pcm_runtime {
Non-Contiguous Buffers - If your hardware supports the page table like emu10k1 or the - buffer descriptors like via82xx, you can use the scatter-gather + If your hardware supports the page table as in emu10k1 or the + buffer descriptors as in via82xx, you can use the scatter-gather (SG) DMA. ALSA provides an interface for handling SG-buffers. The API is provided in <sound/pcm.h>. @@ -5296,7 +5294,7 @@ struct _snd_pcm_runtime { snd_pcm_lib_preallocate_pages_for_all() with SNDRV_DMA_TYPE_DEV_SG in the PCM constructor like other PCI pre-allocator. - You need to pass the snd_dma_pci_data(pci), + You need to pass snd_dma_pci_data(pci), where pci is the struct pci_dev pointer of the chip as well. The struct snd_sg_buf instance is created as @@ -5314,7 +5312,7 @@ struct _snd_pcm_runtime { Then call snd_pcm_lib_malloc_pages() - in hw_params callback + in the hw_params callback as well as in the case of normal PCI buffer. The SG-buffer handler will allocate the non-contiguous kernel pages of the given size and map them onto the virtually contiguous @@ -5335,7 +5333,7 @@ struct _snd_pcm_runtime { - For releasing the data, call + To release the data, call snd_pcm_lib_free_pages() in the hw_free callback as usual. @@ -5390,7 +5388,7 @@ struct _snd_pcm_runtime { - For creating a proc file, call + To create a proc file, call snd_card_proc_new(). @@ -5402,7 +5400,7 @@ struct _snd_pcm_runtime { - where the second argument specifies the proc-file name to be + where the second argument specifies the name of the proc file to be created. The above example will create a file my-file under the card directory, e.g. /proc/asound/card0/my-file. @@ -5417,8 +5415,8 @@ struct _snd_pcm_runtime { When the creation is successful, the function stores a new - instance at the pointer given in the third argument. - It is initialized as a text proc file for read only. For using + instance in the pointer given in the third argument. + It is initialized as a text proc file for read only. To use this proc file as a read-only text file as it is, set the read callback with a private data via snd_info_set_text_ops(). @@ -5470,9 +5468,9 @@ struct _snd_pcm_runtime { - The file permission can be changed afterwards. As default, it's - set as read only for all users. If you want to add the write - permission to the user (root as default), set like below: + The file permissions can be changed afterwards. As default, it's + set as read only for all users. If you want to add write + permission for the user (root as default), do as follows: @@ -5503,7 +5501,7 @@ struct _snd_pcm_runtime { - For a raw-data proc-file, set the attributes like the following: + For a raw-data proc-file, set the attributes as follows: @@ -5524,7 +5522,7 @@ struct _snd_pcm_runtime { The callback is much more complicated than the text-file - version. You need to use a low-level i/o functions such as + version. You need to use a low-level I/O functions such as copy_from/to_user() to transfer the data. @@ -5560,28 +5558,28 @@ struct _snd_pcm_runtime { Power Management If the chip is supposed to work with suspend/resume - functions, you need to add the power-management codes to the - driver. The additional codes for the power-management should be + functions, you need to add power-management code to the + driver. The additional code for power-management should be ifdef'ed with CONFIG_PM. - If the driver supports the suspend/resume - fully, that is, the device can be - properly resumed to the status at the suspend is called, - you can set SNDRV_PCM_INFO_RESUME flag - to pcm info field. Usually, this is possible when the - registers of ths chip can be safely saved and restored to the - RAM. If this is set, the trigger callback is called with - SNDRV_PCM_TRIGGER_RESUME after resume - callback is finished. + If the driver fully supports suspend/resume + that is, the device can be + properly resumed to its state when suspend was called, + you can set the SNDRV_PCM_INFO_RESUME flag + in the pcm info field. Usually, this is possible when the + registers of the chip can be safely saved and restored to + RAM. If this is set, the trigger callback is called with + SNDRV_PCM_TRIGGER_RESUME after the resume + callback completes. - Even if the driver doesn't support PM fully but only the - partial suspend/resume is possible, it's still worthy to - implement suspend/resume callbacks. In such a case, applications + Even if the driver doesn't support PM fully but + partial suspend/resume is still possible, it's still worthy to + implement suspend/resume callbacks. In such a case, applications would reset the status by calling snd_pcm_prepare() and restart the stream appropriately. Hence, you can define suspend/resume callbacks @@ -5590,22 +5588,22 @@ struct _snd_pcm_runtime { - Note that the trigger with SUSPEND can be always called when + Note that the trigger with SUSPEND can always be called when snd_pcm_suspend_all is called, - regardless of SNDRV_PCM_INFO_RESUME flag. + regardless of the SNDRV_PCM_INFO_RESUME flag. The RESUME flag affects only the behavior of snd_pcm_resume(). (Thus, in theory, SNDRV_PCM_TRIGGER_RESUME isn't needed to be handled in the trigger callback when no SNDRV_PCM_INFO_RESUME flag is set. But, - it's better to keep it for compatibility reason.) + it's better to keep it for compatibility reasons.) In the earlier version of ALSA drivers, a common power-management layer was provided, but it has been removed. The driver needs to define the suspend/resume hooks according to - the bus the device is assigned. In the case of PCI driver, the + the bus the device is connected to. In the case of PCI drivers, the callbacks look like below: @@ -5629,7 +5627,7 @@ struct _snd_pcm_runtime { - The scheme of the real suspend job is as following. + The scheme of the real suspend job is as follows. Retrieve the card and the chip data. @@ -5679,11 +5677,11 @@ struct _snd_pcm_runtime { - The scheme of the real resume job is as following. + The scheme of the real resume job is as follows. Retrieve the card and the chip data. - Set up PCI. First, call pci_restore_state(). + Set up PCI. First, call pci_restore_state(). Then enable the pci device again by calling pci_enable_device(). Call pci_set_master() if necessary, too. Re-initialize the chip. @@ -5734,7 +5732,7 @@ struct _snd_pcm_runtime { snd_pcm_suspend_all() or snd_pcm_suspend(). It means that the PCM streams are already stoppped when the register snapshot is - taken. But, remind that you don't have to restart the PCM + taken. But, remember that you don't have to restart the PCM stream in the resume callback. It'll be restarted via trigger call with SNDRV_PCM_TRIGGER_RESUME when necessary. @@ -5795,7 +5793,7 @@ struct _snd_pcm_runtime { - If you need a space for saving the registers, allocate the + If you need a space to save the registers, allocate the buffer for it here, too, since it would be fatal if you cannot allocate a memory in the suspend phase. The allocated buffer should be released in the corresponding @@ -5833,7 +5831,7 @@ struct _snd_pcm_runtime { Module Parameters There are standard module options for ALSA. At least, each - module should have index, + module should have the index, id and enable options. @@ -5841,8 +5839,8 @@ struct _snd_pcm_runtime { If the module supports multiple cards (usually up to 8 = SNDRV_CARDS cards), they should be - arrays. The default initial values are defined already as - constants for ease of programming: + arrays. The default initial values are defined already as + constants for easier programming: @@ -5858,7 +5856,7 @@ struct _snd_pcm_runtime { If the module supports only a single card, they could be single variables, instead. enable option is not - always necessary in this case, but it wouldn't be so bad to have a + always necessary in this case, but it would be better to have a dummy option for compatibility. @@ -5923,22 +5921,22 @@ struct _snd_pcm_runtime { - Suppose that you'll create a new PCI driver for the card + Suppose that you create a new PCI driver for the card xyz. The card module name would be - snd-xyz. The new driver is usually put into alsa-driver + snd-xyz. The new driver is usually put into the alsa-driver tree, alsa-driver/pci directory in the case of PCI cards. Then the driver is evaluated, audited and tested by developers and users. After a certain time, the driver - will go to alsa-kernel tree (to the corresponding directory, + will go to the alsa-kernel tree (to the corresponding directory, such as alsa-kernel/pci) and eventually - integrated into Linux 2.6 tree (the directory would be + will be integrated into the Linux 2.6 tree (the directory would be linux/sound/pci). In the following sections, the driver code is supposed - to be put into alsa-driver tree. The two cases are assumed: + to be put into alsa-driver tree. The two cases are covered: a driver consisting of a single source file and one consisting of several source files. @@ -6033,7 +6031,7 @@ struct _snd_pcm_runtime { Add a new directory (xyz) in - alsa-driver/pci/Makefile like below + alsa-driver/pci/Makefile as below @@ -6102,7 +6100,7 @@ struct _snd_pcm_runtime {
<function>snd_printk()</function> and friends - ALSA provides a verbose version of + ALSA provides a verbose version of the printk() function. If a kernel config CONFIG_SND_VERBOSE_PRINTK is set, this function prints the given message together with the file name @@ -6170,7 +6168,7 @@ struct _snd_pcm_runtime {
<function>snd_BUG()</function> - It shows BUG? message and + It shows the BUG? message and stack trace as well as snd_assert at the point. It's useful to show that a fatal error happens there. @@ -6199,6 +6197,4 @@ struct _snd_pcm_runtime { in the hardware constraints section. - - -- cgit v0.10.2 From 27ed7ceda58d147dea27cd7443915f140c2665a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Oct 2007 18:18:11 +0200 Subject: [ALSA] ad1848 - Fix print format Fixed the print format for debug message. Spotted by Matthew Wilcox. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index a901cd1..9a64035 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -213,7 +213,7 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) for (timeout = 12000; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) udelay(100); - snd_printdd("(1) timeout = %d\n", timeout); + snd_printdd("(1) timeout = %ld\n", timeout); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) -- cgit v0.10.2 From 9674513d60e7f3bdde1e4739ed81b2daaee74df7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 25 Oct 2007 11:46:24 +0200 Subject: [ALSA] via82xx - Fix quirk for Shuttle AK32VN Fix quirk for Shuttle AK32VN. It works better with DXS_SRC, and needs HP_ONLY ac97 quirk. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index cf62d2a..ad58ca5 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1793,6 +1793,12 @@ static struct ac97_quirk ac97_quirks[] = { .name = "m680x", .type = AC97_TUNE_HP_ONLY, /* http://launchpad.net/bugs/38546 */ }, + { + .subvendor = 0x1297, + .subdevice = 0xa232, + .name = "Shuttle AK32VN", + .type = AC97_TUNE_HP_ONLY + }, { } /* terminator */ }; @@ -2364,8 +2370,8 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x10cf, 0x118e, "FSC Laptop", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1106, 0, "ASRock", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xa231, "Shuttle AK31v2", VIA_DXS_SRC), - SND_PCI_QUIRK(0x1297, 0xa232, "Shuttle", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1297, 0xc160, "Shuttle Sk41G", VIA_DXS_ENABLE), + SND_PCI_QUIRK(0x1297, 0xa232, "Shuttle", VIA_DXS_SRC), + SND_PCI_QUIRK(0x1297, 0xc160, "Shuttle Sk41G", VIA_DXS_SRC), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte GA-7VAXP", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1462, 0x3800, "MSI KT266", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1462, 0x7120, "MSI KT4V", VIA_DXS_ENABLE), -- cgit v0.10.2 From f651b50b9d1ab44f7b09d1ef28ba702903732fd3 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Fri, 26 Oct 2007 12:40:47 +0200 Subject: [ALSA] HDA: Add support for Samsung Q1 Ultra Vista edition This patch adds full record and playback support for the Samsung Q1 Ultra - Vista model (different codec than the earlier Q1 Ultra models). Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 69ab91c..8e49c19 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -828,6 +828,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. hippo Hippo (ATI) with jack detection, Sony UX-90s hippo_1 Hippo (Benq) with jack detection sony-assamd Sony ASSAMD + ultra Samsung Q1 Ultra Vista model basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b78f6fc..8de41c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -95,6 +95,7 @@ enum { ALC262_BENQ_ED8, ALC262_SONY_ASSAMD, ALC262_BENQ_T31, + ALC262_ULTRA, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -8082,6 +8083,72 @@ static struct hda_verb alc262_benq_t31_EAPD_verbs[] = { {} }; +/* Samsung Q1 Ultra Vista model setup */ +static struct snd_kcontrol_new alc262_ultra_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_ultra_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Mic is on Node 0x19 */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x24, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {} +}; + +static struct hda_input_mux alc262_ultra_capture_source = { + .num_items = 1, + .items = { + { "Mic", 0x1 }, + }, +}; + +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_ultra_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + unsigned int present; + + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + if (spec->jack_present) { + /* mute internal speaker */ + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +/* unsolicited event for HP jack sensing */ +static void alc262_ultra_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_ultra_automute(codec); +} + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -8487,6 +8554,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", [ALC262_SONY_ASSAMD] = "sony-assamd", + [ALC262_ULTRA] = "ultra", [ALC262_AUTO] = "auto", }; @@ -8517,6 +8585,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), {} }; @@ -8635,6 +8704,19 @@ static struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, }, + [ALC262_ULTRA] = { + .mixers = { alc262_ultra_mixer }, + .init_verbs = { alc262_init_verbs, alc262_ultra_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_ultra_capture_source, + .unsol_event = alc262_ultra_unsol_event, + .init_hook = alc262_ultra_automute, + }, }; static int patch_alc262(struct hda_codec *codec) -- cgit v0.10.2 From 040956fabbc16b9ce746a03d2b589052e1771138 Mon Sep 17 00:00:00 2001 From: Ivan Kuten Date: Fri, 26 Oct 2007 14:53:47 +0200 Subject: [ALSA] soc - ln2440sbc ac97 support This patch adds ac97 support for ln2440sbc board from LittleChips. This board is based on s3c2440 SoC + AC97 Realtek ALC650 codec. Existing s3c2443 implementation is slightly modified because s3c2440 and s3c2443 have different AC97 interrupts. Signed-off-by: Ivan Kuten Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 5632a2e..14def2e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -34,4 +34,12 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710 Say Y if you want to add support for SoC audio on smdk2443 with the WM9710. +config SND_S3C24XX_SOC_LN2440SBC_ALC650 + tristate "SoC AC97 Audio support for LN2440SBC - ALC650" + depends on SND_S3C24XX_SOC + select SND_S3C2443_SOC_AC97 + select SND_SOC_AC97_CODEC + help + Say Y if you want to add support for SoC audio on ln2440sbc + with the ALC650. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 13c92f0..9471919 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -10,6 +10,8 @@ obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o # S3C24XX Machine Support snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o +snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o +obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c new file mode 100644 index 0000000..ec0d1a2 --- /dev/null +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -0,0 +1,86 @@ +/* + * SoC audio for ln2440sbc + * + * Copyright 2007 KonekTel, a.s. + * Author: Ivan Kuten + * ivan.kuten@promwad.com + * + * Heavily based on smdk2443_wm9710.c + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/ac97.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-ac97.h" + +static struct snd_soc_machine ln2440sbc; + +static struct snd_soc_dai_link ln2440sbc_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c2443_ac97_dai[0], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_machine ln2440sbc = { + .name = "LN2440SBC", + .dai_link = ln2440sbc_dai, + .num_links = ARRAY_SIZE(ln2440sbc_dai), +}; + +static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { + .machine = &ln2440sbc, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *ln2440sbc_snd_ac97_device; + +static int __init ln2440sbc_init(void) +{ + int ret; + + ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!ln2440sbc_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(ln2440sbc_snd_ac97_device, + &ln2440sbc_snd_ac97_devdata); + ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; + ret = platform_device_add(ln2440sbc_snd_ac97_device); + + if (ret) + platform_device_put(ln2440sbc_snd_ac97_device); + + return ret; +} + +static void __exit ln2440sbc_exit(void) +{ + platform_device_unregister(ln2440sbc_snd_ac97_device); +} + +module_init(ln2440sbc_init); +module_exit(ln2440sbc_exit); + +/* Module information */ +MODULE_AUTHOR("Ivan Kuten"); +MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 758a263..96605b7 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -253,7 +253,7 @@ static int s3c2443_ac97_probe(struct platform_device *pdev) ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ret = request_irq(IRQ_S3C2443_AC97, s3c2443_ac97_irq, + ret = request_irq(IRQ_S3C244x_AC97, s3c2443_ac97_irq, IRQF_DISABLED, "AC97", NULL); if (ret < 0) { printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n"); @@ -266,7 +266,7 @@ static int s3c2443_ac97_probe(struct platform_device *pdev) static void s3c2443_ac97_remove(struct platform_device *pdev) { - free_irq(IRQ_S3C2443_AC97, NULL); + free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); clk_put(s3c24xx_ac97.ac97_clk); iounmap(s3c24xx_ac97.regs); diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h index 2b835e8..bf03e8e 100644 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ b/sound/soc/s3c24xx/s3c24xx-ac97.h @@ -20,6 +20,12 @@ #define AC_CMD_ADDR(x) (x << 16) #define AC_CMD_DATA(x) (x & 0xffff) +#ifdef CONFIG_CPU_S3C2440 +#define IRQ_S3C244x_AC97 IRQ_S3C2440_AC97 +#else +#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 +#endif + extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/ -- cgit v0.10.2 From e97a516701cfc3c4b27444212208884214d10c20 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 26 Oct 2007 14:56:36 +0200 Subject: [ALSA] hda-codec - Show more information in proc file Show the current EAPD status and volume-knob status in proc file. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e94944f..000c6c4 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -213,7 +213,7 @@ static void print_pin_caps(struct snd_info_buffer *buffer, "SPDIF In", "Digitial In", "Reserved", "Other" }; static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; - unsigned int caps; + unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); snd_iprintf(buffer, " Pincap 0x08%x:", caps); @@ -237,6 +237,11 @@ static void print_pin_caps(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps), get_jack_color(caps)); + if (caps & AC_PINCAP_EAPD) { + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_EAPD_BTLENABLE, 0); + snd_iprintf(buffer, " EAPD: 0x%x\n", val); + } } @@ -284,6 +289,7 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; int conn_len = 0; hda_nid_t conn[HDA_MAX_CONNECTIONS]; + unsigned int pinctls; snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, get_wid_type_name(wid_type), wid_caps); @@ -318,8 +324,8 @@ static void print_codec_info(struct snd_info_entry *entry, wid_caps & AC_WCAP_STEREO, 1); } - if (wid_type == AC_WID_PIN) { - unsigned int pinctls; + switch (wid_type) { + case AC_WID_PIN: print_pin_caps(buffer, codec, nid); pinctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, @@ -332,12 +338,19 @@ static void print_codec_info(struct snd_info_entry *entry, if (pinctls & AC_PINCTL_HP_EN) snd_iprintf(buffer, " HP"); snd_iprintf(buffer, "\n"); - } - - if ((wid_type == AC_WID_AUD_OUT || wid_type == AC_WID_AUD_IN) && - (wid_caps & AC_WCAP_FORMAT_OVRD)) { - snd_iprintf(buffer, " PCM:\n"); - print_pcm_caps(buffer, codec, nid); + break; + case AC_WID_VOL_KNB: + snd_iprintf(buffer, " Volume-Knob: 0x%x\n", + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_VOLUME_KNOB_CONTROL, 0)); + break; + case AC_WID_AUD_OUT: + case AC_WID_AUD_IN: + if (wid_caps & AC_WCAP_FORMAT_OVRD) { + snd_iprintf(buffer, " PCM:\n"); + print_pcm_caps(buffer, codec, nid); + } + break; } if (wid_caps & AC_WCAP_POWER) -- cgit v0.10.2 From 304cd07f92027af7e92d742d9554c3be9fa4d4cf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 26 Oct 2007 15:10:15 +0200 Subject: [ALSA] Introduce slots option to snd module Introduced the global 'slots' option to snd module. This option provides an alternative way to handle the order of multiple sound card instances. It's an easier approach to avoid conflict with hotplug devices, and can be used together with the existing 'order' option of each card driver. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8e49c19..f34821d 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -57,7 +57,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. - Default: 1 - For auto-loading more than one card, specify this option together with snd-card-X aliases. - + slots - Reserve the slot index for the given driver. + This option takes multiple strings. + See "Module Autoloading Support" section for details. Module snd-pcm-oss ------------------ @@ -2140,6 +2142,23 @@ alias sound-slot-1 snd-ens1371 In this example, the interwave card is always loaded as the first card (index 0) and ens1371 as the second (index 1). +Alternative (and new) way to fixate the slot assignment is to use +"slots" option of snd module. In the case above, specify like the +following: + +options snd slots=snd-interwave,snd-ens1371 + +Then, the first slot (#0) is reserved for snd-interwave driver, and +the second (#1) for snd-ens1371. You can omit index option in each +driver if slots option is used (although you can still have them at +the same time as long as they don't conflict). + +The slots option is especially useful for avoiding the possible +hot-plugging and the resultant slot conflict. For example, in the +case above again, the first two slots are already reserved. If any +other driver (e.g. snd-usb-audio) is loaded before snd-interwave or +snd-ens1371, it will be assigned to the third or later slot. + ALSA PCM devices to OSS devices mapping ======================================= diff --git a/sound/core/init.c b/sound/core/init.c index 2cb7099..48d38a7 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -43,6 +43,40 @@ EXPORT_SYMBOL(snd_cards); static DEFINE_MUTEX(snd_card_mutex); +static char *slots[SNDRV_CARDS]; +module_param_array(slots, charp, NULL, 0444); +MODULE_PARM_DESC(slots, "Module names assigned to the slots."); + +/* return non-zero if the given index is already reserved for another + * module via slots option + */ +static int module_slot_mismatch(struct module *module, int idx) +{ +#ifdef MODULE + char *s1, *s2; + if (!module || !module->name || !slots[idx]) + return 0; + s1 = slots[idx]; + s2 = module->name; + /* compare module name strings + * hyphens are handled as equivalent with underscore + */ + for (;;) { + char c1 = *s1++; + char c2 = *s2++; + if (c1 == '-') + c1 = '_'; + if (c2 == '-') + c2 = '_'; + if (c1 != c2) + return 1; + if (!c1) + break; + } +#endif + return 0; +} + #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag); EXPORT_SYMBOL(snd_mixer_oss_notify_callback); @@ -115,6 +149,8 @@ struct snd_card *snd_card_new(int idx, const char *xid, for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) /* idx == -1 == 0xffff means: take any free slot */ if (~snd_cards_lock & idx & 1<= snd_ecards_limit) snd_ecards_limit = idx + 1; -- cgit v0.10.2 From 54720a2481e67e622d5ee40a5130fd45f5c57c14 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 29 Oct 2007 10:48:40 +0100 Subject: [ALSA] ac97_patch: compilation warning fix This patch kills these two compilation warnings if power management is disabled: sound/pci/ac97/ac97_patch.h:86: warning: 'snd_ac97_restore_status' declared 'static' but never defined sound/pci/ac97/ac97_patch.h:87: warning: 'snd_ac97_restore_iec958' declared 'static' but never defined Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h index 9cccc27..47bf8df 100644 --- a/sound/pci/ac97/ac97_patch.h +++ b/sound/pci/ac97/ac97_patch.h @@ -83,8 +83,10 @@ static int snd_ac97_swap_ctl(struct snd_ac97 *ac97, const char *s1, const char *s2, const char *suffix); static void snd_ac97_rename_vol_ctl(struct snd_ac97 *ac97, const char *src, const char *dst); +#ifdef CONFIG_PM static void snd_ac97_restore_status(struct snd_ac97 *ac97); static void snd_ac97_restore_iec958(struct snd_ac97 *ac97); +#endif static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol, -- cgit v0.10.2 From ceac4bf34e14d9040d16b35fd97a92d6e951ccf4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Oct 2007 10:49:43 +0100 Subject: [ALSA] Dreamcast AICA sound - Get rid of annoying compiler warning This patch supresses an annoying compiler warning that the variable err may be used uninitialised. Signed-off by: Adrian McMenamin Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 88dc840..8861d2f 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -237,6 +237,7 @@ static int aica_dma_transfer(int channels, int buffer_size, struct snd_card_aica *dreamcastcard; struct snd_pcm_runtime *runtime; unsigned long flags; + err = 0; dreamcastcard = substream->pcm->private_data; period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels); -- cgit v0.10.2 From 224a033252bba46c5c8b5df625f5e781ca138f48 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Oct 2007 11:49:22 +0100 Subject: [ALSA] opl3 - Use hwdep for patch loading Use the hwdep device for loading OPL2/3 patch data instead of the messy sequencer instrument layer. Due to this change, the sbiload program should be updated, too. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/asound_fm.h b/include/sound/asound_fm.h index 8fbcab7..c2a4b96 100644 --- a/include/sound/asound_fm.h +++ b/include/sound/asound_fm.h @@ -104,6 +104,8 @@ struct snd_dm_fm_params { #define SNDRV_DM_FM_IOCTL_SET_MODE _IOW('H', 0x25, int) /* for OPL3 only */ #define SNDRV_DM_FM_IOCTL_SET_CONNECTION _IOW('H', 0x26, int) +/* SBI patch management */ +#define SNDRV_DM_FM_IOCTL_CLEAR_PATCHES _IO ('H', 0x40) #define SNDRV_DM_FM_OSS_IOCTL_RESET 0x20 #define SNDRV_DM_FM_OSS_IOCTL_PLAY_NOTE 0x21 @@ -112,4 +114,21 @@ struct snd_dm_fm_params { #define SNDRV_DM_FM_OSS_IOCTL_SET_MODE 0x24 #define SNDRV_DM_FM_OSS_IOCTL_SET_OPL 0x25 +/* + * Patch Record - fixed size for write + */ + +#define FM_KEY_SBI "SBI\032" +#define FM_KEY_2OP "2OP\032" +#define FM_KEY_4OP "4OP\032" + +struct sbi_patch { + unsigned char prog; + unsigned char bank; + char key[4]; + char name[25]; + char extension[7]; + unsigned char data[32]; +}; + #endif /* __SOUND_ASOUND_FM_H */ diff --git a/include/sound/opl3.h b/include/sound/opl3.h index 1d14b3f..7ee865d 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -63,7 +63,7 @@ #include "seq_oss_legacy.h" #endif #include "seq_device.h" -#include "ainstr_fm.h" +#include "asound_fm.h" /* * Register numbers for the global registers @@ -240,6 +240,47 @@ struct snd_opl3; /* + * Instrument record, aka "Patch" + */ + +/* FM operator */ +struct fm_operator { + unsigned char am_vib; + unsigned char ksl_level; + unsigned char attack_decay; + unsigned char sustain_release; + unsigned char wave_select; +} __attribute__((packed)); + +/* Instrument data */ +struct fm_instrument { + struct fm_operator op[4]; + unsigned char feedback_connection[2]; + unsigned char echo_delay; + unsigned char echo_atten; + unsigned char chorus_spread; + unsigned char trnsps; + unsigned char fix_dur; + unsigned char modes; + unsigned char fix_key; +}; + +/* type */ +#define FM_PATCH_OPL2 0x01 /* OPL2 2 operators FM instrument */ +#define FM_PATCH_OPL3 0x02 /* OPL3 4 operators FM instrument */ + +/* Instrument record */ +struct fm_patch { + unsigned char prog; + unsigned char bank; + unsigned char type; + struct fm_instrument inst; + char name[24]; + struct fm_patch *next; +}; + + +/* * A structure to keep track of each hardware voice */ struct snd_opl3_voice { @@ -297,8 +338,8 @@ struct snd_opl3 { struct snd_midi_channel_set * oss_chset; #endif - struct snd_seq_kinstr_ops fm_ops; - struct snd_seq_kinstr_list *ilist; +#define OPL3_PATCH_HASH_SIZE 32 + struct fm_patch *patch_table[OPL3_PATCH_HASH_SIZE]; struct snd_opl3_voice voices[MAX_OPL3_VOICES]; /* Voices (OPL3 'channel') */ int use_time; /* allocation counter */ @@ -333,8 +374,19 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, int device, int seq_device, int snd_opl3_open(struct snd_hwdep * hw, struct file *file); int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg); +long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, + loff_t *offset); int snd_opl3_release(struct snd_hwdep * hw, struct file *file); void snd_opl3_reset(struct snd_opl3 * opl3); +int snd_opl3_load_patch(struct snd_opl3 *opl3, + int prog, int bank, int type, + const char *name, + const unsigned char *ext, + const unsigned char *data); +struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, + int create_patch); +void snd_opl3_clear_patches(struct snd_opl3 *opl3); + #endif /* __SOUND_OPL3_H */ diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index a2b9ce0..a657da9 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -327,6 +327,7 @@ static int snd_opl3_free(struct snd_opl3 *opl3) snd_assert(opl3 != NULL, return -ENXIO); if (opl3->private_free) opl3->private_free(opl3); + snd_opl3_clear_patches(opl3); release_and_free_resource(opl3->res_l_port); release_and_free_resource(opl3->res_r_port); kfree(opl3); @@ -521,6 +522,7 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, /* operators - only ioctl */ hw->ops.open = snd_opl3_open; hw->ops.ioctl = snd_opl3_ioctl; + hw->ops.write = snd_opl3_write; hw->ops.release = snd_opl3_release; opl3->seq_dev_num = seq_device; diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 3557b6e..cebcb8b 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -289,8 +289,6 @@ static int snd_opl3_oss_map[MAX_OPL3_VOICES] = { void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) { struct snd_opl3 *opl3; - struct snd_seq_instr wanted; - struct snd_seq_kinstr *kinstr; int instr_4op; int voice; @@ -306,11 +304,13 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) unsigned char voice_offset; unsigned short opl3_reg; unsigned char reg_val; + unsigned char prg, bank; int key = note; unsigned char fnum, blocknum; int i; + struct fm_patch *patch; struct fm_instrument *fm; unsigned long flags; @@ -320,19 +320,17 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) snd_printk("Note on, ch %i, inst %i, note %i, vel %i\n", chan->number, chan->midi_program, note, vel); #endif - wanted.cluster = 0; - wanted.std = SNDRV_SEQ_INSTR_TYPE2_OPL2_3; /* in SYNTH mode, application takes care of voices */ /* in SEQ mode, drum voice numbers are notes on drum channel */ if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { if (chan->drum_channel) { /* percussion instruments are located in bank 128 */ - wanted.bank = 128; - wanted.prg = note; + bank = 128; + prg = note; } else { - wanted.bank = chan->gm_bank_select; - wanted.prg = chan->midi_program; + bank = chan->gm_bank_select; + prg = chan->midi_program; } } else { /* Prepare for OSS mode */ @@ -340,8 +338,8 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) return; /* OSS instruments are located in bank 127 */ - wanted.bank = 127; - wanted.prg = chan->midi_program; + bank = 127; + prg = chan->midi_program; } spin_lock_irqsave(&opl3->voice_lock, flags); @@ -353,15 +351,14 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) } __extra_prg: - kinstr = snd_seq_instr_find(opl3->ilist, &wanted, 1, 0); - if (kinstr == NULL) { + patch = snd_opl3_find_patch(opl3, prg, bank, 0); + if (!patch) { spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } - fm = KINSTR_DATA(kinstr); - - switch (fm->type) { + fm = &patch->inst; + switch (patch->type) { case FM_PATCH_OPL2: instr_4op = 0; break; @@ -371,14 +368,12 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) break; } default: - snd_seq_instr_free_use(opl3->ilist, kinstr); spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } - #ifdef DEBUG_MIDI snd_printk(" --> OPL%i instrument: %s\n", - instr_4op ? 3 : 2, kinstr->name); + instr_4op ? 3 : 2, patch->name); #endif /* in SYNTH mode, application takes care of voices */ /* in SEQ mode, allocate voice on free OPL3 channel */ @@ -569,8 +564,6 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) /* get extra pgm, but avoid possible loops */ extra_prg = (extra_prg) ? 0 : fm->modes; - snd_seq_instr_free_use(opl3->ilist, kinstr); - /* do the bookkeeping */ vp->time = opl3->use_time++; vp->note = key; @@ -601,12 +594,12 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) /* allocate extra program if specified in patch library */ if (extra_prg) { if (extra_prg > 128) { - wanted.bank = 128; + bank = 128; /* percussions start at 35 */ - wanted.prg = extra_prg - 128 + 35 - 1; + prg = extra_prg - 128 + 35 - 1; } else { - wanted.bank = 0; - wanted.prg = extra_prg - 1; + bank = 0; + prg = extra_prg - 1; } #ifdef DEBUG_MIDI snd_printk(" *** allocating extra program\n"); diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 5fd3a4c..239347f 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -195,17 +195,6 @@ static int snd_opl3_close_seq_oss(struct snd_seq_oss_arg *arg) /* load patch */ -/* offsets for SBI params */ -#define AM_VIB 0 -#define KSL_LEVEL 2 -#define ATTACK_DECAY 4 -#define SUSTAIN_RELEASE 6 -#define WAVE_SELECT 8 - -/* offset for SBI instrument */ -#define CONNECTION 10 -#define OFFSET_4OP 11 - /* from sound_config.h */ #define SBFM_MAXINSTR 256 @@ -213,112 +202,42 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, const char __user *buf, int offs, int count) { struct snd_opl3 *opl3; - int err = -EINVAL; + struct sbi_instrument sbi; + char name[32]; + int err, type; snd_assert(arg != NULL, return -ENXIO); opl3 = arg->private_data; - if ((format == FM_PATCH) || (format == OPL3_PATCH)) { - struct sbi_instrument sbi; + if (format == FM_PATCH) + type = FM_PATCH_OPL2; + else if (format == OPL3_PATCH) + type = FM_PATCH_OPL3; + else + return -EINVAL; - size_t size; - struct snd_seq_instr_header *put; - struct snd_seq_instr_data *data; - struct fm_xinstrument *xinstr; + if (count < (int)sizeof(sbi)) { + snd_printk("FM Error: Patch record too short\n"); + return -EINVAL; + } + if (copy_from_user(&sbi, buf, sizeof(sbi))) + return -EFAULT; - struct snd_seq_event ev; - int i; + if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { + snd_printk("FM Error: Invalid instrument number %d\n", + sbi.channel); + return -EINVAL; + } - mm_segment_t fs; + memset(name, 0, sizeof(name)); + sprintf(name, "Chan%d", sbi.channel); - if (count < (int)sizeof(sbi)) { - snd_printk("FM Error: Patch record too short\n"); - return -EINVAL; - } - if (copy_from_user(&sbi, buf, sizeof(sbi))) - return -EFAULT; + err = snd_opl3_load_patch(opl3, sbi.channel, 127, type, name, NULL, + sbi.operators); + if (err < 0) + return err; - if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { - snd_printk("FM Error: Invalid instrument number %d\n", sbi.channel); - return -EINVAL; - } - - size = sizeof(*put) + sizeof(struct fm_xinstrument); - put = kzalloc(size, GFP_KERNEL); - if (put == NULL) - return -ENOMEM; - /* build header */ - data = &put->data; - data->type = SNDRV_SEQ_INSTR_ATYPE_DATA; - strcpy(data->data.format, SNDRV_SEQ_INSTR_ID_OPL2_3); - /* build data section */ - xinstr = (struct fm_xinstrument *)(data + 1); - xinstr->stype = FM_STRU_INSTR; - - for (i = 0; i < 2; i++) { - xinstr->op[i].am_vib = sbi.operators[AM_VIB + i]; - xinstr->op[i].ksl_level = sbi.operators[KSL_LEVEL + i]; - xinstr->op[i].attack_decay = sbi.operators[ATTACK_DECAY + i]; - xinstr->op[i].sustain_release = sbi.operators[SUSTAIN_RELEASE + i]; - xinstr->op[i].wave_select = sbi.operators[WAVE_SELECT + i]; - } - xinstr->feedback_connection[0] = sbi.operators[CONNECTION]; - - if (format == OPL3_PATCH) { - xinstr->type = FM_PATCH_OPL3; - for (i = 0; i < 2; i++) { - xinstr->op[i+2].am_vib = sbi.operators[OFFSET_4OP + AM_VIB + i]; - xinstr->op[i+2].ksl_level = sbi.operators[OFFSET_4OP + KSL_LEVEL + i]; - xinstr->op[i+2].attack_decay = sbi.operators[OFFSET_4OP + ATTACK_DECAY + i]; - xinstr->op[i+2].sustain_release = sbi.operators[OFFSET_4OP + SUSTAIN_RELEASE + i]; - xinstr->op[i+2].wave_select = sbi.operators[OFFSET_4OP + WAVE_SELECT + i]; - } - xinstr->feedback_connection[1] = sbi.operators[OFFSET_4OP + CONNECTION]; - } else { - xinstr->type = FM_PATCH_OPL2; - } - - put->id.instr.std = SNDRV_SEQ_INSTR_TYPE2_OPL2_3; - put->id.instr.bank = 127; - put->id.instr.prg = sbi.channel; - put->cmd = SNDRV_SEQ_INSTR_PUT_CMD_CREATE; - - memset (&ev, 0, sizeof(ev)); - ev.source.client = SNDRV_SEQ_CLIENT_OSS; - ev.dest = arg->addr; - - ev.flags = SNDRV_SEQ_EVENT_LENGTH_VARUSR; - ev.queue = SNDRV_SEQ_QUEUE_DIRECT; - - fs = snd_enter_user(); - __again: - ev.type = SNDRV_SEQ_EVENT_INSTR_PUT; - ev.data.ext.len = size; - ev.data.ext.ptr = put; - - err = snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, &ev, - opl3->seq_client, 0, 0); - if (err == -EBUSY) { - struct snd_seq_instr_header remove; - - memset (&remove, 0, sizeof(remove)); - remove.cmd = SNDRV_SEQ_INSTR_FREE_CMD_SINGLE; - remove.id.instr = put->id.instr; - - /* remove instrument */ - ev.type = SNDRV_SEQ_EVENT_INSTR_FREE; - ev.data.ext.len = sizeof(remove); - ev.data.ext.ptr = &remove; - - snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, &ev, - opl3->seq_client, 0, 0); - goto __again; - } - snd_leave_user(fs); - - kfree(put); - } - return err; + return sizeof(sbi); } /* ioctl */ diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index 96762c9..ff6da16 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -152,15 +152,7 @@ static int snd_opl3_synth_event_input(struct snd_seq_event * ev, int direct, { struct snd_opl3 *opl3 = private_data; - if (ev->type >= SNDRV_SEQ_EVENT_INSTR_BEGIN && - ev->type <= SNDRV_SEQ_EVENT_INSTR_CHANGE) { - if (direct) { - snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, ev, - opl3->seq_client, atomic, hop); - } - } else { - snd_midi_process_event(&opl3_ops, ev, opl3->chset); - } + snd_midi_process_event(&opl3_ops, ev, opl3->chset); return 0; } @@ -249,16 +241,6 @@ static int snd_opl3_seq_new_device(struct snd_seq_device *dev) return err; } - /* initialize instrument list */ - opl3->ilist = snd_seq_instr_list_new(); - if (opl3->ilist == NULL) { - snd_seq_delete_kernel_client(client); - opl3->seq_client = -1; - return -ENOMEM; - } - opl3->ilist->flags = SNDRV_SEQ_INSTR_FLG_DIRECT; - snd_seq_fm_init(&opl3->fm_ops, NULL); - /* setup system timer */ init_timer(&opl3->tlist); opl3->tlist.function = snd_opl3_timer_func; @@ -287,8 +269,6 @@ static int snd_opl3_seq_delete_device(struct snd_seq_device *dev) snd_seq_delete_kernel_client(opl3->seq_client); opl3->seq_client = -1; } - if (opl3->ilist) - snd_seq_instr_list_free(&opl3->ilist); return 0; } diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index a4b3543..d55eefc 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -165,6 +165,10 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, #endif return snd_opl3_set_connection(opl3, (int) arg); + case SNDRV_DM_FM_IOCTL_CLEAR_PATCHES: + snd_opl3_clear_patches(opl3); + return 0; + #ifdef CONFIG_SND_DEBUG default: snd_printk("unknown IOCTL: 0x%x\n", cmd); @@ -188,6 +192,170 @@ int snd_opl3_release(struct snd_hwdep * hw, struct file *file) return 0; } +/* + * write the device - load patches + */ +long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, + loff_t *offset) +{ + struct snd_opl3 *opl3 = hw->private_data; + long result = 0; + int err = 0; + struct sbi_patch inst; + + while (count >= sizeof(inst)) { + unsigned char type; + if (copy_from_user(&inst, buf, sizeof(inst))) + return -EFAULT; + if (!memcmp(inst.key, FM_KEY_SBI, 4) || + !memcmp(inst.key, FM_KEY_2OP, 4)) + type = FM_PATCH_OPL2; + else if (!memcmp(inst.key, FM_KEY_4OP, 4)) + type = FM_PATCH_OPL3; + else /* invalid type */ + break; + err = snd_opl3_load_patch(opl3, inst.prog, inst.bank, type, + inst.name, inst.extension, + inst.data); + if (err < 0) + break; + result += sizeof(inst); + count -= sizeof(inst); + } + return result > 0 ? result : err; +} + + +/* + * Patch management + */ + +/* offsets for SBI params */ +#define AM_VIB 0 +#define KSL_LEVEL 2 +#define ATTACK_DECAY 4 +#define SUSTAIN_RELEASE 6 +#define WAVE_SELECT 8 + +/* offset for SBI instrument */ +#define CONNECTION 10 +#define OFFSET_4OP 11 + +/* + * load a patch, obviously. + * + * loaded on the given program and bank numbers with the given type + * (FM_PATCH_OPLx). + * data is the pointer of SBI record _without_ header (key and name). + * name is the name string of the patch. + * ext is the extension data of 7 bytes long (stored in name of SBI + * data up to offset 25), or NULL to skip. + * return 0 if successful or a negative error code. + */ +int snd_opl3_load_patch(struct snd_opl3 *opl3, + int prog, int bank, int type, + const char *name, + const unsigned char *ext, + const unsigned char *data) +{ + struct fm_patch *patch; + int i; + + patch = snd_opl3_find_patch(opl3, prog, bank, 1); + if (!patch) + return -ENOMEM; + + patch->type = type; + + for (i = 0; i < 2; i++) { + patch->inst.op[i].am_vib = data[AM_VIB + i]; + patch->inst.op[i].ksl_level = data[KSL_LEVEL + i]; + patch->inst.op[i].attack_decay = data[ATTACK_DECAY + i]; + patch->inst.op[i].sustain_release = data[SUSTAIN_RELEASE + i]; + patch->inst.op[i].wave_select = data[WAVE_SELECT + i]; + } + patch->inst.feedback_connection[0] = data[CONNECTION]; + + if (type == FM_PATCH_OPL3) { + for (i = 0; i < 2; i++) { + patch->inst.op[i+2].am_vib = + data[OFFSET_4OP + AM_VIB + i]; + patch->inst.op[i+2].ksl_level = + data[OFFSET_4OP + KSL_LEVEL + i]; + patch->inst.op[i+2].attack_decay = + data[OFFSET_4OP + ATTACK_DECAY + i]; + patch->inst.op[i+2].sustain_release = + data[OFFSET_4OP + SUSTAIN_RELEASE + i]; + patch->inst.op[i+2].wave_select = + data[OFFSET_4OP + WAVE_SELECT + i]; + } + patch->inst.feedback_connection[1] = + data[OFFSET_4OP + CONNECTION]; + } + + if (ext) { + patch->inst.echo_delay = ext[0]; + patch->inst.echo_atten = ext[1]; + patch->inst.chorus_spread = ext[2]; + patch->inst.trnsps = ext[3]; + patch->inst.fix_dur = ext[4]; + patch->inst.modes = ext[5]; + patch->inst.fix_key = ext[6]; + } + + if (name) + strlcpy(patch->name, name, sizeof(patch->name)); + + return 0; +} +EXPORT_SYMBOL(snd_opl3_load_patch); + +/* + * find a patch with the given program and bank numbers, returns its pointer + * if no matching patch is found and create_patch is set, it creates a + * new patch object. + */ +struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, + int create_patch) +{ + /* pretty dumb hash key */ + unsigned int key = (prog + bank) % OPL3_PATCH_HASH_SIZE; + struct fm_patch *patch; + + for (patch = opl3->patch_table[key]; patch; patch = patch->next) { + if (patch->prog == prog && patch->bank == bank) + return patch; + } + if (!create_patch) + return NULL; + + patch = kzalloc(sizeof(*patch), GFP_KERNEL); + if (!patch) + return NULL; + patch->prog = prog; + patch->bank = bank; + patch->next = opl3->patch_table[key]; + opl3->patch_table[key] = patch; + return patch; +} +EXPORT_SYMBOL(snd_opl3_find_patch); + +/* + * Clear all patches of the given OPL3 instance + */ +void snd_opl3_clear_patches(struct snd_opl3 *opl3) +{ + int i; + for (i = 0; i < OPL3_PATCH_HASH_SIZE; i++) { + struct fm_patch *patch, *next; + for (patch = opl3->patch_table[i]; patch; patch = next) { + next = patch->next; + kfree(patch); + } + } + memset(opl3->patch_table, 0, sizeof(opl3->patch_table)); +} + /* ------------------------------ */ void snd_opl3_reset(struct snd_opl3 * opl3) -- cgit v0.10.2 From 05c1afe75fcebf456017ec186811cf1599f4360e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Oct 2007 11:59:15 +0100 Subject: [ALSA] opl3 - simplify exclusive access lock Use the exclusive access lock in hwdep instead of the own one. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/opl3.h b/include/sound/opl3.h index 7ee865d..eea584b 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -320,7 +320,6 @@ struct snd_opl3 { spinlock_t reg_lock; struct snd_card *card; /* The card that this belongs to */ - int used; /* usage flag - exclusive */ unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */ unsigned char rhythm; /* percussion mode flag */ unsigned char max_voices; /* max number of voices */ @@ -353,7 +352,6 @@ struct snd_opl3 { int sys_timer_status; /* system timer run status */ spinlock_t sys_timer_lock; /* Lock for system timer access */ #endif - struct mutex access_mutex; /* locking */ }; /* opl3.c */ diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index a657da9..a127084 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -361,7 +361,6 @@ int snd_opl3_new(struct snd_card *card, opl3->hardware = hardware; spin_lock_init(&opl3->reg_lock); spin_lock_init(&opl3->timer_lock); - mutex_init(&opl3->access_mutex); if ((err = snd_device_new(card, SNDRV_DEV_CODEC, opl3, &ops)) < 0) { snd_opl3_free(opl3); @@ -497,6 +496,7 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, return err; } hw->private_data = opl3; + hw->exclusive = 1; #ifdef CONFIG_SND_OSSEMUL if (device == 0) { hw->oss_type = SNDRV_OSS_DEVICE_TYPE_DMFM; diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index ff6da16..6fd60b7 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -51,14 +51,15 @@ void snd_opl3_synth_use_dec(struct snd_opl3 * opl3) int snd_opl3_synth_setup(struct snd_opl3 * opl3) { int idx; + struct snd_hwdep *hwdep = opl3->hwdep; - mutex_lock(&opl3->access_mutex); - if (opl3->used) { - mutex_unlock(&opl3->access_mutex); + mutex_lock(&hwdep->open_mutex); + if (hwdep->used) { + mutex_unlock(&hwdep->open_mutex); return -EBUSY; } - opl3->used++; - mutex_unlock(&opl3->access_mutex); + hwdep->used++; + mutex_unlock(&hwdep->open_mutex); snd_opl3_reset(opl3); @@ -91,9 +92,11 @@ void snd_opl3_synth_cleanup(struct snd_opl3 * opl3) spin_unlock_irqrestore(&opl3->sys_timer_lock, flags); snd_opl3_reset(opl3); - mutex_lock(&opl3->access_mutex); - opl3->used--; - mutex_unlock(&opl3->access_mutex); + hwdep = opl3->hwdep; + mutex_lock(&hwdep->open_mutex); + hwdep->used--; + mutex_unlock(&hwdep->open_mutex); + wake_up(&hwdep->open_wait); } static int snd_opl3_synth_use(void *private_data, struct snd_seq_port_subscribe * info) diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index d55eefc..a7bf7a4 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -76,16 +76,6 @@ static int snd_opl3_set_connection(struct snd_opl3 * opl3, int connection); */ int snd_opl3_open(struct snd_hwdep * hw, struct file *file) { - struct snd_opl3 *opl3 = hw->private_data; - - mutex_lock(&opl3->access_mutex); - if (opl3->used) { - mutex_unlock(&opl3->access_mutex); - return -EAGAIN; - } - opl3->used++; - mutex_unlock(&opl3->access_mutex); - return 0; } @@ -185,10 +175,6 @@ int snd_opl3_release(struct snd_hwdep * hw, struct file *file) struct snd_opl3 *opl3 = hw->private_data; snd_opl3_reset(opl3); - mutex_lock(&opl3->access_mutex); - opl3->used--; - mutex_unlock(&opl3->access_mutex); - return 0; } -- cgit v0.10.2 From e5723b41abe559bafc52591dcf8ee19cc131d3a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Oct 2007 12:17:17 +0100 Subject: [ALSA] Remove sequencer instrument layer Remove sequencer instrument layer from the tree. This mechanism hasn't been used much with the actual devices. The only reasonable user was OPL3 loader, and now it was rewritten to use hwdep instead. So, let's remove the rest of rotten codes. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/ainstr_fm.h b/include/sound/ainstr_fm.h deleted file mode 100644 index c4afb1f..0000000 --- a/include/sound/ainstr_fm.h +++ /dev/null @@ -1,134 +0,0 @@ -/* - * Advanced Linux Sound Architecture - * - * FM (OPL2/3) Instrument Format - * Copyright (c) 2000 Uros Bizjak - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#ifndef __SOUND_AINSTR_FM_H -#define __SOUND_AINSTR_FM_H - -#ifndef __KERNEL__ -#include -#include -#endif - -/* - * share types (share ID 1) - */ - -#define FM_SHARE_FILE 0 - -/* - * FM operator - */ - -struct fm_operator { - unsigned char am_vib; - unsigned char ksl_level; - unsigned char attack_decay; - unsigned char sustain_release; - unsigned char wave_select; -}; - -/* - * Instrument - */ - -#define FM_PATCH_OPL2 0x01 /* OPL2 2 operators FM instrument */ -#define FM_PATCH_OPL3 0x02 /* OPL3 4 operators FM instrument */ - -struct fm_instrument { - unsigned int share_id[4]; /* share id - zero = no sharing */ - unsigned char type; /* instrument type */ - - struct fm_operator op[4]; - unsigned char feedback_connection[2]; - - unsigned char echo_delay; - unsigned char echo_atten; - unsigned char chorus_spread; - unsigned char trnsps; - unsigned char fix_dur; - unsigned char modes; - unsigned char fix_key; -}; - -/* - * - * Kernel <-> user space - * Hardware (CPU) independent section - * - * * = zero or more - * + = one or more - * - * fm_xinstrument FM_STRU_INSTR - * - */ - -#define FM_STRU_INSTR __cpu_to_be32(('I'<<24)|('N'<<16)|('S'<<8)|'T') - -/* - * FM operator - */ - -struct fm_xoperator { - __u8 am_vib; - __u8 ksl_level; - __u8 attack_decay; - __u8 sustain_release; - __u8 wave_select; -}; - -/* - * Instrument - */ - -struct fm_xinstrument { - __u32 stype; /* structure type */ - - __u32 share_id[4]; /* share id - zero = no sharing */ - __u8 type; /* instrument type */ - - struct fm_xoperator op[4]; /* fm operators */ - __u8 feedback_connection[2]; - - __u8 echo_delay; - __u8 echo_atten; - __u8 chorus_spread; - __u8 trnsps; - __u8 fix_dur; - __u8 modes; - __u8 fix_key; -}; - -#ifdef __KERNEL__ - -#include "seq_instr.h" - -int snd_seq_fm_init(struct snd_seq_kinstr_ops * ops, - struct snd_seq_kinstr_ops * next); - -#endif - -/* typedefs for compatibility to user-space */ -typedef struct fm_xoperator fm_xoperator_t; -typedef struct fm_xinstrument fm_xinstrument_t; - -#endif /* __SOUND_AINSTR_FM_H */ diff --git a/include/sound/ainstr_gf1.h b/include/sound/ainstr_gf1.h deleted file mode 100644 index b62b665..0000000 --- a/include/sound/ainstr_gf1.h +++ /dev/null @@ -1,229 +0,0 @@ -/* - * Advanced Linux Sound Architecture - * - * GF1 (GUS) Patch Instrument Format - * Copyright (c) 1994-99 by Jaroslav Kysela - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#ifndef __SOUND_AINSTR_GF1_H -#define __SOUND_AINSTR_GF1_H - -#ifndef __KERNEL__ -#include -#include -#endif - -/* - * share types (share ID 1) - */ - -#define GF1_SHARE_FILE 0 - -/* - * wave formats - */ - -#define GF1_WAVE_16BIT 0x0001 /* 16-bit wave */ -#define GF1_WAVE_UNSIGNED 0x0002 /* unsigned wave */ -#define GF1_WAVE_INVERT 0x0002 /* same as unsigned wave */ -#define GF1_WAVE_BACKWARD 0x0004 /* backward mode (maybe used for reverb or ping-ping loop) */ -#define GF1_WAVE_LOOP 0x0008 /* loop mode */ -#define GF1_WAVE_BIDIR 0x0010 /* bidirectional mode */ -#define GF1_WAVE_STEREO 0x0100 /* stereo mode */ -#define GF1_WAVE_ULAW 0x0200 /* uLaw compression mode */ - -/* - * Wavetable definitions - */ - -struct gf1_wave { - unsigned int share_id[4]; /* share id - zero = no sharing */ - unsigned int format; /* wave format */ - - struct { - unsigned int number; /* some other ID for this instrument */ - unsigned int memory; /* begin of waveform in onboard memory */ - unsigned char *ptr; /* pointer to waveform in system memory */ - } address; - - unsigned int size; /* size of waveform in samples */ - unsigned int start; /* start offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned int loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned int loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned short loop_repeat; /* loop repeat - 0 = forever */ - - unsigned char flags; /* GF1 patch flags */ - unsigned char pad; - unsigned int sample_rate; /* sample rate in Hz */ - unsigned int low_frequency; /* low frequency range */ - unsigned int high_frequency; /* high frequency range */ - unsigned int root_frequency; /* root frequency range */ - signed short tune; - unsigned char balance; - unsigned char envelope_rate[6]; - unsigned char envelope_offset[6]; - unsigned char tremolo_sweep; - unsigned char tremolo_rate; - unsigned char tremolo_depth; - unsigned char vibrato_sweep; - unsigned char vibrato_rate; - unsigned char vibrato_depth; - unsigned short scale_frequency; - unsigned short scale_factor; /* 0-2048 or 0-2 */ - - struct gf1_wave *next; -}; - -/* - * Instrument - */ - -#define IWFFFF_EXCLUDE_NONE 0x0000 /* exclusion mode - none */ -#define IWFFFF_EXCLUDE_SINGLE 0x0001 /* exclude single - single note from the instrument group */ -#define IWFFFF_EXCLUDE_MULTIPLE 0x0002 /* exclude multiple - stop only same note from this instrument */ - -#define IWFFFF_EFFECT_NONE 0 -#define IWFFFF_EFFECT_REVERB 1 -#define IWFFFF_EFFECT_CHORUS 2 -#define IWFFFF_EFFECT_ECHO 3 - -struct gf1_instrument { - unsigned short exclusion; - unsigned short exclusion_group; /* 0 - none, 1-65535 */ - - unsigned char effect1; /* effect 1 */ - unsigned char effect1_depth; /* 0-127 */ - unsigned char effect2; /* effect 2 */ - unsigned char effect2_depth; /* 0-127 */ - - struct gf1_wave *wave; /* first waveform */ -}; - -/* - * - * Kernel <-> user space - * Hardware (CPU) independent section - * - * * = zero or more - * + = one or more - * - * gf1_xinstrument IWFFFF_STRU_INSTR - * +gf1_xwave IWFFFF_STRU_WAVE - * - */ - -#define GF1_STRU_WAVE __cpu_to_be32(('W'<<24)|('A'<<16)|('V'<<8)|'E') -#define GF1_STRU_INSTR __cpu_to_be32(('I'<<24)|('N'<<16)|('S'<<8)|'T') - -/* - * Wavetable definitions - */ - -struct gf1_xwave { - __u32 stype; /* structure type */ - - __u32 share_id[4]; /* share id - zero = no sharing */ - __u32 format; /* wave format */ - - __u32 size; /* size of waveform in samples */ - __u32 start; /* start offset in samples * 16 (lowest 4 bits - fraction) */ - __u32 loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */ - __u32 loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */ - __u16 loop_repeat; /* loop repeat - 0 = forever */ - - __u8 flags; /* GF1 patch flags */ - __u8 pad; - __u32 sample_rate; /* sample rate in Hz */ - __u32 low_frequency; /* low frequency range */ - __u32 high_frequency; /* high frequency range */ - __u32 root_frequency; /* root frequency range */ - __s16 tune; - __u8 balance; - __u8 envelope_rate[6]; - __u8 envelope_offset[6]; - __u8 tremolo_sweep; - __u8 tremolo_rate; - __u8 tremolo_depth; - __u8 vibrato_sweep; - __u8 vibrato_rate; - __u8 vibrato_depth; - __u16 scale_frequency; - __u16 scale_factor; /* 0-2048 or 0-2 */ -}; - -/* - * Instrument - */ - -struct gf1_xinstrument { - __u32 stype; - - __u16 exclusion; - __u16 exclusion_group; /* 0 - none, 1-65535 */ - - __u8 effect1; /* effect 1 */ - __u8 effect1_depth; /* 0-127 */ - __u8 effect2; /* effect 2 */ - __u8 effect2_depth; /* 0-127 */ -}; - -/* - * Instrument info - */ - -#define GF1_INFO_ENVELOPE (1<<0) -#define GF1_INFO_TREMOLO (1<<1) -#define GF1_INFO_VIBRATO (1<<2) - -struct gf1_info { - unsigned char flags; /* supported wave flags */ - unsigned char pad[3]; - unsigned int features; /* supported features */ - unsigned int max8_len; /* maximum 8-bit wave length */ - unsigned int max16_len; /* maximum 16-bit wave length */ -}; - -#ifdef __KERNEL__ - -#include "seq_instr.h" - -struct snd_gf1_ops { - void *private_data; - int (*info)(void *private_data, struct gf1_info *info); - int (*put_sample)(void *private_data, struct gf1_wave *wave, - char __user *data, long len, int atomic); - int (*get_sample)(void *private_data, struct gf1_wave *wave, - char __user *data, long len, int atomic); - int (*remove_sample)(void *private_data, struct gf1_wave *wave, - int atomic); - void (*notify)(void *private_data, struct snd_seq_kinstr *instr, int what); - struct snd_seq_kinstr_ops kops; -}; - -int snd_seq_gf1_init(struct snd_gf1_ops *ops, - void *private_data, - struct snd_seq_kinstr_ops *next); - -#endif - -/* typedefs for compatibility to user-space */ -typedef struct gf1_xwave gf1_xwave_t; -typedef struct gf1_xinstrument gf1_xinstrument_t; - -#endif /* __SOUND_AINSTR_GF1_H */ diff --git a/include/sound/ainstr_iw.h b/include/sound/ainstr_iw.h deleted file mode 100644 index 11bd250..0000000 --- a/include/sound/ainstr_iw.h +++ /dev/null @@ -1,384 +0,0 @@ -/* - * Advanced Linux Sound Architecture - * - * InterWave FFFF Instrument Format - * Copyright (c) 1994-99 by Jaroslav Kysela - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#ifndef __SOUND_AINSTR_IW_H -#define __SOUND_AINSTR_IW_H - -#ifndef __KERNEL__ -#include -#include -#endif - -/* - * share types (share ID 1) - */ - -#define IWFFFF_SHARE_FILE 0 - -/* - * wave formats - */ - -#define IWFFFF_WAVE_16BIT 0x0001 /* 16-bit wave */ -#define IWFFFF_WAVE_UNSIGNED 0x0002 /* unsigned wave */ -#define IWFFFF_WAVE_INVERT 0x0002 /* same as unsigned wave */ -#define IWFFFF_WAVE_BACKWARD 0x0004 /* backward mode (maybe used for reverb or ping-ping loop) */ -#define IWFFFF_WAVE_LOOP 0x0008 /* loop mode */ -#define IWFFFF_WAVE_BIDIR 0x0010 /* bidirectional mode */ -#define IWFFFF_WAVE_ULAW 0x0020 /* uLaw compressed wave */ -#define IWFFFF_WAVE_RAM 0x0040 /* wave is _preloaded_ in RAM (it is used for ROM simulation) */ -#define IWFFFF_WAVE_ROM 0x0080 /* wave is in ROM */ -#define IWFFFF_WAVE_STEREO 0x0100 /* wave is stereo */ - -/* - * Wavetable definitions - */ - -struct iwffff_wave { - unsigned int share_id[4]; /* share id - zero = no sharing */ - unsigned int format; /* wave format */ - - struct { - unsigned int number; /* some other ID for this wave */ - unsigned int memory; /* begin of waveform in onboard memory */ - unsigned char *ptr; /* pointer to waveform in system memory */ - } address; - - unsigned int size; /* size of waveform in samples */ - unsigned int start; /* start offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned int loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned int loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned short loop_repeat; /* loop repeat - 0 = forever */ - unsigned int sample_ratio; /* sample ratio (44100 * 1024 / rate) */ - unsigned char attenuation; /* 0 - 127 (no corresponding midi controller) */ - unsigned char low_note; /* lower frequency range for this waveform */ - unsigned char high_note; /* higher frequency range for this waveform */ - unsigned char pad; - - struct iwffff_wave *next; -}; - -/* - * Layer - */ - -#define IWFFFF_LFO_SHAPE_TRIANGLE 0 -#define IWFFFF_LFO_SHAPE_POSTRIANGLE 1 - -struct iwffff_lfo { - unsigned short freq; /* (0-2047) 0.01Hz - 21.5Hz */ - signed short depth; /* volume +- (0-255) 0.48675dB/step */ - signed short sweep; /* 0 - 950 deciseconds */ - unsigned char shape; /* see to IWFFFF_LFO_SHAPE_XXXX */ - unsigned char delay; /* 0 - 255 deciseconds */ -}; - -#define IWFFFF_ENV_FLAG_RETRIGGER 0x0001 /* flag - retrigger */ - -#define IWFFFF_ENV_MODE_ONE_SHOT 0x0001 /* mode - one shot */ -#define IWFFFF_ENV_MODE_SUSTAIN 0x0002 /* mode - sustain */ -#define IWFFFF_ENV_MODE_NO_SUSTAIN 0x0003 /* mode - no sustain */ - -#define IWFFFF_ENV_INDEX_VELOCITY 0x0001 /* index - velocity */ -#define IWFFFF_ENV_INDEX_FREQUENCY 0x0002 /* index - frequency */ - -struct iwffff_env_point { - unsigned short offset; - unsigned short rate; -}; - -struct iwffff_env_record { - unsigned short nattack; - unsigned short nrelease; - unsigned short sustain_offset; - unsigned short sustain_rate; - unsigned short release_rate; - unsigned char hirange; - unsigned char pad; - struct iwffff_env_record *next; - /* points are stored here */ - /* count of points = nattack + nrelease */ -}; - -struct iwffff_env { - unsigned char flags; - unsigned char mode; - unsigned char index; - unsigned char pad; - struct iwffff_env_record *record; -}; - -#define IWFFFF_LAYER_FLAG_RETRIGGER 0x0001 /* retrigger */ - -#define IWFFFF_LAYER_VELOCITY_TIME 0x0000 /* velocity mode = time */ -#define IWFFFF_LAYER_VELOCITY_RATE 0x0001 /* velocity mode = rate */ - -#define IWFFFF_LAYER_EVENT_KUP 0x0000 /* layer event - key up */ -#define IWFFFF_LAYER_EVENT_KDOWN 0x0001 /* layer event - key down */ -#define IWFFFF_LAYER_EVENT_RETRIG 0x0002 /* layer event - retrigger */ -#define IWFFFF_LAYER_EVENT_LEGATO 0x0003 /* layer event - legato */ - -struct iwffff_layer { - unsigned char flags; - unsigned char velocity_mode; - unsigned char layer_event; - unsigned char low_range; /* range for layer based */ - unsigned char high_range; /* on either velocity or frequency */ - unsigned char pan; /* pan offset from CC1 (0 left - 127 right) */ - unsigned char pan_freq_scale; /* position based on frequency (0-127) */ - unsigned char attenuation; /* 0-127 (no corresponding midi controller) */ - struct iwffff_lfo tremolo; /* tremolo effect */ - struct iwffff_lfo vibrato; /* vibrato effect */ - unsigned short freq_scale; /* 0-2048, 1024 is equal to semitone scaling */ - unsigned char freq_center; /* center for keyboard frequency scaling */ - unsigned char pad; - struct iwffff_env penv; /* pitch envelope */ - struct iwffff_env venv; /* volume envelope */ - - struct iwffff_wave *wave; - struct iwffff_layer *next; -}; - -/* - * Instrument - */ - -#define IWFFFF_EXCLUDE_NONE 0x0000 /* exclusion mode - none */ -#define IWFFFF_EXCLUDE_SINGLE 0x0001 /* exclude single - single note from the instrument group */ -#define IWFFFF_EXCLUDE_MULTIPLE 0x0002 /* exclude multiple - stop only same note from this instrument */ - -#define IWFFFF_LAYER_NONE 0x0000 /* not layered */ -#define IWFFFF_LAYER_ON 0x0001 /* layered */ -#define IWFFFF_LAYER_VELOCITY 0x0002 /* layered by velocity */ -#define IWFFFF_LAYER_FREQUENCY 0x0003 /* layered by frequency */ - -#define IWFFFF_EFFECT_NONE 0 -#define IWFFFF_EFFECT_REVERB 1 -#define IWFFFF_EFFECT_CHORUS 2 -#define IWFFFF_EFFECT_ECHO 3 - -struct iwffff_instrument { - unsigned short exclusion; - unsigned short layer_type; - unsigned short exclusion_group; /* 0 - none, 1-65535 */ - - unsigned char effect1; /* effect 1 */ - unsigned char effect1_depth; /* 0-127 */ - unsigned char effect2; /* effect 2 */ - unsigned char effect2_depth; /* 0-127 */ - - struct iwffff_layer *layer; /* first layer */ -}; - -/* - * - * Kernel <-> user space - * Hardware (CPU) independent section - * - * * = zero or more - * + = one or more - * - * iwffff_xinstrument IWFFFF_STRU_INSTR - * +iwffff_xlayer IWFFFF_STRU_LAYER - * *iwffff_xenv_record IWFFFF_STRU_ENV_RECT (tremolo) - * *iwffff_xenv_record IWFFFF_STRU_EVN_RECT (vibrato) - * +iwffff_xwave IWFFFF_STRU_WAVE - * - */ - -#define IWFFFF_STRU_WAVE __cpu_to_be32(('W'<<24)|('A'<<16)|('V'<<8)|'E') -#define IWFFFF_STRU_ENV_RECP __cpu_to_be32(('E'<<24)|('N'<<16)|('R'<<8)|'P') -#define IWFFFF_STRU_ENV_RECV __cpu_to_be32(('E'<<24)|('N'<<16)|('R'<<8)|'V') -#define IWFFFF_STRU_LAYER __cpu_to_be32(('L'<<24)|('A'<<16)|('Y'<<8)|'R') -#define IWFFFF_STRU_INSTR __cpu_to_be32(('I'<<24)|('N'<<16)|('S'<<8)|'T') - -/* - * Wavetable definitions - */ - -struct iwffff_xwave { - __u32 stype; /* structure type */ - - __u32 share_id[4]; /* share id - zero = no sharing */ - - __u32 format; /* wave format */ - __u32 offset; /* offset to ROM (address) */ - - __u32 size; /* size of waveform in samples */ - __u32 start; /* start offset in samples * 16 (lowest 4 bits - fraction) */ - __u32 loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */ - __u32 loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */ - __u16 loop_repeat; /* loop repeat - 0 = forever */ - __u32 sample_ratio; /* sample ratio (44100 * 1024 / rate) */ - __u8 attenuation; /* 0 - 127 (no corresponding midi controller) */ - __u8 low_note; /* lower frequency range for this waveform */ - __u8 high_note; /* higher frequency range for this waveform */ - __u8 pad; -}; - -/* - * Layer - */ - -struct iwffff_xlfo { - __u16 freq; /* (0-2047) 0.01Hz - 21.5Hz */ - __s16 depth; /* volume +- (0-255) 0.48675dB/step */ - __s16 sweep; /* 0 - 950 deciseconds */ - __u8 shape; /* see to ULTRA_IW_LFO_SHAPE_XXXX */ - __u8 delay; /* 0 - 255 deciseconds */ -}; - -struct iwffff_xenv_point { - __u16 offset; - __u16 rate; -}; - -struct iwffff_xenv_record { - __u32 stype; - __u16 nattack; - __u16 nrelease; - __u16 sustain_offset; - __u16 sustain_rate; - __u16 release_rate; - __u8 hirange; - __u8 pad; - /* points are stored here.. */ - /* count of points = nattack + nrelease */ -}; - -struct iwffff_xenv { - __u8 flags; - __u8 mode; - __u8 index; - __u8 pad; -}; - -struct iwffff_xlayer { - __u32 stype; - __u8 flags; - __u8 velocity_mode; - __u8 layer_event; - __u8 low_range; /* range for layer based */ - __u8 high_range; /* on either velocity or frequency */ - __u8 pan; /* pan offset from CC1 (0 left - 127 right) */ - __u8 pan_freq_scale; /* position based on frequency (0-127) */ - __u8 attenuation; /* 0-127 (no corresponding midi controller) */ - struct iwffff_xlfo tremolo; /* tremolo effect */ - struct iwffff_xlfo vibrato; /* vibrato effect */ - __u16 freq_scale; /* 0-2048, 1024 is equal to semitone scaling */ - __u8 freq_center; /* center for keyboard frequency scaling */ - __u8 pad; - struct iwffff_xenv penv; /* pitch envelope */ - struct iwffff_xenv venv; /* volume envelope */ -}; - -/* - * Instrument - */ - -struct iwffff_xinstrument { - __u32 stype; - - __u16 exclusion; - __u16 layer_type; - __u16 exclusion_group; /* 0 - none, 1-65535 */ - - __u8 effect1; /* effect 1 */ - __u8 effect1_depth; /* 0-127 */ - __u8 effect2; /* effect 2 */ - __u8 effect2_depth; /* 0-127 */ -}; - -/* - * ROM support - * InterWave ROMs are Little-Endian (x86) - */ - -#define IWFFFF_ROM_HDR_SIZE 512 - -struct iwffff_rom_header { - __u8 iwave[8]; - __u8 revision; - __u8 series_number; - __u8 series_name[16]; - __u8 date[10]; - __u16 vendor_revision_major; - __u16 vendor_revision_minor; - __u32 rom_size; - __u8 copyright[128]; - __u8 vendor_name[64]; - __u8 description[128]; -}; - -/* - * Instrument info - */ - -#define IWFFFF_INFO_LFO_VIBRATO (1<<0) -#define IWFFFF_INFO_LFO_VIBRATO_SHAPE (1<<1) -#define IWFFFF_INFO_LFO_TREMOLO (1<<2) -#define IWFFFF_INFO_LFO_TREMOLO_SHAPE (1<<3) - -struct iwffff_info { - unsigned int format; /* supported format bits */ - unsigned int effects; /* supported effects (1 << IWFFFF_EFFECT*) */ - unsigned int lfos; /* LFO effects */ - unsigned int max8_len; /* maximum 8-bit wave length */ - unsigned int max16_len; /* maximum 16-bit wave length */ -}; - -#ifdef __KERNEL__ - -#include "seq_instr.h" - -struct snd_iwffff_ops { - void *private_data; - int (*info)(void *private_data, struct iwffff_info *info); - int (*put_sample)(void *private_data, struct iwffff_wave *wave, - char __user *data, long len, int atomic); - int (*get_sample)(void *private_data, struct iwffff_wave *wave, - char __user *data, long len, int atomic); - int (*remove_sample)(void *private_data, struct iwffff_wave *wave, - int atomic); - void (*notify)(void *private_data, struct snd_seq_kinstr *instr, int what); - struct snd_seq_kinstr_ops kops; -}; - -int snd_seq_iwffff_init(struct snd_iwffff_ops *ops, - void *private_data, - struct snd_seq_kinstr_ops *next); - -#endif - -/* typedefs for compatibility to user-space */ -typedef struct iwffff_xwave iwffff_xwave_t; -typedef struct iwffff_xlfo iwffff_xlfo_t; -typedef struct iwffff_xenv_point iwffff_xenv_point_t; -typedef struct iwffff_xenv_record iwffff_xenv_record_t; -typedef struct iwffff_xenv iwffff_xenv_t; -typedef struct iwffff_xlayer iwffff_xlayer_t; -typedef struct iwffff_xinstrument iwffff_xinstrument_t; -typedef struct iwffff_rom_header iwffff_rom_header_t; -typedef struct iwffff_info iwffff_info_t; - -#endif /* __SOUND_AINSTR_IW_H */ diff --git a/include/sound/ainstr_simple.h b/include/sound/ainstr_simple.h deleted file mode 100644 index da08e72..0000000 --- a/include/sound/ainstr_simple.h +++ /dev/null @@ -1,159 +0,0 @@ -/* - * Advanced Linux Sound Architecture - * - * Simple (MOD player) Instrument Format - * Copyright (c) 1994-99 by Jaroslav Kysela - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#ifndef __SOUND_AINSTR_SIMPLE_H -#define __SOUND_AINSTR_SIMPLE_H - -#ifndef __KERNEL__ -#include -#include -#endif - -/* - * share types (share ID 1) - */ - -#define SIMPLE_SHARE_FILE 0 - -/* - * wave formats - */ - -#define SIMPLE_WAVE_16BIT 0x0001 /* 16-bit wave */ -#define SIMPLE_WAVE_UNSIGNED 0x0002 /* unsigned wave */ -#define SIMPLE_WAVE_INVERT 0x0002 /* same as unsigned wave */ -#define SIMPLE_WAVE_BACKWARD 0x0004 /* backward mode (maybe used for reverb or ping-ping loop) */ -#define SIMPLE_WAVE_LOOP 0x0008 /* loop mode */ -#define SIMPLE_WAVE_BIDIR 0x0010 /* bidirectional mode */ -#define SIMPLE_WAVE_STEREO 0x0100 /* stereo wave */ -#define SIMPLE_WAVE_ULAW 0x0200 /* uLaw compression mode */ - -/* - * instrument effects - */ - -#define SIMPLE_EFFECT_NONE 0 -#define SIMPLE_EFFECT_REVERB 1 -#define SIMPLE_EFFECT_CHORUS 2 -#define SIMPLE_EFFECT_ECHO 3 - -/* - * instrument info - */ - -struct simple_instrument_info { - unsigned int format; /* supported format bits */ - unsigned int effects; /* supported effects (1 << SIMPLE_EFFECT_*) */ - unsigned int max8_len; /* maximum 8-bit wave length */ - unsigned int max16_len; /* maximum 16-bit wave length */ -}; - -/* - * Instrument - */ - -struct simple_instrument { - unsigned int share_id[4]; /* share id - zero = no sharing */ - unsigned int format; /* wave format */ - - struct { - unsigned int number; /* some other ID for this instrument */ - unsigned int memory; /* begin of waveform in onboard memory */ - unsigned char *ptr; /* pointer to waveform in system memory */ - } address; - - unsigned int size; /* size of waveform in samples */ - unsigned int start; /* start offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned int loop_start; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned int loop_end; /* loop end offset in samples * 16 (lowest 4 bits - fraction) */ - unsigned short loop_repeat; /* loop repeat - 0 = forever */ - - unsigned char effect1; /* effect 1 */ - unsigned char effect1_depth; /* 0-127 */ - unsigned char effect2; /* effect 2 */ - unsigned char effect2_depth; /* 0-127 */ -}; - -/* - * - * Kernel <-> user space - * Hardware (CPU) independent section - * - * * = zero or more - * + = one or more - * - * simple_xinstrument SIMPLE_STRU_INSTR - * - */ - -#define SIMPLE_STRU_INSTR __cpu_to_be32(('I'<<24)|('N'<<16)|('S'<<8)|'T') - -/* - * Instrument - */ - -struct simple_xinstrument { - __u32 stype; - - __u32 share_id[4]; /* share id - zero = no sharing */ - __u32 format; /* wave format */ - - __u32 size; /* size of waveform in samples */ - __u32 start; /* start offset in samples * 16 (lowest 4 bits - fraction) */ - __u32 loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */ - __u32 loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */ - __u16 loop_repeat; /* loop repeat - 0 = forever */ - - __u8 effect1; /* effect 1 */ - __u8 effect1_depth; /* 0-127 */ - __u8 effect2; /* effect 2 */ - __u8 effect2_depth; /* 0-127 */ -}; - -#ifdef __KERNEL__ - -#include "seq_instr.h" - -struct snd_simple_ops { - void *private_data; - int (*info)(void *private_data, struct simple_instrument_info *info); - int (*put_sample)(void *private_data, struct simple_instrument *instr, - char __user *data, long len, int atomic); - int (*get_sample)(void *private_data, struct simple_instrument *instr, - char __user *data, long len, int atomic); - int (*remove_sample)(void *private_data, struct simple_instrument *instr, - int atomic); - void (*notify)(void *private_data, struct snd_seq_kinstr *instr, int what); - struct snd_seq_kinstr_ops kops; -}; - -int snd_seq_simple_init(struct snd_simple_ops *ops, - void *private_data, - struct snd_seq_kinstr_ops *next); - -#endif - -/* typedefs for compatibility to user-space */ -typedef struct simple_xinstrument simple_xinstrument_t; - -#endif /* __SOUND_AINSTR_SIMPLE_H */ diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 64daccb..1505e6d 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -110,18 +110,7 @@ #define SNDRV_SEQ_EVENT_PORT_SUBSCRIBED 66 /* ports connected */ #define SNDRV_SEQ_EVENT_PORT_UNSUBSCRIBED 67 /* ports disconnected */ -/** synthesizer events - * event data type = snd_seq_eve_sample_control - */ -#define SNDRV_SEQ_EVENT_SAMPLE 70 /* sample select */ -#define SNDRV_SEQ_EVENT_SAMPLE_CLUSTER 71 /* sample cluster select */ -#define SNDRV_SEQ_EVENT_SAMPLE_START 72 /* voice start */ -#define SNDRV_SEQ_EVENT_SAMPLE_STOP 73 /* voice stop */ -#define SNDRV_SEQ_EVENT_SAMPLE_FREQ 74 /* playback frequency */ -#define SNDRV_SEQ_EVENT_SAMPLE_VOLUME 75 /* volume and balance */ -#define SNDRV_SEQ_EVENT_SAMPLE_LOOP 76 /* sample loop */ -#define SNDRV_SEQ_EVENT_SAMPLE_POSITION 77 /* sample position */ -#define SNDRV_SEQ_EVENT_SAMPLE_PRIVATE1 78 /* private (hardware dependent) event */ +/* 70-89: synthesizer events - obsoleted */ /** user-defined events with fixed length * event data type = any @@ -137,28 +126,7 @@ #define SNDRV_SEQ_EVENT_USR8 98 #define SNDRV_SEQ_EVENT_USR9 99 -/** instrument layer - * variable length data can be passed directly to the driver - */ -#define SNDRV_SEQ_EVENT_INSTR_BEGIN 100 /* begin of instrument management */ -#define SNDRV_SEQ_EVENT_INSTR_END 101 /* end of instrument management */ -#define SNDRV_SEQ_EVENT_INSTR_INFO 102 /* instrument interface info */ -#define SNDRV_SEQ_EVENT_INSTR_INFO_RESULT 103 /* result */ -#define SNDRV_SEQ_EVENT_INSTR_FINFO 104 /* get format info */ -#define SNDRV_SEQ_EVENT_INSTR_FINFO_RESULT 105 /* get format info */ -#define SNDRV_SEQ_EVENT_INSTR_RESET 106 /* reset instrument memory */ -#define SNDRV_SEQ_EVENT_INSTR_STATUS 107 /* instrument interface status */ -#define SNDRV_SEQ_EVENT_INSTR_STATUS_RESULT 108 /* result */ -#define SNDRV_SEQ_EVENT_INSTR_PUT 109 /* put instrument to port */ -#define SNDRV_SEQ_EVENT_INSTR_GET 110 /* get instrument from port */ -#define SNDRV_SEQ_EVENT_INSTR_GET_RESULT 111 /* result */ -#define SNDRV_SEQ_EVENT_INSTR_FREE 112 /* free instrument(s) */ -#define SNDRV_SEQ_EVENT_INSTR_LIST 113 /* instrument list */ -#define SNDRV_SEQ_EVENT_INSTR_LIST_RESULT 114 /* result */ -#define SNDRV_SEQ_EVENT_INSTR_CLUSTER 115 /* cluster parameters */ -#define SNDRV_SEQ_EVENT_INSTR_CLUSTER_GET 116 /* get cluster parameters */ -#define SNDRV_SEQ_EVENT_INSTR_CLUSTER_RESULT 117 /* result */ -#define SNDRV_SEQ_EVENT_INSTR_CHANGE 118 /* instrument change */ +/* 100-118: instrument layer - obsoleted */ /* 119-129: reserved */ /* 130-139: variable length events @@ -258,78 +226,6 @@ struct snd_seq_ev_ext { void *ptr; /* pointer to data (note: maybe 64-bit) */ } __attribute__((packed)); -/* Instrument cluster type */ -typedef unsigned int snd_seq_instr_cluster_t; - -/* Instrument type */ -struct snd_seq_instr { - snd_seq_instr_cluster_t cluster; - unsigned int std; /* the upper byte means a private instrument (owner - client #) */ - unsigned short bank; - unsigned short prg; -}; - - /* sample number */ -struct snd_seq_ev_sample { - unsigned int std; - unsigned short bank; - unsigned short prg; -}; - - /* sample cluster */ -struct snd_seq_ev_cluster { - snd_seq_instr_cluster_t cluster; -}; - - /* sample position */ -typedef unsigned int snd_seq_position_t; /* playback position (in samples) * 16 */ - - /* sample stop mode */ -enum { - SAMPLE_STOP_IMMEDIATELY = 0, /* terminate playing immediately */ - SAMPLE_STOP_VENVELOPE = 1, /* finish volume envelope */ - SAMPLE_STOP_LOOP = 2 /* terminate loop and finish wave */ -}; - - /* sample frequency */ -typedef int snd_seq_frequency_t; /* playback frequency in HZ * 16 */ - - /* sample volume control; if any value is set to -1 == do not change */ -struct snd_seq_ev_volume { - signed short volume; /* range: 0-16383 */ - signed short lr; /* left-right balance; range: 0-16383 */ - signed short fr; /* front-rear balance; range: 0-16383 */ - signed short du; /* down-up balance; range: 0-16383 */ -}; - - /* simple loop redefinition */ -struct snd_seq_ev_loop { - unsigned int start; /* loop start (in samples) * 16 */ - unsigned int end; /* loop end (in samples) * 16 */ -}; - -struct snd_seq_ev_sample_control { - unsigned char channel; - unsigned char unused1, unused2, unused3; /* pad */ - union { - struct snd_seq_ev_sample sample; - struct snd_seq_ev_cluster cluster; - snd_seq_position_t position; - int stop_mode; - snd_seq_frequency_t frequency; - struct snd_seq_ev_volume volume; - struct snd_seq_ev_loop loop; - unsigned char raw8[8]; - } param; -}; - - - -/* INSTR_BEGIN event */ -struct snd_seq_ev_instr_begin { - int timeout; /* zero = forever, otherwise timeout in ms */ -}; - struct snd_seq_result { int event; /* processed event type */ int result; @@ -399,8 +295,6 @@ struct snd_seq_event { struct snd_seq_addr addr; struct snd_seq_connect connect; struct snd_seq_result result; - struct snd_seq_ev_instr_begin instr_begin; - struct snd_seq_ev_sample_control sample; struct snd_seq_ev_quote quote; } data; }; @@ -441,8 +335,6 @@ struct snd_seq_event_bounce { #define snd_seq_ev_is_user_type(ev) ((ev)->type >= 90 && (ev)->type < 99) /* fixed length events: 0-99 */ #define snd_seq_ev_is_fixed_type(ev) ((ev)->type < 100) -/* instrument layer events: 100-129 */ -#define snd_seq_ev_is_instr_type(ev) ((ev)->type >= 100 && (ev)->type < 130) /* variable length events: 130-139 */ #define snd_seq_ev_is_variable_type(ev) ((ev)->type >= 130 && (ev)->type < 140) /* reserved for kernel */ @@ -738,136 +630,6 @@ struct snd_seq_query_subs { /* - * Instrument abstraction layer - * - based on events - */ - -/* instrument types */ -#define SNDRV_SEQ_INSTR_ATYPE_DATA 0 /* instrument data */ -#define SNDRV_SEQ_INSTR_ATYPE_ALIAS 1 /* instrument alias */ - -/* instrument ASCII identifiers */ -#define SNDRV_SEQ_INSTR_ID_DLS1 "DLS1" -#define SNDRV_SEQ_INSTR_ID_DLS2 "DLS2" -#define SNDRV_SEQ_INSTR_ID_SIMPLE "Simple Wave" -#define SNDRV_SEQ_INSTR_ID_SOUNDFONT "SoundFont" -#define SNDRV_SEQ_INSTR_ID_GUS_PATCH "GUS Patch" -#define SNDRV_SEQ_INSTR_ID_INTERWAVE "InterWave FFFF" -#define SNDRV_SEQ_INSTR_ID_OPL2_3 "OPL2/3 FM" -#define SNDRV_SEQ_INSTR_ID_OPL4 "OPL4" - -/* instrument types */ -#define SNDRV_SEQ_INSTR_TYPE0_DLS1 (1<<0) /* MIDI DLS v1 */ -#define SNDRV_SEQ_INSTR_TYPE0_DLS2 (1<<1) /* MIDI DLS v2 */ -#define SNDRV_SEQ_INSTR_TYPE1_SIMPLE (1<<0) /* Simple Wave */ -#define SNDRV_SEQ_INSTR_TYPE1_SOUNDFONT (1<<1) /* EMU SoundFont */ -#define SNDRV_SEQ_INSTR_TYPE1_GUS_PATCH (1<<2) /* Gravis UltraSound Patch */ -#define SNDRV_SEQ_INSTR_TYPE1_INTERWAVE (1<<3) /* InterWave FFFF */ -#define SNDRV_SEQ_INSTR_TYPE2_OPL2_3 (1<<0) /* Yamaha OPL2/3 FM */ -#define SNDRV_SEQ_INSTR_TYPE2_OPL4 (1<<1) /* Yamaha OPL4 */ - -/* put commands */ -#define SNDRV_SEQ_INSTR_PUT_CMD_CREATE 0 -#define SNDRV_SEQ_INSTR_PUT_CMD_REPLACE 1 -#define SNDRV_SEQ_INSTR_PUT_CMD_MODIFY 2 -#define SNDRV_SEQ_INSTR_PUT_CMD_ADD 3 -#define SNDRV_SEQ_INSTR_PUT_CMD_REMOVE 4 - -/* get commands */ -#define SNDRV_SEQ_INSTR_GET_CMD_FULL 0 -#define SNDRV_SEQ_INSTR_GET_CMD_PARTIAL 1 - -/* query flags */ -#define SNDRV_SEQ_INSTR_QUERY_FOLLOW_ALIAS (1<<0) - -/* free commands */ -#define SNDRV_SEQ_INSTR_FREE_CMD_ALL 0 -#define SNDRV_SEQ_INSTR_FREE_CMD_PRIVATE 1 -#define SNDRV_SEQ_INSTR_FREE_CMD_CLUSTER 2 -#define SNDRV_SEQ_INSTR_FREE_CMD_SINGLE 3 - -/* size of ROM/RAM */ -typedef unsigned int snd_seq_instr_size_t; - -/* INSTR_INFO */ - -struct snd_seq_instr_info { - int result; /* operation result */ - unsigned int formats[8]; /* bitmap of supported formats */ - int ram_count; /* count of RAM banks */ - snd_seq_instr_size_t ram_sizes[16]; /* size of RAM banks */ - int rom_count; /* count of ROM banks */ - snd_seq_instr_size_t rom_sizes[8]; /* size of ROM banks */ - char reserved[128]; -}; - -/* INSTR_STATUS */ - -struct snd_seq_instr_status { - int result; /* operation result */ - snd_seq_instr_size_t free_ram[16]; /* free RAM in banks */ - int instrument_count; /* count of downloaded instruments */ - char reserved[128]; -}; - -/* INSTR_FORMAT_INFO */ - -struct snd_seq_instr_format_info { - char format[16]; /* format identifier - SNDRV_SEQ_INSTR_ID_* */ - unsigned int len; /* max data length (without this structure) */ -}; - -struct snd_seq_instr_format_info_result { - int result; /* operation result */ - char format[16]; /* format identifier */ - unsigned int len; /* filled data length (without this structure) */ -}; - -/* instrument data */ -struct snd_seq_instr_data { - char name[32]; /* instrument name */ - char reserved[16]; /* for the future use */ - int type; /* instrument type */ - union { - char format[16]; /* format identifier */ - struct snd_seq_instr alias; - } data; -}; - -/* INSTR_PUT/GET, data are stored in one block (extended), header + data */ - -struct snd_seq_instr_header { - union { - struct snd_seq_instr instr; - snd_seq_instr_cluster_t cluster; - } id; /* instrument identifier */ - unsigned int cmd; /* get/put/free command */ - unsigned int flags; /* query flags (only for get) */ - unsigned int len; /* real instrument data length (without header) */ - int result; /* operation result */ - char reserved[16]; /* for the future */ - struct snd_seq_instr_data data; /* instrument data (for put/get result) */ -}; - -/* INSTR_CLUSTER_SET */ - -struct snd_seq_instr_cluster_set { - snd_seq_instr_cluster_t cluster; /* cluster identifier */ - char name[32]; /* cluster name */ - int priority; /* cluster priority */ - char reserved[64]; /* for the future use */ -}; - -/* INSTR_CLUSTER_GET */ - -struct snd_seq_instr_cluster_get { - snd_seq_instr_cluster_t cluster; /* cluster identifier */ - char name[32]; /* cluster name */ - int priority; /* cluster priority */ - char reserved[64]; /* for the future use */ -}; - -/* * IOCTL commands */ diff --git a/include/sound/gus.h b/include/sound/gus.h index e5433d8..841bb8d 100644 --- a/include/sound/gus.h +++ b/include/sound/gus.h @@ -27,13 +27,8 @@ #include "timer.h" #include "seq_midi_emul.h" #include "seq_device.h" -#include "ainstr_iw.h" -#include "ainstr_gf1.h" -#include "ainstr_simple.h" #include -#define SNDRV_SEQ_DEV_ID_GUS "gus-synth" - /* IO ports */ #define GUSP(gus, x) ((gus)->gf1.port + SNDRV_g_u_s_##x) @@ -234,16 +229,6 @@ struct snd_gus_port { struct snd_gus_voice; -struct snd_gus_sample_ops { - void (*sample_start)(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_position_t position); - void (*sample_stop)(struct snd_gus_card *gus, struct snd_gus_voice *voice, int mode); - void (*sample_freq)(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_frequency_t freq); - void (*sample_volume)(struct snd_gus_card *gus, struct snd_gus_voice *voice, struct snd_seq_ev_volume *volume); - void (*sample_loop)(struct snd_gus_card *card, struct snd_gus_voice *voice, struct snd_seq_ev_loop *loop); - void (*sample_pos)(struct snd_gus_card *card, struct snd_gus_voice *voice, snd_seq_position_t position); - void (*sample_private1)(struct snd_gus_card *card, struct snd_gus_voice *voice, unsigned char *data); -}; - #define SNDRV_GF1_VOICE_TYPE_PCM 0 #define SNDRV_GF1_VOICE_TYPE_SYNTH 1 #define SNDRV_GF1_VOICE_TYPE_MIDI 2 @@ -284,12 +269,8 @@ struct snd_gus_voice { struct snd_gus_sample_ops *sample_ops; - struct snd_seq_instr instr; - /* running status / registers */ - struct snd_seq_ev_volume sample_volume; - unsigned short fc_register; unsigned short fc_lfo; unsigned short gf1_volume; @@ -382,10 +363,6 @@ struct snd_gf1 { int seq_client; struct snd_gus_port seq_ports[4]; - struct snd_seq_kinstr_list *ilist; - struct snd_iwffff_ops iwffff_ops; - struct snd_gf1_ops gf1_ops; - struct snd_simple_ops simple_ops; /* timer */ @@ -458,8 +435,6 @@ struct snd_gus_card { struct snd_rawmidi_substream *midi_substream_output; struct snd_rawmidi_substream *midi_substream_input; - struct snd_seq_device *seq_dev; - spinlock_t reg_lock; spinlock_t voice_alloc; spinlock_t active_voice_lock; @@ -647,48 +622,10 @@ void snd_gus_irq_profile_init(struct snd_gus_card *gus); int snd_gf1_rawmidi_new(struct snd_gus_card * gus, int device, struct snd_rawmidi **rrawmidi); -#if 0 -extern void snd_engine_instrument_register(unsigned short mode, - struct _SND_INSTRUMENT_VOICE_COMMANDS *voice_cmds, - struct _SND_INSTRUMENT_NOTE_COMMANDS *note_cmds, - struct _SND_INSTRUMENT_CHANNEL_COMMANDS *channel_cmds); -extern int snd_engine_instrument_register_ask(unsigned short mode); -#endif - /* gus_dram.c */ int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr, unsigned int addr, unsigned int size); int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr, unsigned int addr, unsigned int size, int rom); -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) - -/* gus_sample.c */ -void snd_gus_sample_event(struct snd_seq_event *ev, struct snd_gus_port *p); - -/* gus_simple.c */ -void snd_gf1_simple_init(struct snd_gus_voice *voice); - -/* gus_instr.c */ -int snd_gus_iwffff_put_sample(void *private_data, struct iwffff_wave *wave, - char __user *data, long len, int atomic); -int snd_gus_iwffff_get_sample(void *private_data, struct iwffff_wave *wave, - char __user *data, long len, int atomic); -int snd_gus_iwffff_remove_sample(void *private_data, struct iwffff_wave *wave, - int atomic); -int snd_gus_gf1_put_sample(void *private_data, struct gf1_wave *wave, - char __user *data, long len, int atomic); -int snd_gus_gf1_get_sample(void *private_data, struct gf1_wave *wave, - char __user *data, long len, int atomic); -int snd_gus_gf1_remove_sample(void *private_data, struct gf1_wave *wave, - int atomic); -int snd_gus_simple_put_sample(void *private_data, struct simple_instrument *instr, - char __user *data, long len, int atomic); -int snd_gus_simple_get_sample(void *private_data, struct simple_instrument *instr, - char __user *data, long len, int atomic); -int snd_gus_simple_remove_sample(void *private_data, struct simple_instrument *instr, - int atomic); - -#endif /* CONFIG_SND_SEQUENCER */ - #endif /* __SOUND_GUS_H */ diff --git a/include/sound/seq_instr.h b/include/sound/seq_instr.h deleted file mode 100644 index 93b0c51..0000000 --- a/include/sound/seq_instr.h +++ /dev/null @@ -1,110 +0,0 @@ -#ifndef __SOUND_SEQ_INSTR_H -#define __SOUND_SEQ_INSTR_H - -/* - * Main kernel header file for the ALSA sequencer - * Copyright (c) 1999 by Jaroslav Kysela - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ -#include "seq_kernel.h" - -/* Instrument cluster */ -struct snd_seq_kcluster { - snd_seq_instr_cluster_t cluster; - char name[32]; - int priority; - struct snd_seq_kcluster *next; -}; - -/* return pointer to private data */ -#define KINSTR_DATA(kinstr) (void *)(((char *)kinstr) + sizeof(struct snd_seq_kinstr)) - -/* Instrument structure */ -struct snd_seq_kinstr { - struct snd_seq_instr instr; - char name[32]; - int type; /* instrument type */ - int use; /* use count */ - int busy; /* not useable */ - int add_len; /* additional length */ - struct snd_seq_kinstr_ops *ops; /* operations */ - struct snd_seq_kinstr *next; -}; - -#define SNDRV_SEQ_INSTR_HASH_SIZE 32 - -/* Instrument flags */ -#define SNDRV_SEQ_INSTR_FLG_DIRECT (1<<0) /* accept only direct events */ - -/* List of all instruments */ -struct snd_seq_kinstr_list { - struct snd_seq_kinstr *hash[SNDRV_SEQ_INSTR_HASH_SIZE]; - int count; /* count of all instruments */ - - struct snd_seq_kcluster *chash[SNDRV_SEQ_INSTR_HASH_SIZE]; - int ccount; /* count of all clusters */ - - int owner; /* current owner of the instrument list */ - unsigned int flags; - - spinlock_t lock; - spinlock_t ops_lock; - struct mutex ops_mutex; - unsigned long ops_flags; -}; - -#define SNDRV_SEQ_INSTR_NOTIFY_REMOVE 0 -#define SNDRV_SEQ_INSTR_NOTIFY_CHANGE 1 - -struct snd_seq_kinstr_ops { - void *private_data; - long add_len; /* additional length */ - char *instr_type; - int (*info)(void *private_data, char *info_data, long len); - int (*put)(void *private_data, struct snd_seq_kinstr *kinstr, - char __user *instr_data, long len, int atomic, int cmd); - int (*get)(void *private_data, struct snd_seq_kinstr *kinstr, - char __user *instr_data, long len, int atomic, int cmd); - int (*get_size)(void *private_data, struct snd_seq_kinstr *kinstr, long *size); - int (*remove)(void *private_data, struct snd_seq_kinstr *kinstr, int atomic); - void (*notify)(void *private_data, struct snd_seq_kinstr *kinstr, int what); - struct snd_seq_kinstr_ops *next; -}; - - -/* instrument operations */ -struct snd_seq_kinstr_list *snd_seq_instr_list_new(void); -void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list); -int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list, - struct snd_seq_instr_header *ifree, - int client, - int atomic); -struct snd_seq_kinstr *snd_seq_instr_find(struct snd_seq_kinstr_list *list, - struct snd_seq_instr *instr, - int exact, - int follow_alias); -void snd_seq_instr_free_use(struct snd_seq_kinstr_list *list, - struct snd_seq_kinstr *instr); -int snd_seq_instr_event(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int client, - int atomic, - int hop); - -#endif /* __SOUND_SEQ_INSTR_H */ diff --git a/include/sound/trident.h b/include/sound/trident.h index 9752243..9f191a0 100644 --- a/include/sound/trident.h +++ b/include/sound/trident.h @@ -26,19 +26,12 @@ #include "pcm.h" #include "mpu401.h" #include "ac97_codec.h" -#include "seq_midi_emul.h" -#include "seq_device.h" #include "util_mem.h" -//#include "ainstr_iw.h" -//#include "ainstr_gf1.h" -#include "ainstr_simple.h" #define TRIDENT_DEVICE_ID_DX ((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_DX) #define TRIDENT_DEVICE_ID_NX ((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_NX) #define TRIDENT_DEVICE_ID_SI7018 ((PCI_VENDOR_ID_SI<<16)|PCI_DEVICE_ID_SI_7018) -#define SNDRV_SEQ_DEV_ID_TRIDENT "trident-synth" - #define SNDRV_TRIDENT_VOICE_TYPE_PCM 0 #define SNDRV_TRIDENT_VOICE_TYPE_SYNTH 1 #define SNDRV_TRIDENT_VOICE_TYPE_MIDI 2 @@ -257,16 +250,6 @@ struct snd_trident; struct snd_trident_voice; struct snd_trident_pcm_mixer; -struct snd_trident_sample_ops { - void (*sample_start)(struct snd_trident *gus, struct snd_trident_voice *voice, snd_seq_position_t position); - void (*sample_stop)(struct snd_trident *gus, struct snd_trident_voice *voice, int mode); - void (*sample_freq)(struct snd_trident *gus, struct snd_trident_voice *voice, snd_seq_frequency_t freq); - void (*sample_volume)(struct snd_trident *gus, struct snd_trident_voice *voice, struct snd_seq_ev_volume *volume); - void (*sample_loop)(struct snd_trident *card, struct snd_trident_voice *voice, struct snd_seq_ev_loop *loop); - void (*sample_pos)(struct snd_trident *card, struct snd_trident_voice *voice, snd_seq_position_t position); - void (*sample_private1)(struct snd_trident *card, struct snd_trident_voice *voice, unsigned char *data); -}; - struct snd_trident_port { struct snd_midi_channel_set * chset; struct snd_trident * trident; @@ -300,7 +283,6 @@ struct snd_trident_voice { unsigned char port; unsigned char index; - struct snd_seq_instr instr; struct snd_trident_sample_ops *sample_ops; /* channel parameters */ @@ -354,9 +336,6 @@ struct snd_4dwave { int seq_client; struct snd_trident_port seq_ports[4]; - struct snd_simple_ops simple_ops; - struct snd_seq_kinstr_list *ilist; - struct snd_trident_voice voices[64]; int ChanSynthCount; /* number of allocated synth channels */ @@ -416,7 +395,6 @@ struct snd_trident { struct snd_pcm *foldback; /* Foldback PCM */ struct snd_pcm *spdif; /* SPDIF PCM */ struct snd_rawmidi *rmidi; - struct snd_seq_device *seq_dev; struct snd_ac97_bus *ac97_bus; struct snd_ac97 *ac97; diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index ceef14a..0695937 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -3,7 +3,6 @@ # Copyright (c) 1999 by Jaroslav Kysela # -obj-$(CONFIG_SND) += instr/ ifeq ($(CONFIG_SND_SEQUENCER_OSS),y) obj-$(CONFIG_SND_SEQUENCER) += oss/ endif @@ -15,7 +14,6 @@ snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \ snd-seq-midi-objs := seq_midi.o snd-seq-midi-emul-objs := seq_midi_emul.o snd-seq-midi-event-objs := seq_midi_event.o -snd-seq-instr-objs := seq_instr.o snd-seq-dummy-objs := seq_dummy.o snd-seq-virmidi-objs := seq_virmidi.o @@ -36,9 +34,7 @@ obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o # Toplevel Module Dependency obj-$(CONFIG_SND_VIRMIDI) += snd-seq-virmidi.o snd-seq-midi-event.o obj-$(call sequencer,$(CONFIG_SND_RAWMIDI)) += snd-seq-midi.o snd-seq-midi-event.o -obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o snd-seq-instr.o -obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o snd-seq-instr.o -obj-$(call sequencer,$(CONFIG_SND_GUS_SYNTH)) += snd-seq-midi-emul.o snd-seq-instr.o +obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o +obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-seq-midi-emul.o snd-seq-virmidi.o obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-seq-midi-emul.o snd-seq-virmidi.o -obj-$(call sequencer,$(CONFIG_SND_TRIDENT)) += snd-seq-midi-emul.o snd-seq-instr.o diff --git a/sound/core/seq/instr/Makefile b/sound/core/seq/instr/Makefile deleted file mode 100644 index 6089603..0000000 --- a/sound/core/seq/instr/Makefile +++ /dev/null @@ -1,23 +0,0 @@ -# -# Makefile for ALSA -# Copyright (c) 1999 by Jaroslav Kysela -# - -snd-ainstr-fm-objs := ainstr_fm.o -snd-ainstr-simple-objs := ainstr_simple.o -snd-ainstr-gf1-objs := ainstr_gf1.o -snd-ainstr-iw-objs := ainstr_iw.o - -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - -# Toplevel Module Dependency -obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-ainstr-fm.o -obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-ainstr-fm.o -obj-$(call sequencer,$(CONFIG_SND_GUS_SYNTH)) += snd-ainstr-gf1.o snd-ainstr-simple.o snd-ainstr-iw.o -obj-$(call sequencer,$(CONFIG_SND_TRIDENT)) += snd-ainstr-simple.o diff --git a/sound/core/seq/instr/ainstr_fm.c b/sound/core/seq/instr/ainstr_fm.c deleted file mode 100644 index f80fab8..0000000 --- a/sound/core/seq/instr/ainstr_fm.c +++ /dev/null @@ -1,155 +0,0 @@ -/* - * FM (OPL2/3) Instrument routines - * Copyright (c) 2000 Uros Bizjak - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include -#include - -MODULE_AUTHOR("Uros Bizjak "); -MODULE_DESCRIPTION("Advanced Linux Sound Architecture FM Instrument support."); -MODULE_LICENSE("GPL"); - -static int snd_seq_fm_put(void *private_data, struct snd_seq_kinstr *instr, - char __user *instr_data, long len, int atomic, int cmd) -{ - struct fm_instrument *ip; - struct fm_xinstrument ix; - int idx; - - if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE) - return -EINVAL; - /* copy instrument data */ - if (len < (long)sizeof(ix)) - return -EINVAL; - if (copy_from_user(&ix, instr_data, sizeof(ix))) - return -EFAULT; - if (ix.stype != FM_STRU_INSTR) - return -EINVAL; - ip = (struct fm_instrument *)KINSTR_DATA(instr); - ip->share_id[0] = le32_to_cpu(ix.share_id[0]); - ip->share_id[1] = le32_to_cpu(ix.share_id[1]); - ip->share_id[2] = le32_to_cpu(ix.share_id[2]); - ip->share_id[3] = le32_to_cpu(ix.share_id[3]); - ip->type = ix.type; - for (idx = 0; idx < 4; idx++) { - ip->op[idx].am_vib = ix.op[idx].am_vib; - ip->op[idx].ksl_level = ix.op[idx].ksl_level; - ip->op[idx].attack_decay = ix.op[idx].attack_decay; - ip->op[idx].sustain_release = ix.op[idx].sustain_release; - ip->op[idx].wave_select = ix.op[idx].wave_select; - } - for (idx = 0; idx < 2; idx++) { - ip->feedback_connection[idx] = ix.feedback_connection[idx]; - } - ip->echo_delay = ix.echo_delay; - ip->echo_atten = ix.echo_atten; - ip->chorus_spread = ix.chorus_spread; - ip->trnsps = ix.trnsps; - ip->fix_dur = ix.fix_dur; - ip->modes = ix.modes; - ip->fix_key = ix.fix_key; - return 0; -} - -static int snd_seq_fm_get(void *private_data, struct snd_seq_kinstr *instr, - char __user *instr_data, long len, int atomic, - int cmd) -{ - struct fm_instrument *ip; - struct fm_xinstrument ix; - int idx; - - if (cmd != SNDRV_SEQ_INSTR_GET_CMD_FULL) - return -EINVAL; - if (len < (long)sizeof(ix)) - return -ENOMEM; - memset(&ix, 0, sizeof(ix)); - ip = (struct fm_instrument *)KINSTR_DATA(instr); - ix.stype = FM_STRU_INSTR; - ix.share_id[0] = cpu_to_le32(ip->share_id[0]); - ix.share_id[1] = cpu_to_le32(ip->share_id[1]); - ix.share_id[2] = cpu_to_le32(ip->share_id[2]); - ix.share_id[3] = cpu_to_le32(ip->share_id[3]); - ix.type = ip->type; - for (idx = 0; idx < 4; idx++) { - ix.op[idx].am_vib = ip->op[idx].am_vib; - ix.op[idx].ksl_level = ip->op[idx].ksl_level; - ix.op[idx].attack_decay = ip->op[idx].attack_decay; - ix.op[idx].sustain_release = ip->op[idx].sustain_release; - ix.op[idx].wave_select = ip->op[idx].wave_select; - } - for (idx = 0; idx < 2; idx++) { - ix.feedback_connection[idx] = ip->feedback_connection[idx]; - } - if (copy_to_user(instr_data, &ix, sizeof(ix))) - return -EFAULT; - ix.echo_delay = ip->echo_delay; - ix.echo_atten = ip->echo_atten; - ix.chorus_spread = ip->chorus_spread; - ix.trnsps = ip->trnsps; - ix.fix_dur = ip->fix_dur; - ix.modes = ip->modes; - ix.fix_key = ip->fix_key; - return 0; -} - -static int snd_seq_fm_get_size(void *private_data, struct snd_seq_kinstr *instr, - long *size) -{ - *size = sizeof(struct fm_xinstrument); - return 0; -} - -int snd_seq_fm_init(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_ops *next) -{ - memset(ops, 0, sizeof(*ops)); - // ops->private_data = private_data; - ops->add_len = sizeof(struct fm_instrument); - ops->instr_type = SNDRV_SEQ_INSTR_ID_OPL2_3; - ops->put = snd_seq_fm_put; - ops->get = snd_seq_fm_get; - ops->get_size = snd_seq_fm_get_size; - // ops->remove = snd_seq_fm_remove; - // ops->notify = snd_seq_fm_notify; - ops->next = next; - return 0; -} - -/* - * Init part - */ - -static int __init alsa_ainstr_fm_init(void) -{ - return 0; -} - -static void __exit alsa_ainstr_fm_exit(void) -{ -} - -module_init(alsa_ainstr_fm_init) -module_exit(alsa_ainstr_fm_exit) - -EXPORT_SYMBOL(snd_seq_fm_init); diff --git a/sound/core/seq/instr/ainstr_gf1.c b/sound/core/seq/instr/ainstr_gf1.c deleted file mode 100644 index 4940026..0000000 --- a/sound/core/seq/instr/ainstr_gf1.c +++ /dev/null @@ -1,359 +0,0 @@ -/* - * GF1 (GUS) Patch - Instrument routines - * Copyright (c) 1999 by Jaroslav Kysela - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include -#include -#include - -MODULE_AUTHOR("Jaroslav Kysela "); -MODULE_DESCRIPTION("Advanced Linux Sound Architecture GF1 (GUS) Patch support."); -MODULE_LICENSE("GPL"); - -static unsigned int snd_seq_gf1_size(unsigned int size, unsigned int format) -{ - unsigned int result = size; - - if (format & GF1_WAVE_16BIT) - result <<= 1; - if (format & GF1_WAVE_STEREO) - result <<= 1; - return format; -} - -static int snd_seq_gf1_copy_wave_from_stream(struct snd_gf1_ops *ops, - struct gf1_instrument *ip, - char __user **data, - long *len, - int atomic) -{ - struct gf1_wave *wp, *prev; - struct gf1_xwave xp; - int err; - gfp_t gfp_mask; - unsigned int real_size; - - gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL; - if (*len < (long)sizeof(xp)) - return -EINVAL; - if (copy_from_user(&xp, *data, sizeof(xp))) - return -EFAULT; - *data += sizeof(xp); - *len -= sizeof(xp); - wp = kzalloc(sizeof(*wp), gfp_mask); - if (wp == NULL) - return -ENOMEM; - wp->share_id[0] = le32_to_cpu(xp.share_id[0]); - wp->share_id[1] = le32_to_cpu(xp.share_id[1]); - wp->share_id[2] = le32_to_cpu(xp.share_id[2]); - wp->share_id[3] = le32_to_cpu(xp.share_id[3]); - wp->format = le32_to_cpu(xp.format); - wp->size = le32_to_cpu(xp.size); - wp->start = le32_to_cpu(xp.start); - wp->loop_start = le32_to_cpu(xp.loop_start); - wp->loop_end = le32_to_cpu(xp.loop_end); - wp->loop_repeat = le16_to_cpu(xp.loop_repeat); - wp->flags = xp.flags; - wp->sample_rate = le32_to_cpu(xp.sample_rate); - wp->low_frequency = le32_to_cpu(xp.low_frequency); - wp->high_frequency = le32_to_cpu(xp.high_frequency); - wp->root_frequency = le32_to_cpu(xp.root_frequency); - wp->tune = le16_to_cpu(xp.tune); - wp->balance = xp.balance; - memcpy(wp->envelope_rate, xp.envelope_rate, 6); - memcpy(wp->envelope_offset, xp.envelope_offset, 6); - wp->tremolo_sweep = xp.tremolo_sweep; - wp->tremolo_rate = xp.tremolo_rate; - wp->tremolo_depth = xp.tremolo_depth; - wp->vibrato_sweep = xp.vibrato_sweep; - wp->vibrato_rate = xp.vibrato_rate; - wp->vibrato_depth = xp.vibrato_depth; - wp->scale_frequency = le16_to_cpu(xp.scale_frequency); - wp->scale_factor = le16_to_cpu(xp.scale_factor); - real_size = snd_seq_gf1_size(wp->size, wp->format); - if ((long)real_size > *len) { - kfree(wp); - return -ENOMEM; - } - if (ops->put_sample) { - err = ops->put_sample(ops->private_data, wp, - *data, real_size, atomic); - if (err < 0) { - kfree(wp); - return err; - } - } - *data += real_size; - *len -= real_size; - prev = ip->wave; - if (prev) { - while (prev->next) prev = prev->next; - prev->next = wp; - } else { - ip->wave = wp; - } - return 0; -} - -static void snd_seq_gf1_wave_free(struct snd_gf1_ops *ops, - struct gf1_wave *wave, - int atomic) -{ - if (ops->remove_sample) - ops->remove_sample(ops->private_data, wave, atomic); - kfree(wave); -} - -static void snd_seq_gf1_instr_free(struct snd_gf1_ops *ops, - struct gf1_instrument *ip, - int atomic) -{ - struct gf1_wave *wave; - - while ((wave = ip->wave) != NULL) { - ip->wave = wave->next; - snd_seq_gf1_wave_free(ops, wave, atomic); - } -} - -static int snd_seq_gf1_put(void *private_data, struct snd_seq_kinstr *instr, - char __user *instr_data, long len, int atomic, - int cmd) -{ - struct snd_gf1_ops *ops = private_data; - struct gf1_instrument *ip; - struct gf1_xinstrument ix; - int err; - gfp_t gfp_mask; - - if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE) - return -EINVAL; - gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL; - /* copy instrument data */ - if (len < (long)sizeof(ix)) - return -EINVAL; - if (copy_from_user(&ix, instr_data, sizeof(ix))) - return -EFAULT; - if (ix.stype != GF1_STRU_INSTR) - return -EINVAL; - instr_data += sizeof(ix); - len -= sizeof(ix); - ip = (struct gf1_instrument *)KINSTR_DATA(instr); - ip->exclusion = le16_to_cpu(ix.exclusion); - ip->exclusion_group = le16_to_cpu(ix.exclusion_group); - ip->effect1 = ix.effect1; - ip->effect1_depth = ix.effect1_depth; - ip->effect2 = ix.effect2; - ip->effect2_depth = ix.effect2_depth; - /* copy layers */ - while (len > (long)sizeof(__u32)) { - __u32 stype; - - if (copy_from_user(&stype, instr_data, sizeof(stype))) - return -EFAULT; - if (stype != GF1_STRU_WAVE) { - snd_seq_gf1_instr_free(ops, ip, atomic); - return -EINVAL; - } - err = snd_seq_gf1_copy_wave_from_stream(ops, - ip, - &instr_data, - &len, - atomic); - if (err < 0) { - snd_seq_gf1_instr_free(ops, ip, atomic); - return err; - } - } - return 0; -} - -static int snd_seq_gf1_copy_wave_to_stream(struct snd_gf1_ops *ops, - struct gf1_instrument *ip, - char __user **data, - long *len, - int atomic) -{ - struct gf1_wave *wp; - struct gf1_xwave xp; - int err; - unsigned int real_size; - - for (wp = ip->wave; wp; wp = wp->next) { - if (*len < (long)sizeof(xp)) - return -ENOMEM; - memset(&xp, 0, sizeof(xp)); - xp.stype = GF1_STRU_WAVE; - xp.share_id[0] = cpu_to_le32(wp->share_id[0]); - xp.share_id[1] = cpu_to_le32(wp->share_id[1]); - xp.share_id[2] = cpu_to_le32(wp->share_id[2]); - xp.share_id[3] = cpu_to_le32(wp->share_id[3]); - xp.format = cpu_to_le32(wp->format); - xp.size = cpu_to_le32(wp->size); - xp.start = cpu_to_le32(wp->start); - xp.loop_start = cpu_to_le32(wp->loop_start); - xp.loop_end = cpu_to_le32(wp->loop_end); - xp.loop_repeat = cpu_to_le32(wp->loop_repeat); - xp.flags = wp->flags; - xp.sample_rate = cpu_to_le32(wp->sample_rate); - xp.low_frequency = cpu_to_le32(wp->low_frequency); - xp.high_frequency = cpu_to_le32(wp->high_frequency); - xp.root_frequency = cpu_to_le32(wp->root_frequency); - xp.tune = cpu_to_le16(wp->tune); - xp.balance = wp->balance; - memcpy(xp.envelope_rate, wp->envelope_rate, 6); - memcpy(xp.envelope_offset, wp->envelope_offset, 6); - xp.tremolo_sweep = wp->tremolo_sweep; - xp.tremolo_rate = wp->tremolo_rate; - xp.tremolo_depth = wp->tremolo_depth; - xp.vibrato_sweep = wp->vibrato_sweep; - xp.vibrato_rate = wp->vibrato_rate; - xp.vibrato_depth = wp->vibrato_depth; - xp.scale_frequency = cpu_to_le16(wp->scale_frequency); - xp.scale_factor = cpu_to_le16(wp->scale_factor); - if (copy_to_user(*data, &xp, sizeof(xp))) - return -EFAULT; - *data += sizeof(xp); - *len -= sizeof(xp); - real_size = snd_seq_gf1_size(wp->size, wp->format); - if (*len < (long)real_size) - return -ENOMEM; - if (ops->get_sample) { - err = ops->get_sample(ops->private_data, wp, - *data, real_size, atomic); - if (err < 0) - return err; - } - *data += wp->size; - *len -= wp->size; - } - return 0; -} - -static int snd_seq_gf1_get(void *private_data, struct snd_seq_kinstr *instr, - char __user *instr_data, long len, int atomic, - int cmd) -{ - struct snd_gf1_ops *ops = private_data; - struct gf1_instrument *ip; - struct gf1_xinstrument ix; - - if (cmd != SNDRV_SEQ_INSTR_GET_CMD_FULL) - return -EINVAL; - if (len < (long)sizeof(ix)) - return -ENOMEM; - memset(&ix, 0, sizeof(ix)); - ip = (struct gf1_instrument *)KINSTR_DATA(instr); - ix.stype = GF1_STRU_INSTR; - ix.exclusion = cpu_to_le16(ip->exclusion); - ix.exclusion_group = cpu_to_le16(ip->exclusion_group); - ix.effect1 = cpu_to_le16(ip->effect1); - ix.effect1_depth = cpu_to_le16(ip->effect1_depth); - ix.effect2 = ip->effect2; - ix.effect2_depth = ip->effect2_depth; - if (copy_to_user(instr_data, &ix, sizeof(ix))) - return -EFAULT; - instr_data += sizeof(ix); - len -= sizeof(ix); - return snd_seq_gf1_copy_wave_to_stream(ops, - ip, - &instr_data, - &len, - atomic); -} - -static int snd_seq_gf1_get_size(void *private_data, struct snd_seq_kinstr *instr, - long *size) -{ - long result; - struct gf1_instrument *ip; - struct gf1_wave *wp; - - *size = 0; - ip = (struct gf1_instrument *)KINSTR_DATA(instr); - result = sizeof(struct gf1_xinstrument); - for (wp = ip->wave; wp; wp = wp->next) { - result += sizeof(struct gf1_xwave); - result += wp->size; - } - *size = result; - return 0; -} - -static int snd_seq_gf1_remove(void *private_data, - struct snd_seq_kinstr *instr, - int atomic) -{ - struct snd_gf1_ops *ops = private_data; - struct gf1_instrument *ip; - - ip = (struct gf1_instrument *)KINSTR_DATA(instr); - snd_seq_gf1_instr_free(ops, ip, atomic); - return 0; -} - -static void snd_seq_gf1_notify(void *private_data, - struct snd_seq_kinstr *instr, - int what) -{ - struct snd_gf1_ops *ops = private_data; - - if (ops->notify) - ops->notify(ops->private_data, instr, what); -} - -int snd_seq_gf1_init(struct snd_gf1_ops *ops, - void *private_data, - struct snd_seq_kinstr_ops *next) -{ - memset(ops, 0, sizeof(*ops)); - ops->private_data = private_data; - ops->kops.private_data = ops; - ops->kops.add_len = sizeof(struct gf1_instrument); - ops->kops.instr_type = SNDRV_SEQ_INSTR_ID_GUS_PATCH; - ops->kops.put = snd_seq_gf1_put; - ops->kops.get = snd_seq_gf1_get; - ops->kops.get_size = snd_seq_gf1_get_size; - ops->kops.remove = snd_seq_gf1_remove; - ops->kops.notify = snd_seq_gf1_notify; - ops->kops.next = next; - return 0; -} - -/* - * Init part - */ - -static int __init alsa_ainstr_gf1_init(void) -{ - return 0; -} - -static void __exit alsa_ainstr_gf1_exit(void) -{ -} - -module_init(alsa_ainstr_gf1_init) -module_exit(alsa_ainstr_gf1_exit) - -EXPORT_SYMBOL(snd_seq_gf1_init); diff --git a/sound/core/seq/instr/ainstr_iw.c b/sound/core/seq/instr/ainstr_iw.c deleted file mode 100644 index 6c40eb7..0000000 --- a/sound/core/seq/instr/ainstr_iw.c +++ /dev/null @@ -1,623 +0,0 @@ -/* - * IWFFFF - AMD InterWave (tm) - Instrument routines - * Copyright (c) 1999 by Jaroslav Kysela - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include -#include -#include - -MODULE_AUTHOR("Jaroslav Kysela "); -MODULE_DESCRIPTION("Advanced Linux Sound Architecture IWFFFF support."); -MODULE_LICENSE("GPL"); - -static unsigned int snd_seq_iwffff_size(unsigned int size, unsigned int format) -{ - unsigned int result = size; - - if (format & IWFFFF_WAVE_16BIT) - result <<= 1; - if (format & IWFFFF_WAVE_STEREO) - result <<= 1; - return result; -} - -static void snd_seq_iwffff_copy_lfo_from_stream(struct iwffff_lfo *fp, - struct iwffff_xlfo *fx) -{ - fp->freq = le16_to_cpu(fx->freq); - fp->depth = le16_to_cpu(fx->depth); - fp->sweep = le16_to_cpu(fx->sweep); - fp->shape = fx->shape; - fp->delay = fx->delay; -} - -static int snd_seq_iwffff_copy_env_from_stream(__u32 req_stype, - struct iwffff_layer *lp, - struct iwffff_env *ep, - struct iwffff_xenv *ex, - char __user **data, - long *len, - gfp_t gfp_mask) -{ - __u32 stype; - struct iwffff_env_record *rp, *rp_last; - struct iwffff_xenv_record rx; - struct iwffff_env_point *pp; - struct iwffff_xenv_point px; - int points_size, idx; - - ep->flags = ex->flags; - ep->mode = ex->mode; - ep->index = ex->index; - rp_last = NULL; - while (1) { - if (*len < (long)sizeof(__u32)) - return -EINVAL; - if (copy_from_user(&stype, *data, sizeof(stype))) - return -EFAULT; - if (stype == IWFFFF_STRU_WAVE) - return 0; - if (req_stype != stype) { - if (stype == IWFFFF_STRU_ENV_RECP || - stype == IWFFFF_STRU_ENV_RECV) - return 0; - } - if (*len < (long)sizeof(rx)) - return -EINVAL; - if (copy_from_user(&rx, *data, sizeof(rx))) - return -EFAULT; - *data += sizeof(rx); - *len -= sizeof(rx); - points_size = (le16_to_cpu(rx.nattack) + le16_to_cpu(rx.nrelease)) * 2 * sizeof(__u16); - if (points_size > *len) - return -EINVAL; - rp = kzalloc(sizeof(*rp) + points_size, gfp_mask); - if (rp == NULL) - return -ENOMEM; - rp->nattack = le16_to_cpu(rx.nattack); - rp->nrelease = le16_to_cpu(rx.nrelease); - rp->sustain_offset = le16_to_cpu(rx.sustain_offset); - rp->sustain_rate = le16_to_cpu(rx.sustain_rate); - rp->release_rate = le16_to_cpu(rx.release_rate); - rp->hirange = rx.hirange; - pp = (struct iwffff_env_point *)(rp + 1); - for (idx = 0; idx < rp->nattack + rp->nrelease; idx++) { - if (copy_from_user(&px, *data, sizeof(px))) - return -EFAULT; - *data += sizeof(px); - *len -= sizeof(px); - pp->offset = le16_to_cpu(px.offset); - pp->rate = le16_to_cpu(px.rate); - } - if (ep->record == NULL) { - ep->record = rp; - } else { - rp_last = rp; - } - rp_last = rp; - } - return 0; -} - -static int snd_seq_iwffff_copy_wave_from_stream(struct snd_iwffff_ops *ops, - struct iwffff_layer *lp, - char __user **data, - long *len, - int atomic) -{ - struct iwffff_wave *wp, *prev; - struct iwffff_xwave xp; - int err; - gfp_t gfp_mask; - unsigned int real_size; - - gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL; - if (*len < (long)sizeof(xp)) - return -EINVAL; - if (copy_from_user(&xp, *data, sizeof(xp))) - return -EFAULT; - *data += sizeof(xp); - *len -= sizeof(xp); - wp = kzalloc(sizeof(*wp), gfp_mask); - if (wp == NULL) - return -ENOMEM; - wp->share_id[0] = le32_to_cpu(xp.share_id[0]); - wp->share_id[1] = le32_to_cpu(xp.share_id[1]); - wp->share_id[2] = le32_to_cpu(xp.share_id[2]); - wp->share_id[3] = le32_to_cpu(xp.share_id[3]); - wp->format = le32_to_cpu(xp.format); - wp->address.memory = le32_to_cpu(xp.offset); - wp->size = le32_to_cpu(xp.size); - wp->start = le32_to_cpu(xp.start); - wp->loop_start = le32_to_cpu(xp.loop_start); - wp->loop_end = le32_to_cpu(xp.loop_end); - wp->loop_repeat = le16_to_cpu(xp.loop_repeat); - wp->sample_ratio = le32_to_cpu(xp.sample_ratio); - wp->attenuation = xp.attenuation; - wp->low_note = xp.low_note; - wp->high_note = xp.high_note; - real_size = snd_seq_iwffff_size(wp->size, wp->format); - if (!(wp->format & IWFFFF_WAVE_ROM)) { - if ((long)real_size > *len) { - kfree(wp); - return -ENOMEM; - } - } - if (ops->put_sample) { - err = ops->put_sample(ops->private_data, wp, - *data, real_size, atomic); - if (err < 0) { - kfree(wp); - return err; - } - } - if (!(wp->format & IWFFFF_WAVE_ROM)) { - *data += real_size; - *len -= real_size; - } - prev = lp->wave; - if (prev) { - while (prev->next) prev = prev->next; - prev->next = wp; - } else { - lp->wave = wp; - } - return 0; -} - -static void snd_seq_iwffff_env_free(struct snd_iwffff_ops *ops, - struct iwffff_env *env, - int atomic) -{ - struct iwffff_env_record *rec; - - while ((rec = env->record) != NULL) { - env->record = rec->next; - kfree(rec); - } -} - -static void snd_seq_iwffff_wave_free(struct snd_iwffff_ops *ops, - struct iwffff_wave *wave, - int atomic) -{ - if (ops->remove_sample) - ops->remove_sample(ops->private_data, wave, atomic); - kfree(wave); -} - -static void snd_seq_iwffff_instr_free(struct snd_iwffff_ops *ops, - struct iwffff_instrument *ip, - int atomic) -{ - struct iwffff_layer *layer; - struct iwffff_wave *wave; - - while ((layer = ip->layer) != NULL) { - ip->layer = layer->next; - snd_seq_iwffff_env_free(ops, &layer->penv, atomic); - snd_seq_iwffff_env_free(ops, &layer->venv, atomic); - while ((wave = layer->wave) != NULL) { - layer->wave = wave->next; - snd_seq_iwffff_wave_free(ops, wave, atomic); - } - kfree(layer); - } -} - -static int snd_seq_iwffff_put(void *private_data, struct snd_seq_kinstr *instr, - char __user *instr_data, long len, int atomic, - int cmd) -{ - struct snd_iwffff_ops *ops = private_data; - struct iwffff_instrument *ip; - struct iwffff_xinstrument ix; - struct iwffff_layer *lp, *prev_lp; - struct iwffff_xlayer lx; - int err; - gfp_t gfp_mask; - - if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE) - return -EINVAL; - gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL; - /* copy instrument data */ - if (len < (long)sizeof(ix)) - return -EINVAL; - if (copy_from_user(&ix, instr_data, sizeof(ix))) - return -EFAULT; - if (ix.stype != IWFFFF_STRU_INSTR) - return -EINVAL; - instr_data += sizeof(ix); - len -= sizeof(ix); - ip = (struct iwffff_instrument *)KINSTR_DATA(instr); - ip->exclusion = le16_to_cpu(ix.exclusion); - ip->layer_type = le16_to_cpu(ix.layer_type); - ip->exclusion_group = le16_to_cpu(ix.exclusion_group); - ip->effect1 = ix.effect1; - ip->effect1_depth = ix.effect1_depth; - ip->effect2 = ix.effect2; - ip->effect2_depth = ix.effect2_depth; - /* copy layers */ - prev_lp = NULL; - while (len > 0) { - if (len < (long)sizeof(struct iwffff_xlayer)) { - snd_seq_iwffff_instr_free(ops, ip, atomic); - return -EINVAL; - } - if (copy_from_user(&lx, instr_data, sizeof(lx))) - return -EFAULT; - instr_data += sizeof(lx); - len -= sizeof(lx); - if (lx.stype != IWFFFF_STRU_LAYER) { - snd_seq_iwffff_instr_free(ops, ip, atomic); - return -EINVAL; - } - lp = kzalloc(sizeof(*lp), gfp_mask); - if (lp == NULL) { - snd_seq_iwffff_instr_free(ops, ip, atomic); - return -ENOMEM; - } - if (prev_lp) { - prev_lp->next = lp; - } else { - ip->layer = lp; - } - prev_lp = lp; - lp->flags = lx.flags; - lp->velocity_mode = lx.velocity_mode; - lp->layer_event = lx.layer_event; - lp->low_range = lx.low_range; - lp->high_range = lx.high_range; - lp->pan = lx.pan; - lp->pan_freq_scale = lx.pan_freq_scale; - lp->attenuation = lx.attenuation; - snd_seq_iwffff_copy_lfo_from_stream(&lp->tremolo, &lx.tremolo); - snd_seq_iwffff_copy_lfo_from_stream(&lp->vibrato, &lx.vibrato); - lp->freq_scale = le16_to_cpu(lx.freq_scale); - lp->freq_center = lx.freq_center; - err = snd_seq_iwffff_copy_env_from_stream(IWFFFF_STRU_ENV_RECP, - lp, - &lp->penv, &lx.penv, - &instr_data, &len, - gfp_mask); - if (err < 0) { - snd_seq_iwffff_instr_free(ops, ip, atomic); - return err; - } - err = snd_seq_iwffff_copy_env_from_stream(IWFFFF_STRU_ENV_RECV, - lp, - &lp->venv, &lx.venv, - &instr_data, &len, - gfp_mask); - if (err < 0) { - snd_seq_iwffff_instr_free(ops, ip, atomic); - return err; - } - while (len > (long)sizeof(__u32)) { - __u32 stype; - - if (copy_from_user(&stype, instr_data, sizeof(stype))) - return -EFAULT; - if (stype != IWFFFF_STRU_WAVE) - break; - err = snd_seq_iwffff_copy_wave_from_stream(ops, - lp, - &instr_data, - &len, - atomic); - if (err < 0) { - snd_seq_iwffff_instr_free(ops, ip, atomic); - return err; - } - } - } - return 0; -} - -static void snd_seq_iwffff_copy_lfo_to_stream(struct iwffff_xlfo *fx, - struct iwffff_lfo *fp) -{ - fx->freq = cpu_to_le16(fp->freq); - fx->depth = cpu_to_le16(fp->depth); - fx->sweep = cpu_to_le16(fp->sweep); - fp->shape = fx->shape; - fp->delay = fx->delay; -} - -static int snd_seq_iwffff_copy_env_to_stream(__u32 req_stype, - struct iwffff_layer *lp, - struct iwffff_xenv *ex, - struct iwffff_env *ep, - char __user **data, - long *len) -{ - struct iwffff_env_record *rp; - struct iwffff_xenv_record rx; - struct iwffff_env_point *pp; - struct iwffff_xenv_point px; - int points_size, idx; - - ex->flags = ep->flags; - ex->mode = ep->mode; - ex->index = ep->index; - for (rp = ep->record; rp; rp = rp->next) { - if (*len < (long)sizeof(rx)) - return -ENOMEM; - memset(&rx, 0, sizeof(rx)); - rx.stype = req_stype; - rx.nattack = cpu_to_le16(rp->nattack); - rx.nrelease = cpu_to_le16(rp->nrelease); - rx.sustain_offset = cpu_to_le16(rp->sustain_offset); - rx.sustain_rate = cpu_to_le16(rp->sustain_rate); - rx.release_rate = cpu_to_le16(rp->release_rate); - rx.hirange = cpu_to_le16(rp->hirange); - if (copy_to_user(*data, &rx, sizeof(rx))) - return -EFAULT; - *data += sizeof(rx); - *len -= sizeof(rx); - points_size = (rp->nattack + rp->nrelease) * 2 * sizeof(__u16); - if (*len < points_size) - return -ENOMEM; - pp = (struct iwffff_env_point *)(rp + 1); - for (idx = 0; idx < rp->nattack + rp->nrelease; idx++) { - px.offset = cpu_to_le16(pp->offset); - px.rate = cpu_to_le16(pp->rate); - if (copy_to_user(*data, &px, sizeof(px))) - return -EFAULT; - *data += sizeof(px); - *len -= sizeof(px); - } - } - return 0; -} - -static int snd_seq_iwffff_copy_wave_to_stream(struct snd_iwffff_ops *ops, - struct iwffff_layer *lp, - char __user **data, - long *len, - int atomic) -{ - struct iwffff_wave *wp; - struct iwffff_xwave xp; - int err; - unsigned int real_size; - - for (wp = lp->wave; wp; wp = wp->next) { - if (*len < (long)sizeof(xp)) - return -ENOMEM; - memset(&xp, 0, sizeof(xp)); - xp.stype = IWFFFF_STRU_WAVE; - xp.share_id[0] = cpu_to_le32(wp->share_id[0]); - xp.share_id[1] = cpu_to_le32(wp->share_id[1]); - xp.share_id[2] = cpu_to_le32(wp->share_id[2]); - xp.share_id[3] = cpu_to_le32(wp->share_id[3]); - xp.format = cpu_to_le32(wp->format); - if (wp->format & IWFFFF_WAVE_ROM) - xp.offset = cpu_to_le32(wp->address.memory); - xp.size = cpu_to_le32(wp->size); - xp.start = cpu_to_le32(wp->start); - xp.loop_start = cpu_to_le32(wp->loop_start); - xp.loop_end = cpu_to_le32(wp->loop_end); - xp.loop_repeat = cpu_to_le32(wp->loop_repeat); - xp.sample_ratio = cpu_to_le32(wp->sample_ratio); - xp.attenuation = wp->attenuation; - xp.low_note = wp->low_note; - xp.high_note = wp->high_note; - if (copy_to_user(*data, &xp, sizeof(xp))) - return -EFAULT; - *data += sizeof(xp); - *len -= sizeof(xp); - real_size = snd_seq_iwffff_size(wp->size, wp->format); - if (!(wp->format & IWFFFF_WAVE_ROM)) { - if (*len < (long)real_size) - return -ENOMEM; - } - if (ops->get_sample) { - err = ops->get_sample(ops->private_data, wp, - *data, real_size, atomic); - if (err < 0) - return err; - } - if (!(wp->format & IWFFFF_WAVE_ROM)) { - *data += real_size; - *len -= real_size; - } - } - return 0; -} - -static int snd_seq_iwffff_get(void *private_data, struct snd_seq_kinstr *instr, - char __user *instr_data, long len, int atomic, int cmd) -{ - struct snd_iwffff_ops *ops = private_data; - struct iwffff_instrument *ip; - struct iwffff_xinstrument ix; - struct iwffff_layer *lp; - struct iwffff_xlayer lx; - char __user *layer_instr_data; - int err; - - if (cmd != SNDRV_SEQ_INSTR_GET_CMD_FULL) - return -EINVAL; - if (len < (long)sizeof(ix)) - return -ENOMEM; - memset(&ix, 0, sizeof(ix)); - ip = (struct iwffff_instrument *)KINSTR_DATA(instr); - ix.stype = IWFFFF_STRU_INSTR; - ix.exclusion = cpu_to_le16(ip->exclusion); - ix.layer_type = cpu_to_le16(ip->layer_type); - ix.exclusion_group = cpu_to_le16(ip->exclusion_group); - ix.effect1 = cpu_to_le16(ip->effect1); - ix.effect1_depth = cpu_to_le16(ip->effect1_depth); - ix.effect2 = ip->effect2; - ix.effect2_depth = ip->effect2_depth; - if (copy_to_user(instr_data, &ix, sizeof(ix))) - return -EFAULT; - instr_data += sizeof(ix); - len -= sizeof(ix); - for (lp = ip->layer; lp; lp = lp->next) { - if (len < (long)sizeof(lx)) - return -ENOMEM; - memset(&lx, 0, sizeof(lx)); - lx.stype = IWFFFF_STRU_LAYER; - lx.flags = lp->flags; - lx.velocity_mode = lp->velocity_mode; - lx.layer_event = lp->layer_event; - lx.low_range = lp->low_range; - lx.high_range = lp->high_range; - lx.pan = lp->pan; - lx.pan_freq_scale = lp->pan_freq_scale; - lx.attenuation = lp->attenuation; - snd_seq_iwffff_copy_lfo_to_stream(&lx.tremolo, &lp->tremolo); - snd_seq_iwffff_copy_lfo_to_stream(&lx.vibrato, &lp->vibrato); - layer_instr_data = instr_data; - instr_data += sizeof(lx); - len -= sizeof(lx); - err = snd_seq_iwffff_copy_env_to_stream(IWFFFF_STRU_ENV_RECP, - lp, - &lx.penv, &lp->penv, - &instr_data, &len); - if (err < 0) - return err; - err = snd_seq_iwffff_copy_env_to_stream(IWFFFF_STRU_ENV_RECV, - lp, - &lx.venv, &lp->venv, - &instr_data, &len); - if (err < 0) - return err; - /* layer structure updating is now finished */ - if (copy_to_user(layer_instr_data, &lx, sizeof(lx))) - return -EFAULT; - err = snd_seq_iwffff_copy_wave_to_stream(ops, - lp, - &instr_data, - &len, - atomic); - if (err < 0) - return err; - } - return 0; -} - -static long snd_seq_iwffff_env_size_in_stream(struct iwffff_env *ep) -{ - long result = 0; - struct iwffff_env_record *rp; - - for (rp = ep->record; rp; rp = rp->next) { - result += sizeof(struct iwffff_xenv_record); - result += (rp->nattack + rp->nrelease) * 2 * sizeof(__u16); - } - return 0; -} - -static long snd_seq_iwffff_wave_size_in_stream(struct iwffff_layer *lp) -{ - long result = 0; - struct iwffff_wave *wp; - - for (wp = lp->wave; wp; wp = wp->next) { - result += sizeof(struct iwffff_xwave); - if (!(wp->format & IWFFFF_WAVE_ROM)) - result += wp->size; - } - return result; -} - -static int snd_seq_iwffff_get_size(void *private_data, struct snd_seq_kinstr *instr, - long *size) -{ - long result; - struct iwffff_instrument *ip; - struct iwffff_layer *lp; - - *size = 0; - ip = (struct iwffff_instrument *)KINSTR_DATA(instr); - result = sizeof(struct iwffff_xinstrument); - for (lp = ip->layer; lp; lp = lp->next) { - result += sizeof(struct iwffff_xlayer); - result += snd_seq_iwffff_env_size_in_stream(&lp->penv); - result += snd_seq_iwffff_env_size_in_stream(&lp->venv); - result += snd_seq_iwffff_wave_size_in_stream(lp); - } - *size = result; - return 0; -} - -static int snd_seq_iwffff_remove(void *private_data, - struct snd_seq_kinstr *instr, - int atomic) -{ - struct snd_iwffff_ops *ops = private_data; - struct iwffff_instrument *ip; - - ip = (struct iwffff_instrument *)KINSTR_DATA(instr); - snd_seq_iwffff_instr_free(ops, ip, atomic); - return 0; -} - -static void snd_seq_iwffff_notify(void *private_data, - struct snd_seq_kinstr *instr, - int what) -{ - struct snd_iwffff_ops *ops = private_data; - - if (ops->notify) - ops->notify(ops->private_data, instr, what); -} - -int snd_seq_iwffff_init(struct snd_iwffff_ops *ops, - void *private_data, - struct snd_seq_kinstr_ops *next) -{ - memset(ops, 0, sizeof(*ops)); - ops->private_data = private_data; - ops->kops.private_data = ops; - ops->kops.add_len = sizeof(struct iwffff_instrument); - ops->kops.instr_type = SNDRV_SEQ_INSTR_ID_INTERWAVE; - ops->kops.put = snd_seq_iwffff_put; - ops->kops.get = snd_seq_iwffff_get; - ops->kops.get_size = snd_seq_iwffff_get_size; - ops->kops.remove = snd_seq_iwffff_remove; - ops->kops.notify = snd_seq_iwffff_notify; - ops->kops.next = next; - return 0; -} - -/* - * Init part - */ - -static int __init alsa_ainstr_iw_init(void) -{ - return 0; -} - -static void __exit alsa_ainstr_iw_exit(void) -{ -} - -module_init(alsa_ainstr_iw_init) -module_exit(alsa_ainstr_iw_exit) - -EXPORT_SYMBOL(snd_seq_iwffff_init); diff --git a/sound/core/seq/instr/ainstr_simple.c b/sound/core/seq/instr/ainstr_simple.c deleted file mode 100644 index 78f68be..0000000 --- a/sound/core/seq/instr/ainstr_simple.c +++ /dev/null @@ -1,215 +0,0 @@ -/* - * Simple (MOD player) - Instrument routines - * Copyright (c) 1999 by Jaroslav Kysela - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include -#include -#include - -MODULE_AUTHOR("Jaroslav Kysela "); -MODULE_DESCRIPTION("Advanced Linux Sound Architecture Simple Instrument support."); -MODULE_LICENSE("GPL"); - -static unsigned int snd_seq_simple_size(unsigned int size, unsigned int format) -{ - unsigned int result = size; - - if (format & SIMPLE_WAVE_16BIT) - result <<= 1; - if (format & SIMPLE_WAVE_STEREO) - result <<= 1; - return result; -} - -static void snd_seq_simple_instr_free(struct snd_simple_ops *ops, - struct simple_instrument *ip, - int atomic) -{ - if (ops->remove_sample) - ops->remove_sample(ops->private_data, ip, atomic); -} - -static int snd_seq_simple_put(void *private_data, struct snd_seq_kinstr *instr, - char __user *instr_data, long len, - int atomic, int cmd) -{ - struct snd_simple_ops *ops = private_data; - struct simple_instrument *ip; - struct simple_xinstrument ix; - int err; - gfp_t gfp_mask; - unsigned int real_size; - - if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE) - return -EINVAL; - gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL; - /* copy instrument data */ - if (len < (long)sizeof(ix)) - return -EINVAL; - if (copy_from_user(&ix, instr_data, sizeof(ix))) - return -EFAULT; - if (ix.stype != SIMPLE_STRU_INSTR) - return -EINVAL; - instr_data += sizeof(ix); - len -= sizeof(ix); - ip = (struct simple_instrument *)KINSTR_DATA(instr); - ip->share_id[0] = le32_to_cpu(ix.share_id[0]); - ip->share_id[1] = le32_to_cpu(ix.share_id[1]); - ip->share_id[2] = le32_to_cpu(ix.share_id[2]); - ip->share_id[3] = le32_to_cpu(ix.share_id[3]); - ip->format = le32_to_cpu(ix.format); - ip->size = le32_to_cpu(ix.size); - ip->start = le32_to_cpu(ix.start); - ip->loop_start = le32_to_cpu(ix.loop_start); - ip->loop_end = le32_to_cpu(ix.loop_end); - ip->loop_repeat = le16_to_cpu(ix.loop_repeat); - ip->effect1 = ix.effect1; - ip->effect1_depth = ix.effect1_depth; - ip->effect2 = ix.effect2; - ip->effect2_depth = ix.effect2_depth; - real_size = snd_seq_simple_size(ip->size, ip->format); - if (len < (long)real_size) - return -EINVAL; - if (ops->put_sample) { - err = ops->put_sample(ops->private_data, ip, - instr_data, real_size, atomic); - if (err < 0) - return err; - } - return 0; -} - -static int snd_seq_simple_get(void *private_data, struct snd_seq_kinstr *instr, - char __user *instr_data, long len, - int atomic, int cmd) -{ - struct snd_simple_ops *ops = private_data; - struct simple_instrument *ip; - struct simple_xinstrument ix; - int err; - unsigned int real_size; - - if (cmd != SNDRV_SEQ_INSTR_GET_CMD_FULL) - return -EINVAL; - if (len < (long)sizeof(ix)) - return -ENOMEM; - memset(&ix, 0, sizeof(ix)); - ip = (struct simple_instrument *)KINSTR_DATA(instr); - ix.stype = SIMPLE_STRU_INSTR; - ix.share_id[0] = cpu_to_le32(ip->share_id[0]); - ix.share_id[1] = cpu_to_le32(ip->share_id[1]); - ix.share_id[2] = cpu_to_le32(ip->share_id[2]); - ix.share_id[3] = cpu_to_le32(ip->share_id[3]); - ix.format = cpu_to_le32(ip->format); - ix.size = cpu_to_le32(ip->size); - ix.start = cpu_to_le32(ip->start); - ix.loop_start = cpu_to_le32(ip->loop_start); - ix.loop_end = cpu_to_le32(ip->loop_end); - ix.loop_repeat = cpu_to_le32(ip->loop_repeat); - ix.effect1 = cpu_to_le16(ip->effect1); - ix.effect1_depth = cpu_to_le16(ip->effect1_depth); - ix.effect2 = ip->effect2; - ix.effect2_depth = ip->effect2_depth; - if (copy_to_user(instr_data, &ix, sizeof(ix))) - return -EFAULT; - instr_data += sizeof(ix); - len -= sizeof(ix); - real_size = snd_seq_simple_size(ip->size, ip->format); - if (len < (long)real_size) - return -ENOMEM; - if (ops->get_sample) { - err = ops->get_sample(ops->private_data, ip, - instr_data, real_size, atomic); - if (err < 0) - return err; - } - return 0; -} - -static int snd_seq_simple_get_size(void *private_data, struct snd_seq_kinstr *instr, - long *size) -{ - struct simple_instrument *ip; - - ip = (struct simple_instrument *)KINSTR_DATA(instr); - *size = sizeof(struct simple_xinstrument) + snd_seq_simple_size(ip->size, ip->format); - return 0; -} - -static int snd_seq_simple_remove(void *private_data, - struct snd_seq_kinstr *instr, - int atomic) -{ - struct snd_simple_ops *ops = private_data; - struct simple_instrument *ip; - - ip = (struct simple_instrument *)KINSTR_DATA(instr); - snd_seq_simple_instr_free(ops, ip, atomic); - return 0; -} - -static void snd_seq_simple_notify(void *private_data, - struct snd_seq_kinstr *instr, - int what) -{ - struct snd_simple_ops *ops = private_data; - - if (ops->notify) - ops->notify(ops->private_data, instr, what); -} - -int snd_seq_simple_init(struct snd_simple_ops *ops, - void *private_data, - struct snd_seq_kinstr_ops *next) -{ - memset(ops, 0, sizeof(*ops)); - ops->private_data = private_data; - ops->kops.private_data = ops; - ops->kops.add_len = sizeof(struct simple_instrument); - ops->kops.instr_type = SNDRV_SEQ_INSTR_ID_SIMPLE; - ops->kops.put = snd_seq_simple_put; - ops->kops.get = snd_seq_simple_get; - ops->kops.get_size = snd_seq_simple_get_size; - ops->kops.remove = snd_seq_simple_remove; - ops->kops.notify = snd_seq_simple_notify; - ops->kops.next = next; - return 0; -} - -/* - * Init part - */ - -static int __init alsa_ainstr_simple_init(void) -{ - return 0; -} - -static void __exit alsa_ainstr_simple_exit(void) -{ -} - -module_init(alsa_ainstr_simple_init) -module_exit(alsa_ainstr_simple_exit) - -EXPORT_SYMBOL(snd_seq_simple_init); diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 2e3fa25..69421ca 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -966,8 +966,7 @@ static int check_event_type_and_length(struct snd_seq_event *ev) return -EINVAL; break; case SNDRV_SEQ_EVENT_LENGTH_VARUSR: - if (! snd_seq_ev_is_instr_type(ev) || - ! snd_seq_ev_is_direct(ev)) + if (! snd_seq_ev_is_direct(ev)) return -EINVAL; break; } diff --git a/sound/core/seq/seq_instr.c b/sound/core/seq/seq_instr.c deleted file mode 100644 index 9a6fd56..0000000 --- a/sound/core/seq/seq_instr.c +++ /dev/null @@ -1,655 +0,0 @@ -/* - * Generic Instrument routines for ALSA sequencer - * Copyright (c) 1999 by Jaroslav Kysela - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include "seq_clientmgr.h" -#include -#include - -MODULE_AUTHOR("Jaroslav Kysela "); -MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer instrument library."); -MODULE_LICENSE("GPL"); - - -static void snd_instr_lock_ops(struct snd_seq_kinstr_list *list) -{ - if (!(list->flags & SNDRV_SEQ_INSTR_FLG_DIRECT)) { - spin_lock_irqsave(&list->ops_lock, list->ops_flags); - } else { - mutex_lock(&list->ops_mutex); - } -} - -static void snd_instr_unlock_ops(struct snd_seq_kinstr_list *list) -{ - if (!(list->flags & SNDRV_SEQ_INSTR_FLG_DIRECT)) { - spin_unlock_irqrestore(&list->ops_lock, list->ops_flags); - } else { - mutex_unlock(&list->ops_mutex); - } -} - -static struct snd_seq_kinstr *snd_seq_instr_new(int add_len, int atomic) -{ - struct snd_seq_kinstr *instr; - - instr = kzalloc(sizeof(struct snd_seq_kinstr) + add_len, atomic ? GFP_ATOMIC : GFP_KERNEL); - if (instr == NULL) - return NULL; - instr->add_len = add_len; - return instr; -} - -static int snd_seq_instr_free(struct snd_seq_kinstr *instr, int atomic) -{ - int result = 0; - - if (instr == NULL) - return -EINVAL; - if (instr->ops && instr->ops->remove) - result = instr->ops->remove(instr->ops->private_data, instr, 1); - if (!result) - kfree(instr); - return result; -} - -struct snd_seq_kinstr_list *snd_seq_instr_list_new(void) -{ - struct snd_seq_kinstr_list *list; - - list = kzalloc(sizeof(struct snd_seq_kinstr_list), GFP_KERNEL); - if (list == NULL) - return NULL; - spin_lock_init(&list->lock); - spin_lock_init(&list->ops_lock); - mutex_init(&list->ops_mutex); - list->owner = -1; - return list; -} - -void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list_ptr) -{ - struct snd_seq_kinstr_list *list; - struct snd_seq_kinstr *instr; - struct snd_seq_kcluster *cluster; - int idx; - unsigned long flags; - - if (list_ptr == NULL) - return; - list = *list_ptr; - *list_ptr = NULL; - if (list == NULL) - return; - - for (idx = 0; idx < SNDRV_SEQ_INSTR_HASH_SIZE; idx++) { - while ((instr = list->hash[idx]) != NULL) { - list->hash[idx] = instr->next; - list->count--; - spin_lock_irqsave(&list->lock, flags); - while (instr->use) { - spin_unlock_irqrestore(&list->lock, flags); - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&list->lock, flags); - } - spin_unlock_irqrestore(&list->lock, flags); - if (snd_seq_instr_free(instr, 0)<0) - snd_printk(KERN_WARNING "instrument free problem\n"); - } - while ((cluster = list->chash[idx]) != NULL) { - list->chash[idx] = cluster->next; - list->ccount--; - kfree(cluster); - } - } - kfree(list); -} - -static int instr_free_compare(struct snd_seq_kinstr *instr, - struct snd_seq_instr_header *ifree, - unsigned int client) -{ - switch (ifree->cmd) { - case SNDRV_SEQ_INSTR_FREE_CMD_ALL: - /* all, except private for other clients */ - if ((instr->instr.std & 0xff000000) == 0) - return 0; - if (((instr->instr.std >> 24) & 0xff) == client) - return 0; - return 1; - case SNDRV_SEQ_INSTR_FREE_CMD_PRIVATE: - /* all my private instruments */ - if ((instr->instr.std & 0xff000000) == 0) - return 1; - if (((instr->instr.std >> 24) & 0xff) == client) - return 0; - return 1; - case SNDRV_SEQ_INSTR_FREE_CMD_CLUSTER: - /* all my private instruments */ - if ((instr->instr.std & 0xff000000) == 0) { - if (instr->instr.cluster == ifree->id.cluster) - return 0; - return 1; - } - if (((instr->instr.std >> 24) & 0xff) == client) { - if (instr->instr.cluster == ifree->id.cluster) - return 0; - } - return 1; - } - return 1; -} - -int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list, - struct snd_seq_instr_header *ifree, - int client, - int atomic) -{ - struct snd_seq_kinstr *instr, *prev, *next, *flist; - int idx; - unsigned long flags; - - snd_instr_lock_ops(list); - for (idx = 0; idx < SNDRV_SEQ_INSTR_HASH_SIZE; idx++) { - spin_lock_irqsave(&list->lock, flags); - instr = list->hash[idx]; - prev = flist = NULL; - while (instr) { - while (instr && instr_free_compare(instr, ifree, (unsigned int)client)) { - prev = instr; - instr = instr->next; - } - if (instr == NULL) - continue; - if (instr->ops && instr->ops->notify) - instr->ops->notify(instr->ops->private_data, instr, SNDRV_SEQ_INSTR_NOTIFY_REMOVE); - next = instr->next; - if (prev == NULL) { - list->hash[idx] = next; - } else { - prev->next = next; - } - list->count--; - instr->next = flist; - flist = instr; - instr = next; - } - spin_unlock_irqrestore(&list->lock, flags); - while (flist) { - instr = flist; - flist = instr->next; - while (instr->use) { - schedule_timeout_uninterruptible(1); - barrier(); - } - if (snd_seq_instr_free(instr, atomic)<0) - snd_printk(KERN_WARNING "instrument free problem\n"); - instr = next; - } - } - snd_instr_unlock_ops(list); - return 0; -} - -static int compute_hash_instr_key(struct snd_seq_instr *instr) -{ - int result; - - result = instr->bank | (instr->prg << 16); - result += result >> 24; - result += result >> 16; - result += result >> 8; - return result & (SNDRV_SEQ_INSTR_HASH_SIZE-1); -} - -#if 0 -static int compute_hash_cluster_key(snd_seq_instr_cluster_t cluster) -{ - int result; - - result = cluster; - result += result >> 24; - result += result >> 16; - result += result >> 8; - return result & (SNDRV_SEQ_INSTR_HASH_SIZE-1); -} -#endif - -static int compare_instr(struct snd_seq_instr *i1, struct snd_seq_instr *i2, int exact) -{ - if (exact) { - if (i1->cluster != i2->cluster || - i1->bank != i2->bank || - i1->prg != i2->prg) - return 1; - if ((i1->std & 0xff000000) != (i2->std & 0xff000000)) - return 1; - if (!(i1->std & i2->std)) - return 1; - return 0; - } else { - unsigned int client_check; - - if (i2->cluster && i1->cluster != i2->cluster) - return 1; - client_check = i2->std & 0xff000000; - if (client_check) { - if ((i1->std & 0xff000000) != client_check) - return 1; - } else { - if ((i1->std & i2->std) != i2->std) - return 1; - } - return i1->bank != i2->bank || i1->prg != i2->prg; - } -} - -struct snd_seq_kinstr *snd_seq_instr_find(struct snd_seq_kinstr_list *list, - struct snd_seq_instr *instr, - int exact, - int follow_alias) -{ - unsigned long flags; - int depth = 0; - struct snd_seq_kinstr *result; - - if (list == NULL || instr == NULL) - return NULL; - spin_lock_irqsave(&list->lock, flags); - __again: - result = list->hash[compute_hash_instr_key(instr)]; - while (result) { - if (!compare_instr(&result->instr, instr, exact)) { - if (follow_alias && (result->type == SNDRV_SEQ_INSTR_ATYPE_ALIAS)) { - instr = (struct snd_seq_instr *)KINSTR_DATA(result); - if (++depth > 10) - goto __not_found; - goto __again; - } - result->use++; - spin_unlock_irqrestore(&list->lock, flags); - return result; - } - result = result->next; - } - __not_found: - spin_unlock_irqrestore(&list->lock, flags); - return NULL; -} - -void snd_seq_instr_free_use(struct snd_seq_kinstr_list *list, - struct snd_seq_kinstr *instr) -{ - unsigned long flags; - - if (list == NULL || instr == NULL) - return; - spin_lock_irqsave(&list->lock, flags); - if (instr->use <= 0) { - snd_printk(KERN_ERR "free_use: fatal!!! use = %i, name = '%s'\n", instr->use, instr->name); - } else { - instr->use--; - } - spin_unlock_irqrestore(&list->lock, flags); -} - -static struct snd_seq_kinstr_ops *instr_ops(struct snd_seq_kinstr_ops *ops, - char *instr_type) -{ - while (ops) { - if (!strcmp(ops->instr_type, instr_type)) - return ops; - ops = ops->next; - } - return NULL; -} - -static int instr_result(struct snd_seq_event *ev, - int type, int result, - int atomic) -{ - struct snd_seq_event sev; - - memset(&sev, 0, sizeof(sev)); - sev.type = SNDRV_SEQ_EVENT_RESULT; - sev.flags = SNDRV_SEQ_TIME_STAMP_REAL | SNDRV_SEQ_EVENT_LENGTH_FIXED | - SNDRV_SEQ_PRIORITY_NORMAL; - sev.source = ev->dest; - sev.dest = ev->source; - sev.data.result.event = type; - sev.data.result.result = result; -#if 0 - printk("instr result - type = %i, result = %i, queue = %i, source.client:port = %i:%i, dest.client:port = %i:%i\n", - type, result, - sev.queue, - sev.source.client, sev.source.port, - sev.dest.client, sev.dest.port); -#endif - return snd_seq_kernel_client_dispatch(sev.source.client, &sev, atomic, 0); -} - -static int instr_begin(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - unsigned long flags; - - spin_lock_irqsave(&list->lock, flags); - if (list->owner >= 0 && list->owner != ev->source.client) { - spin_unlock_irqrestore(&list->lock, flags); - return instr_result(ev, SNDRV_SEQ_EVENT_INSTR_BEGIN, -EBUSY, atomic); - } - list->owner = ev->source.client; - spin_unlock_irqrestore(&list->lock, flags); - return instr_result(ev, SNDRV_SEQ_EVENT_INSTR_BEGIN, 0, atomic); -} - -static int instr_end(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - unsigned long flags; - - /* TODO: timeout handling */ - spin_lock_irqsave(&list->lock, flags); - if (list->owner == ev->source.client) { - list->owner = -1; - spin_unlock_irqrestore(&list->lock, flags); - return instr_result(ev, SNDRV_SEQ_EVENT_INSTR_END, 0, atomic); - } - spin_unlock_irqrestore(&list->lock, flags); - return instr_result(ev, SNDRV_SEQ_EVENT_INSTR_END, -EINVAL, atomic); -} - -static int instr_info(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - return -ENXIO; -} - -static int instr_format_info(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - return -ENXIO; -} - -static int instr_reset(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - return -ENXIO; -} - -static int instr_status(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - return -ENXIO; -} - -static int instr_put(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - unsigned long flags; - struct snd_seq_instr_header put; - struct snd_seq_kinstr *instr; - int result = -EINVAL, len, key; - - if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARUSR) - goto __return; - - if (ev->data.ext.len < sizeof(struct snd_seq_instr_header)) - goto __return; - if (copy_from_user(&put, (void __user *)ev->data.ext.ptr, - sizeof(struct snd_seq_instr_header))) { - result = -EFAULT; - goto __return; - } - snd_instr_lock_ops(list); - if (put.id.instr.std & 0xff000000) { /* private instrument */ - put.id.instr.std &= 0x00ffffff; - put.id.instr.std |= (unsigned int)ev->source.client << 24; - } - if ((instr = snd_seq_instr_find(list, &put.id.instr, 1, 0))) { - snd_seq_instr_free_use(list, instr); - snd_instr_unlock_ops(list); - result = -EBUSY; - goto __return; - } - ops = instr_ops(ops, put.data.data.format); - if (ops == NULL) { - snd_instr_unlock_ops(list); - goto __return; - } - len = ops->add_len; - if (put.data.type == SNDRV_SEQ_INSTR_ATYPE_ALIAS) - len = sizeof(struct snd_seq_instr); - instr = snd_seq_instr_new(len, atomic); - if (instr == NULL) { - snd_instr_unlock_ops(list); - result = -ENOMEM; - goto __return; - } - instr->ops = ops; - instr->instr = put.id.instr; - strlcpy(instr->name, put.data.name, sizeof(instr->name)); - instr->type = put.data.type; - if (instr->type == SNDRV_SEQ_INSTR_ATYPE_DATA) { - result = ops->put(ops->private_data, - instr, - (void __user *)ev->data.ext.ptr + sizeof(struct snd_seq_instr_header), - ev->data.ext.len - sizeof(struct snd_seq_instr_header), - atomic, - put.cmd); - if (result < 0) { - snd_seq_instr_free(instr, atomic); - snd_instr_unlock_ops(list); - goto __return; - } - } - key = compute_hash_instr_key(&instr->instr); - spin_lock_irqsave(&list->lock, flags); - instr->next = list->hash[key]; - list->hash[key] = instr; - list->count++; - spin_unlock_irqrestore(&list->lock, flags); - snd_instr_unlock_ops(list); - result = 0; - __return: - instr_result(ev, SNDRV_SEQ_EVENT_INSTR_PUT, result, atomic); - return result; -} - -static int instr_get(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - return -ENXIO; -} - -static int instr_free(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - struct snd_seq_instr_header ifree; - struct snd_seq_kinstr *instr, *prev; - int result = -EINVAL; - unsigned long flags; - unsigned int hash; - - if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARUSR) - goto __return; - - if (ev->data.ext.len < sizeof(struct snd_seq_instr_header)) - goto __return; - if (copy_from_user(&ifree, (void __user *)ev->data.ext.ptr, - sizeof(struct snd_seq_instr_header))) { - result = -EFAULT; - goto __return; - } - if (ifree.cmd == SNDRV_SEQ_INSTR_FREE_CMD_ALL || - ifree.cmd == SNDRV_SEQ_INSTR_FREE_CMD_PRIVATE || - ifree.cmd == SNDRV_SEQ_INSTR_FREE_CMD_CLUSTER) { - result = snd_seq_instr_list_free_cond(list, &ifree, ev->dest.client, atomic); - goto __return; - } - if (ifree.cmd == SNDRV_SEQ_INSTR_FREE_CMD_SINGLE) { - if (ifree.id.instr.std & 0xff000000) { - ifree.id.instr.std &= 0x00ffffff; - ifree.id.instr.std |= (unsigned int)ev->source.client << 24; - } - hash = compute_hash_instr_key(&ifree.id.instr); - snd_instr_lock_ops(list); - spin_lock_irqsave(&list->lock, flags); - instr = list->hash[hash]; - prev = NULL; - while (instr) { - if (!compare_instr(&instr->instr, &ifree.id.instr, 1)) - goto __free_single; - prev = instr; - instr = instr->next; - } - result = -ENOENT; - spin_unlock_irqrestore(&list->lock, flags); - snd_instr_unlock_ops(list); - goto __return; - - __free_single: - if (prev) { - prev->next = instr->next; - } else { - list->hash[hash] = instr->next; - } - if (instr->ops && instr->ops->notify) - instr->ops->notify(instr->ops->private_data, instr, - SNDRV_SEQ_INSTR_NOTIFY_REMOVE); - while (instr->use) { - spin_unlock_irqrestore(&list->lock, flags); - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&list->lock, flags); - } - spin_unlock_irqrestore(&list->lock, flags); - result = snd_seq_instr_free(instr, atomic); - snd_instr_unlock_ops(list); - goto __return; - } - - __return: - instr_result(ev, SNDRV_SEQ_EVENT_INSTR_FREE, result, atomic); - return result; -} - -static int instr_list(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - return -ENXIO; -} - -static int instr_cluster(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int atomic, int hop) -{ - return -ENXIO; -} - -int snd_seq_instr_event(struct snd_seq_kinstr_ops *ops, - struct snd_seq_kinstr_list *list, - struct snd_seq_event *ev, - int client, - int atomic, - int hop) -{ - int direct = 0; - - snd_assert(ops != NULL && list != NULL && ev != NULL, return -EINVAL); - if (snd_seq_ev_is_direct(ev)) { - direct = 1; - switch (ev->type) { - case SNDRV_SEQ_EVENT_INSTR_BEGIN: - return instr_begin(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_END: - return instr_end(ops, list, ev, atomic, hop); - } - } - if ((list->flags & SNDRV_SEQ_INSTR_FLG_DIRECT) && !direct) - return -EINVAL; - switch (ev->type) { - case SNDRV_SEQ_EVENT_INSTR_INFO: - return instr_info(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_FINFO: - return instr_format_info(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_RESET: - return instr_reset(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_STATUS: - return instr_status(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_PUT: - return instr_put(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_GET: - return instr_get(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_FREE: - return instr_free(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_LIST: - return instr_list(ops, list, ev, atomic, hop); - case SNDRV_SEQ_EVENT_INSTR_CLUSTER: - return instr_cluster(ops, list, ev, atomic, hop); - } - return -EINVAL; -} - -/* - * Init part - */ - -static int __init alsa_seq_instr_init(void) -{ - return 0; -} - -static void __exit alsa_seq_instr_exit(void) -{ -} - -module_init(alsa_seq_instr_init) -module_exit(alsa_seq_instr_exit) - -EXPORT_SYMBOL(snd_seq_instr_list_new); -EXPORT_SYMBOL(snd_seq_instr_list_free); -EXPORT_SYMBOL(snd_seq_instr_list_free_cond); -EXPORT_SYMBOL(snd_seq_instr_find); -EXPORT_SYMBOL(snd_seq_instr_free_use); -EXPORT_SYMBOL(snd_seq_instr_event); diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 17b3e6f..6645fc5 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -229,13 +229,6 @@ snd_midi_process_event(struct snd_midi_op *ops, case SNDRV_SEQ_EVENT_PORT_START: case SNDRV_SEQ_EVENT_PORT_EXIT: case SNDRV_SEQ_EVENT_PORT_CHANGE: - case SNDRV_SEQ_EVENT_SAMPLE: - case SNDRV_SEQ_EVENT_SAMPLE_START: - case SNDRV_SEQ_EVENT_SAMPLE_STOP: - case SNDRV_SEQ_EVENT_SAMPLE_FREQ: - case SNDRV_SEQ_EVENT_SAMPLE_VOLUME: - case SNDRV_SEQ_EVENT_SAMPLE_LOOP: - case SNDRV_SEQ_EVENT_SAMPLE_POSITION: case SNDRV_SEQ_EVENT_ECHO: not_yet: default: diff --git a/sound/isa/gus/Makefile b/sound/isa/gus/Makefile index df3d59f2..6cd4ee0 100644 --- a/sound/isa/gus/Makefile +++ b/sound/isa/gus/Makefile @@ -9,7 +9,6 @@ snd-gus-lib-objs := gus_main.o \ gus_pcm.o gus_mixer.o \ gus_uart.o \ gus_reset.o -snd-gus-synth-objs := gus_synth.o gus_sample.o gus_simple.o gus_instr.o snd-gusclassic-objs := gusclassic.o snd-gusextreme-objs := gusextreme.o @@ -17,20 +16,9 @@ snd-gusmax-objs := gusmax.o snd-interwave-objs := interwave.o snd-interwave-stb-objs := interwave-stb.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_GUSCLASSIC) += snd-gusclassic.o snd-gus-lib.o obj-$(CONFIG_SND_GUSMAX) += snd-gusmax.o snd-gus-lib.o obj-$(CONFIG_SND_GUSEXTREME) += snd-gusextreme.o snd-gus-lib.o obj-$(CONFIG_SND_INTERWAVE) += snd-interwave.o snd-gus-lib.o obj-$(CONFIG_SND_INTERWAVE_STB) += snd-interwave-stb.o snd-gus-lib.o -obj-$(call sequencer,$(CONFIG_SND_GUS_SYNTH)) += snd-gus-synth.o - -obj-m := $(sort $(obj-m)) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index b14d5d6..e4453e5 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -104,12 +104,6 @@ static int snd_gus_free(struct snd_gus_card *gus) { if (gus->gf1.res_port2 == NULL) goto __hw_end; -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) - if (gus->seq_dev) { - snd_device_free(gus->card, gus->seq_dev); - gus->seq_dev = NULL; - } -#endif snd_gf1_stop(gus); snd_gus_init_dma_irq(gus, 0); __hw_end: @@ -408,14 +402,6 @@ static int snd_gus_check_version(struct snd_gus_card * gus) return 0; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) -static void snd_gus_seq_dev_free(struct snd_seq_device *seq_dev) -{ - struct snd_gus_card *gus = seq_dev->private_data; - gus->seq_dev = NULL; -} -#endif - int snd_gus_initialize(struct snd_gus_card *gus) { int err; @@ -430,15 +416,6 @@ int snd_gus_initialize(struct snd_gus_card *gus) } if ((err = snd_gus_init_dma_irq(gus, 1)) < 0) return err; -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) - if (snd_seq_device_new(gus->card, 1, SNDRV_SEQ_DEV_ID_GUS, - sizeof(struct snd_gus_card *), &gus->seq_dev) >= 0) { - strcpy(gus->seq_dev->name, "GUS"); - *(struct snd_gus_card **)SNDRV_SEQ_DEVICE_ARGPTR(gus->seq_dev) = gus; - gus->seq_dev->private_data = gus; - gus->seq_dev->private_free = snd_gus_seq_dev_free; - } -#endif snd_gf1_start(gus); gus->initialized = 1; return 0; diff --git a/sound/isa/gus/gus_sample.c b/sound/isa/gus/gus_sample.c deleted file mode 100644 index cba0829..0000000 --- a/sound/isa/gus/gus_sample.c +++ /dev/null @@ -1,165 +0,0 @@ -/* - * Routines for Gravis UltraSound soundcards - Sample support - * Copyright (c) by Jaroslav Kysela - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include - -/* - * - */ - -static void select_instrument(struct snd_gus_card * gus, struct snd_gus_voice * v) -{ - struct snd_seq_kinstr *instr; - -#if 0 - printk("select instrument: cluster = %li, std = 0x%x, bank = %i, prg = %i\n", - v->instr.cluster, - v->instr.std, - v->instr.bank, - v->instr.prg); -#endif - instr = snd_seq_instr_find(gus->gf1.ilist, &v->instr, 0, 1); - if (instr != NULL) { - if (instr->ops) { - if (!strcmp(instr->ops->instr_type, SNDRV_SEQ_INSTR_ID_SIMPLE)) - snd_gf1_simple_init(v); - } - snd_seq_instr_free_use(gus->gf1.ilist, instr); - } -} - -/* - * - */ - -static void event_sample(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_stop) - v->sample_ops->sample_stop(p->gus, v, SAMPLE_STOP_IMMEDIATELY); - v->instr.std = ev->data.sample.param.sample.std; - if (v->instr.std & 0xff000000) { /* private instrument */ - v->instr.std &= 0x00ffffff; - v->instr.std |= (unsigned int)ev->source.client << 24; - } - v->instr.bank = ev->data.sample.param.sample.bank; - v->instr.prg = ev->data.sample.param.sample.prg; - select_instrument(p->gus, v); -} - -static void event_cluster(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_stop) - v->sample_ops->sample_stop(p->gus, v, SAMPLE_STOP_IMMEDIATELY); - v->instr.cluster = ev->data.sample.param.cluster.cluster; - select_instrument(p->gus, v); -} - -static void event_start(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_start) - v->sample_ops->sample_start(p->gus, v, ev->data.sample.param.position); -} - -static void event_stop(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_stop) - v->sample_ops->sample_stop(p->gus, v, ev->data.sample.param.stop_mode); -} - -static void event_freq(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_freq) - v->sample_ops->sample_freq(p->gus, v, ev->data.sample.param.frequency); -} - -static void event_volume(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_volume) - v->sample_ops->sample_volume(p->gus, v, &ev->data.sample.param.volume); -} - -static void event_loop(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_loop) - v->sample_ops->sample_loop(p->gus, v, &ev->data.sample.param.loop); -} - -static void event_position(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_pos) - v->sample_ops->sample_pos(p->gus, v, ev->data.sample.param.position); -} - -static void event_private1(struct snd_seq_event *ev, struct snd_gus_port *p, - struct snd_gus_voice *v) -{ - if (v->sample_ops && v->sample_ops->sample_private1) - v->sample_ops->sample_private1(p->gus, v, (unsigned char *)&ev->data.sample.param.raw8); -} - -typedef void (gus_sample_event_handler_t)(struct snd_seq_event *ev, - struct snd_gus_port *p, - struct snd_gus_voice *v); -static gus_sample_event_handler_t *gus_sample_event_handlers[9] = { - event_sample, - event_cluster, - event_start, - event_stop, - event_freq, - event_volume, - event_loop, - event_position, - event_private1 -}; - -void snd_gus_sample_event(struct snd_seq_event *ev, struct snd_gus_port *p) -{ - int idx, voice; - struct snd_gus_card *gus = p->gus; - struct snd_gus_voice *v; - unsigned long flags; - - idx = ev->type - SNDRV_SEQ_EVENT_SAMPLE; - if (idx < 0 || idx > 8) - return; - for (voice = 0; voice < 32; voice++) { - v = &gus->gf1.voices[voice]; - if (v->use && v->client == ev->source.client && - v->port == ev->source.port && - v->index == ev->data.sample.channel) { - spin_lock_irqsave(&gus->event_lock, flags); - gus_sample_event_handlers[idx](ev, p, v); - spin_unlock_irqrestore(&gus->event_lock, flags); - return; - } - } -} diff --git a/sound/isa/gus/gus_simple.c b/sound/isa/gus/gus_simple.c deleted file mode 100644 index 39d121e..0000000 --- a/sound/isa/gus/gus_simple.c +++ /dev/null @@ -1,634 +0,0 @@ -/* - * Routines for Gravis UltraSound soundcards - Simple instrument handlers - * Copyright (c) by Jaroslav Kysela - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include "gus_tables.h" - -/* - * - */ - -static void interrupt_wave(struct snd_gus_card *gus, struct snd_gus_voice *voice); -static void interrupt_volume(struct snd_gus_card *gus, struct snd_gus_voice *voice); -static void interrupt_effect(struct snd_gus_card *gus, struct snd_gus_voice *voice); - -static void sample_start(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_position_t position); -static void sample_stop(struct snd_gus_card *gus, struct snd_gus_voice *voice, int mode); -static void sample_freq(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_frequency_t freq); -static void sample_volume(struct snd_gus_card *card, struct snd_gus_voice *voice, struct snd_seq_ev_volume *volume); -static void sample_loop(struct snd_gus_card *card, struct snd_gus_voice *voice, struct snd_seq_ev_loop *loop); -static void sample_pos(struct snd_gus_card *card, struct snd_gus_voice *voice, snd_seq_position_t position); -static void sample_private1(struct snd_gus_card *card, struct snd_gus_voice *voice, unsigned char *data); - -static struct snd_gus_sample_ops sample_ops = { - sample_start, - sample_stop, - sample_freq, - sample_volume, - sample_loop, - sample_pos, - sample_private1 -}; - -#if 0 - -static void note_stop(struct snd_gus_card *gus, struct snd_gus_voice *voice, int wait); -static void note_wait(struct snd_gus_card *gus, struct snd_gus_voice *voice); -static void note_off(struct snd_gus_card *gus, struct snd_gus_voice *voice); -static void note_volume(struct snd_gus_card *card, struct snd_gus_voice *voice); -static void note_pitchbend(struct snd_gus_card *card, struct snd_gus_voice *voice); -static void note_vibrato(struct snd_gus_card *card, struct snd_gus_voice *voice); -static void note_tremolo(struct snd_gus_card *card, struct snd_gus_voice *voice); - -static struct snd_gus_note_handlers note_commands = { - note_stop, - note_wait, - note_off, - note_volume, - note_pitchbend, - note_vibrato, - note_tremolo -}; - -static void chn_trigger_down(struct snd_gus_card *card, ultra_channel_t *channel, ultra_instrument_t *instrument, unsigned char note, unsigned char velocity, unsigned char priority ); -static void chn_trigger_up( ultra_card_t *card, ultra_note_t *note ); -static void chn_control( ultra_card_t *card, ultra_channel_t *channel, unsigned short p1, unsigned short p2 ); - -static struct ULTRA_STRU_INSTRUMENT_CHANNEL_COMMANDS channel_commands = { - chn_trigger_down, - chn_trigger_up, - chn_control -}; - -#endif - -static void do_volume_envelope(struct snd_gus_card *card, struct snd_gus_voice *voice); -static void do_pan_envelope(struct snd_gus_card *card, struct snd_gus_voice *voice); - -/* - * - */ - -static void interrupt_wave(struct snd_gus_card *gus, struct snd_gus_voice *voice) -{ - spin_lock(&gus->event_lock); - snd_gf1_stop_voice(gus, voice->number); - spin_lock(&gus->reg_lock); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, 0); - spin_unlock(&gus->reg_lock); - voice->flags &= ~SNDRV_GF1_VFLG_RUNNING; - spin_unlock(&gus->event_lock); -} - -static void interrupt_volume(struct snd_gus_card *gus, struct snd_gus_voice *voice) -{ - spin_lock(&gus->event_lock); - if (voice->flags & SNDRV_GF1_VFLG_RUNNING) - do_volume_envelope(gus, voice); - else - snd_gf1_stop_voice(gus, voice->number); - spin_unlock(&gus->event_lock); -} - -static void interrupt_effect(struct snd_gus_card *gus, struct snd_gus_voice *voice) -{ - spin_lock(&gus->event_lock); - if ((voice->flags & (SNDRV_GF1_VFLG_RUNNING|SNDRV_GF1_VFLG_EFFECT_TIMER1)) == - (SNDRV_GF1_VFLG_RUNNING|SNDRV_GF1_VFLG_EFFECT_TIMER1)) - do_pan_envelope(gus, voice); - spin_unlock(&gus->event_lock); -} - -/* - * - */ - -static void do_volume_envelope(struct snd_gus_card *gus, struct snd_gus_voice *voice) -{ - unsigned short next, rate, old_volume; - int program_next_ramp; - unsigned long flags; - - if (!gus->gf1.volume_ramp) { - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); - snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, voice->gf1_volume); - /* printk("gf1_volume = 0x%x\n", voice->gf1_volume); */ - spin_unlock_irqrestore(&gus->reg_lock, flags); - return; - } - program_next_ramp = 0; - rate = next = 0; - while (1) { - program_next_ramp = 0; - rate = next = 0; - switch (voice->venv_state) { - case VENV_BEFORE: - voice->venv_state = VENV_ATTACK; - voice->venv_value_next = 0; - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); - snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, SNDRV_GF1_MIN_VOLUME); - spin_unlock_irqrestore(&gus->reg_lock, flags); - break; - case VENV_ATTACK: - voice->venv_state = VENV_SUSTAIN; - program_next_ramp++; - next = 255; - rate = gus->gf1.volume_ramp; - break; - case VENV_SUSTAIN: - voice->venv_state = VENV_RELEASE; - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); - snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, ((int)voice->gf1_volume * (int)voice->venv_value_next) / 255); - spin_unlock_irqrestore(&gus->reg_lock, flags); - return; - case VENV_RELEASE: - voice->venv_state = VENV_DONE; - program_next_ramp++; - next = 0; - rate = gus->gf1.volume_ramp; - break; - case VENV_DONE: - snd_gf1_stop_voice(gus, voice->number); - voice->flags &= ~SNDRV_GF1_VFLG_RUNNING; - return; - case VENV_VOLUME: - program_next_ramp++; - next = voice->venv_value_next; - rate = gus->gf1.volume_ramp; - voice->venv_state = voice->venv_state_prev; - break; - } - voice->venv_value_next = next; - if (!program_next_ramp) - continue; - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); - old_volume = snd_gf1_read16(gus, SNDRV_GF1_VW_VOLUME) >> 8; - if (!rate) { - spin_unlock_irqrestore(&gus->reg_lock, flags); - continue; - } - next = (((int)voice->gf1_volume * (int)next) / 255) >> 8; - if (old_volume < SNDRV_GF1_MIN_OFFSET) - old_volume = SNDRV_GF1_MIN_OFFSET; - if (next < SNDRV_GF1_MIN_OFFSET) - next = SNDRV_GF1_MIN_OFFSET; - if (next > SNDRV_GF1_MAX_OFFSET) - next = SNDRV_GF1_MAX_OFFSET; - if (old_volume == next) { - spin_unlock_irqrestore(&gus->reg_lock, flags); - continue; - } - voice->volume_control &= ~0xc3; - voice->volume_control |= 0x20; - if (old_volume > next) { - snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_START, next); - snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_END, old_volume); - voice->volume_control |= 0x40; - } else { - snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_START, old_volume); - snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_END, next); - } - snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_RATE, rate); - snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, voice->volume_control); - if (!gus->gf1.enh_mode) { - snd_gf1_delay(gus); - snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, voice->volume_control); - } - spin_unlock_irqrestore(&gus->reg_lock, flags); - return; - } -} - -static void do_pan_envelope(struct snd_gus_card *gus, struct snd_gus_voice *voice) -{ - unsigned long flags; - unsigned char old_pan; - -#if 0 - snd_gf1_select_voice(gus, voice->number); - printk(" -%i- do_pan_envelope - flags = 0x%x (0x%x -> 0x%x)\n", - voice->number, - voice->flags, - voice->gf1_pan, - snd_gf1_i_read8(gus, SNDRV_GF1_VB_PAN) & 0x0f); -#endif - if (gus->gf1.enh_mode) { - voice->flags &= ~(SNDRV_GF1_VFLG_EFFECT_TIMER1|SNDRV_GF1_VFLG_PAN); - return; - } - if (!gus->gf1.smooth_pan) { - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_write8(gus, SNDRV_GF1_VB_PAN, voice->gf1_pan); - spin_unlock_irqrestore(&gus->reg_lock, flags); - return; - } - if (!(voice->flags & SNDRV_GF1_VFLG_PAN)) /* before */ - voice->flags |= SNDRV_GF1_VFLG_EFFECT_TIMER1|SNDRV_GF1_VFLG_PAN; - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - old_pan = snd_gf1_read8(gus, SNDRV_GF1_VB_PAN) & 0x0f; - if (old_pan > voice->gf1_pan ) - old_pan--; - if (old_pan < voice->gf1_pan) - old_pan++; - snd_gf1_write8(gus, SNDRV_GF1_VB_PAN, old_pan); - spin_unlock_irqrestore(&gus->reg_lock, flags); - if (old_pan == voice->gf1_pan) /* the goal was reached */ - voice->flags &= ~(SNDRV_GF1_VFLG_EFFECT_TIMER1|SNDRV_GF1_VFLG_PAN); -#if 0 - snd_gf1_select_voice(gus, voice->number); - printk(" -%i- (1) do_pan_envelope - flags = 0x%x (0x%x -> 0x%x)\n", - voice->number, - voice->flags, - voice->gf1_pan, - snd_gf1_i_read8(gus, GF1_VB_PAN) & 0x0f); -#endif -} - -static void set_enhanced_pan(struct snd_gus_card *gus, struct snd_gus_voice *voice, unsigned short pan) -{ - unsigned long flags; - unsigned short vlo, vro; - - vlo = SNDRV_GF1_ATTEN((SNDRV_GF1_ATTEN_TABLE_SIZE-1) - pan); - vro = SNDRV_GF1_ATTEN(pan); - if (pan != SNDRV_GF1_ATTEN_TABLE_SIZE - 1 && pan != 0) { - vlo >>= 1; - vro >>= 1; - } - vlo <<= 4; - vro <<= 4; -#if 0 - printk("vlo = 0x%x (0x%x), vro = 0x%x (0x%x)\n", - vlo, snd_gf1_i_read16(gus, GF1_VW_OFFSET_LEFT), - vro, snd_gf1_i_read16(gus, GF1_VW_OFFSET_RIGHT)); -#endif - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_write16(gus, SNDRV_GF1_VW_OFFSET_LEFT_FINAL, vlo); - snd_gf1_write16(gus, SNDRV_GF1_VW_OFFSET_RIGHT_FINAL, vro); - spin_unlock_irqrestore(&gus->reg_lock, flags); - voice->vlo = vlo; - voice->vro = vro; -} - -/* - * - */ - -static void sample_start(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_position_t position) -{ - unsigned long flags; - unsigned int begin, addr, addr_end, addr_start; - int w_16; - struct simple_instrument *simple; - struct snd_seq_kinstr *instr; - - instr = snd_seq_instr_find(gus->gf1.ilist, &voice->instr, 0, 1); - if (instr == NULL) - return; - voice->instr = instr->instr; /* copy ID to speedup aliases */ - simple = KINSTR_DATA(instr); - begin = simple->address.memory << 4; - w_16 = simple->format & SIMPLE_WAVE_16BIT ? 0x04 : 0; - addr_start = simple->loop_start; - if (simple->format & SIMPLE_WAVE_LOOP) { - addr_end = simple->loop_end; - } else { - addr_end = (simple->size << 4) - (w_16 ? 40 : 24); - } - if (simple->format & SIMPLE_WAVE_BACKWARD) { - addr = simple->loop_end; - if (position < simple->loop_end) - addr -= position; - } else { - addr = position; - } - voice->control = 0x00; - voice->mode = 0x20; /* enable offset registers */ - if (simple->format & SIMPLE_WAVE_16BIT) - voice->control |= 0x04; - if (simple->format & SIMPLE_WAVE_BACKWARD) - voice->control |= 0x40; - if (simple->format & SIMPLE_WAVE_LOOP) { - voice->control |= 0x08; - } else { - voice->control |= 0x20; - } - if (simple->format & SIMPLE_WAVE_BIDIR) - voice->control |= 0x10; - if (simple->format & SIMPLE_WAVE_ULAW) - voice->mode |= 0x40; - if (w_16) { - addr = ((addr << 1) & ~0x1f) | (addr & 0x0f); - addr_start = ((addr_start << 1) & ~0x1f) | (addr_start & 0x0f); - addr_end = ((addr_end << 1) & ~0x1f) | (addr_end & 0x0f); - } - addr += begin; - addr_start += begin; - addr_end += begin; - snd_gf1_stop_voice(gus, voice->number); - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_write16(gus, SNDRV_GF1_VW_FREQUENCY, voice->fc_register + voice->fc_lfo); - voice->venv_state = VENV_BEFORE; - voice->volume_control = 0x03; - snd_gf1_write_addr(gus, SNDRV_GF1_VA_START, addr_start, w_16); - snd_gf1_write_addr(gus, SNDRV_GF1_VA_END, addr_end, w_16); - snd_gf1_write_addr(gus, SNDRV_GF1_VA_CURRENT, addr, w_16); - if (!gus->gf1.enh_mode) { - snd_gf1_write8(gus, SNDRV_GF1_VB_PAN, voice->gf1_pan); - } else { - snd_gf1_write16(gus, SNDRV_GF1_VW_OFFSET_LEFT, voice->vlo); - snd_gf1_write16(gus, SNDRV_GF1_VW_OFFSET_LEFT_FINAL, voice->vlo); - snd_gf1_write16(gus, SNDRV_GF1_VW_OFFSET_RIGHT, voice->vro); - snd_gf1_write16(gus, SNDRV_GF1_VW_OFFSET_RIGHT_FINAL, voice->vro); - snd_gf1_write8(gus, SNDRV_GF1_VB_ACCUMULATOR, voice->effect_accumulator); - snd_gf1_write16(gus, SNDRV_GF1_VW_EFFECT_VOLUME, voice->gf1_effect_volume); - snd_gf1_write16(gus, SNDRV_GF1_VW_EFFECT_VOLUME_FINAL, voice->gf1_effect_volume); - } - spin_unlock_irqrestore(&gus->reg_lock, flags); - do_volume_envelope(gus, voice); - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - if (gus->gf1.enh_mode) - snd_gf1_write8(gus, SNDRV_GF1_VB_MODE, voice->mode); - snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, voice->control); - if (!gus->gf1.enh_mode) { - snd_gf1_delay(gus); - snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, voice->control ); - } - spin_unlock_irqrestore(&gus->reg_lock, flags); -#if 0 - snd_gf1_print_voice_registers(gus); -#endif - voice->flags |= SNDRV_GF1_VFLG_RUNNING; - snd_seq_instr_free_use(gus->gf1.ilist, instr); -} - -static void sample_stop(struct snd_gus_card *gus, struct snd_gus_voice *voice, int mode) -{ - unsigned char control; - unsigned long flags; - - if (!(voice->flags & SNDRV_GF1_VFLG_RUNNING)) - return; - switch (mode) { - default: - if (gus->gf1.volume_ramp > 0) { - if (voice->venv_state < VENV_RELEASE) { - voice->venv_state = VENV_RELEASE; - do_volume_envelope(gus, voice); - } - } - if (mode != SAMPLE_STOP_VENVELOPE) { - snd_gf1_stop_voice(gus, voice->number); - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, SNDRV_GF1_MIN_VOLUME); - spin_unlock_irqrestore(&gus->reg_lock, flags); - voice->flags &= ~SNDRV_GF1_VFLG_RUNNING; - } - break; - case SAMPLE_STOP_LOOP: /* disable loop only */ - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - control = snd_gf1_read8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL); - control &= ~(0x83 | 0x04); - control |= 0x20; - snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, control); - spin_unlock_irqrestore(&gus->reg_lock, flags); - break; - } -} - -static void sample_freq(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_frequency_t freq) -{ - unsigned long flags; - - spin_lock_irqsave(&gus->reg_lock, flags); - voice->fc_register = snd_gf1_translate_freq(gus, freq); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_write16(gus, SNDRV_GF1_VW_FREQUENCY, voice->fc_register + voice->fc_lfo); - spin_unlock_irqrestore(&gus->reg_lock, flags); -} - -static void sample_volume(struct snd_gus_card *gus, struct snd_gus_voice *voice, struct snd_seq_ev_volume *volume) -{ - if (volume->volume >= 0) { - volume->volume &= 0x3fff; - voice->gf1_volume = snd_gf1_lvol_to_gvol_raw(volume->volume << 2) << 4; - voice->venv_state_prev = VENV_SUSTAIN; - voice->venv_state = VENV_VOLUME; - do_volume_envelope(gus, voice); - } - if (volume->lr >= 0) { - volume->lr &= 0x3fff; - if (!gus->gf1.enh_mode) { - voice->gf1_pan = (volume->lr >> 10) & 15; - if (!gus->gf1.full_range_pan) { - if (voice->gf1_pan == 0) - voice->gf1_pan++; - if (voice->gf1_pan == 15) - voice->gf1_pan--; - } - voice->flags &= ~SNDRV_GF1_VFLG_PAN; /* before */ - do_pan_envelope(gus, voice); - } else { - set_enhanced_pan(gus, voice, volume->lr >> 7); - } - } -} - -static void sample_loop(struct snd_gus_card *gus, struct snd_gus_voice *voice, struct snd_seq_ev_loop *loop) -{ - unsigned long flags; - int w_16 = voice->control & 0x04; - unsigned int begin, addr_start, addr_end; - struct simple_instrument *simple; - struct snd_seq_kinstr *instr; - -#if 0 - printk("voice_loop: start = 0x%x, end = 0x%x\n", loop->start, loop->end); -#endif - instr = snd_seq_instr_find(gus->gf1.ilist, &voice->instr, 0, 1); - if (instr == NULL) - return; - voice->instr = instr->instr; /* copy ID to speedup aliases */ - simple = KINSTR_DATA(instr); - begin = simple->address.memory; - addr_start = loop->start; - addr_end = loop->end; - addr_start = (((addr_start << 1) & ~0x1f) | (addr_start & 0x0f)) + begin; - addr_end = (((addr_end << 1) & ~0x1f) | (addr_end & 0x0f)) + begin; - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_write_addr(gus, SNDRV_GF1_VA_START, addr_start, w_16); - snd_gf1_write_addr(gus, SNDRV_GF1_VA_END, addr_end, w_16); - spin_unlock_irqrestore(&gus->reg_lock, flags); - snd_seq_instr_free_use(gus->gf1.ilist, instr); -} - -static void sample_pos(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_position_t position) -{ - unsigned long flags; - int w_16 = voice->control & 0x04; - unsigned int begin, addr; - struct simple_instrument *simple; - struct snd_seq_kinstr *instr; - -#if 0 - printk("voice_loop: start = 0x%x, end = 0x%x\n", loop->start, loop->end); -#endif - instr = snd_seq_instr_find(gus->gf1.ilist, &voice->instr, 0, 1); - if (instr == NULL) - return; - voice->instr = instr->instr; /* copy ID to speedup aliases */ - simple = KINSTR_DATA(instr); - begin = simple->address.memory; - addr = (((position << 1) & ~0x1f) | (position & 0x0f)) + begin; - spin_lock_irqsave(&gus->reg_lock, flags); - snd_gf1_select_voice(gus, voice->number); - snd_gf1_write_addr(gus, SNDRV_GF1_VA_CURRENT, addr, w_16); - spin_unlock_irqrestore(&gus->reg_lock, flags); - snd_seq_instr_free_use(gus->gf1.ilist, instr); -} - -#if 0 - -static unsigned char get_effects_mask( ultra_card_t *card, int value ) -{ - if ( value > 7 ) return 0; - if ( card -> gf1.effects && card -> gf1.effects -> chip_type == ULTRA_EFFECT_CHIP_INTERWAVE ) - return card -> gf1.effects -> chip.interwave.voice_output[ value ]; - return 0; -} - -#endif - -static void sample_private1(struct snd_gus_card *card, struct snd_gus_voice *voice, unsigned char *data) -{ -#if 0 - unsigned long flags; - unsigned char uc; - - switch ( *data ) { - case ULTRA_PRIV1_IW_EFFECT: - uc = get_effects_mask( card, ultra_get_byte( data, 4 ) ); - uc |= get_effects_mask( card, ultra_get_byte( data, 4 ) >> 4 ); - uc |= get_effects_mask( card, ultra_get_byte( data, 5 ) ); - uc |= get_effects_mask( card, ultra_get_byte( data, 5 ) >> 4 ); - voice -> data.simple.effect_accumulator = uc; - voice -> data.simple.effect_volume = ultra_translate_voice_volume( card, ultra_get_word( data, 2 ) ) << 4; - if ( !card -> gf1.enh_mode ) return; - if ( voice -> flags & VFLG_WAIT_FOR_START ) return; - if ( voice -> flags & VFLG_RUNNING ) - { - CLI( &flags ); - gf1_select_voice( card, voice -> number ); - ultra_write8( card, GF1_VB_ACCUMULATOR, voice -> data.simple.effect_accumulator ); - ultra_write16( card, GF1_VW_EFFECT_VOLUME_FINAL, voice -> data.simple.effect_volume ); - STI( &flags ); - } - break; - case ULTRA_PRIV1_IW_LFO: - ultra_lfo_command( card, voice -> number, data ); - } -#endif -} - -#if 0 - -/* - * - */ - -static void note_stop( ultra_card_t *card, ultra_voice_t *voice, int wait ) -{ -} - -static void note_wait( ultra_card_t *card, ultra_voice_t *voice ) -{ -} - -static void note_off( ultra_card_t *card, ultra_voice_t *voice ) -{ -} - -static void note_volume( ultra_card_t *card, ultra_voice_t *voice ) -{ -} - -static void note_pitchbend( ultra_card_t *card, ultra_voice_t *voice ) -{ -} - -static void note_vibrato( ultra_card_t *card, ultra_voice_t *voice ) -{ -} - -static void note_tremolo( ultra_card_t *card, ultra_voice_t *voice ) -{ -} - -/* - * - */ - -static void chn_trigger_down( ultra_card_t *card, ultra_channel_t *channel, ultra_instrument_t *instrument, unsigned char note, unsigned char velocity, unsigned char priority ) -{ -} - -static void chn_trigger_up( ultra_card_t *card, ultra_note_t *note ) -{ -} - -static void chn_control( ultra_card_t *card, ultra_channel_t *channel, unsigned short p1, unsigned short p2 ) -{ -} - -/* - * - */ - -#endif - -void snd_gf1_simple_init(struct snd_gus_voice *voice) -{ - voice->handler_wave = interrupt_wave; - voice->handler_volume = interrupt_volume; - voice->handler_effect = interrupt_effect; - voice->volume_change = NULL; - voice->sample_ops = &sample_ops; -} diff --git a/sound/isa/gus/gus_synth.c b/sound/isa/gus/gus_synth.c deleted file mode 100644 index 2c20517..0000000 --- a/sound/isa/gus/gus_synth.c +++ /dev/null @@ -1,314 +0,0 @@ -/* - * Routines for Gravis UltraSound soundcards - Synthesizer - * Copyright (c) by Jaroslav Kysela - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include -#include - -MODULE_AUTHOR("Jaroslav Kysela "); -MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards - Synthesizer"); -MODULE_LICENSE("GPL"); - -/* - * - */ - -static void snd_gus_synth_free_voices(struct snd_gus_card * gus, int client, int port) -{ - int idx; - struct snd_gus_voice * voice; - - for (idx = 0; idx < 32; idx++) { - voice = &gus->gf1.voices[idx]; - if (voice->use && voice->client == client && voice->port == port) - snd_gf1_free_voice(gus, voice); - } -} - -static int snd_gus_synth_use(void *private_data, struct snd_seq_port_subscribe *info) -{ - struct snd_gus_port * port = private_data; - struct snd_gus_card * gus = port->gus; - struct snd_gus_voice * voice; - unsigned int idx; - - if (info->voices > 32) - return -EINVAL; - mutex_lock(&gus->register_mutex); - if (!snd_gus_use_inc(gus)) { - mutex_unlock(&gus->register_mutex); - return -EFAULT; - } - for (idx = 0; idx < info->voices; idx++) { - voice = snd_gf1_alloc_voice(gus, SNDRV_GF1_VOICE_TYPE_SYNTH, info->sender.client, info->sender.port); - if (voice == NULL) { - snd_gus_synth_free_voices(gus, info->sender.client, info->sender.port); - snd_gus_use_dec(gus); - mutex_unlock(&gus->register_mutex); - return -EBUSY; - } - voice->index = idx; - } - mutex_unlock(&gus->register_mutex); - return 0; -} - -static int snd_gus_synth_unuse(void *private_data, struct snd_seq_port_subscribe *info) -{ - struct snd_gus_port * port = private_data; - struct snd_gus_card * gus = port->gus; - - mutex_lock(&gus->register_mutex); - snd_gus_synth_free_voices(gus, info->sender.client, info->sender.port); - snd_gus_use_dec(gus); - mutex_unlock(&gus->register_mutex); - return 0; -} - -/* - * - */ - -static void snd_gus_synth_free_private_instruments(struct snd_gus_port *p, int client) -{ - struct snd_seq_instr_header ifree; - - memset(&ifree, 0, sizeof(ifree)); - ifree.cmd = SNDRV_SEQ_INSTR_FREE_CMD_PRIVATE; - snd_seq_instr_list_free_cond(p->gus->gf1.ilist, &ifree, client, 0); -} - -static int snd_gus_synth_event_input(struct snd_seq_event *ev, int direct, - void *private_data, int atomic, int hop) -{ - struct snd_gus_port * p = private_data; - - snd_assert(p != NULL, return -EINVAL); - if (ev->type >= SNDRV_SEQ_EVENT_SAMPLE && - ev->type <= SNDRV_SEQ_EVENT_SAMPLE_PRIVATE1) { - snd_gus_sample_event(ev, p); - return 0; - } - if (ev->source.client == SNDRV_SEQ_CLIENT_SYSTEM && - ev->source.port == SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE) { - if (ev->type == SNDRV_SEQ_EVENT_CLIENT_EXIT) { - snd_gus_synth_free_private_instruments(p, ev->data.addr.client); - return 0; - } - } - if (direct) { - if (ev->type >= SNDRV_SEQ_EVENT_INSTR_BEGIN) { - snd_seq_instr_event(&p->gus->gf1.iwffff_ops.kops, - p->gus->gf1.ilist, - ev, - p->gus->gf1.seq_client, - atomic, hop); - return 0; - } - } - return 0; -} - -static void snd_gus_synth_instr_notify(void *private_data, - struct snd_seq_kinstr *instr, - int what) -{ - unsigned int idx; - struct snd_gus_card *gus = private_data; - struct snd_gus_voice *pvoice; - unsigned long flags; - - spin_lock_irqsave(&gus->event_lock, flags); - for (idx = 0; idx < 32; idx++) { - pvoice = &gus->gf1.voices[idx]; - if (pvoice->use && !memcmp(&pvoice->instr, &instr->instr, sizeof(pvoice->instr))) { - if (pvoice->sample_ops && pvoice->sample_ops->sample_stop) { - pvoice->sample_ops->sample_stop(gus, pvoice, SAMPLE_STOP_IMMEDIATELY); - } else { - snd_gf1_stop_voice(gus, pvoice->number); - pvoice->flags &= ~SNDRV_GF1_VFLG_RUNNING; - } - } - } - spin_unlock_irqrestore(&gus->event_lock, flags); -} - -/* - * - */ - -static void snd_gus_synth_free_port(void *private_data) -{ - struct snd_gus_port * p = private_data; - - if (p) - snd_midi_channel_free_set(p->chset); -} - -static int snd_gus_synth_create_port(struct snd_gus_card * gus, int idx) -{ - struct snd_gus_port * p; - struct snd_seq_port_callback callbacks; - char name[32]; - int result; - - p = &gus->gf1.seq_ports[idx]; - p->chset = snd_midi_channel_alloc_set(16); - if (p->chset == NULL) - return -ENOMEM; - p->chset->private_data = p; - p->gus = gus; - p->client = gus->gf1.seq_client; - - memset(&callbacks, 0, sizeof(callbacks)); - callbacks.owner = THIS_MODULE; - callbacks.use = snd_gus_synth_use; - callbacks.unuse = snd_gus_synth_unuse; - callbacks.event_input = snd_gus_synth_event_input; - callbacks.private_free = snd_gus_synth_free_port; - callbacks.private_data = p; - - sprintf(name, "%s port %i", gus->interwave ? "AMD InterWave" : "GF1", idx); - p->chset->port = snd_seq_event_port_attach(gus->gf1.seq_client, - &callbacks, - SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE, - SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE | - SNDRV_SEQ_PORT_TYPE_SYNTH | - SNDRV_SEQ_PORT_TYPE_HARDWARE | - SNDRV_SEQ_PORT_TYPE_SYNTHESIZER, - 16, 0, - name); - if (p->chset->port < 0) { - result = p->chset->port; - snd_gus_synth_free_port(p); - return result; - } - p->port = p->chset->port; - return 0; -} - -/* - * - */ - -static int snd_gus_synth_new_device(struct snd_seq_device *dev) -{ - struct snd_gus_card *gus; - int client, i; - struct snd_seq_port_subscribe sub; - struct snd_iwffff_ops *iwops; - struct snd_gf1_ops *gf1ops; - struct snd_simple_ops *simpleops; - - gus = *(struct snd_gus_card **)SNDRV_SEQ_DEVICE_ARGPTR(dev); - if (gus == NULL) - return -EINVAL; - - mutex_init(&gus->register_mutex); - gus->gf1.seq_client = -1; - - /* allocate new client */ - client = gus->gf1.seq_client = - snd_seq_create_kernel_client(gus->card, 1, gus->interwave ? - "AMD InterWave" : "GF1"); - if (client < 0) - return client; - - for (i = 0; i < 4; i++) - snd_gus_synth_create_port(gus, i); - - gus->gf1.ilist = snd_seq_instr_list_new(); - if (gus->gf1.ilist == NULL) { - snd_seq_delete_kernel_client(client); - gus->gf1.seq_client = -1; - return -ENOMEM; - } - gus->gf1.ilist->flags = SNDRV_SEQ_INSTR_FLG_DIRECT; - - simpleops = &gus->gf1.simple_ops; - snd_seq_simple_init(simpleops, gus, NULL); - simpleops->put_sample = snd_gus_simple_put_sample; - simpleops->get_sample = snd_gus_simple_get_sample; - simpleops->remove_sample = snd_gus_simple_remove_sample; - simpleops->notify = snd_gus_synth_instr_notify; - - gf1ops = &gus->gf1.gf1_ops; - snd_seq_gf1_init(gf1ops, gus, &simpleops->kops); - gf1ops->put_sample = snd_gus_gf1_put_sample; - gf1ops->get_sample = snd_gus_gf1_get_sample; - gf1ops->remove_sample = snd_gus_gf1_remove_sample; - gf1ops->notify = snd_gus_synth_instr_notify; - - iwops = &gus->gf1.iwffff_ops; - snd_seq_iwffff_init(iwops, gus, &gf1ops->kops); - iwops->put_sample = snd_gus_iwffff_put_sample; - iwops->get_sample = snd_gus_iwffff_get_sample; - iwops->remove_sample = snd_gus_iwffff_remove_sample; - iwops->notify = snd_gus_synth_instr_notify; - - memset(&sub, 0, sizeof(sub)); - sub.sender.client = SNDRV_SEQ_CLIENT_SYSTEM; - sub.sender.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; - sub.dest.client = client; - sub.dest.port = 0; - snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, &sub); - - return 0; -} - -static int snd_gus_synth_delete_device(struct snd_seq_device *dev) -{ - struct snd_gus_card *gus; - - gus = *(struct snd_gus_card **)SNDRV_SEQ_DEVICE_ARGPTR(dev); - if (gus == NULL) - return -EINVAL; - - if (gus->gf1.seq_client >= 0) { - snd_seq_delete_kernel_client(gus->gf1.seq_client); - gus->gf1.seq_client = -1; - } - if (gus->gf1.ilist) - snd_seq_instr_list_free(&gus->gf1.ilist); - return 0; -} - -static int __init alsa_gus_synth_init(void) -{ - static struct snd_seq_dev_ops ops = { - snd_gus_synth_new_device, - snd_gus_synth_delete_device - }; - - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_GUS, &ops, - sizeof(struct snd_gus_card *)); -} - -static void __exit alsa_gus_synth_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_GUS); -} - -module_init(alsa_gus_synth_init) -module_exit(alsa_gus_synth_exit) diff --git a/sound/pci/trident/Makefile b/sound/pci/trident/Makefile index 65f2c21..88676b5 100644 --- a/sound/pci/trident/Makefile +++ b/sound/pci/trident/Makefile @@ -4,16 +4,6 @@ # snd-trident-objs := trident.o trident_main.o trident_memory.o -snd-trident-synth-objs := trident_synth.o - -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) # Toplevel Module Dependency obj-$(CONFIG_SND_TRIDENT) += snd-trident.o -obj-$(call sequencer,$(CONFIG_SND_TRIDENT)) += snd-trident-synth.o diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 8488456..6193c7e 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -155,13 +155,6 @@ static int __devinit snd_trident_probe(struct pci_dev *pci, return err; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) - if ((err = snd_trident_attach_synthesizer(trident)) < 0) { - snd_card_free(card); - return err; - } -#endif - snd_trident_create_gameport(trident); if ((err = snd_card_register(card)) < 0) { diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index a235e03..59a3195 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3313,12 +3313,6 @@ static void snd_trident_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "Memory Free : %d\n", snd_util_mem_avail(trident->tlb.memhdr)); } } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) - snd_iprintf(buffer,"\nWavetable Synth\n"); - snd_iprintf(buffer, "Memory Maximum : %d\n", trident->synth.max_size); - snd_iprintf(buffer, "Memory Used : %d\n", trident->synth.current_size); - snd_iprintf(buffer, "Memory Free : %d\n", (trident->synth.max_size-trident->synth.current_size)); -#endif } static void __devinit snd_trident_proc_init(struct snd_trident * trident) @@ -3815,28 +3809,6 @@ static irqreturn_t snd_trident_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } -/*--------------------------------------------------------------------------- - snd_trident_attach_synthesizer - - Description: Attach synthesizer hooks - - Paramters: trident - device specific private data for 4DWave card - - Returns: None. - - ---------------------------------------------------------------------------*/ -int snd_trident_attach_synthesizer(struct snd_trident *trident) -{ -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) - if (snd_seq_device_new(trident->card, 1, SNDRV_SEQ_DEV_ID_TRIDENT, - sizeof(struct snd_trident *), &trident->seq_dev) >= 0) { - strcpy(trident->seq_dev->name, "4DWave"); - *(struct snd_trident **)SNDRV_SEQ_DEVICE_ARGPTR(trident->seq_dev) = trident; - } -#endif - return 0; -} - struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, int type, int client, int port) { struct snd_trident_voice *pvoice; diff --git a/sound/pci/trident/trident_synth.c b/sound/pci/trident/trident_synth.c deleted file mode 100644 index 9b7dee8..0000000 --- a/sound/pci/trident/trident_synth.c +++ /dev/null @@ -1,1024 +0,0 @@ -/* - * Routines for Trident 4DWave NX/DX soundcards - Synthesizer - * Copyright (c) by Scott McNab - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -MODULE_AUTHOR("Scott McNab "); -MODULE_DESCRIPTION("Routines for Trident 4DWave NX/DX soundcards - Synthesizer"); -MODULE_LICENSE("GPL"); - -/* linear to log pan conversion table (4.2 channel attenuation format) */ -static unsigned int pan_table[63] = { - 7959, 7733, 7514, 7301, 7093, 6892, 6697, 6507, - 6322, 6143, 5968, 5799, 5634, 5475, 5319, 5168, - 5022, 4879, 4741, 4606, 4475, 4349, 4225, 4105, - 3989, 3876, 3766, 3659, 3555, 3454, 3356, 3261, - 3168, 3078, 2991, 2906, 2824, 2744, 2666, 2590, - 2517, 2445, 2376, 2308, 2243, 2179, 2117, 2057, - 1999, 1942, 1887, 1833, 1781, 1731, 1682, 1634, - 1588, 1543, 1499, 1456, 1415, 1375, 1336 -}; - -#define LOG_TABLE_SIZE 386 - -/* Linear half-attenuation to log conversion table in the format: - * {linear volume, logarithmic attenuation equivalent}, ... - * - * Provides conversion from a linear half-volume value in the range - * [0,8192] to a logarithmic attenuation value in the range 0 to 6.02dB. - * Halving the linear volume is equivalent to an additional 6dB of - * logarithmic attenuation. The algorithm used in log_from_linear() - * therefore uses this table as follows: - * - * - loop and for every time the volume is less than half the maximum - * volume (16384), add another 6dB and halve the maximum value used - * for this comparison. - * - when the volume is greater than half the maximum volume, take - * the difference of the volume to half volume (in the range [0,8192]) - * and look up the log_table[] to find the nearest entry. - * - take the logarithic component of this entry and add it to the - * resulting attenuation. - * - * Thus this routine provides a linear->log conversion for a range of - * [0,16384] using only 386 table entries - * - * Note: although this table stores log attenuation in 8.8 format, values - * were only calculated for 6 bits fractional precision, since that is - * the most precision offered by the trident hardware. - */ - -static unsigned short log_table[LOG_TABLE_SIZE*2] = -{ - 4, 0x0604, 19, 0x0600, 34, 0x05fc, - 49, 0x05f8, 63, 0x05f4, 78, 0x05f0, 93, 0x05ec, 108, 0x05e8, - 123, 0x05e4, 138, 0x05e0, 153, 0x05dc, 168, 0x05d8, 183, 0x05d4, - 198, 0x05d0, 213, 0x05cc, 228, 0x05c8, 244, 0x05c4, 259, 0x05c0, - 274, 0x05bc, 289, 0x05b8, 304, 0x05b4, 320, 0x05b0, 335, 0x05ac, - 350, 0x05a8, 366, 0x05a4, 381, 0x05a0, 397, 0x059c, 412, 0x0598, - 428, 0x0594, 443, 0x0590, 459, 0x058c, 474, 0x0588, 490, 0x0584, - 506, 0x0580, 521, 0x057c, 537, 0x0578, 553, 0x0574, 568, 0x0570, - 584, 0x056c, 600, 0x0568, 616, 0x0564, 632, 0x0560, 647, 0x055c, - 663, 0x0558, 679, 0x0554, 695, 0x0550, 711, 0x054c, 727, 0x0548, - 743, 0x0544, 759, 0x0540, 776, 0x053c, 792, 0x0538, 808, 0x0534, - 824, 0x0530, 840, 0x052c, 857, 0x0528, 873, 0x0524, 889, 0x0520, - 906, 0x051c, 922, 0x0518, 938, 0x0514, 955, 0x0510, 971, 0x050c, - 988, 0x0508, 1004, 0x0504, 1021, 0x0500, 1037, 0x04fc, 1054, 0x04f8, - 1071, 0x04f4, 1087, 0x04f0, 1104, 0x04ec, 1121, 0x04e8, 1138, 0x04e4, - 1154, 0x04e0, 1171, 0x04dc, 1188, 0x04d8, 1205, 0x04d4, 1222, 0x04d0, - 1239, 0x04cc, 1256, 0x04c8, 1273, 0x04c4, 1290, 0x04c0, 1307, 0x04bc, - 1324, 0x04b8, 1341, 0x04b4, 1358, 0x04b0, 1376, 0x04ac, 1393, 0x04a8, - 1410, 0x04a4, 1427, 0x04a0, 1445, 0x049c, 1462, 0x0498, 1479, 0x0494, - 1497, 0x0490, 1514, 0x048c, 1532, 0x0488, 1549, 0x0484, 1567, 0x0480, - 1584, 0x047c, 1602, 0x0478, 1620, 0x0474, 1637, 0x0470, 1655, 0x046c, - 1673, 0x0468, 1690, 0x0464, 1708, 0x0460, 1726, 0x045c, 1744, 0x0458, - 1762, 0x0454, 1780, 0x0450, 1798, 0x044c, 1816, 0x0448, 1834, 0x0444, - 1852, 0x0440, 1870, 0x043c, 1888, 0x0438, 1906, 0x0434, 1924, 0x0430, - 1943, 0x042c, 1961, 0x0428, 1979, 0x0424, 1997, 0x0420, 2016, 0x041c, - 2034, 0x0418, 2053, 0x0414, 2071, 0x0410, 2089, 0x040c, 2108, 0x0408, - 2127, 0x0404, 2145, 0x0400, 2164, 0x03fc, 2182, 0x03f8, 2201, 0x03f4, - 2220, 0x03f0, 2239, 0x03ec, 2257, 0x03e8, 2276, 0x03e4, 2295, 0x03e0, - 2314, 0x03dc, 2333, 0x03d8, 2352, 0x03d4, 2371, 0x03d0, 2390, 0x03cc, - 2409, 0x03c8, 2428, 0x03c4, 2447, 0x03c0, 2466, 0x03bc, 2485, 0x03b8, - 2505, 0x03b4, 2524, 0x03b0, 2543, 0x03ac, 2562, 0x03a8, 2582, 0x03a4, - 2601, 0x03a0, 2621, 0x039c, 2640, 0x0398, 2660, 0x0394, 2679, 0x0390, - 2699, 0x038c, 2718, 0x0388, 2738, 0x0384, 2758, 0x0380, 2777, 0x037c, - 2797, 0x0378, 2817, 0x0374, 2837, 0x0370, 2857, 0x036c, 2876, 0x0368, - 2896, 0x0364, 2916, 0x0360, 2936, 0x035c, 2956, 0x0358, 2976, 0x0354, - 2997, 0x0350, 3017, 0x034c, 3037, 0x0348, 3057, 0x0344, 3077, 0x0340, - 3098, 0x033c, 3118, 0x0338, 3138, 0x0334, 3159, 0x0330, 3179, 0x032c, - 3200, 0x0328, 3220, 0x0324, 3241, 0x0320, 3261, 0x031c, 3282, 0x0318, - 3303, 0x0314, 3323, 0x0310, 3344, 0x030c, 3365, 0x0308, 3386, 0x0304, - 3406, 0x0300, 3427, 0x02fc, 3448, 0x02f8, 3469, 0x02f4, 3490, 0x02f0, - 3511, 0x02ec, 3532, 0x02e8, 3553, 0x02e4, 3575, 0x02e0, 3596, 0x02dc, - 3617, 0x02d8, 3638, 0x02d4, 3660, 0x02d0, 3681, 0x02cc, 3702, 0x02c8, - 3724, 0x02c4, 3745, 0x02c0, 3767, 0x02bc, 3788, 0x02b8, 3810, 0x02b4, - 3831, 0x02b0, 3853, 0x02ac, 3875, 0x02a8, 3896, 0x02a4, 3918, 0x02a0, - 3940, 0x029c, 3962, 0x0298, 3984, 0x0294, 4006, 0x0290, 4028, 0x028c, - 4050, 0x0288, 4072, 0x0284, 4094, 0x0280, 4116, 0x027c, 4138, 0x0278, - 4160, 0x0274, 4182, 0x0270, 4205, 0x026c, 4227, 0x0268, 4249, 0x0264, - 4272, 0x0260, 4294, 0x025c, 4317, 0x0258, 4339, 0x0254, 4362, 0x0250, - 4384, 0x024c, 4407, 0x0248, 4430, 0x0244, 4453, 0x0240, 4475, 0x023c, - 4498, 0x0238, 4521, 0x0234, 4544, 0x0230, 4567, 0x022c, 4590, 0x0228, - 4613, 0x0224, 4636, 0x0220, 4659, 0x021c, 4682, 0x0218, 4705, 0x0214, - 4728, 0x0210, 4752, 0x020c, 4775, 0x0208, 4798, 0x0204, 4822, 0x0200, - 4845, 0x01fc, 4869, 0x01f8, 4892, 0x01f4, 4916, 0x01f0, 4939, 0x01ec, - 4963, 0x01e8, 4987, 0x01e4, 5010, 0x01e0, 5034, 0x01dc, 5058, 0x01d8, - 5082, 0x01d4, 5106, 0x01d0, 5130, 0x01cc, 5154, 0x01c8, 5178, 0x01c4, - 5202, 0x01c0, 5226, 0x01bc, 5250, 0x01b8, 5274, 0x01b4, 5299, 0x01b0, - 5323, 0x01ac, 5347, 0x01a8, 5372, 0x01a4, 5396, 0x01a0, 5420, 0x019c, - 5445, 0x0198, 5469, 0x0194, 5494, 0x0190, 5519, 0x018c, 5543, 0x0188, - 5568, 0x0184, 5593, 0x0180, 5618, 0x017c, 5643, 0x0178, 5668, 0x0174, - 5692, 0x0170, 5717, 0x016c, 5743, 0x0168, 5768, 0x0164, 5793, 0x0160, - 5818, 0x015c, 5843, 0x0158, 5868, 0x0154, 5894, 0x0150, 5919, 0x014c, - 5945, 0x0148, 5970, 0x0144, 5995, 0x0140, 6021, 0x013c, 6047, 0x0138, - 6072, 0x0134, 6098, 0x0130, 6124, 0x012c, 6149, 0x0128, 6175, 0x0124, - 6201, 0x0120, 6227, 0x011c, 6253, 0x0118, 6279, 0x0114, 6305, 0x0110, - 6331, 0x010c, 6357, 0x0108, 6384, 0x0104, 6410, 0x0100, 6436, 0x00fc, - 6462, 0x00f8, 6489, 0x00f4, 6515, 0x00f0, 6542, 0x00ec, 6568, 0x00e8, - 6595, 0x00e4, 6621, 0x00e0, 6648, 0x00dc, 6675, 0x00d8, 6702, 0x00d4, - 6728, 0x00d0, 6755, 0x00cc, 6782, 0x00c8, 6809, 0x00c4, 6836, 0x00c0, - 6863, 0x00bc, 6890, 0x00b8, 6917, 0x00b4, 6945, 0x00b0, 6972, 0x00ac, - 6999, 0x00a8, 7027, 0x00a4, 7054, 0x00a0, 7081, 0x009c, 7109, 0x0098, - 7136, 0x0094, 7164, 0x0090, 7192, 0x008c, 7219, 0x0088, 7247, 0x0084, - 7275, 0x0080, 7303, 0x007c, 7331, 0x0078, 7359, 0x0074, 7387, 0x0070, - 7415, 0x006c, 7443, 0x0068, 7471, 0x0064, 7499, 0x0060, 7527, 0x005c, - 7556, 0x0058, 7584, 0x0054, 7613, 0x0050, 7641, 0x004c, 7669, 0x0048, - 7698, 0x0044, 7727, 0x0040, 7755, 0x003c, 7784, 0x0038, 7813, 0x0034, - 7842, 0x0030, 7870, 0x002c, 7899, 0x0028, 7928, 0x0024, 7957, 0x0020, - 7986, 0x001c, 8016, 0x0018, 8045, 0x0014, 8074, 0x0010, 8103, 0x000c, - 8133, 0x0008, 8162, 0x0004, 8192, 0x0000 -}; - -static unsigned short lookup_volume_table( unsigned short value ) -{ - /* This code is an optimised version of: - * int i = 0; - * while( volume_table[i*2] < value ) - * i++; - * return volume_table[i*2+1]; - */ - unsigned short *ptr = log_table; - while( *ptr < value ) - ptr += 2; - return *(ptr+1); -} - -/* this function calculates a 8.8 fixed point logarithmic attenuation - * value from a linear volume value in the range 0 to 16384 */ -static unsigned short log_from_linear( unsigned short value ) -{ - if (value >= 16384) - return 0x0000; - if (value) { - unsigned short result = 0; - int v, c; - for( c = 0, v = 8192; c < 14; c++, v >>= 1 ) { - if( value >= v ) { - result += lookup_volume_table( (value - v) << c ); - return result; - } - result += 0x0605; /* 6.0205 (result of -20*log10(0.5)) */ - } - } - return 0xffff; -} - -/* - * Sample handling operations - */ - -static void sample_start(struct snd_trident * trident, struct snd_trident_voice * voice, snd_seq_position_t position); -static void sample_stop(struct snd_trident * trident, struct snd_trident_voice * voice, int mode); -static void sample_freq(struct snd_trident * trident, struct snd_trident_voice * voice, snd_seq_frequency_t freq); -static void sample_volume(struct snd_trident * trident, struct snd_trident_voice * voice, struct snd_seq_ev_volume * volume); -static void sample_loop(struct snd_trident * trident, struct snd_trident_voice * voice, struct snd_seq_ev_loop * loop); -static void sample_pos(struct snd_trident * trident, struct snd_trident_voice * voice, snd_seq_position_t position); -static void sample_private1(struct snd_trident * trident, struct snd_trident_voice * voice, unsigned char *data); - -static struct snd_trident_sample_ops sample_ops = -{ - sample_start, - sample_stop, - sample_freq, - sample_volume, - sample_loop, - sample_pos, - sample_private1 -}; - -static void snd_trident_simple_init(struct snd_trident_voice * voice) -{ - //voice->handler_wave = interrupt_wave; - //voice->handler_volume = interrupt_volume; - //voice->handler_effect = interrupt_effect; - //voice->volume_change = NULL; - voice->sample_ops = &sample_ops; -} - -static void sample_start(struct snd_trident * trident, struct snd_trident_voice * voice, snd_seq_position_t position) -{ - struct simple_instrument *simple; - struct snd_seq_kinstr *instr; - unsigned long flags; - unsigned int loop_start, loop_end, sample_start, sample_end, start_offset; - unsigned int value; - unsigned int shift = 0; - - instr = snd_seq_instr_find(trident->synth.ilist, &voice->instr, 0, 1); - if (instr == NULL) - return; - voice->instr = instr->instr; /* copy ID to speedup aliases */ - simple = KINSTR_DATA(instr); - - spin_lock_irqsave(&trident->reg_lock, flags); - - if (trident->device == TRIDENT_DEVICE_ID_SI7018) - voice->GVSel = 1; /* route to Wave volume */ - - voice->CTRL = 0; - voice->Alpha = 0; - voice->FMS = 0; - - loop_start = simple->loop_start >> 4; - loop_end = simple->loop_end >> 4; - sample_start = (simple->start + position) >> 4; - if( sample_start >= simple->size ) - sample_start = simple->start >> 4; - sample_end = simple->size; - start_offset = position >> 4; - - if (simple->format & SIMPLE_WAVE_16BIT) { - voice->CTRL |= 8; - shift++; - } - if (simple->format & SIMPLE_WAVE_STEREO) { - voice->CTRL |= 4; - shift++; - } - if (!(simple->format & SIMPLE_WAVE_UNSIGNED)) - voice->CTRL |= 2; - - voice->LBA = simple->address.memory; - - if (simple->format & SIMPLE_WAVE_LOOP) { - voice->CTRL |= 1; - voice->LBA += loop_start << shift; - if( start_offset >= loop_start ) { - voice->CSO = start_offset - loop_start; - voice->negCSO = 0; - } else { - voice->CSO = loop_start - start_offset; - voice->negCSO = 1; - } - voice->ESO = loop_end - loop_start - 1; - } else { - voice->LBA += start_offset << shift; - voice->CSO = sample_start; - voice->ESO = sample_end - 1; - voice->negCSO = 0; - } - - if (voice->flags & SNDRV_TRIDENT_VFLG_RUNNING) { - snd_trident_stop_voice(trident, voice->number); - voice->flags &= ~SNDRV_TRIDENT_VFLG_RUNNING; - } - - /* set CSO sign */ - value = inl(TRID_REG(trident, T4D_SIGN_CSO_A)); - if( voice->negCSO ) { - value |= 1 << (voice->number&31); - } else { - value &= ~(1 << (voice->number&31)); - } - outl(value,TRID_REG(trident, T4D_SIGN_CSO_A)); - - voice->Attribute = 0; - snd_trident_write_voice_regs(trident, voice); - snd_trident_start_voice(trident, voice->number); - voice->flags |= SNDRV_TRIDENT_VFLG_RUNNING; - spin_unlock_irqrestore(&trident->reg_lock, flags); - snd_seq_instr_free_use(trident->synth.ilist, instr); -} - -static void sample_stop(struct snd_trident * trident, struct snd_trident_voice * voice, int mode) -{ - unsigned long flags; - - if (!(voice->flags & SNDRV_TRIDENT_VFLG_RUNNING)) - return; - - switch (mode) { - default: - spin_lock_irqsave(&trident->reg_lock, flags); - snd_trident_stop_voice(trident, voice->number); - voice->flags &= ~SNDRV_TRIDENT_VFLG_RUNNING; - spin_unlock_irqrestore(&trident->reg_lock, flags); - break; - case SAMPLE_STOP_LOOP: /* disable loop only */ - voice->CTRL &= ~1; - spin_lock_irqsave(&trident->reg_lock, flags); - outb((unsigned char) voice->number, TRID_REG(trident, T4D_LFO_GC_CIR)); - outw((((voice->CTRL << 12) | (voice->EC & 0x0fff)) & 0xffff), CH_GVSEL_PAN_VOL_CTRL_EC); - spin_unlock_irqrestore(&trident->reg_lock, flags); - break; - } -} - -static void sample_freq(struct snd_trident * trident, struct snd_trident_voice * voice, snd_seq_frequency_t freq) -{ - unsigned long flags; - freq >>= 4; - - spin_lock_irqsave(&trident->reg_lock, flags); - if (freq == 44100) - voice->Delta = 0xeb3; - else if (freq == 8000) - voice->Delta = 0x2ab; - else if (freq == 48000) - voice->Delta = 0x1000; - else - voice->Delta = (((freq << 12) + freq) / 48000) & 0x0000ffff; - - outb((unsigned char) voice->number, TRID_REG(trident, T4D_LFO_GC_CIR)); - if (trident->device == TRIDENT_DEVICE_ID_NX) { - outb((unsigned char) voice->Delta, TRID_REG(trident, CH_NX_DELTA_CSO + 3)); - outb((unsigned char) (voice->Delta >> 8), TRID_REG(trident, CH_NX_DELTA_ESO + 3)); - } else { - outw((unsigned short) voice->Delta, TRID_REG(trident, CH_DX_ESO_DELTA)); - } - - spin_unlock_irqrestore(&trident->reg_lock, flags); -} - -static void sample_volume(struct snd_trident * trident, struct snd_trident_voice * voice, struct snd_seq_ev_volume * volume) -{ - unsigned long flags; - unsigned short value; - - spin_lock_irqsave(&trident->reg_lock, flags); - voice->GVSel = 0; /* use global music volume */ - voice->FMC = 0x03; /* fixme: can we do something useful with FMC? */ - if (volume->volume >= 0) { - volume->volume &= 0x3fff; - /* linear volume -> logarithmic attenuation conversion - * uses EC register for greater resolution (6.6 bits) than Vol register (5.3 bits) - * Vol register used when additional attenuation is required */ - voice->RVol = 0; - voice->CVol = 0; - value = log_from_linear( volume->volume ); - voice->Vol = 0; - voice->EC = (value & 0x3fff) >> 2; - if (value > 0x3fff) { - voice->EC |= 0xfc0; - if (value < 0x5f00 ) - voice->Vol = ((value >> 8) - 0x3f) << 5; - else { - voice->Vol = 0x3ff; - voice->EC = 0xfff; - } - } - } - if (volume->lr >= 0) { - volume->lr &= 0x3fff; - /* approximate linear pan by attenuating channels */ - if (volume->lr >= 0x2000) { /* attenuate left (pan right) */ - value = 0x3fff - volume->lr; - for (voice->Pan = 0; voice->Pan < 63; voice->Pan++ ) - if (value >= pan_table[voice->Pan] ) - break; - } else { /* attenuate right (pan left) */ - for (voice->Pan = 0; voice->Pan < 63; voice->Pan++ ) - if ((unsigned int)volume->lr >= pan_table[voice->Pan] ) - break; - voice->Pan |= 0x40; - } - } - outb((unsigned char) voice->number, TRID_REG(trident, T4D_LFO_GC_CIR)); - outl((voice->GVSel << 31) | ((voice->Pan & 0x0000007f) << 24) | - ((voice->Vol & 0x000000ff) << 16) | ((voice->CTRL & 0x0000000f) << 12) | - (voice->EC & 0x00000fff), TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); - value = ((voice->FMC & 0x03) << 14) | ((voice->RVol & 0x7f) << 7) | (voice->CVol & 0x7f); - outw(value, TRID_REG(trident, CH_DX_FMC_RVOL_CVOL)); - spin_unlock_irqrestore(&trident->reg_lock, flags); -} - -static void sample_loop(struct snd_trident * trident, struct snd_trident_voice * voice, struct snd_seq_ev_loop * loop) -{ - unsigned long flags; - struct simple_instrument *simple; - struct snd_seq_kinstr *instr; - unsigned int loop_start, loop_end; - - instr = snd_seq_instr_find(trident->synth.ilist, &voice->instr, 0, 1); - if (instr == NULL) - return; - voice->instr = instr->instr; /* copy ID to speedup aliases */ - simple = KINSTR_DATA(instr); - - loop_start = loop->start >> 4; - loop_end = loop->end >> 4; - - spin_lock_irqsave(&trident->reg_lock, flags); - - voice->LBA = simple->address.memory + loop_start; - voice->CSO = 0; - voice->ESO = loop_end - loop_start - 1; - - outb((unsigned char) voice->number, TRID_REG(trident, T4D_LFO_GC_CIR)); - outb((voice->LBA >> 16), TRID_REG(trident, CH_LBA + 2)); - outw((voice->LBA & 0xffff), TRID_REG(trident, CH_LBA)); - if (trident->device == TRIDENT_DEVICE_ID_NX) { - outb((voice->ESO >> 16), TRID_REG(trident, CH_NX_DELTA_ESO + 2)); - outw((voice->ESO & 0xffff), TRID_REG(trident, CH_NX_DELTA_ESO)); - outb((voice->CSO >> 16), TRID_REG(trident, CH_NX_DELTA_CSO + 2)); - outw((voice->CSO & 0xffff), TRID_REG(trident, CH_NX_DELTA_CSO)); - } else { - outw((voice->ESO & 0xffff), TRID_REG(trident, CH_DX_ESO_DELTA + 2)); - outw((voice->CSO & 0xffff), TRID_REG(trident, CH_DX_CSO_ALPHA_FMS + 2)); - } - - spin_unlock_irqrestore(&trident->reg_lock, flags); - snd_seq_instr_free_use(trident->synth.ilist, instr); -} - -static void sample_pos(struct snd_trident * trident, struct snd_trident_voice * voice, snd_seq_position_t position) -{ - unsigned long flags; - struct simple_instrument *simple; - struct snd_seq_kinstr *instr; - unsigned int value; - - instr = snd_seq_instr_find(trident->synth.ilist, &voice->instr, 0, 1); - if (instr == NULL) - return; - voice->instr = instr->instr; /* copy ID to speedup aliases */ - simple = KINSTR_DATA(instr); - - spin_lock_irqsave(&trident->reg_lock, flags); - - if (simple->format & SIMPLE_WAVE_LOOP) { - if( position >= simple->loop_start ) { - voice->CSO = (position - simple->loop_start) >> 4; - voice->negCSO = 0; - } else { - voice->CSO = (simple->loop_start - position) >> 4; - voice->negCSO = 1; - } - } else { - voice->CSO = position >> 4; - voice->negCSO = 0; - } - - /* set CSO sign */ - value = inl(TRID_REG(trident, T4D_SIGN_CSO_A)); - if( voice->negCSO ) { - value |= 1 << (voice->number&31); - } else { - value &= ~(1 << (voice->number&31)); - } - outl(value,TRID_REG(trident, T4D_SIGN_CSO_A)); - - - outb((unsigned char) voice->number, TRID_REG(trident, T4D_LFO_GC_CIR)); - if (trident->device == TRIDENT_DEVICE_ID_NX) { - outw((voice->CSO & 0xffff), TRID_REG(trident, CH_NX_DELTA_CSO)); - outb((voice->CSO >> 16), TRID_REG(trident, CH_NX_DELTA_CSO + 2)); - } else { - outw((voice->CSO & 0xffff), TRID_REG(trident, CH_DX_CSO_ALPHA_FMS) + 2); - } - - spin_unlock_irqrestore(&trident->reg_lock, flags); - snd_seq_instr_free_use(trident->synth.ilist, instr); -} - -static void sample_private1(struct snd_trident * trident, struct snd_trident_voice * voice, unsigned char *data) -{ -} - -/* - * Memory management / sample loading - */ - -static int snd_trident_simple_put_sample(void *private_data, - struct simple_instrument * instr, - char __user *data, long len, int atomic) -{ - struct snd_trident *trident = private_data; - int size = instr->size; - int shift = 0; - - if (instr->format & SIMPLE_WAVE_BACKWARD || - instr->format & SIMPLE_WAVE_BIDIR || - instr->format & SIMPLE_WAVE_ULAW) - return -EINVAL; /* not supported */ - - if (instr->format & SIMPLE_WAVE_16BIT) - shift++; - if (instr->format & SIMPLE_WAVE_STEREO) - shift++; - size <<= shift; - - if (trident->synth.current_size + size > trident->synth.max_size) - return -ENOMEM; - - if (!access_ok(VERIFY_READ, data, size)) - return -EFAULT; - - if (trident->tlb.entries) { - struct snd_util_memblk *memblk; - memblk = snd_trident_synth_alloc(trident, size); - if (memblk == NULL) - return -ENOMEM; - if (snd_trident_synth_copy_from_user(trident, memblk, 0, data, size) ) { - snd_trident_synth_free(trident, memblk); - return -EFAULT; - } - instr->address.ptr = (unsigned char*)memblk; - instr->address.memory = memblk->offset; - } else { - struct snd_dma_buffer dmab; - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci), - size, &dmab) < 0) - return -ENOMEM; - - if (copy_from_user(dmab.area, data, size)) { - snd_dma_free_pages(&dmab); - return -EFAULT; - } - instr->address.ptr = dmab.area; - instr->address.memory = dmab.addr; - } - - trident->synth.current_size += size; - return 0; -} - -static int snd_trident_simple_get_sample(void *private_data, - struct simple_instrument * instr, - char __user *data, long len, int atomic) -{ - //struct snd_trident *trident = private_data; - int size = instr->size; - int shift = 0; - - if (instr->format & SIMPLE_WAVE_16BIT) - shift++; - if (instr->format & SIMPLE_WAVE_STEREO) - shift++; - size <<= shift; - - if (!access_ok(VERIFY_WRITE, data, size)) - return -EFAULT; - - /* FIXME: not implemented yet */ - - return -EBUSY; -} - -static int snd_trident_simple_remove_sample(void *private_data, - struct simple_instrument * instr, - int atomic) -{ - struct snd_trident *trident = private_data; - int size = instr->size; - - if (instr->format & SIMPLE_WAVE_16BIT) - size <<= 1; - if (instr->format & SIMPLE_WAVE_STEREO) - size <<= 1; - - if (trident->tlb.entries) { - struct snd_util_memblk *memblk = (struct snd_util_memblk *)instr->address.ptr; - if (memblk) - snd_trident_synth_free(trident, memblk); - else - return -EFAULT; - } else { - struct snd_dma_buffer dmab; - dmab.dev.type = SNDRV_DMA_TYPE_DEV; - dmab.dev.dev = snd_dma_pci_data(trident->pci); - dmab.area = instr->address.ptr; - dmab.addr = instr->address.memory; - dmab.bytes = size; - snd_dma_free_pages(&dmab); - } - - trident->synth.current_size -= size; - if (trident->synth.current_size < 0) /* shouldn't need this check... */ - trident->synth.current_size = 0; - - return 0; -} - -static void select_instrument(struct snd_trident * trident, struct snd_trident_voice * v) -{ - struct snd_seq_kinstr *instr; - instr = snd_seq_instr_find(trident->synth.ilist, &v->instr, 0, 1); - if (instr != NULL) { - if (instr->ops) { - if (!strcmp(instr->ops->instr_type, SNDRV_SEQ_INSTR_ID_SIMPLE)) - snd_trident_simple_init(v); - } - snd_seq_instr_free_use(trident->synth.ilist, instr); - } -} - -/* - - */ - -static void event_sample(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_stop) - v->sample_ops->sample_stop(p->trident, v, SAMPLE_STOP_IMMEDIATELY); - v->instr.std = ev->data.sample.param.sample.std; - if (v->instr.std & 0xff000000) { /* private instrument */ - v->instr.std &= 0x00ffffff; - v->instr.std |= (unsigned int)ev->source.client << 24; - } - v->instr.bank = ev->data.sample.param.sample.bank; - v->instr.prg = ev->data.sample.param.sample.prg; - select_instrument(p->trident, v); -} - -static void event_cluster(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_stop) - v->sample_ops->sample_stop(p->trident, v, SAMPLE_STOP_IMMEDIATELY); - v->instr.cluster = ev->data.sample.param.cluster.cluster; - select_instrument(p->trident, v); -} - -static void event_start(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_start) - v->sample_ops->sample_start(p->trident, v, ev->data.sample.param.position); -} - -static void event_stop(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_stop) - v->sample_ops->sample_stop(p->trident, v, ev->data.sample.param.stop_mode); -} - -static void event_freq(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_freq) - v->sample_ops->sample_freq(p->trident, v, ev->data.sample.param.frequency); -} - -static void event_volume(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_volume) - v->sample_ops->sample_volume(p->trident, v, &ev->data.sample.param.volume); -} - -static void event_loop(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_loop) - v->sample_ops->sample_loop(p->trident, v, &ev->data.sample.param.loop); -} - -static void event_position(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_pos) - v->sample_ops->sample_pos(p->trident, v, ev->data.sample.param.position); -} - -static void event_private1(struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v) -{ - if (v->sample_ops && v->sample_ops->sample_private1) - v->sample_ops->sample_private1(p->trident, v, (unsigned char *) &ev->data.sample.param.raw8); -} - -typedef void (trident_sample_event_handler_t) (struct snd_seq_event * ev, struct snd_trident_port * p, struct snd_trident_voice * v); - -static trident_sample_event_handler_t *trident_sample_event_handlers[9] = -{ - event_sample, - event_cluster, - event_start, - event_stop, - event_freq, - event_volume, - event_loop, - event_position, - event_private1 -}; - -static void snd_trident_sample_event(struct snd_seq_event * ev, struct snd_trident_port * p) -{ - int idx, voice; - struct snd_trident *trident = p->trident; - struct snd_trident_voice *v; - unsigned long flags; - - idx = ev->type - SNDRV_SEQ_EVENT_SAMPLE; - if (idx < 0 || idx > 8) - return; - for (voice = 0; voice < 64; voice++) { - v = &trident->synth.voices[voice]; - if (v->use && v->client == ev->source.client && - v->port == ev->source.port && - v->index == ev->data.sample.channel) { - spin_lock_irqsave(&trident->event_lock, flags); - trident_sample_event_handlers[idx] (ev, p, v); - spin_unlock_irqrestore(&trident->event_lock, flags); - return; - } - } -} - -/* - - */ - -static void snd_trident_synth_free_voices(struct snd_trident * trident, int client, int port) -{ - int idx; - struct snd_trident_voice *voice; - - for (idx = 0; idx < 32; idx++) { - voice = &trident->synth.voices[idx]; - if (voice->use && voice->client == client && voice->port == port) - snd_trident_free_voice(trident, voice); - } -} - -static int snd_trident_synth_use(void *private_data, struct snd_seq_port_subscribe * info) -{ - struct snd_trident_port *port = private_data; - struct snd_trident *trident = port->trident; - struct snd_trident_voice *voice; - unsigned int idx; - unsigned long flags; - - if (info->voices > 32) - return -EINVAL; - spin_lock_irqsave(&trident->reg_lock, flags); - for (idx = 0; idx < info->voices; idx++) { - voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_SYNTH, info->sender.client, info->sender.port); - if (voice == NULL) { - snd_trident_synth_free_voices(trident, info->sender.client, info->sender.port); - spin_unlock_irqrestore(&trident->reg_lock, flags); - return -EBUSY; - } - voice->index = idx; - voice->Vol = 0x3ff; - voice->EC = 0x0fff; - } -#if 0 - for (idx = 0; idx < info->midi_voices; idx++) { - port->midi_has_voices = 1; - voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_MIDI, info->sender.client, info->sender.port); - if (voice == NULL) { - snd_trident_synth_free_voices(trident, info->sender.client, info->sender.port); - spin_unlock_irqrestore(&trident->reg_lock, flags); - return -EBUSY; - } - voice->Vol = 0x3ff; - voice->EC = 0x0fff; - } -#endif - spin_unlock_irqrestore(&trident->reg_lock, flags); - return 0; -} - -static int snd_trident_synth_unuse(void *private_data, struct snd_seq_port_subscribe * info) -{ - struct snd_trident_port *port = private_data; - struct snd_trident *trident = port->trident; - unsigned long flags; - - spin_lock_irqsave(&trident->reg_lock, flags); - snd_trident_synth_free_voices(trident, info->sender.client, info->sender.port); - spin_unlock_irqrestore(&trident->reg_lock, flags); - return 0; -} - -/* - - */ - -static void snd_trident_synth_free_private_instruments(struct snd_trident_port * p, int client) -{ - struct snd_seq_instr_header ifree; - - memset(&ifree, 0, sizeof(ifree)); - ifree.cmd = SNDRV_SEQ_INSTR_FREE_CMD_PRIVATE; - snd_seq_instr_list_free_cond(p->trident->synth.ilist, &ifree, client, 0); -} - -static int snd_trident_synth_event_input(struct snd_seq_event * ev, int direct, void *private_data, int atomic, int hop) -{ - struct snd_trident_port *p = (struct snd_trident_port *) private_data; - - if (p == NULL) - return -EINVAL; - if (ev->type >= SNDRV_SEQ_EVENT_SAMPLE && - ev->type <= SNDRV_SEQ_EVENT_SAMPLE_PRIVATE1) { - snd_trident_sample_event(ev, p); - return 0; - } - if (ev->source.client == SNDRV_SEQ_CLIENT_SYSTEM && - ev->source.port == SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE) { - if (ev->type == SNDRV_SEQ_EVENT_CLIENT_EXIT) { - snd_trident_synth_free_private_instruments(p, ev->data.addr.client); - return 0; - } - } - if (direct) { - if (ev->type >= SNDRV_SEQ_EVENT_INSTR_BEGIN) { - snd_seq_instr_event(&p->trident->synth.simple_ops.kops, - p->trident->synth.ilist, ev, - p->trident->synth.seq_client, atomic, hop); - return 0; - } - } - return 0; -} - -static void snd_trident_synth_instr_notify(void *private_data, - struct snd_seq_kinstr * instr, - int what) -{ - int idx; - struct snd_trident *trident = private_data; - struct snd_trident_voice *pvoice; - unsigned long flags; - - spin_lock_irqsave(&trident->event_lock, flags); - for (idx = 0; idx < 64; idx++) { - pvoice = &trident->synth.voices[idx]; - if (pvoice->use && !memcmp(&pvoice->instr, &instr->instr, sizeof(pvoice->instr))) { - if (pvoice->sample_ops && pvoice->sample_ops->sample_stop) { - pvoice->sample_ops->sample_stop(trident, pvoice, SAMPLE_STOP_IMMEDIATELY); - } else { - snd_trident_stop_voice(trident, pvoice->number); - pvoice->flags &= ~SNDRV_TRIDENT_VFLG_RUNNING; - } - } - } - spin_unlock_irqrestore(&trident->event_lock, flags); -} - -/* - - */ - -static void snd_trident_synth_free_port(void *private_data) -{ - struct snd_trident_port *p = (struct snd_trident_port *) private_data; - - if (p) - snd_midi_channel_free_set(p->chset); -} - -static int snd_trident_synth_create_port(struct snd_trident * trident, int idx) -{ - struct snd_trident_port *p; - struct snd_seq_port_callback callbacks; - char name[32]; - char *str; - int result; - - p = &trident->synth.seq_ports[idx]; - p->chset = snd_midi_channel_alloc_set(16); - if (p->chset == NULL) - return -ENOMEM; - p->chset->private_data = p; - p->trident = trident; - p->client = trident->synth.seq_client; - - memset(&callbacks, 0, sizeof(callbacks)); - callbacks.owner = THIS_MODULE; - callbacks.use = snd_trident_synth_use; - callbacks.unuse = snd_trident_synth_unuse; - callbacks.event_input = snd_trident_synth_event_input; - callbacks.private_free = snd_trident_synth_free_port; - callbacks.private_data = p; - - str = "???"; - switch (trident->device) { - case TRIDENT_DEVICE_ID_DX: str = "Trident 4DWave-DX"; break; - case TRIDENT_DEVICE_ID_NX: str = "Trident 4DWave-NX"; break; - case TRIDENT_DEVICE_ID_SI7018: str = "SiS 7018"; break; - } - sprintf(name, "%s port %i", str, idx); - p->chset->port = snd_seq_event_port_attach(trident->synth.seq_client, - &callbacks, - SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE, - SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE | - SNDRV_SEQ_PORT_TYPE_SYNTH | - SNDRV_SEQ_PORT_TYPE_HARDWARE | - SNDRV_SEQ_PORT_TYPE_SYNTHESIZER, - 16, 0, - name); - if (p->chset->port < 0) { - result = p->chset->port; - snd_trident_synth_free_port(p); - return result; - } - p->port = p->chset->port; - return 0; -} - -/* - - */ - -static int snd_trident_synth_new_device(struct snd_seq_device *dev) -{ - struct snd_trident *trident; - int client, i; - struct snd_seq_port_subscribe sub; - struct snd_simple_ops *simpleops; - char *str; - - trident = *(struct snd_trident **)SNDRV_SEQ_DEVICE_ARGPTR(dev); - if (trident == NULL) - return -EINVAL; - - trident->synth.seq_client = -1; - - /* allocate new client */ - str = "???"; - switch (trident->device) { - case TRIDENT_DEVICE_ID_DX: str = "Trident 4DWave-DX"; break; - case TRIDENT_DEVICE_ID_NX: str = "Trident 4DWave-NX"; break; - case TRIDENT_DEVICE_ID_SI7018: str = "SiS 7018"; break; - } - client = trident->synth.seq_client = - snd_seq_create_kernel_client(trident->card, 1, str); - if (client < 0) - return client; - - for (i = 0; i < 4; i++) - snd_trident_synth_create_port(trident, i); - - trident->synth.ilist = snd_seq_instr_list_new(); - if (trident->synth.ilist == NULL) { - snd_seq_delete_kernel_client(client); - trident->synth.seq_client = -1; - return -ENOMEM; - } - trident->synth.ilist->flags = SNDRV_SEQ_INSTR_FLG_DIRECT; - - simpleops = &trident->synth.simple_ops; - snd_seq_simple_init(simpleops, trident, NULL); - simpleops->put_sample = snd_trident_simple_put_sample; - simpleops->get_sample = snd_trident_simple_get_sample; - simpleops->remove_sample = snd_trident_simple_remove_sample; - simpleops->notify = snd_trident_synth_instr_notify; - - memset(&sub, 0, sizeof(sub)); - sub.sender.client = SNDRV_SEQ_CLIENT_SYSTEM; - sub.sender.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; - sub.dest.client = client; - sub.dest.port = 0; - snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, &sub); - - return 0; -} - -static int snd_trident_synth_delete_device(struct snd_seq_device *dev) -{ - struct snd_trident *trident; - - trident = *(struct snd_trident **)SNDRV_SEQ_DEVICE_ARGPTR(dev); - if (trident == NULL) - return -EINVAL; - - if (trident->synth.seq_client >= 0) { - snd_seq_delete_kernel_client(trident->synth.seq_client); - trident->synth.seq_client = -1; - } - if (trident->synth.ilist) - snd_seq_instr_list_free(&trident->synth.ilist); - return 0; -} - -static int __init alsa_trident_synth_init(void) -{ - static struct snd_seq_dev_ops ops = - { - snd_trident_synth_new_device, - snd_trident_synth_delete_device - }; - - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_TRIDENT, &ops, - sizeof(struct snd_trident *)); -} - -static void __exit alsa_trident_synth_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_TRIDENT); -} - -module_init(alsa_trident_synth_init) -module_exit(alsa_trident_synth_exit) -- cgit v0.10.2 From 045765253c610cb5acebb22ae94d759f586d9521 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Oct 2007 12:43:40 +0100 Subject: [ALSA] opl3 - Fix build errors I applied a wrong patch for 'opl3 - simplify exclusive access lock'. Fixed now. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/opl3.h b/include/sound/opl3.h index eea584b..d7e33ce 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -318,6 +318,7 @@ struct snd_opl3 { void *private_data; void (*private_free)(struct snd_opl3 *); + struct snd_hwdep *hwdep; spinlock_t reg_lock; struct snd_card *card; /* The card that this belongs to */ unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */ diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index a127084..ebe4359 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -525,6 +525,7 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, hw->ops.write = snd_opl3_write; hw->ops.release = snd_opl3_release; + opl3->hwdep = hw; opl3->seq_dev_num = seq_device; #if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) if (snd_seq_device_new(card, seq_device, SNDRV_SEQ_DEV_ID_OPL3, diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index 6fd60b7..2d33f53 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -82,6 +82,7 @@ int snd_opl3_synth_setup(struct snd_opl3 * opl3) void snd_opl3_synth_cleanup(struct snd_opl3 * opl3) { unsigned long flags; + struct snd_hwdep *hwdep; /* Stop system timer */ spin_lock_irqsave(&opl3->sys_timer_lock, flags); -- cgit v0.10.2 From 69b1f1e8337fc94a7ea0730588960e82676dc2dc Mon Sep 17 00:00:00 2001 From: Timofei Bondarenko Date: Tue, 30 Oct 2007 15:28:14 +0100 Subject: [ALSA] usb-audio - SB Live24-External better handling This patch improves support for 'SB Live 24-bit Extarnal' USB card. 1) This card can go into muted state when a headphones connected or disconnected. So notify mixer about changes in headphone jack. 2) Add LED controls and procfs support just as in similar Audigy 2 NX card. 3) Rename 'PCM Capture' conrol to 'Mic Capture' to reflect reality: the card may adjust microphone input level only. Signed-off-by: Timofei Bondarenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 5e32969..1f1e91c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1703,6 +1703,11 @@ static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, case 19: /* speaker out jacks */ case 20: /* headphones out jack */ break; + /* live24ext: 4 = line-in jack */ + case 3: /* hp-out jack (may actuate Mute) */ + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id); + break; default: snd_printd(KERN_DEBUG "memory change in unknown unit %d\n", unitid); break; @@ -1951,6 +1956,9 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { + if (i > 1 && /* Live24ext has 2 LEDs only */ + mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + break; err = snd_ctl_add(mixer->chip->card, snd_ctl_new1(&snd_audigy2nx_controls[i], mixer)); if (err < 0) @@ -1963,28 +1971,42 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - static const struct { + static const struct sb_jack { int unitid; const char *name; - } jacks[] = { + } jacks_audigy2nx[] = { {4, "dig in "}, {7, "line in"}, {19, "spk out"}, {20, "hph out"}, + {-1, NULL} + }, jacks_live24ext[] = { + {4, "line in"}, /* &1=Line, &2=Mic*/ + {3, "hph out"}, /* headphones */ + {0, "RC "}, /* last command, 6 bytes see rc_config above */ + {-1, NULL} }; + const struct sb_jack *jacks; struct usb_mixer_interface *mixer = entry->private_data; int i, err; u8 buf[3]; snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname); - for (i = 0; i < ARRAY_SIZE(jacks); ++i) { + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) + jacks = jacks_audigy2nx; + else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + jacks = jacks_live24ext; + else + return; + + for (i = 0; jacks[i].name; ++i) { snd_iprintf(buffer, "%s: ", jacks[i].name); err = snd_usb_ctl_msg(mixer->chip->dev, usb_rcvctrlpipe(mixer->chip->dev, 0), GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE, 0, jacks[i].unitid << 8, buf, 3, 100); - if (err == 3 && buf[0] == 3) + if (err == 3 && (buf[0] == 3 || buf[0] == 6)) snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); else snd_iprintf(buffer, "?\n"); @@ -2022,7 +2044,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif) if ((err = snd_usb_soundblaster_remote_init(mixer)) < 0) goto _error; - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) { + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) { struct snd_info_entry *entry; if ((err = snd_audigy2nx_controls_create(mixer)) < 0) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index 7c4dcb3..d755be0 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -187,6 +187,13 @@ static struct usbmix_selector_map audigy2nx_selectors[] = { { 0 } /* terminator */ }; +/* Creative SoundBlaster Live! 24-bit External */ +static struct usbmix_name_map live24ext_map[] = { + /* 2: PCM Playback Volume */ + { 5, "Mic Capture" }, /* FU, default PCM Capture Volume */ + { 0 } /* terminator */ +}; + /* LineX FM Transmitter entry - needed to bypass controls bug */ static struct usbmix_name_map linex_map[] = { /* 1: IT pcm */ @@ -273,6 +280,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = audigy2nx_map, .selector_map = audigy2nx_selectors, }, + { + .id = USB_ID(0x041e, 0x3040), + .map = live24ext_map, + }, { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), -- cgit v0.10.2 From f889fa91ad47e6fcb530abf1184ca9f1473d3c72 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 31 Oct 2007 15:49:32 +0100 Subject: [ALSA] hda-codec - Improve the auto-configuration Some small improvements on autocfg stuff: - sort HP pins by sequence number, too - move sole mic pin to AUTO_PIN_MIC instead of AUTO_PIN_FRONT_MIC - ditto for line-in pin Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index dacabe5..39240e0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2608,11 +2608,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, short seq, assoc_line_out, assoc_speaker; short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)]; short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)]; + short sequences_hp[ARRAY_SIZE(cfg->hp_pins)]; memset(cfg, 0, sizeof(*cfg)); memset(sequences_line_out, 0, sizeof(sequences_line_out)); memset(sequences_speaker, 0, sizeof(sequences_speaker)); + memset(sequences_hp, 0, sizeof(sequences_hp)); assoc_line_out = assoc_speaker = 0; nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid_start); @@ -2667,9 +2669,12 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->speaker_outs++; break; case AC_JACK_HP_OUT: + seq = get_defcfg_sequence(def_conf); + assoc = get_defcfg_association(def_conf); if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) continue; cfg->hp_pins[cfg->hp_outs] = nid; + sequences_hp[cfg->hp_outs] = (assoc << 4) | seq; cfg->hp_outs++; break; case AC_JACK_MIC_IN: { @@ -2713,7 +2718,24 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->line_outs); sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker, cfg->speaker_outs); + sort_pins_by_sequence(cfg->hp_pins, sequences_hp, + cfg->hp_outs); + /* if we have only one mic, make it AUTO_PIN_MIC */ + if (!cfg->input_pins[AUTO_PIN_MIC] && + cfg->input_pins[AUTO_PIN_FRONT_MIC]) { + cfg->input_pins[AUTO_PIN_MIC] = + cfg->input_pins[AUTO_PIN_FRONT_MIC]; + cfg->input_pins[AUTO_PIN_FRONT_MIC] = 0; + } + /* ditto for line-in */ + if (!cfg->input_pins[AUTO_PIN_LINE] && + cfg->input_pins[AUTO_PIN_FRONT_LINE]) { + cfg->input_pins[AUTO_PIN_LINE] = + cfg->input_pins[AUTO_PIN_FRONT_LINE]; + cfg->input_pins[AUTO_PIN_FRONT_LINE] = 0; + } + /* * FIX-UP: if no line-outs are detected, try to use speaker or HP pin * as a primary output -- cgit v0.10.2 From 755c48abd8ec228a9e3c3bf9b36850ef5a12cc9c Mon Sep 17 00:00:00 2001 From: Timofei Bondarenko Date: Wed, 31 Oct 2007 17:36:20 +0100 Subject: [ALSA] cmipci at 96kHz This patch adds support for 88.2k, 96k, and 128k samplerates on cmi8738-55 chip. Analog playback works fine on all channels. Analog capture works well too, though the extra samples seems interpolated by hardware. spdif playback and capture works fine. Signed-off-by: Timofei Bondarenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1fa5f00..3a2d942 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -150,6 +150,8 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_CH0_SRATE_176K 0x00000200 #define CM_CH0_SRATE_96K 0x00000200 /* model 055? */ #define CM_CH0_SRATE_88K 0x00000100 +#define CM_CH0_SRATE_128K 0x00000300 +#define CM_CH0_SRATE_MASK 0x00000300 #define CM_SPDIF_INVERSE2 0x00000080 /* model 055? */ #define CM_DBLSPDS 0x00000040 /* double SPDIF sample rate 88.2/96 */ @@ -473,6 +475,7 @@ struct cmipci { unsigned int can_ac3_sw: 1; unsigned int can_ac3_hw: 1; unsigned int can_multi_ch: 1; + unsigned int can_96k: 1; /* samplerate above 48k */ unsigned int do_soft_ac3: 1; unsigned int spdif_playback_avail: 1; /* spdif ready? */ @@ -603,8 +606,6 @@ static unsigned int snd_cmipci_rate_freq(unsigned int rate) { unsigned int i; - if (rate > 48000) - rate /= 2; for (i = 0; i < ARRAY_SIZE(rates); i++) { if (rates[i] == rate) return i; @@ -782,7 +783,7 @@ static int set_dac_channels(struct cmipci *cm, struct cmipci_pcm *rec, int chann static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, struct snd_pcm_substream *substream) { - unsigned int reg, freq, val; + unsigned int reg, freq, freq_ext, val; unsigned int period_size; struct snd_pcm_runtime *runtime = substream->runtime; @@ -830,7 +831,17 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, //snd_printd("cmipci: functrl0 = %08x\n", cm->ctrl); /* set sample rate */ - freq = snd_cmipci_rate_freq(runtime->rate); + freq = 0; + freq_ext = 0; + if (runtime->rate > 48000) + switch (runtime->rate) { + case 88200: freq_ext = CM_CH0_SRATE_88K; break; + case 96000: freq_ext = CM_CH0_SRATE_96K; break; + case 128000: freq_ext = CM_CH0_SRATE_128K; break; + default: snd_BUG(); break; + } + else + freq = snd_cmipci_rate_freq(runtime->rate); val = snd_cmipci_read(cm, CM_REG_FUNCTRL1); if (rec->ch) { val &= ~CM_DSFC_MASK; @@ -851,15 +862,9 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, val &= ~CM_CH0FMT_MASK; val |= rec->fmt << CM_CH0FMT_SHIFT; } - if (cm->chip_version == 68) { - if (runtime->rate == 88200) - val |= CM_CH0_SRATE_88K << (rec->ch * 2); - else - val &= ~(CM_CH0_SRATE_88K << (rec->ch * 2)); - if (runtime->rate == 96000) - val |= CM_CH0_SRATE_96K << (rec->ch * 2); - else - val &= ~(CM_CH0_SRATE_96K << (rec->ch * 2)); + if (cm->can_96k) { + val &= ~(CM_CH0_SRATE_MASK << (rec->ch * 2)); + val |= freq_ext << (rec->ch * 2); } snd_cmipci_write(cm, CM_REG_CHFORMAT, val); //snd_printd("cmipci: chformat = %08x\n", val); @@ -1280,7 +1285,7 @@ static int snd_cmipci_playback_prepare(struct snd_pcm_substream *substream) int rate = substream->runtime->rate; int err, do_spdif, do_ac3 = 0; - do_spdif = (rate >= 44100 && + do_spdif = (rate >= 44100 && rate <= 96000 && substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE && substream->runtime->channels == 2); if (do_spdif && cm->can_ac3_hw) @@ -1336,10 +1341,8 @@ static void snd_cmipci_silence_hack(struct cmipci *cm, struct cmipci_pcm *rec) val = snd_cmipci_read(cm, CM_REG_CHFORMAT); val &= ~(CM_CH0FMT_MASK << (rec->ch * 2)); val |= (3 << CM_CH0FMT_SHIFT) << (rec->ch * 2); - if (cm->chip_version == 68) { - val &= ~(CM_CH0_SRATE_88K << (rec->ch * 2)); - val &= ~(CM_CH0_SRATE_96K << (rec->ch * 2)); - } + if (cm->can_96k) + val &= ~(CM_CH0_SRATE_MASK << (rec->ch * 2)); snd_cmipci_write(cm, CM_REG_CHFORMAT, val); /* start stream (we don't need interrupts) */ @@ -1391,6 +1394,12 @@ static int snd_cmipci_capture_spdif_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&cm->reg_lock); snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_CAPTURE_SPDF); + if (cm->can_96k) { + if (substream->runtime->rate > 48000) + snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS); + else + snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS); + } spin_unlock_irq(&cm->reg_lock); return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_CAPT], substream); @@ -1567,6 +1576,14 @@ static struct snd_pcm_hardware snd_cmipci_capture_spdif = .fifo_size = 0, }; +static unsigned int rate_constraints[] = { 5512, 8000, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, 96000, 128000 }; +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rate_constraints), + .list = rate_constraints, + .mask = 0, +}; + /* * check device open/close */ @@ -1636,6 +1653,13 @@ static int snd_cmipci_playback_open(struct snd_pcm_substream *substream) runtime->hw.rates |= SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000; runtime->hw.rate_max = 96000; + } else if (cm->chip_version == 55) { + err = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (err < 0) + return err; + runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; + runtime->hw.rate_max = 128000; } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); cm->dig_pcm_status = cm->dig_status; @@ -1654,6 +1678,13 @@ static int snd_cmipci_capture_open(struct snd_pcm_substream *substream) if (cm->chip_version == 68) { // 8768 only supports 44k/48k recording runtime->hw.rate_min = 41000; runtime->hw.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; + } else if (cm->chip_version == 55) { + err = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (err < 0) + return err; + runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; + runtime->hw.rate_max = 128000; } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); return 0; @@ -1685,6 +1716,13 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream) runtime->hw.rates |= SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000; runtime->hw.rate_max = 96000; + } else if (cm->chip_version == 55) { + err = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (err < 0) + return err; + runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; + runtime->hw.rate_max = 128000; } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); return 0; @@ -1704,7 +1742,7 @@ static int snd_cmipci_playback_spdif_open(struct snd_pcm_substream *substream) runtime->hw.formats |= SNDRV_PCM_FMTBIT_S32_LE; snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); } - if (cm->chip_version == 68) { + if (cm->can_96k) { runtime->hw.rates |= SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000; runtime->hw.rate_max = 96000; @@ -1726,6 +1764,11 @@ static int snd_cmipci_capture_spdif_open(struct snd_pcm_substream *substream) if ((err = open_device_check(cm, CM_OPEN_SPDIF_CAPTURE, substream)) < 0) /* use channel B */ return err; runtime->hw = snd_cmipci_capture_spdif; + if (cm->can_96k && !(cm->chip_version == 68)) { + runtime->hw.rates |= SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + runtime->hw.rate_max = 96000; + } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x40000); return 0; } @@ -2785,9 +2828,11 @@ static void __devinit query_chip(struct cmipci *cm) } else if (detect & CM_CHIP_8768) { cm->chip_version = 68; cm->max_channels = 8; + cm->can_96k = 1; } else { cm->chip_version = 55; cm->max_channels = 6; + cm->can_96k = 1; } cm->can_ac3_hw = 1; cm->can_multi_ch = 1; -- cgit v0.10.2 From d9f9b8baa0fbdba93b0ebb8e5b3ec042a6b4a8fb Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Mon, 5 Nov 2007 15:13:51 +0100 Subject: [ALSA] HDA: Add Asus VX1 support Simple patch to add the Asus VX1 laptop to the Analog Devices pci quirk list. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 196ad3c..cfe064a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -841,6 +841,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD), + SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP), -- cgit v0.10.2 From f1f208d0b4fb79f99d2ca5031c61ff5b52e42e75 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 5 Nov 2007 15:30:13 +0100 Subject: [ALSA] hda: STAC9228 DMIC Added support for the dmics and enabled EAPD for several laptops with STAC9228 cards. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 876e1d4..4dc09ef 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2723,14 +2723,21 @@ static int patch_stac927x(struct hda_codec *codec) spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; spec->dmux_nid = 0x1b; + + /* Enable DMIC0 */ + stac92xx_set_config_reg(codec, 0x13, 0x90a60040); + + /* GPIO2 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000004; break; default: - spec->num_dmics = 0; + spec->num_dmics = 0; + + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; } spec->multiout.dac_nids = spec->dac_nids; - /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); -- cgit v0.10.2 From a9f00d8df2115b396f13ea74b835f18215a871cc Mon Sep 17 00:00:00 2001 From: Joachim Foerster Date: Mon, 5 Nov 2007 16:06:01 +0100 Subject: [ALSA] Xilinx ML403 AC97 Controller Reference device driver Add ALSA support for the opb_ac97_controller_ref_v1_00_a ip core found in Xilinx' ML403 reference design. Known issue: Currently this driver hits a WARN_ON_ONCE(1) statement in kernel/irq/resend.c (line 70). According to Linus (http://lkml.org/lkml/2007/8/5/5) this may be ignored, right? I haven't had a look into this 'problem' yet. Signed-off-by: Joachim Foerster Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 83529b0..75d4fe0 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -120,4 +120,16 @@ config SND_PORTMAN2X4 To compile this driver as a module, choose M here: the module will be called snd-portman2x4. +config SND_ML403_AC97CR + tristate "Xilinx ML403 AC97 Controller Reference" + depends on SND && XILINX_VIRTEX + select SND_AC97_CODEC + help + Say Y here to include support for the + opb_ac97_controller_ref_v1_00_a ip core found in Xilinx' ML403 + reference design. + + To compile this driver as a module, choose M here: the module + will be called snd-ml403_ac97cr. + endmenu diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index 80aeff5..8e55300 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -9,6 +9,7 @@ snd-mts64-objs := mts64.o snd-portman2x4-objs := portman2x4.o snd-serial-u16550-objs := serial-u16550.o snd-virmidi-objs := virmidi.o +snd-ml403-ac97cr-objs := ml403-ac97cr.o pcm-indirect2.o # Toplevel Module Dependency obj-$(CONFIG_SND_DUMMY) += snd-dummy.o @@ -17,5 +18,6 @@ obj-$(CONFIG_SND_SERIAL_U16550) += snd-serial-u16550.o obj-$(CONFIG_SND_MTPAV) += snd-mtpav.o obj-$(CONFIG_SND_MTS64) += snd-mts64.o obj-$(CONFIG_SND_PORTMAN2X4) += snd-portman2x4.o +obj-$(CONFIG_SND_ML403_AC97CR) += snd-ml403-ac97cr.o obj-$(CONFIG_SND) += opl3/ opl4/ mpu401/ vx/ diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c new file mode 100644 index 0000000..2222315 --- /dev/null +++ b/sound/drivers/ml403-ac97cr.c @@ -0,0 +1,1353 @@ +/* + * ALSA driver for Xilinx ML403 AC97 Controller Reference + * IP: opb_ac97_controller_ref_v1_00_a (EDK 8.1i) + * IP: opb_ac97_controller_ref_v1_00_a (EDK 9.1i) + * + * Copyright (c) by 2007 Joachim Foerster + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/* Some notes / status of this driver: + * + * - Don't wonder about some strange implementations of things - especially the + * (heavy) shadowing of codec registers, with which I tried to reduce read + * accesses to a minimum, because after a variable amount of accesses, the AC97 + * controller doesn't raise the register access finished bit anymore ... + * + * - Capture support works - basically, but after ~30s (with rates > ~20kHz) + * ALSA stops reading captured samples from the intermediate buffer and + * therefore a overrun happens - ATM I don't know what's wrong. + * + * - Playback support seems to be pretty stable - no issues here. + */ + +#include +#include +#include + +#include + +#include +#include +#include + +/* HZ */ +#include +/* jiffies, time_*() */ +#include +/* schedule_timeout*() */ +#include +/* spin_lock*() */ +#include +/* struct mutex, mutex_init(), mutex_*lock() */ +#include + +/* snd_printk(), snd_printd() */ +#include +#include +#include +#include +#include + +#include "pcm-indirect2.h" + + +#define SND_ML403_AC97CR_DRIVER "ml403-ac97cr" + +MODULE_AUTHOR("Joachim Foerster "); +MODULE_DESCRIPTION("Xilinx ML403 AC97 Controller Reference"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Xilinx,ML403 AC97 Controller Reference}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for ML403 AC97 Controller Reference."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for ML403 AC97 Controller Reference."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable this ML403 AC97 Controller Reference."); + +/* Special feature options */ +/*#define CODEC_WRITE_CHECK_RAF*/ /* don't return after a write to a codec + * register, while RAF bit is not set + */ +/* Debug options for code which may be removed completely in a final version */ +#ifdef CONFIG_SND_DEBUG +/*#define CODEC_STAT*/ /* turn on some minimal "statistics" + * about codec register usage + */ +#define SND_PCM_INDIRECT2_STAT /* turn on some "statistics" about the + * process of copying bytes from the + * intermediate buffer to the hardware + * fifo and the other way round + */ +#endif + +/* Definition of a "level/facility dependent" printk(); may be removed + * completely in a final version + */ +#undef PDEBUG +#ifdef CONFIG_SND_DEBUG +/* "facilities" for PDEBUG */ +#define UNKNOWN (1<<0) +#define CODEC_SUCCESS (1<<1) +#define CODEC_FAKE (1<<2) +#define INIT_INFO (1<<3) +#define INIT_FAILURE (1<<4) +#define WORK_INFO (1<<5) +#define WORK_FAILURE (1<<6) + +#define PDEBUG_FACILITIES (UNKNOWN | INIT_FAILURE | WORK_FAILURE) + +#define PDEBUG(fac, fmt, args...) do { \ + if (fac & PDEBUG_FACILITIES) \ + snd_printd(KERN_DEBUG SND_ML403_AC97CR_DRIVER ": " \ + fmt, ##args); \ + } while (0) +#else +#define PDEBUG(fac, fmt, args...) /* nothing */ +#endif + + + +/* Defines for "waits"/timeouts (portions of HZ=250 on arch/ppc by default) */ +#define CODEC_TIMEOUT_ON_INIT 5 /* timeout for checking for codec + * readiness (after insmod) + */ +#ifndef CODEC_WRITE_CHECK_RAF +#define CODEC_WAIT_AFTER_WRITE 100 /* general, static wait after a write + * access to a codec register, may be + * 0 to completely remove wait + */ +#else +#define CODEC_TIMEOUT_AFTER_WRITE 5 /* timeout after a write access to a + * codec register, if RAF bit is used + */ +#endif +#define CODEC_TIMEOUT_AFTER_READ 5 /* timeout after a read access to a + * codec register (checking RAF bit) + */ + +/* Infrastructure for codec register shadowing */ +#define LM4550_REG_OK (1<<0) /* register exists */ +#define LM4550_REG_DONEREAD (1<<1) /* read register once, value should be + * the same currently in the register + */ +#define LM4550_REG_NOSAVE (1<<2) /* values written to this register will + * not be saved in the register + */ +#define LM4550_REG_NOSHADOW (1<<3) /* don't do register shadowing, use plain + * hardware access + */ +#define LM4550_REG_READONLY (1<<4) /* register is read only */ +#define LM4550_REG_FAKEPROBE (1<<5) /* fake write _and_ read actions during + * probe() correctly + */ +#define LM4550_REG_FAKEREAD (1<<6) /* fake read access, always return + * default value + */ +#define LM4550_REG_ALLFAKE (LM4550_REG_FAKEREAD | LM4550_REG_FAKEPROBE) + +struct lm4550_reg { + u16 value; + u16 flag; + u16 wmask; + u16 def; +}; + +struct lm4550_reg lm4550_regfile[64] = { + [AC97_RESET / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_NOSAVE \ + | LM4550_REG_FAKEREAD, + .def = 0x0D50}, + [AC97_MASTER / 2] = {.flag = LM4550_REG_OK + | LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8000}, + [AC97_HEADPHONE / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8000}, + [AC97_MASTER_MONO / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x801F, + .def = 0x8000}, + [AC97_PC_BEEP / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x801E, + .def = 0x0}, + [AC97_PHONE / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x801F, + .def = 0x8008}, + [AC97_MIC / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x805F, + .def = 0x8008}, + [AC97_LINE / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8808}, + [AC97_CD / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8808}, + [AC97_VIDEO / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8808}, + [AC97_AUX / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8808}, + [AC97_PCM / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8008}, + [AC97_REC_SEL / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x707, + .def = 0x0}, + [AC97_REC_GAIN / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .wmask = 0x8F0F, + .def = 0x8000}, + [AC97_GENERAL_PURPOSE / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .def = 0x0, + .wmask = 0xA380}, + [AC97_3D_CONTROL / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEREAD \ + | LM4550_REG_READONLY, + .def = 0x0101}, + [AC97_POWERDOWN / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_NOSHADOW \ + | LM4550_REG_NOSAVE, + .wmask = 0xFF00}, + /* may not write ones to + * REF/ANL/DAC/ADC bits + * FIXME: Is this ok? + */ + [AC97_EXTENDED_ID / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEREAD \ + | LM4550_REG_READONLY, + .def = 0x0201}, /* primary codec */ + [AC97_EXTENDED_STATUS / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_NOSHADOW \ + | LM4550_REG_NOSAVE, + .wmask = 0x1}, + [AC97_PCM_FRONT_DAC_RATE / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .def = 0xBB80, + .wmask = 0xFFFF}, + [AC97_PCM_LR_ADC_RATE / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_FAKEPROBE, + .def = 0xBB80, + .wmask = 0xFFFF}, + [AC97_VENDOR_ID1 / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_READONLY \ + | LM4550_REG_FAKEREAD, + .def = 0x4E53}, + [AC97_VENDOR_ID2 / 2] = {.flag = LM4550_REG_OK \ + | LM4550_REG_READONLY \ + | LM4550_REG_FAKEREAD, + .def = 0x4350} +}; + +#define LM4550_RF_OK(reg) (lm4550_regfile[reg / 2].flag & LM4550_REG_OK) + +static void lm4550_regfile_init(void) +{ + int i; + for (i = 0; i < 64; i++) + if (lm4550_regfile[i].flag & LM4550_REG_FAKEPROBE) + lm4550_regfile[i].value = lm4550_regfile[i].def; +} + +static void lm4550_regfile_write_values_after_init(struct snd_ac97 *ac97) +{ + int i; + for (i = 0; i < 64; i++) + if ((lm4550_regfile[i].flag & LM4550_REG_FAKEPROBE) && + (lm4550_regfile[i].value != lm4550_regfile[i].def)) { + PDEBUG(CODEC_FAKE, "lm4550_regfile_write_values_after_" + "init(): reg=0x%x value=0x%x / %d is different " + "from def=0x%x / %d\n", + i, lm4550_regfile[i].value, + lm4550_regfile[i].value, lm4550_regfile[i].def, + lm4550_regfile[i].def); + snd_ac97_write(ac97, i * 2, lm4550_regfile[i].value); + lm4550_regfile[i].flag |= LM4550_REG_DONEREAD; + } +} + + +/* direct registers */ +#define CR_REG(ml403_ac97cr, x) ((ml403_ac97cr)->port + CR_REG_##x) + +#define CR_REG_PLAYFIFO 0x00 +#define CR_PLAYDATA(a) ((a) & 0xFFFF) + +#define CR_REG_RECFIFO 0x04 +#define CR_RECDATA(a) ((a) & 0xFFFF) + +#define CR_REG_STATUS 0x08 +#define CR_RECOVER (1<<7) +#define CR_PLAYUNDER (1<<6) +#define CR_CODECREADY (1<<5) +#define CR_RAF (1<<4) +#define CR_RECEMPTY (1<<3) +#define CR_RECFULL (1<<2) +#define CR_PLAYHALF (1<<1) +#define CR_PLAYFULL (1<<0) + +#define CR_REG_RESETFIFO 0x0C +#define CR_RECRESET (1<<1) +#define CR_PLAYRESET (1<<0) + +#define CR_REG_CODEC_ADDR 0x10 +/* UG082 says: + * #define CR_CODEC_ADDR(a) ((a) << 1) + * #define CR_CODEC_READ (1<<0) + * #define CR_CODEC_WRITE (0<<0) + */ +/* RefDesign example says: */ +#define CR_CODEC_ADDR(a) ((a) << 0) +#define CR_CODEC_READ (1<<7) +#define CR_CODEC_WRITE (0<<7) + +#define CR_REG_CODEC_DATAREAD 0x14 +#define CR_CODEC_DATAREAD(v) ((v) & 0xFFFF) + +#define CR_REG_CODEC_DATAWRITE 0x18 +#define CR_CODEC_DATAWRITE(v) ((v) & 0xFFFF) + +#define CR_FIFO_SIZE 32 + +struct snd_ml403_ac97cr { + /* lock for access to (controller) registers */ + spinlock_t reg_lock; + /* mutex for the whole sequence of accesses to (controller) registers + * which affect codec registers + */ + struct mutex cdc_mutex; + + int irq; /* for playback */ + int enable_irq; /* for playback */ + + int capture_irq; + int enable_capture_irq; + + struct resource *res_port; + void *port; + + struct snd_ac97 *ac97; + int ac97_fake; +#ifdef CODEC_STAT + int ac97_read; + int ac97_write; +#endif + + struct platform_device *pfdev; + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + + struct snd_pcm_indirect2 ind_rec; /* for playback */ + struct snd_pcm_indirect2 capture_ind2_rec; +}; + +static struct snd_pcm_hardware snd_ml403_ac97cr_playback = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_48000), + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = CR_FIFO_SIZE/2, + .period_bytes_max = (64*1024), + .periods_min = 2, + .periods_max = (128*1024)/(CR_FIFO_SIZE/2), + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_ml403_ac97cr_capture = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_48000), + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = CR_FIFO_SIZE/2, + .period_bytes_max = (64*1024), + .periods_min = 2, + .periods_max = (128*1024)/(CR_FIFO_SIZE/2), + .fifo_size = 0, +}; + +static size_t +snd_ml403_ac97cr_playback_ind2_zero(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int copied_words = 0; + u32 full = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + spin_lock(&ml403_ac97cr->reg_lock); + while ((full = (in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_PLAYFULL)) != CR_PLAYFULL) { + out_be32(CR_REG(ml403_ac97cr, PLAYFIFO), 0); + copied_words++; + } + rec->hw_ready = 0; + spin_unlock(&ml403_ac97cr->reg_lock); + + return (size_t) (copied_words * 2); +} + +static size_t +snd_ml403_ac97cr_playback_ind2_copy(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + size_t bytes) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + u16 *src; + int copied_words = 0; + u32 full = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + src = (u16 *)(substream->runtime->dma_area + rec->sw_data); + + spin_lock(&ml403_ac97cr->reg_lock); + while (((full = (in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_PLAYFULL)) != CR_PLAYFULL) && (bytes > 1)) { + out_be32(CR_REG(ml403_ac97cr, PLAYFIFO), + CR_PLAYDATA(src[copied_words])); + copied_words++; + bytes = bytes - 2; + } + if (full != CR_PLAYFULL) + rec->hw_ready = 1; + else + rec->hw_ready = 0; + spin_unlock(&ml403_ac97cr->reg_lock); + + return (size_t) (copied_words * 2); +} + +static size_t +snd_ml403_ac97cr_capture_ind2_null(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int copied_words = 0; + u32 empty = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + spin_lock(&ml403_ac97cr->reg_lock); + while ((empty = (in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_RECEMPTY)) != CR_RECEMPTY) { + volatile u32 trash; + + trash = CR_RECDATA(in_be32(CR_REG(ml403_ac97cr, RECFIFO))); + /* Hmmmm, really necessary? Don't want call to in_be32() + * to be optimised away! + */ + trash++; + copied_words++; + } + rec->hw_ready = 0; + spin_unlock(&ml403_ac97cr->reg_lock); + + return (size_t) (copied_words * 2); +} + +static size_t +snd_ml403_ac97cr_capture_ind2_copy(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, size_t bytes) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + u16 *dst; + int copied_words = 0; + u32 empty = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + dst = (u16 *)(substream->runtime->dma_area + rec->sw_data); + + spin_lock(&ml403_ac97cr->reg_lock); + while (((empty = (in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_RECEMPTY)) != CR_RECEMPTY) && (bytes > 1)) { + dst[copied_words] = CR_RECDATA(in_be32(CR_REG(ml403_ac97cr, + RECFIFO))); + copied_words++; + bytes = bytes - 2; + } + if (empty != CR_RECEMPTY) + rec->hw_ready = 1; + else + rec->hw_ready = 0; + spin_unlock(&ml403_ac97cr->reg_lock); + + return (size_t) (copied_words * 2); +} + +static snd_pcm_uframes_t +snd_ml403_ac97cr_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_indirect2 *ind2_rec = NULL; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + if (substream == ml403_ac97cr->playback_substream) + ind2_rec = &ml403_ac97cr->ind_rec; + if (substream == ml403_ac97cr->capture_substream) + ind2_rec = &ml403_ac97cr->capture_ind2_rec; + + if (ind2_rec != NULL) + return snd_pcm_indirect2_pointer(substream, ind2_rec); + return (snd_pcm_uframes_t) 0; +} + +static int +snd_ml403_ac97cr_pcm_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int err = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + PDEBUG(WORK_INFO, "trigger(playback): START\n"); + ml403_ac97cr->ind_rec.hw_ready = 1; + + /* clear play FIFO */ + out_be32(CR_REG(ml403_ac97cr, RESETFIFO), CR_PLAYRESET); + + /* enable play irq */ + ml403_ac97cr->enable_irq = 1; + enable_irq(ml403_ac97cr->irq); + break; + case SNDRV_PCM_TRIGGER_STOP: + PDEBUG(WORK_INFO, "trigger(playback): STOP\n"); + ml403_ac97cr->ind_rec.hw_ready = 0; +#ifdef SND_PCM_INDIRECT2_STAT + snd_pcm_indirect2_stat(substream, &ml403_ac97cr->ind_rec); +#endif + /* disable play irq */ + disable_irq_nosync(ml403_ac97cr->irq); + ml403_ac97cr->enable_irq = 0; + break; + default: + err = -EINVAL; + break; + } + PDEBUG(WORK_INFO, "trigger(playback): (done)\n"); + return err; +} + +static int +snd_ml403_ac97cr_pcm_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int err = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + PDEBUG(WORK_INFO, "trigger(capture): START\n"); + ml403_ac97cr->capture_ind2_rec.hw_ready = 0; + + /* clear record FIFO */ + out_be32(CR_REG(ml403_ac97cr, RESETFIFO), CR_RECRESET); + + /* enable record irq */ + ml403_ac97cr->enable_capture_irq = 1; + enable_irq(ml403_ac97cr->capture_irq); + break; + case SNDRV_PCM_TRIGGER_STOP: + PDEBUG(WORK_INFO, "trigger(capture): STOP\n"); + ml403_ac97cr->capture_ind2_rec.hw_ready = 0; +#ifdef SND_PCM_INDIRECT2_STAT + snd_pcm_indirect2_stat(substream, + &ml403_ac97cr->capture_ind2_rec); +#endif + /* disable capture irq */ + disable_irq_nosync(ml403_ac97cr->capture_irq); + ml403_ac97cr->enable_capture_irq = 0; + break; + default: + err = -EINVAL; + break; + } + PDEBUG(WORK_INFO, "trigger(capture): (done)\n"); + return err; +} + +static int +snd_ml403_ac97cr_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_runtime *runtime; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + + PDEBUG(WORK_INFO, + "prepare(): period_bytes=%d, minperiod_bytes=%d\n", + snd_pcm_lib_period_bytes(substream), CR_FIFO_SIZE / 2); + + /* set sampling rate */ + snd_ac97_set_rate(ml403_ac97cr->ac97, AC97_PCM_FRONT_DAC_RATE, + runtime->rate); + PDEBUG(WORK_INFO, "prepare(): rate=%d\n", runtime->rate); + + /* init struct for intermediate buffer */ + memset(&ml403_ac97cr->ind_rec, 0, + sizeof(struct snd_pcm_indirect2)); + ml403_ac97cr->ind_rec.hw_buffer_size = CR_FIFO_SIZE; + ml403_ac97cr->ind_rec.sw_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + ml403_ac97cr->ind_rec.min_periods = -1; + ml403_ac97cr->ind_rec.min_multiple = + snd_pcm_lib_period_bytes(substream) / (CR_FIFO_SIZE / 2); + PDEBUG(WORK_INFO, "prepare(): hw_buffer_size=%d, " + "sw_buffer_size=%d, min_multiple=%d\n", + CR_FIFO_SIZE, ml403_ac97cr->ind_rec.sw_buffer_size, + ml403_ac97cr->ind_rec.min_multiple); + return 0; +} + +static int +snd_ml403_ac97cr_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_runtime *runtime; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + + PDEBUG(WORK_INFO, + "prepare(capture): period_bytes=%d, minperiod_bytes=%d\n", + snd_pcm_lib_period_bytes(substream), CR_FIFO_SIZE / 2); + + /* set sampling rate */ + snd_ac97_set_rate(ml403_ac97cr->ac97, AC97_PCM_LR_ADC_RATE, + runtime->rate); + PDEBUG(WORK_INFO, "prepare(capture): rate=%d\n", runtime->rate); + + /* init struct for intermediate buffer */ + memset(&ml403_ac97cr->capture_ind2_rec, 0, + sizeof(struct snd_pcm_indirect2)); + ml403_ac97cr->capture_ind2_rec.hw_buffer_size = CR_FIFO_SIZE; + ml403_ac97cr->capture_ind2_rec.sw_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + ml403_ac97cr->capture_ind2_rec.min_multiple = + snd_pcm_lib_period_bytes(substream) / (CR_FIFO_SIZE / 2); + PDEBUG(WORK_INFO, "prepare(capture): hw_buffer_size=%d, " + "sw_buffer_size=%d, min_multiple=%d\n", CR_FIFO_SIZE, + ml403_ac97cr->capture_ind2_rec.sw_buffer_size, + ml403_ac97cr->capture_ind2_rec.min_multiple); + return 0; +} + +static int snd_ml403_ac97cr_hw_free(struct snd_pcm_substream *substream) +{ + PDEBUG(WORK_INFO, "hw_free()\n"); + return snd_pcm_lib_free_pages(substream); +} + +static int +snd_ml403_ac97cr_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + PDEBUG(WORK_INFO, "hw_params(): desired buffer bytes=%d, desired " + "period bytes=%d\n", + params_buffer_bytes(hw_params), params_period_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_ml403_ac97cr_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_runtime *runtime; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + + PDEBUG(WORK_INFO, "open(playback)\n"); + ml403_ac97cr->playback_substream = substream; + runtime->hw = snd_ml403_ac97cr_playback; + + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + CR_FIFO_SIZE / 2); + return 0; +} + +static int snd_ml403_ac97cr_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_runtime *runtime; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + + PDEBUG(WORK_INFO, "open(capture)\n"); + ml403_ac97cr->capture_substream = substream; + runtime->hw = snd_ml403_ac97cr_capture; + + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + CR_FIFO_SIZE / 2); + return 0; +} + +static int snd_ml403_ac97cr_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + PDEBUG(WORK_INFO, "close(playback)\n"); + ml403_ac97cr->playback_substream = NULL; + return 0; +} + +static int snd_ml403_ac97cr_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + PDEBUG(WORK_INFO, "close(capture)\n"); + ml403_ac97cr->capture_substream = NULL; + return 0; +} + +static struct snd_pcm_ops snd_ml403_ac97cr_playback_ops = { + .open = snd_ml403_ac97cr_playback_open, + .close = snd_ml403_ac97cr_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ml403_ac97cr_hw_params, + .hw_free = snd_ml403_ac97cr_hw_free, + .prepare = snd_ml403_ac97cr_pcm_playback_prepare, + .trigger = snd_ml403_ac97cr_pcm_playback_trigger, + .pointer = snd_ml403_ac97cr_pcm_pointer, +}; + +static struct snd_pcm_ops snd_ml403_ac97cr_capture_ops = { + .open = snd_ml403_ac97cr_capture_open, + .close = snd_ml403_ac97cr_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ml403_ac97cr_hw_params, + .hw_free = snd_ml403_ac97cr_hw_free, + .prepare = snd_ml403_ac97cr_pcm_capture_prepare, + .trigger = snd_ml403_ac97cr_pcm_capture_trigger, + .pointer = snd_ml403_ac97cr_pcm_pointer, +}; + +static irqreturn_t snd_ml403_ac97cr_irq(int irq, void *dev_id) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct platform_device *pfdev; + int cmp_irq; + + ml403_ac97cr = (struct snd_ml403_ac97cr *)dev_id; + if (ml403_ac97cr == NULL) + return IRQ_NONE; + + pfdev = ml403_ac97cr->pfdev; + + /* playback interrupt */ + cmp_irq = platform_get_irq(pfdev, 0); + if (irq == cmp_irq) { + if (ml403_ac97cr->enable_irq) + snd_pcm_indirect2_playback_interrupt( + ml403_ac97cr->playback_substream, + &ml403_ac97cr->ind_rec, + snd_ml403_ac97cr_playback_ind2_copy, + snd_ml403_ac97cr_playback_ind2_zero); + else + goto __disable_irq; + } else { + /* record interrupt */ + cmp_irq = platform_get_irq(pfdev, 1); + if (irq == cmp_irq) { + if (ml403_ac97cr->enable_capture_irq) + snd_pcm_indirect2_capture_interrupt( + ml403_ac97cr->capture_substream, + &ml403_ac97cr->capture_ind2_rec, + snd_ml403_ac97cr_capture_ind2_copy, + snd_ml403_ac97cr_capture_ind2_null); + else + goto __disable_irq; + } else + return IRQ_NONE; + } + return IRQ_HANDLED; + +__disable_irq: + PDEBUG(INIT_INFO, "irq(): irq %d is meant to be disabled! So, now try " + "to disable it _really_!\n", irq); + disable_irq_nosync(irq); + return IRQ_HANDLED; +} + +static unsigned short +snd_ml403_ac97cr_codec_read(struct snd_ac97 *ac97, unsigned short reg) +{ + struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data; +#ifdef CODEC_STAT + u32 stat; + u32 rafaccess = 0; +#endif + unsigned long end_time; + u16 value = 0; + + if (!LM4550_RF_OK(reg)) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "access to unknown/unused codec register 0x%x " + "ignored!\n", reg); + return 0; + } + /* check if we can fake/answer this access from our shadow register */ + if ((lm4550_regfile[reg / 2].flag & + (LM4550_REG_DONEREAD | LM4550_REG_ALLFAKE)) && + !(lm4550_regfile[reg / 2].flag & LM4550_REG_NOSHADOW)) { + if (lm4550_regfile[reg / 2].flag & LM4550_REG_FAKEREAD) { + PDEBUG(CODEC_FAKE, "codec_read(): faking read from " + "reg=0x%x, val=0x%x / %d\n", + reg, lm4550_regfile[reg / 2].def, + lm4550_regfile[reg / 2].def); + return lm4550_regfile[reg / 2].def; + } else if ((lm4550_regfile[reg / 2].flag & + LM4550_REG_FAKEPROBE) && + ml403_ac97cr->ac97_fake) { + PDEBUG(CODEC_FAKE, "codec_read(): faking read from " + "reg=0x%x, val=0x%x / %d (probe)\n", + reg, lm4550_regfile[reg / 2].value, + lm4550_regfile[reg / 2].value); + return lm4550_regfile[reg / 2].value; + } else { +#ifdef CODEC_STAT + PDEBUG(CODEC_FAKE, "codec_read(): read access " + "answered by shadow register 0x%x (value=0x%x " + "/ %d) (cw=%d cr=%d)\n", + reg, lm4550_regfile[reg / 2].value, + lm4550_regfile[reg / 2].value, + ml403_ac97cr->ac97_write, + ml403_ac97cr->ac97_read); +#else + PDEBUG(CODEC_FAKE, "codec_read(): read access " + "answered by shadow register 0x%x (value=0x%x " + "/ %d)\n", + reg, lm4550_regfile[reg / 2].value, + lm4550_regfile[reg / 2].value); +#endif + return lm4550_regfile[reg / 2].value; + } + } + /* if we are here, we _have_ to access the codec really, no faking */ + if (mutex_lock_interruptible(&ml403_ac97cr->cdc_mutex) != 0) + return 0; +#ifdef CODEC_STAT + ml403_ac97cr->ac97_read++; +#endif + spin_lock(&ml403_ac97cr->reg_lock); + out_be32(CR_REG(ml403_ac97cr, CODEC_ADDR), + CR_CODEC_ADDR(reg) | CR_CODEC_READ); + spin_unlock(&ml403_ac97cr->reg_lock); + end_time = jiffies + (HZ / CODEC_TIMEOUT_AFTER_READ); + do { + spin_lock(&ml403_ac97cr->reg_lock); +#ifdef CODEC_STAT + rafaccess++; + stat = in_be32(CR_REG(ml403_ac97cr, STATUS)); + if ((stat & CR_RAF) == CR_RAF) { + value = CR_CODEC_DATAREAD( + in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD))); + PDEBUG(CODEC_SUCCESS, "codec_read(): (done) reg=0x%x, " + "value=0x%x / %d (STATUS=0x%x)\n", + reg, value, value, stat); +#else + if ((in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_RAF) == CR_RAF) { + value = CR_CODEC_DATAREAD( + in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD))); + PDEBUG(CODEC_SUCCESS, "codec_read(): (done) " + "reg=0x%x, value=0x%x / %d\n", + reg, value, value); +#endif + lm4550_regfile[reg / 2].value = value; + lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD; + spin_unlock(&ml403_ac97cr->reg_lock); + mutex_unlock(&ml403_ac97cr->cdc_mutex); + return value; + } + spin_unlock(&ml403_ac97cr->reg_lock); + schedule_timeout_uninterruptible(1); + } while (time_after(end_time, jiffies)); + /* read the DATAREAD register anyway, see comment below */ + spin_lock(&ml403_ac97cr->reg_lock); + value = + CR_CODEC_DATAREAD(in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD))); + spin_unlock(&ml403_ac97cr->reg_lock); +#ifdef CODEC_STAT + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "timeout while codec read! " + "(reg=0x%x, last STATUS=0x%x, DATAREAD=0x%x / %d, %d) " + "(cw=%d, cr=%d)\n", + reg, stat, value, value, rafaccess, + ml403_ac97cr->ac97_write, ml403_ac97cr->ac97_read); +#else + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "timeout while codec read! " + "(reg=0x%x, DATAREAD=0x%x / %d)\n", + reg, value, value); +#endif + /* BUG: This is PURE speculation! But after _most_ read timeouts the + * value in the register is ok! + */ + lm4550_regfile[reg / 2].value = value; + lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD; + mutex_unlock(&ml403_ac97cr->cdc_mutex); + return value; +} + +static void +snd_ml403_ac97cr_codec_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data; + +#ifdef CODEC_STAT + u32 stat; + u32 rafaccess = 0; +#endif +#ifdef CODEC_WRITE_CHECK_RAF + unsigned long end_time; +#endif + + if (!LM4550_RF_OK(reg)) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "access to unknown/unused codec register 0x%x " + "ignored!\n", reg); + return; + } + if (lm4550_regfile[reg / 2].flag & LM4550_REG_READONLY) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "write access to read only codec register 0x%x " + "ignored!\n", reg); + return; + } + if ((val & lm4550_regfile[reg / 2].wmask) != val) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "write access to codec register 0x%x " + "with bad value 0x%x / %d!\n", + reg, val, val); + val = val & lm4550_regfile[reg / 2].wmask; + } + if (((lm4550_regfile[reg / 2].flag & LM4550_REG_FAKEPROBE) && + ml403_ac97cr->ac97_fake) && + !(lm4550_regfile[reg / 2].flag & LM4550_REG_NOSHADOW)) { + PDEBUG(CODEC_FAKE, "codec_write(): faking write to reg=0x%x, " + "val=0x%x / %d\n", reg, val, val); + lm4550_regfile[reg / 2].value = (val & + lm4550_regfile[reg / 2].wmask); + return; + } + if (mutex_lock_interruptible(&ml403_ac97cr->cdc_mutex) != 0) + return; +#ifdef CODEC_STAT + ml403_ac97cr->ac97_write++; +#endif + spin_lock(&ml403_ac97cr->reg_lock); + out_be32(CR_REG(ml403_ac97cr, CODEC_DATAWRITE), + CR_CODEC_DATAWRITE(val)); + out_be32(CR_REG(ml403_ac97cr, CODEC_ADDR), + CR_CODEC_ADDR(reg) | CR_CODEC_WRITE); + spin_unlock(&ml403_ac97cr->reg_lock); +#ifdef CODEC_WRITE_CHECK_RAF + /* check CR_CODEC_RAF bit to see if write access to register is done; + * loop until bit is set or timeout happens + */ + end_time = jiffies + HZ / CODEC_TIMEOUT_AFTER_WRITE; + do { + spin_lock(&ml403_ac97cr->reg_lock); +#ifdef CODEC_STAT + rafaccess++; + stat = in_be32(CR_REG(ml403_ac97cr, STATUS)) + if ((stat & CR_RAF) == CR_RAF) { +#else + if ((in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_RAF) == CR_RAF) { +#endif + PDEBUG(CODEC_SUCCESS, "codec_write(): (done) " + "reg=0x%x, value=%d / 0x%x\n", + reg, val, val); + if (!(lm4550_regfile[reg / 2].flag & + LM4550_REG_NOSHADOW) && + !(lm4550_regfile[reg / 2].flag & + LM4550_REG_NOSAVE)) + lm4550_regfile[reg / 2].value = val; + lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD; + spin_unlock(&ml403_ac97cr->reg_lock); + mutex_unlock(&ml403_ac97cr->cdc_mutex); + return; + } + spin_unlock(&ml403_ac97cr->reg_lock); + schedule_timeout_uninterruptible(1); + } while (time_after(end_time, jiffies)); +#ifdef CODEC_STAT + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "timeout while codec write " + "(reg=0x%x, val=0x%x / %d, last STATUS=0x%x, %d) " + "(cw=%d, cr=%d)\n", + reg, val, val, stat, rafaccess, ml403_ac97cr->ac97_write, + ml403_ac97cr->ac97_read); +#else + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "timeout while codec write (reg=0x%x, val=0x%x / %d)\n", + reg, val, val); +#endif +#else /* CODEC_WRITE_CHECK_RAF */ +#if CODEC_WAIT_AFTER_WRITE > 0 + /* officially, in AC97 spec there is no possibility for a AC97 + * controller to determine, if write access is done or not - so: How + * is Xilinx able to provide a RAF bit for write access? + * => very strange, thus just don't check RAF bit (compare with + * Xilinx's example app in EDK 8.1i) and wait + */ + schedule_timeout_uninterruptible(HZ / CODEC_WAIT_AFTER_WRITE); +#endif + PDEBUG(CODEC_SUCCESS, "codec_write(): (done) " + "reg=0x%x, value=%d / 0x%x (no RAF check)\n", + reg, val, val); +#endif + mutex_unlock(&ml403_ac97cr->cdc_mutex); + return; +} + +static int __devinit +snd_ml403_ac97cr_chip_init(struct snd_ml403_ac97cr *ml403_ac97cr) +{ + unsigned long end_time; + PDEBUG(INIT_INFO, "chip_init():\n"); + end_time = jiffies + HZ / CODEC_TIMEOUT_ON_INIT; + do { + if (in_be32(CR_REG(ml403_ac97cr, STATUS)) & CR_CODECREADY) { + /* clear both hardware FIFOs */ + out_be32(CR_REG(ml403_ac97cr, RESETFIFO), + CR_RECRESET | CR_PLAYRESET); + PDEBUG(INIT_INFO, "chip_init(): (done)\n"); + return 0; + } + schedule_timeout_uninterruptible(1); + } while (time_after(end_time, jiffies)); + snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " + "timeout while waiting for codec, " + "not ready!\n"); + return -EBUSY; +} + +static int snd_ml403_ac97cr_free(struct snd_ml403_ac97cr *ml403_ac97cr) +{ + PDEBUG(INIT_INFO, "free():\n"); + /* irq release */ + if (ml403_ac97cr->irq >= 0) + free_irq(ml403_ac97cr->irq, ml403_ac97cr); + if (ml403_ac97cr->capture_irq >= 0) + free_irq(ml403_ac97cr->capture_irq, ml403_ac97cr); + /* give back "port" */ + if (ml403_ac97cr->port != NULL) + iounmap(ml403_ac97cr->port); + kfree(ml403_ac97cr); + PDEBUG(INIT_INFO, "free(): (done)\n"); + return 0; +} + +static int snd_ml403_ac97cr_dev_free(struct snd_device *snddev) +{ + struct snd_ml403_ac97cr *ml403_ac97cr = snddev->device_data; + PDEBUG(INIT_INFO, "dev_free():\n"); + return snd_ml403_ac97cr_free(ml403_ac97cr); +} + +static int __devinit +snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, + struct snd_ml403_ac97cr **rml403_ac97cr) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_ml403_ac97cr_dev_free, + }; + struct resource *resource; + int irq; + + *rml403_ac97cr = NULL; + ml403_ac97cr = kzalloc(sizeof(*ml403_ac97cr), GFP_KERNEL); + if (ml403_ac97cr == NULL) + return -ENOMEM; + spin_lock_init(&ml403_ac97cr->reg_lock); + mutex_init(&ml403_ac97cr->cdc_mutex); + ml403_ac97cr->card = card; + ml403_ac97cr->pfdev = pfdev; + ml403_ac97cr->irq = -1; + ml403_ac97cr->enable_irq = 0; + ml403_ac97cr->capture_irq = -1; + ml403_ac97cr->enable_capture_irq = 0; + ml403_ac97cr->port = NULL; + ml403_ac97cr->res_port = NULL; + + PDEBUG(INIT_INFO, "Trying to reserve resources now ...\n"); + resource = platform_get_resource(pfdev, IORESOURCE_MEM, 0); + /* get "port" */ + ml403_ac97cr->port = ioremap_nocache(resource->start, + (resource->end) - + (resource->start) + 1); + if (ml403_ac97cr->port == NULL) { + snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " + "unable to remap memory region (%x to %x)\n", + resource->start, resource->end); + snd_ml403_ac97cr_free(ml403_ac97cr); + return -EBUSY; + } + snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": " + "remap controller memory region to " + "0x%x done\n", (unsigned int)ml403_ac97cr->port); + /* get irq */ + irq = platform_get_irq(pfdev, 0); + if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " + "unable to grab IRQ %d\n", + irq); + snd_ml403_ac97cr_free(ml403_ac97cr); + return -EBUSY; + } + ml403_ac97cr->irq = irq; + snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": " + "request (playback) irq %d done\n", + ml403_ac97cr->irq); + irq = platform_get_irq(pfdev, 1); + if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " + "unable to grab IRQ %d\n", + irq); + snd_ml403_ac97cr_free(ml403_ac97cr); + return -EBUSY; + } + ml403_ac97cr->capture_irq = irq; + snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": " + "request (capture) irq %d done\n", + ml403_ac97cr->capture_irq); + + err = snd_ml403_ac97cr_chip_init(ml403_ac97cr); + if (err < 0) { + snd_ml403_ac97cr_free(ml403_ac97cr); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ml403_ac97cr, &ops); + if (err < 0) { + PDEBUG(INIT_FAILURE, "probe(): snd_device_new() failed!\n"); + snd_ml403_ac97cr_free(ml403_ac97cr); + return err; + } + + snd_card_set_dev(card, &pfdev->dev); + + *rml403_ac97cr = ml403_ac97cr; + return 0; +} + +static void snd_ml403_ac97cr_mixer_free(struct snd_ac97 *ac97) +{ + struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data; + PDEBUG(INIT_INFO, "mixer_free():\n"); + ml403_ac97cr->ac97 = NULL; + PDEBUG(INIT_INFO, "mixer_free(): (done)\n"); +} + +static int __devinit +snd_ml403_ac97cr_mixer(struct snd_ml403_ac97cr *ml403_ac97cr) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops ops = { + .write = snd_ml403_ac97cr_codec_write, + .read = snd_ml403_ac97cr_codec_read, + }; + PDEBUG(INIT_INFO, "mixer():\n"); + err = snd_ac97_bus(ml403_ac97cr->card, 0, &ops, NULL, &bus); + if (err < 0) + return err; + + memset(&ac97, 0, sizeof(ac97)); + ml403_ac97cr->ac97_fake = 1; + lm4550_regfile_init(); +#ifdef CODEC_STAT + ml403_ac97cr->ac97_read = 0; + ml403_ac97cr->ac97_write = 0; +#endif + ac97.private_data = ml403_ac97cr; + ac97.private_free = snd_ml403_ac97cr_mixer_free; + ac97.scaps = AC97_SCAP_AUDIO | AC97_SCAP_SKIP_MODEM | + AC97_SCAP_NO_SPDIF; + err = snd_ac97_mixer(bus, &ac97, &ml403_ac97cr->ac97); + ml403_ac97cr->ac97_fake = 0; + lm4550_regfile_write_values_after_init(ml403_ac97cr->ac97); + PDEBUG(INIT_INFO, "mixer(): (done) snd_ac97_mixer()=%d\n", err); + return err; +} + +static int __devinit +snd_ml403_ac97cr_pcm(struct snd_ml403_ac97cr *ml403_ac97cr, int device, + struct snd_pcm **rpcm) +{ + struct snd_pcm *pcm; + int err; + + if (rpcm) + *rpcm = NULL; + err = snd_pcm_new(ml403_ac97cr->card, "ML403AC97CR/1", device, 1, 1, + &pcm); + if (err < 0) + return err; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_ml403_ac97cr_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_ml403_ac97cr_capture_ops); + pcm->private_data = ml403_ac97cr; + pcm->info_flags = 0; + strcpy(pcm->name, "ML403AC97CR DAC/ADC"); + ml403_ac97cr->pcm = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 64 * 1024, + 128 * 1024); + if (rpcm) + *rpcm = pcm; + return 0; +} + +static int __devinit snd_ml403_ac97cr_probe(struct platform_device *pfdev) +{ + struct snd_card *card; + struct snd_ml403_ac97cr *ml403_ac97cr = NULL; + int err; + int dev = pfdev->id; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) + return -ENOENT; + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + err = snd_ml403_ac97cr_create(card, pfdev, &ml403_ac97cr); + if (err < 0) { + PDEBUG(INIT_FAILURE, "probe(): create failed!\n"); + snd_card_free(card); + return err; + } + PDEBUG(INIT_INFO, "probe(): create done\n"); + card->private_data = ml403_ac97cr; + err = snd_ml403_ac97cr_mixer(ml403_ac97cr); + if (err < 0) { + snd_card_free(card); + return err; + } + PDEBUG(INIT_INFO, "probe(): mixer done\n"); + err = snd_ml403_ac97cr_pcm(ml403_ac97cr, 0, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + PDEBUG(INIT_INFO, "probe(): PCM done\n"); + strcpy(card->driver, SND_ML403_AC97CR_DRIVER); + strcpy(card->shortname, "ML403 AC97 Controller Reference"); + sprintf(card->longname, "%s %s at 0x%lx, irq %i & %i, device %i", + card->shortname, card->driver, + (unsigned long)ml403_ac97cr->port, ml403_ac97cr->irq, + ml403_ac97cr->capture_irq, dev + 1); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pfdev, card); + PDEBUG(INIT_INFO, "probe(): (done)\n"); + return 0; +} + +static int snd_ml403_ac97cr_remove(struct platform_device *pfdev) +{ + snd_card_free(platform_get_drvdata(pfdev)); + platform_set_drvdata(pfdev, NULL); + return 0; +} + +static struct platform_driver snd_ml403_ac97cr_driver = { + .probe = snd_ml403_ac97cr_probe, + .remove = snd_ml403_ac97cr_remove, + .driver = { + .name = SND_ML403_AC97CR_DRIVER, + }, +}; + +static int __init alsa_card_ml403_ac97cr_init(void) +{ + return platform_driver_register(&snd_ml403_ac97cr_driver); +} + +static void __exit alsa_card_ml403_ac97cr_exit(void) +{ + platform_driver_unregister(&snd_ml403_ac97cr_driver); +} + +module_init(alsa_card_ml403_ac97cr_init) +module_exit(alsa_card_ml403_ac97cr_exit) diff --git a/sound/drivers/pcm-indirect2.c b/sound/drivers/pcm-indirect2.c new file mode 100644 index 0000000..6a829cd --- /dev/null +++ b/sound/drivers/pcm-indirect2.c @@ -0,0 +1,591 @@ +/* + * Helper functions for indirect PCM data transfer to a simple FIFO in + * hardware (small, no possibility to read "hardware io position", + * updating position done by interrupt, ...) + * + * Copyright (c) by 2007 Joachim Foerster + * + * Based on "pcm-indirect.h" (alsa-driver-1.0.13) by + * + * Copyright (c) by Takashi Iwai + * Jaroslav Kysela + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +/* #dependency of sound/core.h# */ +#include +/* snd_printk/d() */ +#include +/* struct snd_pcm_substream, struct snd_pcm_runtime, snd_pcm_uframes_t + * snd_pcm_period_elapsed() */ +#include + +#include "pcm-indirect2.h" + +#ifdef SND_PCM_INDIRECT2_STAT +/* jiffies */ +#include + +void snd_pcm_indirect2_stat(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int i; + int j; + int k; + int seconds = (rec->lastbytetime - rec->firstbytetime) / HZ; + + snd_printk(KERN_DEBUG "STAT: mul_elapsed: %u, mul_elapsed_real: %d, " + "irq_occured: %d\n", + rec->mul_elapsed, rec->mul_elapsed_real, rec->irq_occured); + snd_printk(KERN_DEBUG "STAT: min_multiple: %d (irqs/period)\n", + rec->min_multiple); + snd_printk(KERN_DEBUG "STAT: firstbytetime: %lu, lastbytetime: %lu, " + "firstzerotime: %lu\n", + rec->firstbytetime, rec->lastbytetime, rec->firstzerotime); + snd_printk(KERN_DEBUG "STAT: bytes2hw: %u Bytes => (by runtime->rate) " + "length: %d s\n", + rec->bytes2hw, rec->bytes2hw / 2 / 2 / runtime->rate); + snd_printk(KERN_DEBUG "STAT: (by measurement) length: %d => " + "rate: %d Bytes/s = %d Frames/s|Hz\n", + seconds, rec->bytes2hw / seconds, + rec->bytes2hw / 2 / 2 / seconds); + snd_printk(KERN_DEBUG + "STAT: zeros2hw: %u = %d ms ~ %d * %d zero copies\n", + rec->zeros2hw, ((rec->zeros2hw / 2 / 2) * 1000) / + runtime->rate, + rec->zeros2hw / (rec->hw_buffer_size / 2), + (rec->hw_buffer_size / 2)); + snd_printk(KERN_DEBUG "STAT: pointer_calls: %u, lastdifftime: %u\n", + rec->pointer_calls, rec->lastdifftime); + snd_printk(KERN_DEBUG "STAT: sw_io: %d, sw_data: %d\n", rec->sw_io, + rec->sw_data); + snd_printk(KERN_DEBUG "STAT: byte_sizes[]:\n"); + k = 0; + for (j = 0; j < 8; j++) { + for (i = j * 8; i < (j + 1) * 8; i++) + if (rec->byte_sizes[i] != 0) { + snd_printk(KERN_DEBUG "%u: %u", + i, rec->byte_sizes[i]); + k++; + } + if (((k % 8) == 0) && (k != 0)) { + snd_printk(KERN_DEBUG "\n"); + k = 0; + } + } + snd_printk(KERN_DEBUG "\n"); + snd_printk(KERN_DEBUG "STAT: zero_sizes[]:\n"); + for (j = 0; j < 8; j++) { + k = 0; + for (i = j * 8; i < (j + 1) * 8; i++) + if (rec->zero_sizes[i] != 0) + snd_printk(KERN_DEBUG "%u: %u", + i, rec->zero_sizes[i]); + else + k++; + if (!k) + snd_printk(KERN_DEBUG "\n"); + } + snd_printk(KERN_DEBUG "\n"); + snd_printk(KERN_DEBUG "STAT: min_adds[]:\n"); + for (j = 0; j < 8; j++) { + if (rec->min_adds[j] != 0) + snd_printk(KERN_DEBUG "%u: %u", j, rec->min_adds[j]); + } + snd_printk(KERN_DEBUG "\n"); + snd_printk(KERN_DEBUG "STAT: mul_adds[]:\n"); + for (j = 0; j < 8; j++) { + if (rec->mul_adds[j] != 0) + snd_printk(KERN_DEBUG "%u: %u", j, rec->mul_adds[j]); + } + snd_printk(KERN_DEBUG "\n"); + snd_printk(KERN_DEBUG + "STAT: zero_times_saved: %d, zero_times_notsaved: %d\n", + rec->zero_times_saved, rec->zero_times_notsaved); + /* snd_printk(KERN_DEBUG "STAT: zero_times[]\n"); + i = 0; + for (j = 0; j < 3750; j++) { + if (rec->zero_times[j] != 0) { + snd_printk(KERN_DEBUG "%u: %u", j, rec->zero_times[j]); + i++; + } + if (((i % 8) == 0) && (i != 0)) + snd_printk(KERN_DEBUG "\n"); + } + snd_printk(KERN_DEBUG "\n"); */ + return; +} +#endif + +/* + * _internal_ helper function for playback/capture transfer function + */ +static void +snd_pcm_indirect2_increase_min_periods(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + int isplay, int iscopy, + unsigned int bytes) +{ + if (rec->min_periods >= 0) { + if (iscopy) { + rec->sw_io += bytes; + if (rec->sw_io >= rec->sw_buffer_size) + rec->sw_io -= rec->sw_buffer_size; + } else if (isplay) { + /* If application does not write data in multiples of + * a period, move sw_data to the next correctly aligned + * position, so that sw_io can converge to it (in the + * next step). + */ + if (!rec->check_alignment) { + if (rec->bytes2hw % + snd_pcm_lib_period_bytes(substream)) { + unsigned bytes2hw_aligned = + (1 + + (rec->bytes2hw / + snd_pcm_lib_period_bytes + (substream))) * + snd_pcm_lib_period_bytes + (substream); + rec->sw_data = + bytes2hw_aligned % + rec->sw_buffer_size; +#ifdef SND_PCM_INDIRECT2_STAT + snd_printk(KERN_DEBUG + "STAT: @re-align: aligned " + "bytes2hw to next period " + "size boundary: %d " + "(instead of %d)\n", + bytes2hw_aligned, + rec->bytes2hw); + snd_printk(KERN_DEBUG + "STAT: @re-align: sw_data " + "moves to: %d\n", + rec->sw_data); +#endif + } + rec->check_alignment = 1; + } + /* We are at the end and are copying zeros into the + * fifo. + * Now, we have to make sure that sw_io is increased + * until the position of sw_data: Filling the fifo with + * the first zeros means, the last bytes were played. + */ + if (rec->sw_io != rec->sw_data) { + unsigned int diff; + if (rec->sw_data > rec->sw_io) + diff = rec->sw_data - rec->sw_io; + else + diff = (rec->sw_buffer_size - + rec->sw_io) + + rec->sw_data; + if (bytes >= diff) + rec->sw_io = rec->sw_data; + else { + rec->sw_io += bytes; + if (rec->sw_io >= rec->sw_buffer_size) + rec->sw_io -= + rec->sw_buffer_size; + } + } + } + rec->min_period_count += bytes; + if (rec->min_period_count >= (rec->hw_buffer_size / 2)) { + rec->min_periods += (rec->min_period_count / + (rec->hw_buffer_size / 2)); +#ifdef SND_PCM_INDIRECT2_STAT + if ((rec->min_period_count / + (rec->hw_buffer_size / 2)) > 7) + snd_printk(KERN_DEBUG + "STAT: more than 7 (%d) min_adds " + "at once - too big to save!\n", + (rec->min_period_count / + (rec->hw_buffer_size / 2))); + else + rec->min_adds[(rec->min_period_count / + (rec->hw_buffer_size / 2))]++; +#endif + rec->min_period_count = (rec->min_period_count % + (rec->hw_buffer_size / 2)); + } + } else if (isplay && iscopy) + rec->min_periods = 0; +} + +/* + * helper function for playback/capture pointer callback + */ +snd_pcm_uframes_t +snd_pcm_indirect2_pointer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec) +{ +#ifdef SND_PCM_INDIRECT2_STAT + rec->pointer_calls++; +#endif + return bytes_to_frames(substream->runtime, rec->sw_io); +} + +/* + * _internal_ helper function for playback interrupt callback + */ +static void +snd_pcm_indirect2_playback_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t zero) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr; + + /* runtime->control->appl_ptr: position where ALSA will write next time + * rec->appl_ptr: position where ALSA was last time + * diff: obviously ALSA wrote that much bytes into the intermediate + * buffer since we checked last time + */ + snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr; + + if (diff) { +#ifdef SND_PCM_INDIRECT2_STAT + rec->lastdifftime = jiffies; +#endif + if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2)) + diff += runtime->boundary; + /* number of bytes "added" by ALSA increases the number of + * bytes which are ready to "be transfered to HW"/"played" + * Then, set rec->appl_ptr to not count bytes twice next time. + */ + rec->sw_ready += (int)frames_to_bytes(runtime, diff); + rec->appl_ptr = appl_ptr; + } + if (rec->hw_ready && (rec->sw_ready <= 0)) { + unsigned int bytes; + +#ifdef SND_PCM_INDIRECT2_STAT + if (rec->firstzerotime == 0) { + rec->firstzerotime = jiffies; + snd_printk(KERN_DEBUG + "STAT: @firstzerotime: mul_elapsed: %d, " + "min_period_count: %d\n", + rec->mul_elapsed, rec->min_period_count); + snd_printk(KERN_DEBUG + "STAT: @firstzerotime: sw_io: %d, " + "sw_data: %d, appl_ptr: %u\n", + rec->sw_io, rec->sw_data, + (unsigned int)appl_ptr); + } + if ((jiffies - rec->firstzerotime) < 3750) { + rec->zero_times[(jiffies - rec->firstzerotime)]++; + rec->zero_times_saved++; + } else + rec->zero_times_notsaved++; +#endif + bytes = zero(substream, rec); + +#ifdef SND_PCM_INDIRECT2_STAT + rec->zeros2hw += bytes; + if (bytes < 64) + rec->zero_sizes[bytes]++; + else + snd_printk(KERN_DEBUG + "STAT: %d zero Bytes copied to hardware at " + "once - too big to save!\n", + bytes); +#endif + snd_pcm_indirect2_increase_min_periods(substream, rec, 1, 0, + bytes); + return; + } + while (rec->hw_ready && (rec->sw_ready > 0)) { + /* sw_to_end: max. number of bytes that can be read/take from + * the current position (sw_data) in _one_ step + */ + unsigned int sw_to_end = rec->sw_buffer_size - rec->sw_data; + + /* bytes: number of bytes we have available (for reading) */ + unsigned int bytes = rec->sw_ready; + + if (sw_to_end < bytes) + bytes = sw_to_end; + if (!bytes) + break; + +#ifdef SND_PCM_INDIRECT2_STAT + if (rec->firstbytetime == 0) + rec->firstbytetime = jiffies; + rec->lastbytetime = jiffies; +#endif + /* copy bytes from intermediate buffer position sw_data to the + * HW and return number of bytes actually written + * Furthermore, set hw_ready to 0, if the fifo isn't empty + * now => more could be transfered to fifo + */ + bytes = copy(substream, rec, bytes); + rec->bytes2hw += bytes; + +#ifdef SND_PCM_INDIRECT2_STAT + if (bytes < 64) + rec->byte_sizes[bytes]++; + else + snd_printk(KERN_DEBUG + "STAT: %d Bytes copied to hardware at once " + "- too big to save!\n", + bytes); +#endif + /* increase sw_data by the number of actually written bytes + * (= number of taken bytes from intermediate buffer) + */ + rec->sw_data += bytes; + if (rec->sw_data == rec->sw_buffer_size) + rec->sw_data = 0; + /* now sw_data is the position where ALSA is going to write + * in the intermediate buffer next time = position we are going + * to read from next time + */ + + snd_pcm_indirect2_increase_min_periods(substream, rec, 1, 1, + bytes); + + /* we read bytes from intermediate buffer, so we need to say + * that the number of bytes ready for transfer are decreased + * now + */ + rec->sw_ready -= bytes; + } + return; +} + +/* + * helper function for playback interrupt routine + */ +void +snd_pcm_indirect2_playback_interrupt(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t zero) +{ +#ifdef SND_PCM_INDIRECT2_STAT + rec->irq_occured++; +#endif + /* hardware played some bytes, so there is room again (in fifo) */ + rec->hw_ready = 1; + + /* don't call ack() now, instead call transfer() function directly + * (normally called by ack() ) + */ + snd_pcm_indirect2_playback_transfer(substream, rec, copy, zero); + + if (rec->min_periods >= rec->min_multiple) { +#ifdef SND_PCM_INDIRECT2_STAT + if ((rec->min_periods / rec->min_multiple) > 7) + snd_printk(KERN_DEBUG + "STAT: more than 7 (%d) mul_adds - too big " + "to save!\n", + (rec->min_periods / rec->min_multiple)); + else + rec->mul_adds[(rec->min_periods / + rec->min_multiple)]++; + rec->mul_elapsed_real += (rec->min_periods / + rec->min_multiple); + rec->mul_elapsed++; +#endif + rec->min_periods = 0; + snd_pcm_period_elapsed(substream); + } +} + +/* + * _internal_ helper function for capture interrupt callback + */ +static void +snd_pcm_indirect2_capture_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t null) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr; + snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr; + + if (diff) { +#ifdef SND_PCM_INDIRECT2_STAT + rec->lastdifftime = jiffies; +#endif + if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2)) + diff += runtime->boundary; + rec->sw_ready -= frames_to_bytes(runtime, diff); + rec->appl_ptr = appl_ptr; + } + /* if hardware has something, but the intermediate buffer is full + * => skip contents of buffer + */ + if (rec->hw_ready && (rec->sw_ready >= (int)rec->sw_buffer_size)) { + unsigned int bytes; + +#ifdef SND_PCM_INDIRECT2_STAT + if (rec->firstzerotime == 0) { + rec->firstzerotime = jiffies; + snd_printk(KERN_DEBUG "STAT: (capture) " + "@firstzerotime: mul_elapsed: %d, " + "min_period_count: %d\n", + rec->mul_elapsed, rec->min_period_count); + snd_printk(KERN_DEBUG "STAT: (capture) " + "@firstzerotime: sw_io: %d, sw_data: %d, " + "appl_ptr: %u\n", + rec->sw_io, rec->sw_data, + (unsigned int)appl_ptr); + } + if ((jiffies - rec->firstzerotime) < 3750) { + rec->zero_times[(jiffies - rec->firstzerotime)]++; + rec->zero_times_saved++; + } else + rec->zero_times_notsaved++; +#endif + bytes = null(substream, rec); + +#ifdef SND_PCM_INDIRECT2_STAT + rec->zeros2hw += bytes; + if (bytes < 64) + rec->zero_sizes[bytes]++; + else + snd_printk(KERN_DEBUG + "STAT: (capture) %d zero Bytes copied to " + "hardware at once - too big to save!\n", + bytes); +#endif + snd_pcm_indirect2_increase_min_periods(substream, rec, 0, 0, + bytes); + /* report an overrun */ + rec->sw_io = SNDRV_PCM_POS_XRUN; + return; + } + while (rec->hw_ready && (rec->sw_ready < (int)rec->sw_buffer_size)) { + /* sw_to_end: max. number of bytes that we can write to the + * intermediate buffer (until it's end) + */ + size_t sw_to_end = rec->sw_buffer_size - rec->sw_data; + + /* bytes: max. number of bytes, which may be copied to the + * intermediate buffer without overflow (in _one_ step) + */ + size_t bytes = rec->sw_buffer_size - rec->sw_ready; + + /* limit number of bytes (for transfer) by available room in + * the intermediate buffer + */ + if (sw_to_end < bytes) + bytes = sw_to_end; + if (!bytes) + break; + +#ifdef SND_PCM_INDIRECT2_STAT + if (rec->firstbytetime == 0) + rec->firstbytetime = jiffies; + rec->lastbytetime = jiffies; +#endif + /* copy bytes from the intermediate buffer (position sw_data) + * to the HW at most and return number of bytes actually copied + * from HW + * Furthermore, set hw_ready to 0, if the fifo is empty now. + */ + bytes = copy(substream, rec, bytes); + rec->bytes2hw += bytes; + +#ifdef SND_PCM_INDIRECT2_STAT + if (bytes < 64) + rec->byte_sizes[bytes]++; + else + snd_printk(KERN_DEBUG + "STAT: (capture) %d Bytes copied to " + "hardware at once - too big to save!\n", + bytes); +#endif + /* increase sw_data by the number of actually copied bytes from + * HW + */ + rec->sw_data += bytes; + if (rec->sw_data == rec->sw_buffer_size) + rec->sw_data = 0; + + snd_pcm_indirect2_increase_min_periods(substream, rec, 0, 1, + bytes); + + /* number of bytes in the intermediate buffer, which haven't + * been fetched by ALSA yet. + */ + rec->sw_ready += bytes; + } + return; +} + +/* + * helper function for capture interrupt routine + */ +void +snd_pcm_indirect2_capture_interrupt(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t null) +{ +#ifdef SND_PCM_INDIRECT2_STAT + rec->irq_occured++; +#endif + /* hardware recorded some bytes, so there is something to read from the + * record fifo: + */ + rec->hw_ready = 1; + + /* don't call ack() now, instead call transfer() function directly + * (normally called by ack() ) + */ + snd_pcm_indirect2_capture_transfer(substream, rec, copy, null); + + if (rec->min_periods >= rec->min_multiple) { + +#ifdef SND_PCM_INDIRECT2_STAT + if ((rec->min_periods / rec->min_multiple) > 7) + snd_printk(KERN_DEBUG + "STAT: more than 7 (%d) mul_adds - " + "too big to save!\n", + (rec->min_periods / rec->min_multiple)); + else + rec->mul_adds[(rec->min_periods / + rec->min_multiple)]++; + rec->mul_elapsed_real += (rec->min_periods / + rec->min_multiple); + rec->mul_elapsed++; + + if (!(rec->mul_elapsed % 4)) { + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int appl_ptr = + frames_to_bytes(runtime, + (unsigned int)runtime->control-> + appl_ptr) % rec->sw_buffer_size; + int diff = rec->sw_data - appl_ptr; + if (diff < 0) + diff += rec->sw_buffer_size; + snd_printk(KERN_DEBUG + "STAT: mul_elapsed: %d, sw_data: %u, " + "appl_ptr (bytes): %u, diff: %d\n", + rec->mul_elapsed, rec->sw_data, appl_ptr, + diff); + } +#endif + rec->min_periods = 0; + snd_pcm_period_elapsed(substream); + } +} diff --git a/sound/drivers/pcm-indirect2.h b/sound/drivers/pcm-indirect2.h new file mode 100644 index 0000000..2ea6e46 --- /dev/null +++ b/sound/drivers/pcm-indirect2.h @@ -0,0 +1,140 @@ +/* + * Helper functions for indirect PCM data transfer to a simple FIFO in + * hardware (small, no possibility to read "hardware io position", + * updating position done by interrupt, ...) + * + * Copyright (c) by 2007 Joachim Foerster + * + * Based on "pcm-indirect.h" (alsa-driver-1.0.13) by + * + * Copyright (c) by Takashi Iwai + * Jaroslav Kysela + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef __SOUND_PCM_INDIRECT2_H +#define __SOUND_PCM_INDIRECT2_H + +/* struct snd_pcm_substream, struct snd_pcm_runtime, snd_pcm_uframes_t */ +#include + +/* Debug options for code which may be removed completely in a final version */ +#ifdef CONFIG_SND_DEBUG +#define SND_PCM_INDIRECT2_STAT /* turn on some "statistics" about the + * process of copying bytes from the + * intermediate buffer to the hardware + * fifo and the other way round + */ +#endif + +struct snd_pcm_indirect2 { + unsigned int hw_buffer_size; /* Byte size of hardware buffer */ + int hw_ready; /* playback: 1 = hw fifo has room left, + * 0 = hw fifo is full + */ + unsigned int min_multiple; + int min_periods; /* counts number of min. periods until + * min_multiple is reached + */ + int min_period_count; /* counts bytes to count number of + * min. periods + */ + + unsigned int sw_buffer_size; /* Byte size of software buffer */ + + /* sw_data: position in intermediate buffer, where we will read (or + * write) from/to next time (to transfer data to/from HW) + */ + unsigned int sw_data; /* Offset to next dst (or src) in sw + * ring buffer + */ + /* easiest case (playback): + * sw_data is nearly the same as ~ runtime->control->appl_ptr, with the + * exception that sw_data is "behind" by the number if bytes ALSA wrote + * to the intermediate buffer last time. + * A call to ack() callback synchronizes both indirectly. + */ + + /* We have no real sw_io pointer here. Usually sw_io is pointing to the + * current playback/capture position _inside_ the hardware. Devices + * with plain FIFOs often have no possibility to publish this position. + * So we say: if sw_data is updated, that means bytes were copied to + * the hardware, we increase sw_io by that amount, because there have + * to be as much bytes which were played. So sw_io will stay behind + * sw_data all the time and has to converge to sw_data at the end of + * playback. + */ + unsigned int sw_io; /* Current software pointer in bytes */ + + /* sw_ready: number of bytes ALSA copied to the intermediate buffer, so + * it represents the number of bytes which wait for transfer to the HW + */ + int sw_ready; /* Bytes ready to be transferred to/from hw */ + + /* appl_ptr: last known position of ALSA (where ALSA is going to write + * next time into the intermediate buffer + */ + snd_pcm_uframes_t appl_ptr; /* Last seen appl_ptr */ + + unsigned int bytes2hw; + int check_alignment; + +#ifdef SND_PCM_INDIRECT2_STAT + unsigned int zeros2hw; + unsigned int mul_elapsed; + unsigned int mul_elapsed_real; + unsigned long firstbytetime; + unsigned long lastbytetime; + unsigned long firstzerotime; + unsigned int byte_sizes[64]; + unsigned int zero_sizes[64]; + unsigned int min_adds[8]; + unsigned int mul_adds[8]; + unsigned int zero_times[3750]; /* = 15s */ + unsigned int zero_times_saved; + unsigned int zero_times_notsaved; + unsigned int irq_occured; + unsigned int pointer_calls; + unsigned int lastdifftime; +#endif +}; + +typedef size_t (*snd_pcm_indirect2_copy_t) (struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + size_t bytes); +typedef size_t (*snd_pcm_indirect2_zero_t) (struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec); + +#ifdef SND_PCM_INDIRECT2_STAT +void snd_pcm_indirect2_stat(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec); +#endif + +snd_pcm_uframes_t +snd_pcm_indirect2_pointer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec); +void +snd_pcm_indirect2_playback_interrupt(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t zero); +void +snd_pcm_indirect2_capture_interrupt(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t null); + +#endif /* __SOUND_PCM_INDIRECT2_H */ -- cgit v0.10.2 From dddefd0d706da7d981e8e397231df257f0122a49 Mon Sep 17 00:00:00 2001 From: Joachim Foerster Date: Mon, 5 Nov 2007 15:48:36 +0100 Subject: [ALSA] [ML403-AC97CR] Fix capture/periodic overrun bug We have to do fairly accurate counting of the minimal periods, instead of being lazy and just setting the number to zero as soon as one period elapses. Signed-off-by: Joachim Foerster Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 2222315..c76a24e 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -28,11 +28,9 @@ * accesses to a minimum, because after a variable amount of accesses, the AC97 * controller doesn't raise the register access finished bit anymore ... * - * - Capture support works - basically, but after ~30s (with rates > ~20kHz) - * ALSA stops reading captured samples from the intermediate buffer and - * therefore a overrun happens - ATM I don't know what's wrong. - * * - Playback support seems to be pretty stable - no issues here. + * - Capture support "works" now, too. Overruns don't happen any longer so often. + * But there might still be some ... */ #include diff --git a/sound/drivers/pcm-indirect2.c b/sound/drivers/pcm-indirect2.c index 6a829cd..660157d 100644 --- a/sound/drivers/pcm-indirect2.c +++ b/sound/drivers/pcm-indirect2.c @@ -403,7 +403,7 @@ snd_pcm_indirect2_playback_interrupt(struct snd_pcm_substream *substream, rec->min_multiple); rec->mul_elapsed++; #endif - rec->min_periods = 0; + rec->min_periods = (rec->min_periods % rec->min_multiple); snd_pcm_period_elapsed(substream); } } @@ -568,24 +568,8 @@ snd_pcm_indirect2_capture_interrupt(struct snd_pcm_substream *substream, rec->mul_elapsed_real += (rec->min_periods / rec->min_multiple); rec->mul_elapsed++; - - if (!(rec->mul_elapsed % 4)) { - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int appl_ptr = - frames_to_bytes(runtime, - (unsigned int)runtime->control-> - appl_ptr) % rec->sw_buffer_size; - int diff = rec->sw_data - appl_ptr; - if (diff < 0) - diff += rec->sw_buffer_size; - snd_printk(KERN_DEBUG - "STAT: mul_elapsed: %d, sw_data: %u, " - "appl_ptr (bytes): %u, diff: %d\n", - rec->mul_elapsed, rec->sw_data, appl_ptr, - diff); - } #endif - rec->min_periods = 0; + rec->min_periods = (rec->min_periods % rec->min_multiple); snd_pcm_period_elapsed(substream); } } -- cgit v0.10.2 From 2797f724cdde5b3c630f6422a1cc3a21772728dd Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Mon, 5 Nov 2007 18:21:56 +0100 Subject: [ALSA] HDA-Intel - Add support for RV610/RV630 HDMI audio The Audio interface on HD2400/HD2600 cards isn't currently detected by snd-hda-intel. I added missing pci device ids for RV610 and RV630, but I only had a HD2400 pro card to test, where now the audio interface is detected (and no need to change patch_atihdmi.c, as the codec vendor id remains 0x1002aa01 for which we already have an entry there). I also couldn't test if sound pass-trough is ok (and I don't know how to), but at least now the device is detected. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3fa0f97..a829c59 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -104,6 +104,8 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{ATI, RS690}," "{ATI, RS780}," "{ATI, R600}," + "{ATI, RV630}," + "{ATI, RV610}," "{VIA, VT8251}," "{VIA, VT8237A}," "{SiS, SIS966}," @@ -1942,6 +1944,8 @@ static struct pci_device_id azx_ids[] = { { 0x1002, 0x7919, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS690 HDMI */ { 0x1002, 0x960c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS780 HDMI */ { 0x1002, 0xaa00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI R600 HDMI */ + { 0x1002, 0xaa08, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV630 HDMI */ + { 0x1002, 0xaa10, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV610 HDMI */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ -- cgit v0.10.2 From e035b841015251062316cb60b47d1f11d2703f6d Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 6 Nov 2007 11:53:55 +0100 Subject: [ALSA] hda: Added new IDT codec family Added initial support for the STAC92HD71BXX family of codecs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4dc09ef..f6e00c4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -62,6 +62,11 @@ enum { }; enum { + STAC_92HD71BXX_REF, + STAC_92HD71BXX_MODELS +}; + +enum { STAC_925x_REF, STAC_M2_2, STAC_MA6, @@ -171,6 +176,23 @@ static hda_nid_t stac9200_dac_nids[1] = { 0x02, }; +static hda_nid_t stac92hd71bxx_adc_nids[2] = { + 0x12, 0x13, +}; + +static hda_nid_t stac92hd71bxx_mux_nids[2] = { + 0x1a, 0x1b +}; + +static hda_nid_t stac92hd71bxx_dac_nids[2] = { + 0x10, /*0x11, */ +}; + +#define STAC92HD71BXX_NUM_DMICS 2 +static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { + 0x18, 0x19, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -237,6 +259,11 @@ static hda_nid_t stac922x_pin_nids[10] = { 0x0f, 0x10, 0x11, 0x15, 0x1b, }; +static hda_nid_t stac92hd71bxx_pin_nids[10] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x14, 0x18, 0x19, 0x1e, +}; + static hda_nid_t stac927x_pin_nids[14] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, @@ -359,6 +386,20 @@ static struct hda_verb stac9200_eapd_init[] = { {} }; +static struct hda_verb stac92hd71bxx_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* connect headphone jack to dac1 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* unmute mono out node */ + { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} +}; + static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -434,6 +475,21 @@ static struct snd_kcontrol_new stac9200_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), + STAC_INPUT_SOURCE(2), + STAC_VOLKNOB(0x28), + + /* hardware gain controls */ + HDA_CODEC_VOLUME("Digital Mic 1 Volume", 0x18, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Digital Mic 2 Volume", 0x19, 0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Capture Volume", 0x1c, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1c, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Mux Volume", 0x1a, 0, HDA_OUTPUT), + { } /* end */ +}; + static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), @@ -786,6 +842,27 @@ static struct snd_pci_quirk stac925x_cfg_tbl[] = { {} /* terminator */ }; +static unsigned int ref92hd71bxx_pin_configs[10] = { + 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, + 0x0181302e, 0x01114010, 0x01a19020, 0x90a000f0, + 0x90a000f0, 0x01452050, +}; + +static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, +}; + +static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD71BXX_REF), + {} /* terminator */ +}; + static unsigned int ref922x_pin_configs[10] = { 0x01014010, 0x01016011, 0x01012012, 0x0221401f, 0x01813122, 0x01011014, 0x01441030, 0x01c41030, @@ -2560,6 +2637,77 @@ static int patch_stac925x(struct hda_codec *codec) return 0; } +static int patch_stac92hd71bxx(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + int err = 0; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->pin_nids = stac92hd71bxx_pin_nids; + spec->board_config = snd_hda_check_board_config(codec, + STAC_92HD71BXX_MODELS, + stac92hd71bxx_models, + stac92hd71bxx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + " STAC92HD71BXX, using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else { + spec->pin_configs = stac92hd71bxx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + stac92xx_enable_gpio_mask(codec); + + spec->init = stac92hd71bxx_core_init; + spec->mixer = stac92hd71bxx_mixer; + + spec->mux_nids = stac92hd71bxx_mux_nids; + spec->adc_nids = stac92hd71bxx_adc_nids; + spec->dmic_nids = stac92hd71bxx_dmic_nids; + spec->dmux_nid = 0x1c; + + spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); + spec->num_dmics = STAC92HD71BXX_NUM_DMICS; + + spec->multiout.num_dacs = 2; + spec->multiout.hp_nid = 0x11; + spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + + err = stac92xx_parse_auto_config(codec, 0x21, 0x23); + if (!err) { + if (spec->board_config < 0) { + printk(KERN_WARNING "hda_codec: No auto-config is " + "available, default to model=ref\n"); + spec->board_config = STAC_92HD71BXX_REF; + goto again; + } + err = -EINVAL; + } + + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +}; + static int patch_stac922x(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -3128,5 +3276,6 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d76b0, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, {} /* terminator */ }; -- cgit v0.10.2 From 690eceb58ca900a55ae0cec5e8770ac166a881d8 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Tue, 6 Nov 2007 11:56:17 +0100 Subject: [ALSA] ASoC: sh: improve generated code for HAC module (AC97) Change loops in ac97_read/write functions to count down to zero rather than up. Gcc will then use the 'dt' (decrement-and-test) op instead of an increment/compare op-pair. Signed-off-by: Manuel Lauss Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 8e3f039..34b77b9 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -105,7 +105,7 @@ static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, unsigned int to1, to2, i; unsigned short adr; - for (i = 0; i < AC97_READ_RETRY; ++i) { + for (i = AC97_READ_RETRY; i; i--) { *v = 0; /* wait for HAC to receive something from the codec */ for (to1 = TMO_E4; @@ -132,7 +132,7 @@ static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, udelay(21); } HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); - return (i < AC97_READ_RETRY); + return i; } static unsigned short hac_read_codec_aux(struct hac_priv *hac, @@ -141,7 +141,7 @@ static unsigned short hac_read_codec_aux(struct hac_priv *hac, unsigned short val; unsigned int i, to; - for (i = 0; i < AC97_READ_RETRY; i++) { + for (i = AC97_READ_RETRY; i; i--) { /* send_read_request */ local_irq_disable(); HACREG(HACTSR) &= ~(TSR_CMDAMT); @@ -159,10 +159,7 @@ static unsigned short hac_read_codec_aux(struct hac_priv *hac, break; } - if (i == AC97_READ_RETRY) - return ~0; - - return val; + return i ? val : ~0; } static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, @@ -172,7 +169,7 @@ static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct hac_priv *hac = &hac_cpu_data[unit_id]; unsigned int i, to; /* write_codec_aux */ - for (i = 0; i < AC97_WRITE_RETRY; i++) { + for (i = AC97_WRITE_RETRY; i; i--) { /* send_write_request */ local_irq_disable(); HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); -- cgit v0.10.2 From 9b35947fcd697001332d0bb2adf6fbc419f5dd4f Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 7 Nov 2007 13:03:12 +0100 Subject: [ALSA] hda: STAC92HD71 codec mixer Added analog loopback support and missing ADC capture mixer for the STAC92HD71 codec family. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f6e00c4..d498ce1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -391,6 +391,11 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* connect ports 0d and 0f to audio mixer */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + /* unmute dac0 input in audio mixer */ + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -465,6 +470,17 @@ static struct hda_verb stac9205_core_init[] = { .private_value = verb_read | (verb_write << 16), \ } +#define STAC_VOLKNOB(knob_nid) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Master Playback Volume", \ + .count = 1, \ + .info = stac92xx_volknob_info, \ + .get = stac92xx_volknob_get, \ + .put = stac92xx_volknob_put, \ + .private_value = 127 | (knob_nid << 16), \ + } + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -481,12 +497,19 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_VOLKNOB(0x28), /* hardware gain controls */ - HDA_CODEC_VOLUME("Digital Mic 1 Volume", 0x18, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Digital Mic 2 Volume", 0x19, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x1c, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1c, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x1a, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */ }; -- cgit v0.10.2 From b98f9334e25c463fbfbb1cc468c66845e1c8e46a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2007 14:18:01 +0100 Subject: [ALSA] hda-codec - new PCI SSID for HP machines Added new PCI SSIDs for HP machines with ALC262 codec. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8de41c3..1126068 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8564,10 +8564,13 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL), -- cgit v0.10.2 From feb77712b0e780f04507cdda0529088ff0f3286f Mon Sep 17 00:00:00 2001 From: Timofei Bondarenko Date: Wed, 7 Nov 2007 15:49:57 +0100 Subject: [ALSA] cmipci - utilize ADC48K44K bit Setting the ADC48K44K greatly improves capture quality at 48k sampling rate. With this bit clear ADC does ZOH interpolation of every 22th sample at 48k. At frequencies higher than 48k there ADC performs a little better with ADC48K44K bit set. At 44.1k ADC performs a little better with this bit clear. At frequencies below 44.1k there is no difference. Signed-off-by: Timofei Bondarenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 3a2d942..d1f23eb 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -869,6 +869,13 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, snd_cmipci_write(cm, CM_REG_CHFORMAT, val); //snd_printd("cmipci: chformat = %08x\n", val); + if (!rec->is_dac && cm->chip_version) { + if (runtime->rate > 44100) + snd_cmipci_set_bit(cm, CM_REG_EXT_MISC, CM_ADC48K44K); + else + snd_cmipci_clear_bit(cm, CM_REG_EXT_MISC, CM_ADC48K44K); + } + rec->running = 0; spin_unlock_irq(&cm->reg_lock); -- cgit v0.10.2 From b46be727286a93056db851ababc85c0ac3f2f91c Mon Sep 17 00:00:00 2001 From: Timofei Bondarenko Date: Wed, 7 Nov 2007 15:50:52 +0100 Subject: [ALSA] cmipci - allow capture of raw spdif subframes Enable capturing of raw 32bit IEC958_SUBFRAME. The 24-bits PCM data can be obtained using iec958 plugin. Known problem: captured stream may begin with either left or right subframe. Since the iec958 plugin doesn't decode preamble it may swap the channels sometime. Signed-off-by: Timofei Bondarenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index d1f23eb..187203e 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1407,6 +1407,11 @@ static int snd_cmipci_capture_spdif_prepare(struct snd_pcm_substream *substream) else snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS); } + if (snd_pcm_format_width(substream->runtime->format) > 16) + snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL); + else + snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL); + spin_unlock_irq(&cm->reg_lock); return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_CAPT], substream); @@ -1418,6 +1423,7 @@ static int snd_cmipci_capture_spdif_hw_free(struct snd_pcm_substream *subs) spin_lock_irq(&cm->reg_lock); snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_CAPTURE_SPDF); + snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL); spin_unlock_irq(&cm->reg_lock); return snd_cmipci_hw_free(subs); @@ -1569,7 +1575,8 @@ static struct snd_pcm_hardware snd_cmipci_capture_spdif = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .rate_min = 44100, .rate_max = 48000, -- cgit v0.10.2 From b222fe50f28ea14b7fde202d143c534dedf9bca9 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 7 Nov 2007 15:54:45 +0100 Subject: [ALSA] hda: STAC9228 Subsystem update Added more laptops subsystem id's that have STAC9228 DMIC support. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d498ce1..ee74617 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2889,8 +2889,10 @@ static int patch_stac927x(struct hda_codec *codec) } switch (codec->subsystem_id) { - case 0x1028020A: /* STAC 9228 */ - case 0x10280209: /* STAC 9228 */ + case 0x10280242: /* STAC 9228 */ + case 0x102801f3: + case 0x1028020A: + case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; spec->dmux_nid = 0x1b; -- cgit v0.10.2 From 6bab53377eecc19d3d66bcbf2ec8d2f8d99abc43 Mon Sep 17 00:00:00 2001 From: Nicolas Kaiser Date: Wed, 7 Nov 2007 18:31:43 +0100 Subject: [ALSA] sound/pci: remove line duplications in defines Remove line duplications in defines. Acked-by: Thomas Sailer Signed-off-by: Nicolas Kaiser Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index 75da174..74175fc 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -272,7 +272,6 @@ #define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */ #define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */ #define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ -#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ #define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */ #define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */ #define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */ diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 3224577..93dfedc 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -732,7 +732,6 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)"); #define MINISRC_IN_BUFFER_SIZE ( 0x50 * 2 ) #define MINISRC_OUT_BUFFER_SIZE ( 0x50 * 2 * 2) -#define MINISRC_OUT_BUFFER_SIZE ( 0x50 * 2 * 2) #define MINISRC_TMP_BUFFER_SIZE ( 112 + ( MINISRC_BIQUAD_STAGE * 3 + 4 ) * 2 * 2 ) #define MINISRC_BIQUAD_STAGE 2 #define MINISRC_COEF_LOC 0x175 -- cgit v0.10.2 From 85db3848c59610c47f64f67677e875abfcff1c3f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2007 09:09:58 +0100 Subject: [ALSA] Update SNDRV_HWDEP_IFACE_LAST Updated the forgotten SNDRV_HWDEP_IFACE_LAST to point the really last member. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/asound.h b/include/sound/asound.h index af9d11d..3ad5341 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -95,7 +95,7 @@ enum { SNDRV_HWDEP_IFACE_HDA, /* HD-audio */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_SB_RC + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_HDA }; struct snd_hwdep_info { -- cgit v0.10.2 From 6a7b8cf4766803c2e16cf118a2998c7095c2c7e3 Mon Sep 17 00:00:00 2001 From: Stanislav Brabec Date: Mon, 12 Nov 2007 12:11:10 +0100 Subject: [ALSA] use convenient treble scale on WM8750 On Zaurus SL-C3200 (terrier/spitz) based on WM8750, treble scale is inconveniently reverted (increase level = decrease treble), in opposite to bass scale, which uses convenient scale. Fix ALSA WM8750 mixer treble to use convenient treble scale (increase = increase treble level) From: Stanislav Brabec Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 28684ee..f8797de 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -189,7 +189,7 @@ SOC_ENUM("Bass Boost", wm8750_enum[0]), SOC_ENUM("Bass Filter", wm8750_enum[1]), SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1), -SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 0), +SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 1), SOC_ENUM("Treble Cut-off", wm8750_enum[2]), SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0), -- cgit v0.10.2 From 5dbc94791005608c57674fba04dd6b5e19ba9342 Mon Sep 17 00:00:00 2001 From: Jiri Olsa Date: Mon, 12 Nov 2007 12:15:42 +0100 Subject: [ALSA] sound: remove dead config symbol from sound code remove dead config symbols from sound code Signed-off-by: Jiri Olsa Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h index 6b40ee6..e3005a6 100644 --- a/include/sound/cs46xx.h +++ b/include/sound/cs46xx.h @@ -1708,9 +1708,6 @@ struct snd_cs46xx { struct gameport *gameport; -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO - int current_gpio; -#endif #ifdef CONFIG_SND_CS46XX_NEW_DSP struct mutex spos_mutex; diff --git a/include/sound/driver.h b/include/sound/driver.h index 5ccb6c5..1889929 100644 --- a/include/sound/driver.h +++ b/include/sound/driver.h @@ -38,10 +38,6 @@ #define CONFIG_SND_MAJOR 116 #endif -#ifndef CONFIG_SND_DEBUG -#undef CONFIG_SND_DEBUG_MEMORY -#endif - #ifdef ALSA_BUILD #include "adriver.h" #endif diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 2c7bfc9..8c44fef 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2084,71 +2084,6 @@ static int snd_cs46xx_spdif_stream_put(struct snd_kcontrol *kcontrol, #endif /* CONFIG_SND_CS46XX_NEW_DSP */ -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO -static int snd_cs46xx_egpio_select_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 8; - return 0; -} - -static int snd_cs46xx_egpio_select_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = chip->current_gpio; - - return 0; -} - -static int snd_cs46xx_egpio_select_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int change = (chip->current_gpio != ucontrol->value.integer.value[0]); - chip->current_gpio = ucontrol->value.integer.value[0]; - - return change; -} - - -static int snd_cs46xx_egpio_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value; - - snd_printdd ("put: reg = %04x, gpio %02x\n",reg,chip->current_gpio); - ucontrol->value.integer.value[0] = - (snd_cs46xx_peekBA0(chip, reg) & (1 << chip->current_gpio)) ? 1 : 0; - - return 0; -} - -static int snd_cs46xx_egpio_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value; - int val = snd_cs46xx_peekBA0(chip, reg); - int oldval = val; - snd_printdd ("put: reg = %04x, gpio %02x\n",reg,chip->current_gpio); - - if (ucontrol->value.integer.value[0]) - val |= (1 << chip->current_gpio); - else - val &= ~(1 << chip->current_gpio); - - snd_cs46xx_pokeBA0(chip, reg,val); - snd_printdd ("put: val %08x oldval %08x\n",val,oldval); - - return (oldval != val); -} -#endif /* CONFIG_SND_CS46XX_DEBUG_GPIO */ - static struct snd_kcontrol_new snd_cs46xx_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2241,40 +2176,6 @@ static struct snd_kcontrol_new snd_cs46xx_controls[] __devinitdata = { }, #endif -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO select", - .info = snd_cs46xx_egpio_select_info, - .get = snd_cs46xx_egpio_select_get, - .put = snd_cs46xx_egpio_select_put, - .private_value = 0, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO Input/Output", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIODR, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO CMOS/Open drain", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIOPTR, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO On/Off", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIOSR, -}, -#endif }; #ifdef CONFIG_SND_CS46XX_NEW_DSP -- cgit v0.10.2 From 27caac81f33c85bfc745c75a8bc1f8107077ef56 Mon Sep 17 00:00:00 2001 From: Nicolas Kaiser Date: Mon, 12 Nov 2007 12:25:02 +0100 Subject: [ALSA] sound/pci: remove duplicated defines Remove duplicated defines. (From their use it looks like 'midiDataOutx are written to rather than read from.) Signed-off-by: Nicolas Kaiser Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index c4af57f..f47172f 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -163,9 +163,6 @@ enum MonitorModeSelector { // this is the upper word of the PCI control reg. #define DEV_VEND_ID_OFFSET 0x70 // location of the device and vendor ID register -#define COMMAND_ACK_DELAY 13 // number of RTC ticks to wait for an acknowledgement - // from the card after sending a command. -#define INTERCOMMAND_DELAY 40 #define MAX_COMMAND_RETRIES 5 // maximum number of times the driver will attempt // to send a command before giving up. #define COMMAND_ACK_MASK 0x8000 // the MSB is set in the command acknowledgment from diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index ff26a36..5aa57ae 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -104,8 +104,6 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_statusRegister 0 #define HDSP_timecode 128 #define HDSP_status2Register 192 -#define HDSP_midiDataOut0 352 -#define HDSP_midiDataOut1 356 #define HDSP_midiDataIn0 360 #define HDSP_midiDataIn1 364 #define HDSP_midiStatusOut0 384 -- cgit v0.10.2 From 7f9310c1c98abfd85d070f9122695901531e48a4 Mon Sep 17 00:00:00 2001 From: Jiang Zhe Date: Mon, 12 Nov 2007 12:43:37 +0100 Subject: [ALSA] hda-codec - Update dell-m82 model pin config Updated dell-m82 model pin config table. The old config doesn't work with Dell 1210 and co. Signed-off-by: Jiang Zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ee74617..e02d3ba 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -932,8 +932,8 @@ static unsigned int dell_922x_m81_pin_configs[10] = { 102801D7 (Dell XPS M1210) */ static unsigned int dell_922x_m82_pin_configs[10] = { - 0x0221121f, 0x408103ff, 0x02111212, 0x90100310, - 0x408003f1, 0x02111211, 0x03451340, 0x40c003f2, + 0x02211211, 0x408103ff, 0x02a1123e, 0x90100310, + 0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2, 0x508003f3, 0x405003f4, }; -- cgit v0.10.2 From bcecd9bd96a5ba598e4b671a55f05b32284987da Mon Sep 17 00:00:00 2001 From: Jiang Zhe Date: Mon, 12 Nov 2007 12:57:03 +0100 Subject: [ALSA] hda-codec - Add workaround for multiple HPs Dell laptops have multiple HP jacks that can be used for multi-channel outputs. The current auto pincfg handles the speaker as the primary output and thus cannot handle the multi-channel configuration for such cases. This patch adds a workaround to fix this issue by swapping the HP and speaker during multi-channel setup routines. Signed-off-by: Jiang Zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e02d3ba..9925683 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2135,6 +2135,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out { struct sigmatel_spec *spec = codec->spec; int err; + int hp_speaker_swap = 0; if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -2143,6 +2144,24 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (! spec->autocfg.line_outs) return 0; /* can't find valid pin config */ + /* If we have no real line-out pin and multiple hp-outs, HPs should + * be set up as multi-channel outputs. + */ + if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT && + spec->autocfg.hp_outs > 1) { + /* Copy hp_outs to line_outs, backup line_outs in + * speaker_outs so that the following routines can handle + * HP pins as primary outputs. + */ + memcpy(spec->autocfg.speaker_pins, spec->autocfg.line_out_pins, + sizeof(spec->autocfg.line_out_pins)); + spec->autocfg.speaker_outs = spec->autocfg.line_outs; + memcpy(spec->autocfg.line_out_pins, spec->autocfg.hp_pins, + sizeof(spec->autocfg.hp_pins)); + spec->autocfg.line_outs = spec->autocfg.hp_outs; + hp_speaker_swap = 1; + } + if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) return err; if (spec->multiout.num_dacs == 0) @@ -2154,6 +2173,19 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; + if (hp_speaker_swap == 1) { + /* Restore the hp_outs and line_outs */ + memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, + sizeof(spec->autocfg.line_out_pins)); + spec->autocfg.hp_outs = spec->autocfg.line_outs; + memcpy(spec->autocfg.line_out_pins, spec->autocfg.speaker_pins, + sizeof(spec->autocfg.speaker_pins)); + spec->autocfg.line_outs = spec->autocfg.speaker_outs; + memset(spec->autocfg.speaker_pins, 0, + sizeof(spec->autocfg.speaker_pins)); + spec->autocfg.speaker_outs = 0; + } + err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg); if (err < 0) -- cgit v0.10.2 From 40c1d30871346c7428d3876fc4c6b593b1b875f2 Mon Sep 17 00:00:00 2001 From: Jiang Zhe Date: Mon, 12 Nov 2007 13:05:16 +0100 Subject: [ALSA] hda-codec - Avoid wrong speaker-auto mute via mic jack When a mic jack is set up as the multiple I/O, it may issue the automute function wrongly. This patch fixes the wrong automute detection. Signed-off-by: Jiang Zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9925683..c4447b1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1600,6 +1600,13 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); } + + /* check the auto-mute again: we need to mute/unmute the speaker + * appropriately according to the pin direction + */ + if (spec->hp_detect) + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + return 1; } @@ -2483,13 +2490,20 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, pin_ctl & ~flag); } -static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) +static int get_hp_pin_presence(struct hda_codec *codec, hda_nid_t nid) { if (!nid) return 0; if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00) - & (1 << 31)) - return 1; + & (1 << 31)) { + unsigned int pinctl; + pinctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (pinctl & AC_PINCTL_IN_EN) + return 0; /* mic- or line-input */ + else + return 1; /* HP-output */ + } return 0; } @@ -2501,7 +2515,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) presence = 0; for (i = 0; i < cfg->hp_outs; i++) { - presence = get_pin_presence(codec, cfg->hp_pins[i]); + presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); if (presence) break; } @@ -3179,7 +3193,7 @@ static int stac9872_vaio_init(struct hda_codec *codec) static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res) { - if (get_pin_presence(codec, 0x0a)) { + if (get_hp_pin_presence(codec, 0x0a)) { stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); } else { -- cgit v0.10.2 From 83a28a09c8d02d29c17e903ca713d9b8f4159ad4 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Mon, 12 Nov 2007 14:55:19 +0000 Subject: [ALSA] emu10k1: Add mixer controls parameter checking. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index ccacd7b..4b85a6b 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1089,6 +1089,7 @@ static int snd_emu10k1_send_volume_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; int change = 0, idx, val; @@ -1141,6 +1142,7 @@ static int snd_emu10k1_attn_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; unsigned long flags; @@ -1158,6 +1160,7 @@ static int snd_emu10k1_attn_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; int change = 0, idx, val; @@ -1210,6 +1213,7 @@ static int snd_emu10k1_efx_send_routing_get(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; int idx; @@ -1229,6 +1233,7 @@ static int snd_emu10k1_efx_send_routing_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch]; int change = 0, idx, val; @@ -1280,6 +1285,7 @@ static int snd_emu10k1_efx_send_volume_get(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; int idx; @@ -1297,6 +1303,7 @@ static int snd_emu10k1_efx_send_volume_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch]; int change = 0, idx, val; @@ -1345,6 +1352,7 @@ static int snd_emu10k1_efx_attn_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; unsigned long flags; @@ -1360,6 +1368,7 @@ static int snd_emu10k1_efx_attn_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + /* FIXME: Check limits */ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch]; int change = 0, val; -- cgit v0.10.2 From 66d2a9d659ccc6ecf51d606fea9d1058d357f453 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 14 Nov 2007 12:06:21 +0100 Subject: [ALSA] hda-codec - Add support of HP Thin Client T5735 Added the support of HP Thin Client T5735 [0x103c 0x302f] with ALC262 codec. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1126068..c6c0bde 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -92,6 +92,7 @@ enum { ALC262_HP_BPC, ALC262_HP_BPC_D7000_WL, ALC262_HP_BPC_D7000_WF, + ALC262_HP_TC_T5735, ALC262_BENQ_ED8, ALC262_SONY_ASSAMD, ALC262_BENQ_T31, @@ -7710,6 +7711,81 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { { } /* end */ }; +static struct hda_bind_ctls alc262_hp_t5735_bind_front_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc262_hp_t5735_bind_front_sw = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_hp_t5735_automute(struct hda_codec *codec, int force) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + + if (force || !spec->sense_updated) { + unsigned int present; + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + spec->sense_updated = 1; + } + if (spec->jack_present) + mute = (0x7080 | ((0)<<8)); /* mute internal speaker */ + else /* unmute internal speaker if necessary */ + mute = (0x7000 | ((0)<<8)); + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_AMP_GAIN_MUTE, mute ); +} + +static void alc262_hp_t5735_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_hp_t5735_automute(codec, 1); +} + +static void alc262_hp_t5735_init_hook(struct hda_codec *codec) +{ + alc262_hp_t5735_automute(codec, 1); +} + +static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { + HDA_BIND_VOL("PCM Playback Volume", &alc262_hp_t5735_bind_front_vol), + HDA_BIND_SW("PCM Playback Switch",&alc262_hp_t5735_bind_front_sw), + HDA_CODEC_VOLUME("LineOut Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("LineOut Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_hp_t5735_verbs[] = { + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + /* bind hp and internal speaker mute (with plug check) */ static int alc262_sony_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -8579,6 +8655,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF), + SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735", + ALC262_HP_TC_T5735), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), @@ -8673,6 +8751,18 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_HP_D7000_capture_source, }, + [ALC262_HP_TC_T5735] = { + .mixers = { alc262_hp_t5735_mixer }, + .init_verbs = { alc262_init_verbs, alc262_hp_t5735_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_hp_t5735_unsol_event, + .init_hook = alc262_hp_t5735_init_hook, + }, [ALC262_BENQ_ED8] = { .mixers = { alc262_base_mixer }, .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs }, -- cgit v0.10.2 From 61dc35de78d2b28e3c28fd7a352ef8c4ff63f42d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Nov 2007 12:26:44 +0100 Subject: [ALSA] hda-codec - Add model=hp-tc-t5735 for ALC262 Added the missing model string for the new support of HP TC T5735. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index f34821d..f5e77c7 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -825,6 +825,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. fujitsu Fujitsu Laptop hp-bpc HP xw4400/6400/8400/9400 laptops hp-bpc-d7000 HP BPC D7000 + hp-tc-t5735 HP Thin Client T5735 benq Benq ED8 benq-t31 Benq T31 hippo Hippo (ATI) with jack detection, Sony UX-90s diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c6c0bde..14c8c01 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8627,6 +8627,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_FUJITSU] = "fujitsu", [ALC262_HP_BPC] = "hp-bpc", [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", + [ALC262_HP_TC_T5735] = "hp-tc-t5735", [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", [ALC262_SONY_ASSAMD] = "sony-assamd", -- cgit v0.10.2 From 3a749730afc224ac11f4eff3df58a42494a0f035 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Wed, 14 Nov 2007 14:30:43 +0100 Subject: [ALSA] sound/pci: Drop unnecessary continue Continue is not needed at the bottom of a loop. The semantic patch implementing this change is as follows: @@ @@ for (...;...;...) { ... if (...) { ... - continue; } } Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 4bb9764..312373c 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2146,7 +2146,6 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock, snd_printk(KERN_ERR "Unable to initialize codec #%d\n", i); if (i == 0) goto __err; - continue; } } /* tune up the primary codec */ -- cgit v0.10.2 From 44d0a879951a90a4eb463fb1d3b91942f97f36ca Mon Sep 17 00:00:00 2001 From: Vladimir Barinov Date: Wed, 14 Nov 2007 17:07:17 +0100 Subject: [ALSA] ASoC TLV320AIC3X codec driver This patch adds ALSA SoC support for TI TLV320AIC3X audio codecs. The features that are supported: o Capture/Playback/Bypass. o 16/20/24/32 bit audio. o 8k - 96k sample rates. o codec master only mode o DAPM. Signed-off-by: Vladimir Barinov Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7824880..898a7d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,3 +37,6 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_TLV320AIC3X + tristate + depends on SND_SOC && I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ad78e3..c6e5338 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o snd-soc-cs4270-objs := cs4270.o +snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o @@ -11,3 +12,4 @@ obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c new file mode 100644 index 0000000..c075a28 --- /dev/null +++ b/sound/soc/codecs/tlv320aic3x.c @@ -0,0 +1,1275 @@ +/* + * ALSA SoC TLV320AIC3X codec driver + * + * Author: Vladimir Barinov, + * Copyright: (C) 2007 MontaVista Software, Inc., + * + * Based on sound/soc/codecs/wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC3X is a driver for a low power stereo audio + * codecs aic31, aic32, aic33. + * + * It supports full aic33 codec functionality. + * The compatibility with aic32, aic31 is as follows: + * aic32 | aic31 + * --------------------------------------- + * MONO_LOUT -> N/A | MONO_LOUT -> N/A + * | IN1L -> LINE1L + * | IN1R -> LINE1R + * | IN2L -> LINE2L + * | IN2R -> LINE2R + * | MIC3L/R -> N/A + * truncated internal functionality in + * accordance with documentation + * --------------------------------------- + * + * Hence the machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tlv320aic3x.h" + +#define AUDIO_NAME "aic3x" +#define AIC3X_VERSION "0.1" + +/* codec private data */ +struct aic3x_priv { + unsigned int sysclk; + int master; +}; + +/* + * AIC3X register cache + * We can't read the AIC3X register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { + 0x00, 0x00, 0x00, 0x10, /* 0 */ + 0x04, 0x00, 0x00, 0x00, /* 4 */ + 0x00, 0x00, 0x00, 0x01, /* 8 */ + 0x00, 0x00, 0x00, 0x80, /* 12 */ + 0x80, 0xff, 0xff, 0x78, /* 16 */ + 0x78, 0x78, 0x78, 0x78, /* 20 */ + 0x78, 0x00, 0x00, 0xfe, /* 24 */ + 0x00, 0x00, 0xfe, 0x00, /* 28 */ + 0x18, 0x18, 0x00, 0x00, /* 32 */ + 0x00, 0x00, 0x00, 0x00, /* 36 */ + 0x00, 0x00, 0x00, 0x80, /* 40 */ + 0x80, 0x00, 0x00, 0x00, /* 44 */ + 0x00, 0x00, 0x00, 0x04, /* 48 */ + 0x00, 0x00, 0x00, 0x00, /* 52 */ + 0x00, 0x00, 0x04, 0x00, /* 56 */ + 0x00, 0x00, 0x00, 0x00, /* 60 */ + 0x00, 0x04, 0x00, 0x00, /* 64 */ + 0x00, 0x00, 0x00, 0x00, /* 68 */ + 0x04, 0x00, 0x00, 0x00, /* 72 */ + 0x00, 0x00, 0x00, 0x00, /* 76 */ + 0x00, 0x00, 0x00, 0x00, /* 80 */ + 0x00, 0x00, 0x00, 0x00, /* 84 */ + 0x00, 0x00, 0x00, 0x00, /* 88 */ + 0x00, 0x00, 0x00, 0x00, /* 92 */ + 0x00, 0x00, 0x00, 0x00, /* 96 */ + 0x00, 0x00, 0x02, /* 100 */ +}; + +/* + * read aic3x register cache + */ +static inline unsigned int aic3x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= AIC3X_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write aic3x register cache + */ +static inline void aic3x_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + if (reg >= AIC3X_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the aic3x register space + */ +static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 aic3x register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + aic3x_write_reg_cache(codec, data[0], data[1]); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw_aic3x, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } + +/* + * All input lines are connected when !0xf and disconnected with 0xf bit field, + * so we have to use specific dapm_put call for input mixer + */ +static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0x0f; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0x01; + unsigned short val, val_mask; + int ret; + struct snd_soc_dapm_path *path; + int found = 0; + + val = (ucontrol->value.integer.value[0] & mask); + + mask = 0xf; + if (val) + val = mask; + + if (invert) + val = mask - val; + val_mask = mask << shift; + val = val << shift; + + mutex_lock(&widget->codec->mutex); + + if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { + /* find dapm widget path assoc with kcontrol */ + list_for_each_entry(path, &widget->codec->dapm_paths, list) { + if (path->kcontrol != kcontrol) + continue; + + /* found, now check type */ + found = 1; + if (val) + /* new connection */ + path->connect = invert ? 0 : 1; + else + /* old connection must be powered down */ + path->connect = invert ? 1 : 0; + break; + } + + if (found) + snd_soc_dapm_sync_endpoints(widget->codec); + } + + ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); + + mutex_unlock(&widget->codec->mutex); + return ret; +} + +static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; +static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; +static const char *aic3x_left_hpcom_mux[] = + { "differential of HPLOUT", "constant VCM", "single-ended" }; +static const char *aic3x_right_hpcom_mux[] = + { "differential of HPROUT", "constant VCM", "single-ended", + "differential of HPLCOM", "external feedback" }; +static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; + +#define LDAC_ENUM 0 +#define RDAC_ENUM 1 +#define LHPCOM_ENUM 2 +#define RHPCOM_ENUM 3 +#define LINE1L_ENUM 4 +#define LINE1R_ENUM 5 +#define LINE2L_ENUM 6 +#define LINE2R_ENUM 7 + +static const struct soc_enum aic3x_enum[] = { + SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), + SOC_ENUM_SINGLE(DAC_LINE_MUX, 4, 3, aic3x_right_dac_mux), + SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux), + SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux), + SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), +}; + +static const struct snd_kcontrol_new aic3x_snd_controls[] = { + /* Output */ + SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1), + + SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, + DACR1_2_RLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, + 0x01, 0), + SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, + PGAR_2_RLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), + + SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, + DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), + SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, + PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, + LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1), + + SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, + DACR1_2_HPROUT_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, + 0x01, 0), + SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, + PGAR_2_HPROUT_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, + LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), + + SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, + DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, + 0x01, 0), + SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, + PGAR_2_HPRCOM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, + LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1), + + /* + * Note: enable Automatic input Gain Controller with care. It can + * adjust PGA to max value when ADC is on and will never go back. + */ + SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + + /* Input */ + SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), + SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), +}; + +/* add non dapm controls */ +static int aic3x_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&aic3x_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Left DAC Mux */ +static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); + +/* Right DAC Mux */ +static const struct snd_kcontrol_new aic3x_right_dac_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[RDAC_ENUM]); + +/* Left HPCOM Mux */ +static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LHPCOM_ENUM]); + +/* Right HPCOM Mux */ +static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]); + +/* Left DAC_L1 Mixer */ +static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), +}; + +/* Right DAC_R1 Mixer */ +static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), +}; + +/* Left PGA Mixer */ +static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), +}; + +/* Right PGA Mixer */ +static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), +}; + +/* Left Line1 Mux */ +static const struct snd_kcontrol_new aic3x_left_line1_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_ENUM]); + +/* Right Line1 Mux */ +static const struct snd_kcontrol_new aic3x_right_line1_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_ENUM]); + +/* Left Line2 Mux */ +static const struct snd_kcontrol_new aic3x_left_line2_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE2L_ENUM]); + +/* Right Line2 Mux */ +static const struct snd_kcontrol_new aic3x_right_line2_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]); + +/* Left PGA Bypass Mixer */ +static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), +}; + +/* Right PGA Bypass Mixer */ +static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), +}; + +/* Left Line2 Bypass Mixer */ +static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), +}; + +/* Right Line2 Bypass Mixer */ +static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), +}; + +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + /* Left DAC to Left Outputs */ + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0), + SND_SOC_DAPM_MUX("Left DAC Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_dac_mux_controls), + SND_SOC_DAPM_MIXER("Left DAC_L1 Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_dac_mixer_controls[0], + ARRAY_SIZE(aic3x_left_dac_mixer_controls)), + SND_SOC_DAPM_MUX("Left HPCOM Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_hpcom_mux_controls), + SND_SOC_DAPM_PGA("Left Line Out", LLOPM_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left HP Out", HPLOUT_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left HP Com", HPLCOM_CTRL, 0, 0, NULL, 0), + + /* Right DAC to Right Outputs */ + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", DAC_PWR, 6, 0), + SND_SOC_DAPM_MUX("Right DAC Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_dac_mux_controls), + SND_SOC_DAPM_MIXER("Right DAC_R1 Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_dac_mixer_controls[0], + ARRAY_SIZE(aic3x_right_dac_mixer_controls)), + SND_SOC_DAPM_MUX("Right HPCOM Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_hpcom_mux_controls), + SND_SOC_DAPM_PGA("Right Line Out", RLOPM_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0), + + /* Mono Output */ + SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), + + /* Left Inputs to Left ADC */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_left_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line1_mux_controls), + SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line2_mux_controls), + + /* Right Inputs to Right ADC */ + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", + LINE1R_2_RADC_CTRL, 2, 0), + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_right_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_line1_mux_controls), + SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_line2_mux_controls), + + /* Mic Bias */ + SND_SOC_DAPM_MICBIAS("Mic Bias 2V", MICBIAS_CTRL, 6, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias 2.5V", MICBIAS_CTRL, 7, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0), + + /* Left PGA to Left Output bypass */ + SND_SOC_DAPM_MIXER("Left PGA Bypass Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_pga_bp_mixer_controls[0], + ARRAY_SIZE(aic3x_left_pga_bp_mixer_controls)), + + /* Right PGA to Right Output bypass */ + SND_SOC_DAPM_MIXER("Right PGA Bypass Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_pga_bp_mixer_controls[0], + ARRAY_SIZE(aic3x_right_pga_bp_mixer_controls)), + + /* Left Line2 to Left Output bypass */ + SND_SOC_DAPM_MIXER("Left Line2 Bypass Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_line2_bp_mixer_controls[0], + ARRAY_SIZE(aic3x_left_line2_bp_mixer_controls)), + + /* Right Line2 to Right Output bypass */ + SND_SOC_DAPM_MIXER("Right Line2 Bypass Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_line2_bp_mixer_controls[0], + ARRAY_SIZE(aic3x_right_line2_bp_mixer_controls)), + + SND_SOC_DAPM_OUTPUT("LLOUT"), + SND_SOC_DAPM_OUTPUT("RLOUT"), + SND_SOC_DAPM_OUTPUT("MONO_LOUT"), + SND_SOC_DAPM_OUTPUT("HPLOUT"), + SND_SOC_DAPM_OUTPUT("HPROUT"), + SND_SOC_DAPM_OUTPUT("HPLCOM"), + SND_SOC_DAPM_OUTPUT("HPRCOM"), + + SND_SOC_DAPM_INPUT("MIC3L"), + SND_SOC_DAPM_INPUT("MIC3R"), + SND_SOC_DAPM_INPUT("LINE1L"), + SND_SOC_DAPM_INPUT("LINE1R"), + SND_SOC_DAPM_INPUT("LINE2L"), + SND_SOC_DAPM_INPUT("LINE2R"), +}; + +static const char *intercon[][3] = { + /* Left Output */ + {"Left DAC Mux", "DAC_L1", "Left DAC"}, + {"Left DAC Mux", "DAC_L2", "Left DAC"}, + {"Left DAC Mux", "DAC_L3", "Left DAC"}, + + {"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"}, + {"Left Line Out", NULL, "Left DAC Mux"}, + {"Left HP Out", NULL, "Left DAC Mux"}, + + {"Left HPCOM Mux", "differential of HPLOUT", "Left DAC_L1 Mixer"}, + {"Left HPCOM Mux", "constant VCM", "Left DAC_L1 Mixer"}, + {"Left HPCOM Mux", "single-ended", "Left DAC_L1 Mixer"}, + + {"Left Line Out", NULL, "Left DAC_L1 Mixer"}, + {"Mono Out", NULL, "Left DAC_L1 Mixer"}, + {"Left HP Out", NULL, "Left DAC_L1 Mixer"}, + {"Left HP Com", NULL, "Left HPCOM Mux"}, + + {"LLOUT", NULL, "Left Line Out"}, + {"LLOUT", NULL, "Left Line Out"}, + {"HPLOUT", NULL, "Left HP Out"}, + {"HPLCOM", NULL, "Left HP Com"}, + + /* Right Output */ + {"Right DAC Mux", "DAC_R1", "Right DAC"}, + {"Right DAC Mux", "DAC_R2", "Right DAC"}, + {"Right DAC Mux", "DAC_R3", "Right DAC"}, + + {"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"}, + {"Right Line Out", NULL, "Right DAC Mux"}, + {"Right HP Out", NULL, "Right DAC Mux"}, + + {"Right HPCOM Mux", "differential of HPROUT", "Right DAC_R1 Mixer"}, + {"Right HPCOM Mux", "constant VCM", "Right DAC_R1 Mixer"}, + {"Right HPCOM Mux", "single-ended", "Right DAC_R1 Mixer"}, + {"Right HPCOM Mux", "differential of HPLCOM", "Right DAC_R1 Mixer"}, + {"Right HPCOM Mux", "external feedback", "Right DAC_R1 Mixer"}, + + {"Right Line Out", NULL, "Right DAC_R1 Mixer"}, + {"Mono Out", NULL, "Right DAC_R1 Mixer"}, + {"Right HP Out", NULL, "Right DAC_R1 Mixer"}, + {"Right HP Com", NULL, "Right HPCOM Mux"}, + + {"RLOUT", NULL, "Right Line Out"}, + {"RLOUT", NULL, "Right Line Out"}, + {"HPROUT", NULL, "Right HP Out"}, + {"HPRCOM", NULL, "Right HP Com"}, + + /* Mono Output */ + {"MONOLOUT", NULL, "Mono Out"}, + {"MONOLOUT", NULL, "Mono Out"}, + + /* Left Input */ + {"Left Line1L Mux", "single-ended", "LINE1L"}, + {"Left Line1L Mux", "differential", "LINE1L"}, + + {"Left Line2L Mux", "single-ended", "LINE2L"}, + {"Left Line2L Mux", "differential", "LINE2L"}, + + {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, + {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, + {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + + {"Left ADC", NULL, "Left PGA Mixer"}, + + /* Right Input */ + {"Right Line1R Mux", "single-ended", "LINE1R"}, + {"Right Line1R Mux", "differential", "LINE1R"}, + + {"Right Line2R Mux", "single-ended", "LINE2R"}, + {"Right Line2R Mux", "differential", "LINE2R"}, + + {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, + {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Right ADC", NULL, "Right PGA Mixer"}, + + /* Left PGA Bypass */ + {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"}, + + {"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"}, + {"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"}, + {"Left HPCOM Mux", "single-ended", "Left PGA Bypass Mixer"}, + + {"Left Line Out", NULL, "Left PGA Bypass Mixer"}, + {"Mono Out", NULL, "Left PGA Bypass Mixer"}, + {"Left HP Out", NULL, "Left PGA Bypass Mixer"}, + + /* Right PGA Bypass */ + {"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"}, + + {"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"}, + {"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"}, + {"Right HPCOM Mux", "single-ended", "Right PGA Bypass Mixer"}, + {"Right HPCOM Mux", "differential of HPLCOM", "Right PGA Bypass Mixer"}, + {"Right HPCOM Mux", "external feedback", "Right PGA Bypass Mixer"}, + + {"Right Line Out", NULL, "Right PGA Bypass Mixer"}, + {"Mono Out", NULL, "Right PGA Bypass Mixer"}, + {"Right HP Out", NULL, "Right PGA Bypass Mixer"}, + + /* Left Line2 Bypass */ + {"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"}, + + {"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"}, + {"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"}, + {"Left HPCOM Mux", "single-ended", "Left Line2 Bypass Mixer"}, + + {"Left Line Out", NULL, "Left Line2 Bypass Mixer"}, + {"Mono Out", NULL, "Left Line2 Bypass Mixer"}, + {"Left HP Out", NULL, "Left Line2 Bypass Mixer"}, + + /* Right Line2 Bypass */ + {"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"}, + + {"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"}, + {"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"}, + {"Right HPCOM Mux", "single-ended", "Right Line2 Bypass Mixer"}, + {"Right HPCOM Mux", "differential of HPLCOM", "Right Line2 Bypass Mixer"}, + {"Right HPCOM Mux", "external feedback", "Right Line2 Bypass Mixer"}, + + {"Right Line Out", NULL, "Right Line2 Bypass Mixer"}, + {"Mono Out", NULL, "Right Line2 Bypass Mixer"}, + {"Right HP Out", NULL, "Right Line2 Bypass Mixer"}, + + /* terminator */ + {NULL, NULL, NULL}, +}; + +static int aic3x_add_widgets(struct snd_soc_codec *codec) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + + /* set up audio path interconnects */ + for (i = 0; intercon[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, intercon[i][0], + intercon[i][1], intercon[i][2]); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct aic3x_rate_divs { + u32 mclk; + u32 rate; + u32 fsref_reg; + u8 sr_reg:4; + u8 pllj_reg; + u16 plld_reg; +}; + +/* AIC3X codec mclk clock divider coefficients */ +static const struct aic3x_rate_divs aic3x_divs[] = { + /* 8k */ + {22579200, 8000, 48000, 0xa, 8, 7075}, + {33868800, 8000, 48000, 0xa, 5, 8049}, + /* 11.025k */ + {22579200, 11025, 44100, 0x6, 8, 0}, + {33868800, 11025, 44100, 0x6, 5, 3333}, + /* 16k */ + {22579200, 16000, 48000, 0x4, 8, 7075}, + {33868800, 16000, 48000, 0x4, 5, 8049}, + /* 22.05k */ + {22579200, 22050, 44100, 0x2, 8, 0}, + {33868800, 22050, 44100, 0x2, 5, 3333}, + /* 32k */ + {22579200, 32000, 48000, 0x1, 8, 7075}, + {33868800, 32000, 48000, 0x1, 5, 8049}, + /* 44.1k */ + {22579200, 44100, 44100, 0x0, 8, 0}, + {33868800, 44100, 44100, 0x0, 5, 3333}, + /* 48k */ + {22579200, 48000, 48000, 0x0, 8, 7075}, + {33868800, 48000, 48000, 0x0, 5, 8049}, + /* 64k */ + {22579200, 96000, 96000, 0x1, 8, 7075}, + {33868800, 96000, 96000, 0x1, 5, 8049}, + /* 88.2k */ + {22579200, 88200, 88200, 0x0, 8, 0}, + {33868800, 88200, 88200, 0x0, 5, 3333}, + /* 96k */ + {22579200, 96000, 96000, 0x0, 8, 7075}, + {33868800, 96000, 96000, 0x0, 5, 8049}, +}; + +static inline int aic3x_get_divs(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(aic3x_divs); i++) { + if (aic3x_divs[i].rate == rate && aic3x_divs[i].mclk == mclk) + return i; + } + + return 0; +} + +static int aic3x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct aic3x_priv *aic3x = codec->private_data; + int i; + u8 data, pll_p, pll_r, pll_j; + u16 pll_d; + + i = aic3x_get_divs(aic3x->sysclk, params_rate(params)); + + /* Route Left DAC to left channel input and + * right DAC to right channel input */ + data = (LDAC2LCH | RDAC2RCH); + switch (aic3x_divs[i].fsref_reg) { + case 44100: + data |= FSREF_44100; + break; + case 48000: + data |= FSREF_48000; + break; + case 88200: + data |= FSREF_44100 | DUAL_RATE_MODE; + break; + case 96000: + data |= FSREF_48000 | DUAL_RATE_MODE; + break; + } + aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data); + + /* codec sample rate select */ + data = aic3x_divs[i].sr_reg; + data |= (data << 4); + aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); + + /* Use PLL for generation Fsref by equation: + * Fsref = (MCLK * K * R)/(2048 * P); + * Fix P = 2 and R = 1 and calculate K, if + * K = J.D, i.e. J - an interger portion of K and D is the fractional + * one with 4 digits of precision; + * Example: + * For MCLK = 22.5792 MHz and Fsref = 48kHz: + * Select P = 2, R= 1, K = 8.7074, which results in J = 8, D = 7074 + */ + pll_p = 2; + pll_r = 1; + pll_j = aic3x_divs[i].pllj_reg; + pll_d = aic3x_divs[i].plld_reg; + + data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); + aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); + aic3x_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); + aic3x_write(codec, AIC3X_PLL_PROGC_REG, (pll_d >> 6) << PLLD_MSB_SHIFT); + aic3x_write(codec, AIC3X_PLL_PROGD_REG, + (pll_d & 0x3F) << PLLD_LSB_SHIFT); + + /* select data word length */ + data = + aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (0x01 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (0x02 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (0x03 << 4); + break; + } + aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data); + + return 0; +} + +static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON; + u8 rdac_reg = aic3x_read_reg_cache(codec, RDAC_VOL) & ~MUTE_ON; + + if (mute) { + aic3x_write(codec, LDAC_VOL, ldac_reg | MUTE_ON); + aic3x_write(codec, RDAC_VOL, rdac_reg | MUTE_ON); + } else { + aic3x_write(codec, LDAC_VOL, ldac_reg); + aic3x_write(codec, RDAC_VOL, rdac_reg); + } + + return 0; +} + +static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic3x_priv *aic3x = codec->private_data; + + switch (freq) { + case 22579200: + case 33868800: + aic3x->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic3x_priv *aic3x = codec->private_data; + u8 iface_areg = 0; + u8 iface_breg = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic3x->master = 1; + iface_areg |= BIT_CLK_MASTER | WORD_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic3x->master = 0; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + iface_breg |= (0x01 << 6); + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_breg |= (0x02 << 6); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_breg |= (0x03 << 6); + break; + default: + return -EINVAL; + } + + /* set iface */ + aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); + aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + + return 0; +} + +static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) +{ + struct aic3x_priv *aic3x = codec->private_data; + u8 reg; + + switch (event) { + case SNDRV_CTL_POWER_D0: + /* all power is driven by DAPM system */ + if (aic3x->master) { + /* enable pll */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, + reg | PLL_ENABLE); + } + break; + case SNDRV_CTL_POWER_D1: + case SNDRV_CTL_POWER_D2: + break; + case SNDRV_CTL_POWER_D3hot: + /* + * all power is driven by DAPM system, + * so output power is safe if bypass was set + */ + if (aic3x->master) { + /* disable pll */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, + reg & ~PLL_ENABLE); + } + break; + case SNDRV_CTL_POWER_D3cold: + /* force all power off */ + reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL); + aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON); + reg = aic3x_read_reg_cache(codec, LINE1R_2_RADC_CTRL); + aic3x_write(codec, LINE1R_2_RADC_CTRL, reg & ~RADC_PWR_ON); + + reg = aic3x_read_reg_cache(codec, DAC_PWR); + aic3x_write(codec, DAC_PWR, reg & ~(LDAC_PWR_ON | RDAC_PWR_ON)); + + reg = aic3x_read_reg_cache(codec, HPLOUT_CTRL); + aic3x_write(codec, HPLOUT_CTRL, reg & ~HPLOUT_PWR_ON); + reg = aic3x_read_reg_cache(codec, HPROUT_CTRL); + aic3x_write(codec, HPROUT_CTRL, reg & ~HPROUT_PWR_ON); + + reg = aic3x_read_reg_cache(codec, HPLCOM_CTRL); + aic3x_write(codec, HPLCOM_CTRL, reg & ~HPLCOM_PWR_ON); + reg = aic3x_read_reg_cache(codec, HPRCOM_CTRL); + aic3x_write(codec, HPRCOM_CTRL, reg & ~HPRCOM_PWR_ON); + + reg = aic3x_read_reg_cache(codec, MONOLOPM_CTRL); + aic3x_write(codec, MONOLOPM_CTRL, reg & ~MONOLOPM_PWR_ON); + + reg = aic3x_read_reg_cache(codec, LLOPM_CTRL); + aic3x_write(codec, LLOPM_CTRL, reg & ~LLOPM_PWR_ON); + reg = aic3x_read_reg_cache(codec, RLOPM_CTRL); + aic3x_write(codec, RLOPM_CTRL, reg & ~RLOPM_PWR_ON); + + if (aic3x->master) { + /* disable pll */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, + reg & ~PLL_ENABLE); + } + break; + } + codec->dapm_state = event; + + return 0; +} + +#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_codec_dai aic3x_dai = { + .name = "aic3x", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC3X_RATES, + .formats = AIC3X_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC3X_RATES, + .formats = AIC3X_FORMATS,}, + .ops = { + .hw_params = aic3x_hw_params, + }, + .dai_ops = { + .digital_mute = aic3x_mute, + .set_sysclk = aic3x_set_dai_sysclk, + .set_fmt = aic3x_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(aic3x_dai); + +static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + return 0; +} + +static int aic3x_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++) { + data[0] = i; + data[1] = cache[i]; + codec->hw_write(codec->control_data, data, 2); + } + + aic3x_dapm_event(codec, codec->suspend_dapm_state); + + return 0; +} + +/* + * initialise the AIC3X driver + * register the mixer and dsp interfaces with the kernel + */ +static int aic3x_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int reg, ret = 0; + + codec->name = "aic3x"; + codec->owner = THIS_MODULE; + codec->read = aic3x_read_reg_cache; + codec->write = aic3x_write; + codec->dapm_event = aic3x_dapm_event; + codec->dai = &aic3x_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(aic3x_reg); + codec->reg_cache = kmemdup(aic3x_reg, sizeof(aic3x_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); + aic3x_write(codec, AIC3X_RESET, SOFT_RESET); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "aic3x: failed to create pcms\n"); + goto pcm_err; + } + + /* DAC default volume and mute */ + aic3x_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); + aic3x_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); + + /* DAC to HP default volume and route to Output mixer */ + aic3x_write(codec, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON); + aic3x_write(codec, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON); + aic3x_write(codec, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON); + aic3x_write(codec, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON); + /* DAC to Line Out default volume and route to Output mixer */ + aic3x_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + aic3x_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + /* DAC to Mono Line Out default volume and route to Output mixer */ + aic3x_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + aic3x_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + + /* unmute all outputs */ + reg = aic3x_read_reg_cache(codec, LLOPM_CTRL); + aic3x_write(codec, LLOPM_CTRL, reg | UNMUTE); + reg = aic3x_read_reg_cache(codec, RLOPM_CTRL); + aic3x_write(codec, RLOPM_CTRL, reg | UNMUTE); + reg = aic3x_read_reg_cache(codec, MONOLOPM_CTRL); + aic3x_write(codec, MONOLOPM_CTRL, reg | UNMUTE); + reg = aic3x_read_reg_cache(codec, HPLOUT_CTRL); + aic3x_write(codec, HPLOUT_CTRL, reg | UNMUTE); + reg = aic3x_read_reg_cache(codec, HPROUT_CTRL); + aic3x_write(codec, HPROUT_CTRL, reg | UNMUTE); + reg = aic3x_read_reg_cache(codec, HPLCOM_CTRL); + aic3x_write(codec, HPLCOM_CTRL, reg | UNMUTE); + reg = aic3x_read_reg_cache(codec, HPRCOM_CTRL); + aic3x_write(codec, HPRCOM_CTRL, reg | UNMUTE); + + /* ADC default volume and unmute */ + aic3x_write(codec, LADC_VOL, DEFAULT_GAIN); + aic3x_write(codec, RADC_VOL, DEFAULT_GAIN); + /* By default route Line1 to ADC PGA mixer */ + aic3x_write(codec, LINE1L_2_LADC_CTRL, 0x0); + aic3x_write(codec, LINE1R_2_RADC_CTRL, 0x0); + + /* PGA to HP Bypass default volume, disconnect from Output Mixer */ + aic3x_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL); + aic3x_write(codec, PGAR_2_HPROUT_VOL, DEFAULT_VOL); + aic3x_write(codec, PGAL_2_HPLCOM_VOL, DEFAULT_VOL); + aic3x_write(codec, PGAR_2_HPRCOM_VOL, DEFAULT_VOL); + /* PGA to Line Out default volume, disconnect from Output Mixer */ + aic3x_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); + aic3x_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); + /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ + aic3x_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + aic3x_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + + /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ + aic3x_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + aic3x_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + aic3x_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + aic3x_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + /* Line2 Line Out default volume, disconnect from Output Mixer */ + aic3x_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + aic3x_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + /* Line2 to Mono Out default volume, disconnect from Output Mixer */ + aic3x_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); + + /* off, with power on */ + aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + aic3x_add_controls(codec); + aic3x_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "aic3x: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *aic3x_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * AIC3X 2 wire address can be up to 4 devices with device addresses + * 0x18, 0x19, 0x1A, 0x1B + */ +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver aic3x_i2c_driver; +static struct i2c_client client_template; + +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = aic3x_socdev; + struct aic3x_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + printk(KERN_ERR "aic3x: failed to attach codec at addr %x\n", + addr); + goto err; + } + + ret = aic3x_init(socdev); + if (ret < 0) { + printk(KERN_ERR "aic3x: failed to initialise AIC3X\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int aic3x_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int aic3x_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, aic3x_codec_probe); +} + +/* machine i2c codec control layer */ +static struct i2c_driver aic3x_i2c_driver = { + .driver = { + .name = "aic3x I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_I2CDEV, + .attach_adapter = aic3x_i2c_attach, + .detach_client = aic3x_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "AIC3X", + .driver = &aic3x_i2c_driver, +}; +#endif + +static int aic3x_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct aic3x_setup_data *setup; + struct snd_soc_codec *codec; + struct aic3x_priv *aic3x; + int ret = 0; + + printk(KERN_INFO "AIC3X Audio Codec %s\n", AIC3X_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); + if (aic3x == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = aic3x; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + aic3x_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t) i2c_master_send; + ret = i2c_add_driver(&aic3x_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +static int aic3x_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + /* power down chip */ + if (codec->control_data) + aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&aic3x_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_aic3x = { + .probe = aic3x_probe, + .remove = aic3x_remove, + .suspend = aic3x_suspend, + .resume = aic3x_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x); + +MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver"); +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h new file mode 100644 index 0000000..d0cdeeb --- /dev/null +++ b/sound/soc/codecs/tlv320aic3x.h @@ -0,0 +1,181 @@ +/* + * ALSA SoC TLV320AIC3X codec driver + * + * Author: Vladimir Barinov, + * Copyright: (C) 2007 MontaVista Software, Inc., + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AIC3X_H +#define _AIC3X_H + +/* AIC3X register space */ +#define AIC3X_CACHEREGNUM 103 + +/* Page select register */ +#define AIC3X_PAGE_SELECT 0 +/* Software reset register */ +#define AIC3X_RESET 1 +/* Codec Sample rate select register */ +#define AIC3X_SAMPLE_RATE_SEL_REG 2 +/* PLL progrramming register A */ +#define AIC3X_PLL_PROGA_REG 3 +/* PLL progrramming register B */ +#define AIC3X_PLL_PROGB_REG 4 +/* PLL progrramming register C */ +#define AIC3X_PLL_PROGC_REG 5 +/* PLL progrramming register D */ +#define AIC3X_PLL_PROGD_REG 6 +/* Codec datapath setup register */ +#define AIC3X_CODEC_DATAPATH_REG 7 +/* Audio serial data interface control register A */ +#define AIC3X_ASD_INTF_CTRLA 8 +/* Audio serial data interface control register B */ +#define AIC3X_ASD_INTF_CTRLB 9 +/* Audio overflow status and PLL R value programming register */ +#define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 + +/* ADC PGA Gain control registers */ +#define LADC_VOL 15 +#define RADC_VOL 16 +/* MIC3 control registers */ +#define MIC3LR_2_LADC_CTRL 17 +#define MIC3LR_2_RADC_CTRL 18 +/* Line1 Input control registers */ +#define LINE1L_2_LADC_CTRL 19 +#define LINE1R_2_RADC_CTRL 22 +/* Line2 Input control registers */ +#define LINE2L_2_LADC_CTRL 20 +#define LINE2R_2_RADC_CTRL 23 +/* MICBIAS Control Register */ +#define MICBIAS_CTRL 25 + +/* AGC Control Registers A, B, C */ +#define LAGC_CTRL_A 26 +#define LAGC_CTRL_B 27 +#define LAGC_CTRL_C 28 +#define RAGC_CTRL_A 29 +#define RAGC_CTRL_B 30 +#define RAGC_CTRL_C 31 + +/* DAC Power and Left High Power Output control registers */ +#define DAC_PWR 37 +#define HPLCOM_CFG 37 +/* Right High Power Output control registers */ +#define HPRCOM_CFG 38 +/* DAC Output Switching control registers */ +#define DAC_LINE_MUX 41 +/* High Power Output Driver Pop Reduction registers */ +#define HPOUT_POP_REDUCTION 42 +/* DAC Digital control registers */ +#define LDAC_VOL 43 +#define RDAC_VOL 44 +/* High Power Output control registers */ +#define LINE2L_2_HPLOUT_VOL 45 +#define LINE2R_2_HPROUT_VOL 62 +#define PGAL_2_HPLOUT_VOL 46 +#define PGAR_2_HPROUT_VOL 63 +#define DACL1_2_HPLOUT_VOL 47 +#define DACR1_2_HPROUT_VOL 64 +#define HPLOUT_CTRL 51 +#define HPROUT_CTRL 65 +/* High Power COM control registers */ +#define LINE2L_2_HPLCOM_VOL 52 +#define LINE2R_2_HPRCOM_VOL 69 +#define PGAL_2_HPLCOM_VOL 53 +#define PGAR_2_HPRCOM_VOL 70 +#define DACL1_2_HPLCOM_VOL 54 +#define DACR1_2_HPRCOM_VOL 71 +#define HPLCOM_CTRL 58 +#define HPRCOM_CTRL 72 +/* Mono Line Output Plus/Minus control registers */ +#define LINE2L_2_MONOLOPM_VOL 73 +#define LINE2R_2_MONOLOPM_VOL 76 +#define PGAL_2_MONOLOPM_VOL 74 +#define PGAR_2_MONOLOPM_VOL 77 +#define DACL1_2_MONOLOPM_VOL 75 +#define DACR1_2_MONOLOPM_VOL 78 +#define MONOLOPM_CTRL 79 +/* Line Output Plus/Minus control registers */ +#define LINE2L_2_LLOPM_VOL 80 +#define LINE2R_2_RLOPM_VOL 90 +#define PGAL_2_LLOPM_VOL 81 +#define PGAR_2_RLOPM_VOL 91 +#define DACL1_2_LLOPM_VOL 82 +#define DACR1_2_RLOPM_VOL 92 +#define LLOPM_CTRL 86 +#define RLOPM_CTRL 93 +/* Clock generation control register */ +#define AIC3X_CLKGEN_CTRL_REG 102 + +/* Page select register bits */ +#define PAGE0_SELECT 0 +#define PAGE1_SELECT 1 + +/* Audio serial data interface control register A bits */ +#define BIT_CLK_MASTER 0x80 +#define WORD_CLK_MASTER 0x40 + +/* Codec Datapath setup register 7 */ +#define FSREF_44100 (1 << 7) +#define FSREF_48000 (0 << 7) +#define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) +#define LDAC2LCH (0x1 << 3) +#define RDAC2RCH (0x1 << 1) + +/* PLL registers bitfields */ +#define PLLP_SHIFT 0 +#define PLLR_SHIFT 0 +#define PLLJ_SHIFT 2 +#define PLLD_MSB_SHIFT 0 +#define PLLD_LSB_SHIFT 2 + +/* Clock generation register bits */ +#define PLL_CLKIN_SHIFT 4 +#define MCLK_SOURCE 0x0 +#define PLL_CLKDIV_SHIFT 0 + +/* Software reset register bits */ +#define SOFT_RESET 0x80 + +/* PLL progrramming register A bits */ +#define PLL_ENABLE 0x80 + +/* Route bits */ +#define ROUTE_ON 0x80 + +/* Mute bits */ +#define UNMUTE 0x08 +#define MUTE_ON 0x80 + +/* Power bits */ +#define LADC_PWR_ON 0x04 +#define RADC_PWR_ON 0x04 +#define LDAC_PWR_ON 0x80 +#define RDAC_PWR_ON 0x40 +#define HPLOUT_PWR_ON 0x01 +#define HPROUT_PWR_ON 0x01 +#define HPLCOM_PWR_ON 0x01 +#define HPRCOM_PWR_ON 0x01 +#define MONOLOPM_PWR_ON 0x01 +#define LLOPM_PWR_ON 0x01 +#define RLOPM_PWR_ON 0x01 + +#define INVERT_VOL(val) (0x7f - val) + +/* Default output volume (inverted) */ +#define DEFAULT_VOL INVERT_VOL(0x50) +/* Default input volume */ +#define DEFAULT_GAIN 0x20 + +struct aic3x_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai aic3x_dai; +extern struct snd_soc_codec_device soc_codec_dev_aic3x; + +#endif /* _AIC3X_H */ -- cgit v0.10.2 From 7e39e2273a9b8182ed1b21af5444e29843fc06ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 13:16:02 +0100 Subject: [ALSA] emu10k1 - Check value ranges in ctl callbacks Check value ranges in ctl callbacks properly. This fixes the unexpected crash due to wrong value assignment. Also, remove invalid comments in the last patch. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 4b85a6b..ccacd7b 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1089,7 +1089,6 @@ static int snd_emu10k1_send_volume_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; int change = 0, idx, val; @@ -1142,7 +1141,6 @@ static int snd_emu10k1_attn_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; unsigned long flags; @@ -1160,7 +1158,6 @@ static int snd_emu10k1_attn_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; int change = 0, idx, val; @@ -1213,7 +1210,6 @@ static int snd_emu10k1_efx_send_routing_get(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; int idx; @@ -1233,7 +1229,6 @@ static int snd_emu10k1_efx_send_routing_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch]; int change = 0, idx, val; @@ -1285,7 +1280,6 @@ static int snd_emu10k1_efx_send_volume_get(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; int idx; @@ -1303,7 +1297,6 @@ static int snd_emu10k1_efx_send_volume_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch]; int change = 0, idx, val; @@ -1352,7 +1345,6 @@ static int snd_emu10k1_efx_attn_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)]; unsigned long flags; @@ -1368,7 +1360,6 @@ static int snd_emu10k1_efx_attn_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - /* FIXME: Check limits */ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch]; int change = 0, val; -- cgit v0.10.2 From 68ea7b2f2d8c1effd662fded04e9a589cb640da6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 15:54:38 +0100 Subject: [ALSA] hda-codec - Check value range in ctl callbacks Check the value ranges in ctl put callbacks properly so that invalid values won't be stored or written to registers. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 39240e0..23d3bef 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2337,7 +2337,8 @@ int snd_hda_ch_mode_put(struct hda_codec *codec, unsigned int mode; mode = ucontrol->value.enumerated.item[0]; - snd_assert(mode < num_chmodes, return -EINVAL); + if (mode >= num_chmodes) + return -EINVAL; if (*max_channelsp == chmode[mode].channels) return 0; /* change the current channel setting */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cfe064a..b2c5380 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -370,7 +370,7 @@ static int ad198x_eapd_put(struct snd_kcontrol *kcontrol, int invert = (kcontrol->private_value >> 8) & 1; hda_nid_t nid = kcontrol->private_value & 0xff; unsigned int eapd; - eapd = ucontrol->value.integer.value[0]; + eapd = !!ucontrol->value.integer.value[0]; if (invert) eapd = !eapd; if (eapd == spec->cur_eapd) @@ -1021,6 +1021,8 @@ static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; + if (ucontrol->value.enumerated.item[0] > 1) + return -EINVAL; if (spec->spdif_route != ucontrol->value.enumerated.item[0]) { spec->spdif_route = ucontrol->value.enumerated.item[0]; snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0, @@ -1966,6 +1968,8 @@ static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol, int change; val = ucontrol->value.enumerated.item[0]; + if (val > 3) + return -EINVAL; if (!val) { sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index beda297..68f23b8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -373,7 +373,7 @@ static int cxt_eapd_put(struct snd_kcontrol *kcontrol, hda_nid_t nid = kcontrol->private_value & 0xff; unsigned int eapd; - eapd = ucontrol->value.integer.value[0]; + eapd = !!ucontrol->value.integer.value[0]; if (invert) eapd = !eapd; if (eapd == spec->cur_eapd) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c4447b1..0817f42 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -349,12 +349,13 @@ static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; unsigned int dac_mode; + unsigned int val; - if (spec->aloopback == ucontrol->value.integer.value[0]) + val = !!ucontrol->value.integer.value[0]; + if (spec->aloopback == val) return 0; - spec->aloopback = ucontrol->value.integer.value[0]; - + spec->aloopback = val; dac_mode = snd_hda_codec_read(codec, codec->afg, 0, kcontrol->private_value & 0xFFFF, 0x0); @@ -373,6 +374,42 @@ static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, return 1; } +static int stac92xx_volknob_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 127; + return 0; +} + +static int stac92xx_volknob_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = kcontrol->private_value & 0xff; + return 0; +} + +static int stac92xx_volknob_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int oval = kcontrol->private_value & 0xff; + unsigned int val; + + val = ucontrol->value.integer.value[0] & 0xff; + if (val == oval) + return 0; + + kcontrol->private_value &= ~0xff; + kcontrol->private_value |= val; + + snd_hda_codec_write_cache(codec, kcontrol->private_value >> 16, 0, + AC_VERB_SET_VOLUME_KNOB_CONTROL, val | 0x80); + return 1; +} + static struct hda_verb stac9200_core_init[] = { /* set dac0mux for dac converter */ { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -1588,7 +1625,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct sigmatel_spec *spec = codec->spec; hda_nid_t nid = kcontrol->private_value >> 8; int io_idx = kcontrol-> private_value & 0xff; - unsigned short val = ucontrol->value.integer.value[0]; + unsigned short val = !!ucontrol->value.integer.value[0]; spec->io_switch[io_idx] = val; @@ -1628,11 +1665,12 @@ static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; hda_nid_t nid = kcontrol->private_value & 0xff; + unsigned int val = !!ucontrol->value.integer.value[0]; - if (spec->clfe_swap == ucontrol->value.integer.value[0]) + if (spec->clfe_swap == val) return 0; - spec->clfe_swap = ucontrol->value.integer.value[0]; + spec->clfe_swap = val; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, spec->clfe_swap ? 0x4 : 0x0); -- cgit v0.10.2 From 9cd17cd2409ddbe9853575569cfd97561fa86b14 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 15:56:07 +0100 Subject: [ALSA] ice1724 - Check value ranges in ctl callbacks Check the value ranges in ctl put callbacks properly. Also fixed the wrong access type to enum elements in aureon.c. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index ec0699c..f83ec2f 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -205,7 +205,7 @@ static int aureon_universe_inmux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = ice->spec.aureon.pca9554_out; + ucontrol->value.enumerated.item[0] = ice->spec.aureon.pca9554_out; return 0; } @@ -216,10 +216,11 @@ static int aureon_universe_inmux_put(struct snd_kcontrol *kcontrol, unsigned char oval, nval; int change; + nval = ucontrol->value.enumerated.item[0]; + if (nval >= 3) + return -EINVAL; snd_ice1712_save_gpio_status(ice); - oval = ice->spec.aureon.pca9554_out; - nval = ucontrol->value.integer.value[0]; if ((change = (oval != nval))) { aureon_pca9554_write(ice, PCA9554_OUT, nval); ice->spec.aureon.pca9554_out = nval; @@ -757,10 +758,13 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ snd_ice1712_save_gpio_status(ice); for (ch = 0; ch < 2; ch++) { - if (ucontrol->value.integer.value[ch] != ice->spec.aureon.master[ch]) { + unsigned int vol = ucontrol->value.integer.value[ch]; + if (vol > WM_VOL_MAX) + continue; + vol |= ice->spec.aureon.master[ch] & WM_VOL_MUTE; + if (vol != ice->spec.aureon.master[ch]) { int dac; - ice->spec.aureon.master[ch] &= WM_VOL_MUTE; - ice->spec.aureon.master[ch] |= ucontrol->value.integer.value[ch]; + ice->spec.aureon.master[ch] = vol; for (dac = 0; dac < ice->num_total_dacs; dac += 2) wm_set_vol(ice, WM_DAC_ATTEN + dac + ch, ice->spec.aureon.vol[dac + ch], @@ -807,10 +811,13 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * ofs = kcontrol->private_value & 0xff; snd_ice1712_save_gpio_status(ice); for (i = 0; i < voices; i++) { - idx = WM_DAC_ATTEN + ofs + i; - if (ucontrol->value.integer.value[i] != ice->spec.aureon.vol[ofs+i]) { - ice->spec.aureon.vol[ofs+i] &= WM_VOL_MUTE; - ice->spec.aureon.vol[ofs+i] |= ucontrol->value.integer.value[i]; + unsigned int vol = ucontrol->value.integer.value[i]; + if (vol > 0x7f) + continue; + vol |= ice->spec.aureon.vol[ofs+i]; + if (vol != ice->spec.aureon.vol[ofs+i]) { + ice->spec.aureon.vol[ofs+i] = vol; + idx = WM_DAC_ATTEN + ofs + i; wm_set_vol(ice, idx, ice->spec.aureon.vol[ofs+i], ice->spec.aureon.master[i]); change = 1; @@ -940,8 +947,10 @@ static int wm_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val unsigned short ovol, nvol; int change = 0; - snd_ice1712_save_gpio_status(ice); nvol = ucontrol->value.integer.value[0]; + if (nvol > PCM_RES) + return -EINVAL; + snd_ice1712_save_gpio_status(ice); nvol = (nvol ? (nvol + PCM_MIN) : 0) & 0xff; ovol = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff; if (ovol != nvol) { @@ -1031,7 +1040,7 @@ static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val snd_ice1712_save_gpio_status(ice); for (i = 0; i < 2; i++) { idx = WM_ADC_GAIN + i; - nvol = ucontrol->value.integer.value[i]; + nvol = ucontrol->value.integer.value[i] & 0x1f; ovol = wm_get(ice, idx); if ((ovol & 0x1f) != nvol) { wm_put(ice, idx, nvol | (ovol & ~0x1f)); diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index 3ac2505..c81efc2 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -326,10 +326,13 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ snd_ice1712_save_gpio_status(ice); for (ch = 0; ch < 2; ch++) { - if (ucontrol->value.integer.value[ch] != ice->spec.phase28.master[ch]) { + unsigned int vol = ucontrol->value.integer.value[ch]; + if (vol > WM_VOL_MAX) + continue; + vol |= ice->spec.phase28.master[ch] & WM_VOL_MUTE; + if (vol != ice->spec.phase28.master[ch]) { int dac; - ice->spec.phase28.master[ch] &= WM_VOL_MUTE; - ice->spec.phase28.master[ch] |= ucontrol->value.integer.value[ch]; + ice->spec.phase28.master[ch] = vol; for (dac = 0; dac < ice->num_total_dacs; dac += 2) wm_set_vol(ice, WM_DAC_ATTEN + dac + ch, ice->spec.phase28.vol[dac + ch], @@ -462,10 +465,14 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * ofs = kcontrol->private_value & 0xff; snd_ice1712_save_gpio_status(ice); for (i = 0; i < voices; i++) { - idx = WM_DAC_ATTEN + ofs + i; - if (ucontrol->value.integer.value[i] != ice->spec.phase28.vol[ofs+i]) { - ice->spec.phase28.vol[ofs+i] &= WM_VOL_MUTE; - ice->spec.phase28.vol[ofs+i] |= ucontrol->value.integer.value[i]; + unsigned int vol; + vol = ucontrol->value.integer.value[i]; + if (vol > 0x7f) + continue; + vol |= ice->spec.phase28.vol[ofs+i] & WM_VOL_MUTE; + if (vol != ice->spec.phase28.vol[ofs+i]) { + ice->spec.phase28.vol[ofs+i] = vol; + idx = WM_DAC_ATTEN + ofs + i; wm_set_vol(ice, idx, ice->spec.phase28.vol[ofs+i], ice->spec.phase28.master[i]); change = 1; @@ -595,8 +602,10 @@ static int wm_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val unsigned short ovol, nvol; int change = 0; - snd_ice1712_save_gpio_status(ice); nvol = ucontrol->value.integer.value[0]; + if (nvol > PCM_RES) + return -EINVAL; + snd_ice1712_save_gpio_status(ice); nvol = (nvol ? (nvol + PCM_MIN) : 0) & 0xff; ovol = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff; if (ovol != nvol) { diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 4180f97..4b21d5c 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -241,7 +241,7 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el for (i = 0; i < 2; ++i) { reg = STAC946X_MIC_L_VOLUME + i; - nvol = ucontrol->value.integer.value[i]; + nvol = ucontrol->value.integer.value[i] & 0x0f; ovol = 0x0f - stac9460_get(ice, reg); change = ((ovol & 0x0f) != nvol); if (change) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 7fcce0a..41a153d 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -178,7 +178,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; - nvol = ucontrol->value.integer.value[0]; + nvol = ucontrol->value.integer.value[0] & 0x7f; tmp = stac9460_get(ice, idx); ovol = 0x7f - (tmp & 0x7f); change = (ovol != nvol); @@ -189,7 +189,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, } else { id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); idx = id + STAC946X_LF_VOLUME; - nvol = ucontrol->value.integer.value[0]; + nvol = ucontrol->value.integer.value[0] & 0x7f; if (id < 6) tmp = stac9460_get(ice, idx); else @@ -317,7 +317,7 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, if (id == 0) { for (i = 0; i < 2; ++i) { reg = STAC946X_MIC_L_VOLUME + i; - nvol = ucontrol->value.integer.value[i]; + nvol = ucontrol->value.integer.value[i] & 0x0f; ovol = 0x0f - stac9460_get(ice, reg); change = ((ovol & 0x0f) != nvol); if (change) @@ -327,7 +327,7 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, } else { for (i = 0; i < 2; ++i) { reg = STAC946X_MIC_L_VOLUME + i; - nvol = ucontrol->value.integer.value[i]; + nvol = ucontrol->value.integer.value[i] & 0x0f; ovol = 0x0f - stac9460_2_get(ice, reg); change = ((ovol & 0x0f) != nvol); if (change) -- cgit v0.10.2 From ab2dac2bdcf562dd616bd1fadddf5078ae7c3d83 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 15:56:47 +0100 Subject: [ALSA] mixart - Check value range in ctl callbacks Check the value ranges in ctl put callbacks properly. Also fixed some coding style issues around that. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 0e16512..5b3c224 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -376,15 +376,27 @@ static int mixart_analog_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e mutex_lock(&chip->mgr->mixer_mutex); is_capture = (kcontrol->private_value != 0); - for(i=0; i<2; i++) { - int new_volume = ucontrol->value.integer.value[i]; - int* stored_volume = is_capture ? &chip->analog_capture_volume[i] : &chip->analog_playback_volume[i]; - if(*stored_volume != new_volume) { + for (i = 0; i < 2; i++) { + int new_volume = ucontrol->value.integer.value[i]; + int *stored_volume = is_capture ? + &chip->analog_capture_volume[i] : + &chip->analog_playback_volume[i]; + if (is_capture) { + if (new_volume < MIXART_ANALOG_CAPTURE_LEVEL_MIN || + new_volume > MIXART_ANALOG_CAPTURE_LEVEL_MAX) + continue; + } else { + if (new_volume < MIXART_ANALOG_PLAYBACK_LEVEL_MIN || + new_volume > MIXART_ANALOG_PLAYBACK_LEVEL_MAX) + continue; + } + if (*stored_volume != new_volume) { *stored_volume = new_volume; changed = 1; } } - if(changed) mixart_update_analog_audio_level(chip, is_capture); + if (changed) + mixart_update_analog_audio_level(chip, is_capture); mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -421,13 +433,16 @@ static int mixart_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); int i, changed = 0; mutex_lock(&chip->mgr->mixer_mutex); - for(i=0; i<2; i++) { - if(chip->analog_playback_active[i] != ucontrol->value.integer.value[i]) { - chip->analog_playback_active[i] = ucontrol->value.integer.value[i]; + for (i = 0; i < 2; i++) { + if (chip->analog_playback_active[i] != + ucontrol->value.integer.value[i]) { + chip->analog_playback_active[i] = + !!ucontrol->value.integer.value[i]; changed = 1; } } - if(changed) mixart_update_analog_audio_level(chip, 0); /* update playback levels */ + if (changed) /* update playback levels */ + mixart_update_analog_audio_level(chip, 0); mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -843,23 +858,33 @@ static int mixart_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem int* stored_volume; int i; mutex_lock(&chip->mgr->mixer_mutex); - if(is_capture) { - if(is_aes) stored_volume = chip->digital_capture_volume[1]; /* AES capture */ - else stored_volume = chip->digital_capture_volume[0]; /* analog capture */ + if (is_capture) { + if (is_aes) /* AES capture */ + stored_volume = chip->digital_capture_volume[1]; + else /* analog capture */ + stored_volume = chip->digital_capture_volume[0]; } else { snd_assert ( idx < MIXART_PLAYBACK_STREAMS ); - if(is_aes) stored_volume = chip->digital_playback_volume[MIXART_PLAYBACK_STREAMS + idx]; /* AES playback */ - else stored_volume = chip->digital_playback_volume[idx]; /* analog playback */ + if (is_aes) /* AES playback */ + stored_volume = chip->digital_playback_volume[MIXART_PLAYBACK_STREAMS + idx]; + else /* analog playback */ + stored_volume = chip->digital_playback_volume[idx]; } - for(i=0; i<2; i++) { - if(stored_volume[i] != ucontrol->value.integer.value[i]) { - stored_volume[i] = ucontrol->value.integer.value[i]; + for (i = 0; i < 2; i++) { + int vol = ucontrol->value.integer.value[i]; + if (vol < MIXART_DIGITAL_LEVEL_MIN || + vol > MIXART_DIGITAL_LEVEL_MAX) + continue; + if (stored_volume[i] != vol) { + stored_volume[i] = vol; changed = 1; } } - if(changed) { - if(is_capture) mixart_update_capture_stream_level(chip, is_aes); - else mixart_update_playback_stream_level(chip, is_aes, idx); + if (changed) { + if (is_capture) + mixart_update_capture_stream_level(chip, is_aes); + else + mixart_update_playback_stream_level(chip, is_aes, idx); } mutex_unlock(&chip->mgr->mixer_mutex); return changed; @@ -905,14 +930,18 @@ static int mixart_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ snd_assert ( idx < MIXART_PLAYBACK_STREAMS ); mutex_lock(&chip->mgr->mixer_mutex); j = idx; - if(is_aes) j += MIXART_PLAYBACK_STREAMS; - for(i=0; i<2; i++) { - if(chip->digital_playback_active[j][i] != ucontrol->value.integer.value[i]) { - chip->digital_playback_active[j][i] = ucontrol->value.integer.value[i]; + if (is_aes) + j += MIXART_PLAYBACK_STREAMS; + for (i = 0; i < 2; i++) { + if (chip->digital_playback_active[j][i] != + ucontrol->value.integer.value[i]) { + chip->digital_playback_active[j][i] = + !!ucontrol->value.integer.value[i]; changed = 1; } } - if(changed) mixart_update_playback_stream_level(chip, is_aes, idx); + if (changed) + mixart_update_playback_stream_level(chip, is_aes, idx); mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -975,9 +1004,11 @@ static int mixart_monitor_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ int changed = 0; int i; mutex_lock(&chip->mgr->mixer_mutex); - for(i=0; i<2; i++) { - if(chip->monitoring_volume[i] != ucontrol->value.integer.value[i]) { - chip->monitoring_volume[i] = ucontrol->value.integer.value[i]; + for (i = 0; i < 2; i++) { + if (chip->monitoring_volume[i] != + ucontrol->value.integer.value[i]) { + chip->monitoring_volume[i] = + !!ucontrol->value.integer.value[i]; mixart_update_monitoring(chip, i); changed = 1; } @@ -1017,24 +1048,35 @@ static int mixart_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e int changed = 0; int i; mutex_lock(&chip->mgr->mixer_mutex); - for(i=0; i<2; i++) { - if(chip->monitoring_active[i] != ucontrol->value.integer.value[i]) { - chip->monitoring_active[i] = ucontrol->value.integer.value[i]; + for (i = 0; i < 2; i++) { + if (chip->monitoring_active[i] != + ucontrol->value.integer.value[i]) { + chip->monitoring_active[i] = + !!ucontrol->value.integer.value[i]; changed |= (1<monitoring_active[0] || chip->monitoring_active[1]; - if(allocate) { - snd_mixart_add_ref_pipe( chip, MIXART_PCM_ANALOG, 0, 1); /* allocate the playback pipe for monitoring */ - snd_mixart_add_ref_pipe( chip, MIXART_PCM_ANALOG, 1, 1); /* allocate the capture pipe for monitoring */ + int allocate = chip->monitoring_active[0] || + chip->monitoring_active[1]; + if (allocate) { + /* allocate the playback pipe for monitoring */ + snd_mixart_add_ref_pipe(chip, MIXART_PCM_ANALOG, 0, 1); + /* allocate the capture pipe for monitoring */ + snd_mixart_add_ref_pipe(chip, MIXART_PCM_ANALOG, 1, 1); } - if(changed & 0x01) mixart_update_monitoring(chip, 0); - if(changed & 0x02) mixart_update_monitoring(chip, 1); - if(!allocate) { - snd_mixart_kill_ref_pipe( chip->mgr, &chip->pipe_in_ana, 1); /* release the capture pipe for monitoring */ - snd_mixart_kill_ref_pipe( chip->mgr, &chip->pipe_out_ana, 1); /* release the playback pipe for monitoring */ + if (changed & 0x01) + mixart_update_monitoring(chip, 0); + if (changed & 0x02) + mixart_update_monitoring(chip, 1); + if (!allocate) { + /* release the capture pipe for monitoring */ + snd_mixart_kill_ref_pipe(chip->mgr, + &chip->pipe_in_ana, 1); + /* release the playback pipe for monitoring */ + snd_mixart_kill_ref_pipe(chip->mgr, + &chip->pipe_out_ana, 1); } } -- cgit v0.10.2 From 4e98d6a7ce934b19bffb309f2522b22384355fef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 15:58:13 +0100 Subject: [ALSA] pci - check value range in ctl callbacks Check the value ranges in ctl put callbacks properly in the rest of PCI drivers. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 98c8b72..50c637e 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -133,6 +133,14 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned char mode = ucontrol->value.enumerated.item[0]; + if (kcontrol->private_value) { + if (mode >= 2) + return -EINVAL; + } else { + if (mode >= 3) + return -EINVAL; + } + if (mode != ac97->channel_mode) { ac97->channel_mode = mode; if (ac97->build_ops->update_jacks) @@ -2142,8 +2150,7 @@ static int snd_ac97_ad1985_vrefout_put(struct snd_kcontrol *kcontrol, struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - if (ucontrol->value.enumerated.item[0] > 3 - || ucontrol->value.enumerated.item[0] < 0) + if (ucontrol->value.enumerated.item[0] > 3) return -EINVAL; val = ctrl2reg[ucontrol->value.enumerated.item[0]] << AC97_AD198X_VREF_SHIFT; diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index f47172f..6586abf 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -1752,22 +1752,22 @@ static int snd_korg1212_control_phase_put(struct snd_kcontrol *kcontrol, i = kcontrol->private_value; - korg1212->volumePhase[i] = u->value.integer.value[0]; + korg1212->volumePhase[i] = !!u->value.integer.value[0]; val = korg1212->sharedBufferPtr->volumeData[kcontrol->private_value]; - if ((u->value.integer.value[0] > 0) != (val < 0)) { + if ((u->value.integer.value[0] != 0) != (val < 0)) { val = abs(val) * (korg1212->volumePhase[i] > 0 ? -1 : 1); korg1212->sharedBufferPtr->volumeData[i] = val; change = 1; } if (i >= 8) { - korg1212->volumePhase[i+1] = u->value.integer.value[1]; + korg1212->volumePhase[i+1] = !!u->value.integer.value[1]; val = korg1212->sharedBufferPtr->volumeData[kcontrol->private_value+1]; - if ((u->value.integer.value[1] > 0) != (val < 0)) { + if ((u->value.integer.value[1] != 0) != (val < 0)) { val = abs(val) * (korg1212->volumePhase[i+1] > 0 ? -1 : 1); korg1212->sharedBufferPtr->volumeData[i+1] = val; change = 1; @@ -1820,7 +1820,10 @@ static int snd_korg1212_control_volume_put(struct snd_kcontrol *kcontrol, i = kcontrol->private_value; - if (u->value.integer.value[0] != abs(korg1212->sharedBufferPtr->volumeData[i])) { + if (u->value.integer.value[0] >= k1212MinVolume && + u->value.integer.value[0] >= k1212MaxVolume && + u->value.integer.value[0] != + abs(korg1212->sharedBufferPtr->volumeData[i])) { val = korg1212->volumePhase[i] > 0 ? -1 : 1; val *= u->value.integer.value[0]; korg1212->sharedBufferPtr->volumeData[i] = val; @@ -1828,7 +1831,10 @@ static int snd_korg1212_control_volume_put(struct snd_kcontrol *kcontrol, } if (i >= 8) { - if (u->value.integer.value[1] != abs(korg1212->sharedBufferPtr->volumeData[i+1])) { + if (u->value.integer.value[1] >= k1212MinVolume && + u->value.integer.value[1] >= k1212MaxVolume && + u->value.integer.value[1] != + abs(korg1212->sharedBufferPtr->volumeData[i+1])) { val = korg1212->volumePhase[i+1] > 0 ? -1 : 1; val *= u->value.integer.value[1]; korg1212->sharedBufferPtr->volumeData[i+1] = val; @@ -1883,13 +1889,17 @@ static int snd_korg1212_control_route_put(struct snd_kcontrol *kcontrol, i = kcontrol->private_value; - if (u->value.enumerated.item[0] != (unsigned) korg1212->sharedBufferPtr->volumeData[i]) { + if (u->value.enumerated.item[0] < kAudioChannels && + u->value.enumerated.item[0] != + (unsigned) korg1212->sharedBufferPtr->volumeData[i]) { korg1212->sharedBufferPtr->routeData[i] = u->value.enumerated.item[0]; change = 1; } if (i >= 8) { - if (u->value.enumerated.item[1] != (unsigned) korg1212->sharedBufferPtr->volumeData[i+1]) { + if (u->value.enumerated.item[1] < kAudioChannels && + u->value.enumerated.item[1] != + (unsigned) korg1212->sharedBufferPtr->volumeData[i+1]) { korg1212->sharedBufferPtr->routeData[i+1] = u->value.enumerated.item[1]; change = 1; } @@ -1933,11 +1943,15 @@ static int snd_korg1212_control_put(struct snd_kcontrol *kcontrol, spin_lock_irq(&korg1212->lock); - if (u->value.integer.value[0] != korg1212->leftADCInSens) { + if (u->value.integer.value[0] >= k1212MinADCSens && + u->value.integer.value[0] <= k1212MaxADCSens && + u->value.integer.value[0] != korg1212->leftADCInSens) { korg1212->leftADCInSens = u->value.integer.value[0]; change = 1; } - if (u->value.integer.value[1] != korg1212->rightADCInSens) { + if (u->value.integer.value[1] >= k1212MinADCSens && + u->value.integer.value[1] <= k1212MaxADCSens && + u->value.integer.value[1] != korg1212->rightADCInSens) { korg1212->rightADCInSens = u->value.integer.value[1]; change = 1; } diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index 5f8d426..4d86545 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -120,8 +120,18 @@ static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol, is_capture = (kcontrol->private_value != 0); for (i = 0; i < 2; i++) { int new_volume = ucontrol->value.integer.value[i]; - int* stored_volume = is_capture ? &chip->analog_capture_volume[i] : + int *stored_volume = is_capture ? + &chip->analog_capture_volume[i] : &chip->analog_playback_volume[i]; + if (is_capture) { + if (new_volume < PCXHR_ANALOG_CAPTURE_LEVEL_MIN || + new_volume > PCXHR_ANALOG_CAPTURE_LEVEL_MAX) + continue; + } else { + if (new_volume < PCXHR_ANALOG_PLAYBACK_LEVEL_MIN || + new_volume > PCXHR_ANALOG_PLAYBACK_LEVEL_MAX) + continue; + } if (*stored_volume != new_volume) { *stored_volume = new_volume; changed = 1; @@ -165,10 +175,13 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol, int i, changed = 0; mutex_lock(&chip->mgr->mixer_mutex); for(i = 0; i < 2; i++) { - if (chip->analog_playback_active[i] != ucontrol->value.integer.value[i]) { - chip->analog_playback_active[i] = ucontrol->value.integer.value[i]; + if (chip->analog_playback_active[i] != + ucontrol->value.integer.value[i]) { + chip->analog_playback_active[i] = + !!ucontrol->value.integer.value[i]; changed = 1; - pcxhr_update_analog_audio_level(chip, 0, i); /* update playback levels */ + /* update playback levels */ + pcxhr_update_analog_audio_level(chip, 0, i); } } mutex_unlock(&chip->mgr->mixer_mutex); @@ -323,20 +336,24 @@ static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol, int i; mutex_lock(&chip->mgr->mixer_mutex); - if (is_capture) - stored_volume = chip->digital_capture_volume; /* digital capture */ - else - stored_volume = chip->digital_playback_volume[idx]; /* digital playback */ + if (is_capture) /* digital capture */ + stored_volume = chip->digital_capture_volume; + else /* digital playback */ + stored_volume = chip->digital_playback_volume[idx]; for (i = 0; i < 2; i++) { - if (stored_volume[i] != ucontrol->value.integer.value[i]) { - stored_volume[i] = ucontrol->value.integer.value[i]; + int vol = ucontrol->value.integer.value[i]; + if (vol < PCXHR_DIGITAL_LEVEL_MIN || + vol > PCXHR_DIGITAL_LEVEL_MAX) + continue; + if (stored_volume[i] != vol) { + stored_volume[i] = vol; changed = 1; if (is_capture) /* update capture volume */ pcxhr_update_audio_pipe_level(chip, 1, i); } } - if (! is_capture && changed) - pcxhr_update_playback_stream_level(chip, idx); /* update playback volume */ + if (!is_capture && changed) /* update playback volume */ + pcxhr_update_playback_stream_level(chip, idx); mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -378,8 +395,10 @@ static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v mutex_lock(&chip->mgr->mixer_mutex); j = idx; for (i = 0; i < 2; i++) { - if (chip->digital_playback_active[j][i] != ucontrol->value.integer.value[i]) { - chip->digital_playback_active[j][i] = ucontrol->value.integer.value[i]; + if (chip->digital_playback_active[j][i] != + ucontrol->value.integer.value[i]) { + chip->digital_playback_active[j][i] = + !!ucontrol->value.integer.value[i]; changed = 1; } } @@ -423,10 +442,13 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol, mutex_lock(&chip->mgr->mixer_mutex); for (i = 0; i < 2; i++) { - if (chip->monitoring_volume[i] != ucontrol->value.integer.value[i]) { - chip->monitoring_volume[i] = ucontrol->value.integer.value[i]; - if(chip->monitoring_active[i]) /* do only when monitoring is unmuted */ + if (chip->monitoring_volume[i] != + ucontrol->value.integer.value[i]) { + chip->monitoring_volume[i] = + !!ucontrol->value.integer.value[i]; + if(chip->monitoring_active[i]) /* update monitoring volume and mute */ + /* do only when monitoring is unmuted */ pcxhr_update_audio_pipe_level(chip, 0, i); changed = 1; } @@ -470,15 +492,17 @@ static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol, mutex_lock(&chip->mgr->mixer_mutex); for (i = 0; i < 2; i++) { - if (chip->monitoring_active[i] != ucontrol->value.integer.value[i]) { - chip->monitoring_active[i] = ucontrol->value.integer.value[i]; + if (chip->monitoring_active[i] != + ucontrol->value.integer.value[i]) { + chip->monitoring_active[i] = + !!ucontrol->value.integer.value[i]; changed |= (1<value.enumerated.item[0] >= 3) + return -EINVAL; mutex_lock(&chip->mgr->mixer_mutex); if (chip->audio_capture_source != ucontrol->value.enumerated.item[0]) { chip->audio_capture_source = ucontrol->value.enumerated.item[0]; @@ -642,8 +668,11 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol); + unsigned int clock_items = 3 + mgr->capture_chips; int rate, ret = 0; + if (ucontrol->value.enumerated.item[0] >= clock_items) + return -EINVAL; mutex_lock(&mgr->mixer_mutex); if (mgr->use_clock_type != ucontrol->value.enumerated.item[0]) { mutex_lock(&mgr->setup_mutex); diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 0b3c532..aff05bd 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -2195,22 +2195,25 @@ snd_rme96_dac_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); int change = 0; + unsigned int vol, maxvol; - if (!RME96_HAS_ANALOG_OUT(rme96)) { + + if (!RME96_HAS_ANALOG_OUT(rme96)) return -EINVAL; - } + maxvol = RME96_185X_MAX_OUT(rme96); spin_lock_irq(&rme96->lock); - if (u->value.integer.value[0] != rme96->vol[0]) { - rme96->vol[0] = u->value.integer.value[0]; - change = 1; - } - if (u->value.integer.value[1] != rme96->vol[1]) { - rme96->vol[1] = u->value.integer.value[1]; - change = 1; - } - if (change) { - snd_rme96_apply_dac_volume(rme96); + vol = u->value.integer.value[0]; + if (vol != rme96->vol[0] && vol <= maxvol) { + rme96->vol[0] = vol; + change = 1; + } + vol = u->value.integer.value[1]; + if (vol != rme96->vol[1] && vol <= maxvol) { + rme96->vol[1] = vol; + change = 1; } + if (change) + snd_rme96_apply_dac_volume(rme96); spin_unlock_irq(&rme96->lock); return change; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 5aa57ae..7956b24 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2119,7 +2119,7 @@ static int snd_hdsp_put_clock_source_lock(struct snd_kcontrol *kcontrol, struct change = (int)ucontrol->value.integer.value[0] != hdsp->clock_source_locked; if (change) - hdsp->clock_source_locked = ucontrol->value.integer.value[0]; + hdsp->clock_source_locked = !!ucontrol->value.integer.value[0]; return change; } diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 55558be..f4f0427 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -877,6 +877,12 @@ static int vx_input_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] < MIC_LEVEL_MAX) + return -EINVAL; + if (ucontrol->value.integer.value[1] < 0 || + ucontrol->value.integer.value[1] < MIC_LEVEL_MAX) + return -EINVAL; mutex_lock(&_chip->mixer_mutex); if (chip->input_level[0] != ucontrol->value.integer.value[0] || chip->input_level[1] != ucontrol->value.integer.value[1]) { @@ -912,6 +918,9 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > MIC_LEVEL_MAX) + return -EINVAL; mutex_lock(&_chip->mixer_mutex); if (chip->mic_level != ucontrol->value.integer.value[0]) { chip->mic_level = ucontrol->value.integer.value[0]; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 1fe39ed..c0789a5 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1735,6 +1735,10 @@ static int snd_ymfpci_pcm_vol_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[1] != chip->pcm_mixer[subs].right) { chip->pcm_mixer[subs].left = ucontrol->value.integer.value[0]; chip->pcm_mixer[subs].right = ucontrol->value.integer.value[1]; + if (chip->pcm_mixer[subs].left > 0x8000) + chip->pcm_mixer[subs].left = 0x8000; + if (chip->pcm_mixer[subs].right > 0x8000) + chip->pcm_mixer[subs].right = 0x8000; substream = (struct snd_pcm_substream *)kcontrol->private_value; spin_lock_irqsave(&chip->voice_lock, flags); -- cgit v0.10.2 From 9c45ba100501c206658f356ea1db3263947b9a30 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 16:05:26 +0100 Subject: [ALSA] ice1724 - Clean up ctl callbacks in se.c Clean up ctl callbacks of SE-200PCI driver. Also make sure to check the value ranges. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ice1712/se.c b/sound/pci/ice1712/se.c index ebfe245..3c8b518 100644 --- a/sound/pci/ice1712/se.c +++ b/sound/pci/ice1712/se.c @@ -415,134 +415,177 @@ static const struct se200pci_control se200pci_cont[] = { } }; -static int se200pci_cont_info(struct snd_kcontrol *kc, - struct snd_ctl_elem_info *uinfo) +static int se200pci_get_enum_count(int n) { - struct snd_ice1712 *ice; - int n; - int c; const char **member; + int c; - ice = snd_kcontrol_chip(kc); - n = kc->private_value; - - if (se200pci_cont[n].type == VOLUME1 || - se200pci_cont[n].type == VOLUME2) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; /* mute */ - uinfo->value.integer.max = 0xff; /* 0dB */ - - } else if (se200pci_cont[n].type == BOOLEAN) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - - } else if (se200pci_cont[n].type == ENUM) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - member = se200pci_cont[n].member; - if (member == NULL) - return -EINVAL; - for (c = 0; member[c]; c++) - ; - - uinfo->count = 1; - uinfo->value.enumerated.items = c; - if (uinfo->value.enumerated.item >= c) - uinfo->value.enumerated.item = c - 1; - strcpy(uinfo->value.enumerated.name, - member[uinfo->value.enumerated.item]); - } + member = se200pci_cont[n].member; + if (!member) + return 0; + for (c = 0; member[c]; c++) + ; + return c; +} +static int se200pci_cont_volume_info(struct snd_kcontrol *kc, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; /* mute */ + uinfo->value.integer.max = 0xff; /* 0dB */ return 0; } -static int se200pci_cont_get(struct snd_kcontrol *kc, - struct snd_ctl_elem_value *uc) +#define se200pci_cont_boolean_info snd_ctl_boolean_mono_info + +static int se200pci_cont_enum_info(struct snd_kcontrol *kc, + struct snd_ctl_elem_info *uinfo) { - struct snd_ice1712 *ice; - int n; + int n, c; - ice = snd_kcontrol_chip(kc); n = kc->private_value; - if (se200pci_cont[n].type == VOLUME1 || - se200pci_cont[n].type == VOLUME2) { - uc->value.integer.value[0] = ice->spec.se.vol[n].ch1; - uc->value.integer.value[1] = ice->spec.se.vol[n].ch2; - - } else if (se200pci_cont[n].type == BOOLEAN || - se200pci_cont[n].type == ENUM) - uc->value.integer.value[0] = ice->spec.se.vol[n].ch1; - + c = se200pci_get_enum_count(n); + if (!c) + return -EINVAL; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = c; + if (uinfo->value.enumerated.item >= c) + uinfo->value.enumerated.item = c - 1; + strcpy(uinfo->value.enumerated.name, + se200pci_cont[n].member[uinfo->value.enumerated.item]); return 0; } -static int se200pci_cont_put(struct snd_kcontrol *kc, - struct snd_ctl_elem_value *uc) +static int se200pci_cont_volume_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) { - struct snd_ice1712 *ice; - int n; - unsigned int vol1, vol2; - int changed; - - ice = snd_kcontrol_chip(kc); - n = kc->private_value; + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + int n = kc->private_value; + uc->value.integer.value[0] = ice->spec.se.vol[n].ch1; + uc->value.integer.value[1] = ice->spec.se.vol[n].ch2; + return 0; +} - changed = 0; - vol1 = 0; vol2 = 0; - if (se200pci_cont[n].type == VOLUME1 || - se200pci_cont[n].type == VOLUME2) { - vol1 = uc->value.integer.value[0]; - vol2 = uc->value.integer.value[1]; - if (ice->spec.se.vol[n].ch1 != vol1) - changed = 1; - if (ice->spec.se.vol[n].ch2 != vol2) - changed = 1; - ice->spec.se.vol[n].ch1 = vol1; - ice->spec.se.vol[n].ch2 = vol2; +static int se200pci_cont_boolean_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + int n = kc->private_value; + uc->value.integer.value[0] = ice->spec.se.vol[n].ch1; + return 0; +} - } else if (se200pci_cont[n].type == BOOLEAN || - se200pci_cont[n].type == ENUM) { - vol1 = uc->value.integer.value[0]; - if (ice->spec.se.vol[n].ch1 != vol1) - changed = 1; - ice->spec.se.vol[n].ch1 = vol1; - } +static int se200pci_cont_enum_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + int n = kc->private_value; + uc->value.enumerated.item[0] = ice->spec.se.vol[n].ch1; + return 0; +} +static void se200pci_cont_update(struct snd_ice1712 *ice, int n) +{ switch (se200pci_cont[n].target) { case WM8766: se200pci_WM8766_set_volume(ice, - se200pci_cont[n].ch, vol1, vol2); + se200pci_cont[n].ch, + ice->spec.se.vol[n].ch1, + ice->spec.se.vol[n].ch2); break; case WM8776in: - se200pci_WM8776_set_input_volume(ice, vol1, vol2); + se200pci_WM8776_set_input_volume(ice, + ice->spec.se.vol[n].ch1, + ice->spec.se.vol[n].ch2); break; case WM8776out: - se200pci_WM8776_set_output_volume(ice, vol1, vol2); + se200pci_WM8776_set_output_volume(ice, + ice->spec.se.vol[n].ch1, + ice->spec.se.vol[n].ch2); break; case WM8776sel: - se200pci_WM8776_set_input_selector(ice, vol1); + se200pci_WM8776_set_input_selector(ice, + ice->spec.se.vol[n].ch1); break; case WM8776agc: - se200pci_WM8776_set_agc(ice, vol1); + se200pci_WM8776_set_agc(ice, ice->spec.se.vol[n].ch1); break; case WM8776afl: - se200pci_WM8776_set_afl(ice, vol1); + se200pci_WM8776_set_afl(ice, ice->spec.se.vol[n].ch1); break; default: break; } +} + +static int se200pci_cont_volume_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + int n = kc->private_value; + unsigned int vol1, vol2; + int changed; + + changed = 0; + vol1 = uc->value.integer.value[0] & 0xff; + vol2 = uc->value.integer.value[1] & 0xff; + if (ice->spec.se.vol[n].ch1 != vol1) { + ice->spec.se.vol[n].ch1 = vol1; + changed = 1; + } + if (ice->spec.se.vol[n].ch2 != vol2) { + ice->spec.se.vol[n].ch2 = vol2; + changed = 1; + } + if (changed) + se200pci_cont_update(ice, n); return changed; } +static int se200pci_cont_boolean_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + int n = kc->private_value; + unsigned int vol1; + + vol1 = !!uc->value.integer.value[0]; + if (ice->spec.se.vol[n].ch1 != vol1) { + ice->spec.se.vol[n].ch1 = vol1; + se200pci_cont_update(ice, n); + return 1; + } + return 0; +} + +static int se200pci_cont_enum_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + int n = kc->private_value; + unsigned int vol1; + + vol1 = uc->value.enumerated.item[0]; + if (vol1 >= se200pci_get_enum_count(n)) + return -EINVAL; + if (ice->spec.se.vol[n].ch1 != vol1) { + ice->spec.se.vol[n].ch1 = vol1; + se200pci_cont_update(ice, n); + return 1; + } + return 0; +} + static const DECLARE_TLV_DB_SCALE(db_scale_gain1, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(db_scale_gain2, -10350, 50, 1); @@ -554,23 +597,36 @@ static int __devinit se200pci_add_controls(struct snd_ice1712 *ice) memset(&cont, 0, sizeof(cont)); cont.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - cont.info = se200pci_cont_info; - cont.get = se200pci_cont_get; - cont.put = se200pci_cont_put; for (i = 0; i < ARRAY_SIZE(se200pci_cont); i++) { cont.private_value = i; cont.name = se200pci_cont[i].name; cont.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; cont.tlv.p = NULL; - if (se200pci_cont[i].type == VOLUME1 || - se200pci_cont[i].type == VOLUME2) { - + switch (se200pci_cont[i].type) { + case VOLUME1: + case VOLUME2: + cont.info = se200pci_cont_volume_info; + cont.get = se200pci_cont_volume_get; + cont.put = se200pci_cont_volume_put; cont.access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - if (se200pci_cont[i].type == VOLUME1) cont.tlv.p = db_scale_gain1; else cont.tlv.p = db_scale_gain2; + break; + case BOOLEAN: + cont.info = se200pci_cont_boolean_info; + cont.get = se200pci_cont_boolean_get; + cont.put = se200pci_cont_boolean_put; + break; + case ENUM: + cont.info = se200pci_cont_enum_info; + cont.get = se200pci_cont_enum_get; + cont.put = se200pci_cont_enum_put; + break; + default: + snd_BUG(); + return -EINVAL; } err = snd_ctl_add(ice->card, snd_ctl_new1(&cont, ice)); if (err < 0) -- cgit v0.10.2 From d05ab185b770de96399766be6bcb5769ab99bc09 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 16:13:32 +0100 Subject: [ALSA] vxpocket - Check value range in ctl callbacks Check the value ranges in ctl put callbacks in vxpocket driver. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c index b8fcd79..a37f0a8 100644 --- a/sound/drivers/vx/vx_mixer.c +++ b/sound/drivers/vx/vx_mixer.c @@ -439,14 +439,19 @@ static int vx_output_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele { struct vx_core *chip = snd_kcontrol_chip(kcontrol); int codec = kcontrol->id.index; + unsigned int val[2], vmax; + + vmax = chip->hw->output_level_max; + val[0] = ucontrol->value.integer.value[0]; + val[1] = ucontrol->value.integer.value[1]; + if (val[0] > vmax || val[1] > vmax) + return -EINVAL; mutex_lock(&chip->mixer_mutex); - if (ucontrol->value.integer.value[0] != chip->output_level[codec][0] || - ucontrol->value.integer.value[1] != chip->output_level[codec][1]) { - vx_set_analog_output_level(chip, codec, - ucontrol->value.integer.value[0], - ucontrol->value.integer.value[1]); - chip->output_level[codec][0] = ucontrol->value.integer.value[0]; - chip->output_level[codec][1] = ucontrol->value.integer.value[1]; + if (val[0] != chip->output_level[codec][0] || + val[1] != chip->output_level[codec][1]) { + vx_set_analog_output_level(chip, codec, val[0], val[1]); + chip->output_level[codec][0] = val[0]; + chip->output_level[codec][1] = val[1]; mutex_unlock(&chip->mixer_mutex); return 1; } @@ -506,6 +511,14 @@ static int vx_audio_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v static int vx_audio_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *chip = snd_kcontrol_chip(kcontrol); + + if (chip->type >= VX_TYPE_VXPOCKET) { + if (ucontrol->value.enumerated.item[0] > 2) + return -EINVAL; + } else { + if (ucontrol->value.enumerated.item[0] > 1) + return -EINVAL; + } mutex_lock(&chip->mixer_mutex); if (chip->audio_source_target != ucontrol->value.enumerated.item[0]) { chip->audio_source_target = ucontrol->value.enumerated.item[0]; @@ -554,6 +567,9 @@ static int vx_clock_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ static int vx_clock_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *chip = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.enumerated.item[0] > 2) + return -EINVAL; mutex_lock(&chip->mixer_mutex); if (chip->clock_mode != ucontrol->value.enumerated.item[0]) { chip->clock_mode = ucontrol->value.enumerated.item[0]; @@ -603,12 +619,17 @@ static int vx_audio_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ struct vx_core *chip = snd_kcontrol_chip(kcontrol); int audio = kcontrol->private_value & 0xff; int capture = (kcontrol->private_value >> 8) & 1; + unsigned int val[2]; + val[0] = ucontrol->value.integer.value[0]; + val[1] = ucontrol->value.integer.value[1]; + if (val[0] > CVAL_MAX || val[1] > CVAL_MAX) + return -EINVAL; mutex_lock(&chip->mixer_mutex); - if (ucontrol->value.integer.value[0] != chip->audio_gain[capture][audio] || - ucontrol->value.integer.value[1] != chip->audio_gain[capture][audio+1]) { - vx_set_audio_gain(chip, audio, capture, ucontrol->value.integer.value[0]); - vx_set_audio_gain(chip, audio+1, capture, ucontrol->value.integer.value[1]); + if (val[0] != chip->audio_gain[capture][audio] || + val[1] != chip->audio_gain[capture][audio+1]) { + vx_set_audio_gain(chip, audio, capture, val[0]); + vx_set_audio_gain(chip, audio+1, capture, val[1]); mutex_unlock(&chip->mixer_mutex); return 1; } @@ -632,13 +653,19 @@ static int vx_audio_monitor_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el { struct vx_core *chip = snd_kcontrol_chip(kcontrol); int audio = kcontrol->private_value & 0xff; + unsigned int val[2]; + + val[0] = ucontrol->value.integer.value[0]; + val[1] = ucontrol->value.integer.value[1]; + if (val[0] > CVAL_MAX || val[1] > CVAL_MAX) + return -EINVAL; mutex_lock(&chip->mixer_mutex); - if (ucontrol->value.integer.value[0] != chip->audio_monitor[audio] || - ucontrol->value.integer.value[1] != chip->audio_monitor[audio+1]) { - vx_set_monitor_level(chip, audio, ucontrol->value.integer.value[0], + if (val[0] != chip->audio_monitor[audio] || + val[1] != chip->audio_monitor[audio+1]) { + vx_set_monitor_level(chip, audio, val[0], chip->audio_monitor_active[audio]); - vx_set_monitor_level(chip, audio+1, ucontrol->value.integer.value[1], + vx_set_monitor_level(chip, audio+1, val[1], chip->audio_monitor_active[audio+1]); mutex_unlock(&chip->mixer_mutex); return 1; @@ -669,8 +696,10 @@ static int vx_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va mutex_lock(&chip->mixer_mutex); if (ucontrol->value.integer.value[0] != chip->audio_active[audio] || ucontrol->value.integer.value[1] != chip->audio_active[audio+1]) { - vx_set_audio_switch(chip, audio, ucontrol->value.integer.value[0]); - vx_set_audio_switch(chip, audio+1, ucontrol->value.integer.value[1]); + vx_set_audio_switch(chip, audio, + !!ucontrol->value.integer.value[0]); + vx_set_audio_switch(chip, audio+1, + !!ucontrol->value.integer.value[1]); mutex_unlock(&chip->mixer_mutex); return 1; } @@ -699,9 +728,9 @@ static int vx_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ if (ucontrol->value.integer.value[0] != chip->audio_monitor_active[audio] || ucontrol->value.integer.value[1] != chip->audio_monitor_active[audio+1]) { vx_set_monitor_level(chip, audio, chip->audio_monitor[audio], - ucontrol->value.integer.value[0]); + !!ucontrol->value.integer.value[0]); vx_set_monitor_level(chip, audio+1, chip->audio_monitor[audio+1], - ucontrol->value.integer.value[1]); + !!ucontrol->value.integer.value[1]); mutex_unlock(&chip->mixer_mutex); return 1; } diff --git a/sound/pcmcia/vx/vxp_mixer.c b/sound/pcmcia/vx/vxp_mixer.c index 1eff158..bf9d3b3 100644 --- a/sound/pcmcia/vx/vxp_mixer.c +++ b/sound/pcmcia/vx/vxp_mixer.c @@ -53,6 +53,10 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip; + unsigned int val = ucontrol->value.integer.value[0]; + + if (val > MIC_LEVEL_MAX) + return -EINVAL; mutex_lock(&_chip->mixer_mutex); if (chip->mic_level != ucontrol->value.integer.value[0]) { vx_set_mic_level(_chip, ucontrol->value.integer.value[0]); @@ -94,10 +98,11 @@ static int vx_mic_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip; + int val = !!ucontrol->value.integer.value[0]; mutex_lock(&_chip->mixer_mutex); - if (chip->mic_level != ucontrol->value.integer.value[0]) { - vx_set_mic_boost(_chip, ucontrol->value.integer.value[0]); - chip->mic_level = ucontrol->value.integer.value[0]; + if (chip->mic_level != val) { + vx_set_mic_boost(_chip, val); + chip->mic_level = val; mutex_unlock(&_chip->mixer_mutex); return 1; } -- cgit v0.10.2 From d4079ac49a08e36d6839a9ceb26aec8c24c9ed82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 16:14:12 +0100 Subject: [ALSA] powermac - Check value range in ctl callbacks Check the value ranges in ctl put callbacks properly in snd-powermac driver. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 05dabe4..b15bfb6 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -175,10 +175,12 @@ static int snd_pmac_awacs_put_volume(struct snd_kcontrol *kcontrol, int inverted = (kcontrol->private_value >> 16) & 1; int val, oldval; unsigned long flags; - int vol[2]; + unsigned int vol[2]; vol[0] = ucontrol->value.integer.value[0]; vol[1] = ucontrol->value.integer.value[1]; + if (vol[0] > 0x0f || vol[1] > 0x0f) + return -EINVAL; if (inverted) { vol[0] = 0x0f - vol[0]; vol[1] = 0x0f - vol[1]; @@ -421,10 +423,14 @@ static int snd_pmac_awacs_put_tone_amp(struct snd_kcontrol *kcontrol, struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); int index = kcontrol->private_value; struct awacs_amp *amp = chip->mixer_data; + unsigned int val; snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - if (ucontrol->value.integer.value[0] != amp->amp_tone[index]) { - amp->amp_tone[index] = ucontrol->value.integer.value[0]; + val = ucontrol->value.integer.value[0]; + if (val > 14) + return -EINVAL; + if (val != amp->amp_tone[index]) { + amp->amp_tone[index] = val; awacs_amp_set_tone(amp, amp->amp_tone[0], amp->amp_tone[1]); return 1; } @@ -456,9 +462,13 @@ static int snd_pmac_awacs_put_master_amp(struct snd_kcontrol *kcontrol, { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); struct awacs_amp *amp = chip->mixer_data; + unsigned int val; snd_assert(amp, return -EINVAL); - if (ucontrol->value.integer.value[0] != amp->amp_master) { - amp->amp_master = ucontrol->value.integer.value[0]; + val = ucontrol->value.integer.value[0]; + if (val > 99) + return -EINVAL; + if (val != amp->amp_master) { + amp->amp_master = val; awacs_amp_set_master(amp, amp->amp_master); return 1; } diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c index 566b5ab..465dd04 100644 --- a/sound/ppc/beep.c +++ b/sound/ppc/beep.c @@ -195,10 +195,13 @@ static int snd_pmac_put_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int oval; + unsigned int oval, nval; snd_assert(chip->beep, return -ENXIO); oval = chip->beep->volume; - chip->beep->volume = ucontrol->value.integer.value[0]; + nval = ucontrol->value.integer.value[0]; + if (nval > 100) + return -EINVAL; + chip->beep->volume = nval; return oval != chip->beep->volume; } diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index e02263f..fec74e8 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -136,6 +136,9 @@ snd_pmac_burgundy_write_volume(struct snd_pmac *chip, unsigned int address, { int hardvolume, lvolume, rvolume; + if (volume[0] < 0 || volume[0] > 100 || + volume[1] < 0 || volume[1] > 100) + return; /* -EINVAL */ lvolume = volume[0] ? volume[0] + BURGUNDY_VOLUME_OFFSET : 0; rvolume = volume[1] ? volume[1] + BURGUNDY_VOLUME_OFFSET : 0; @@ -301,14 +304,14 @@ static int snd_pmac_burgundy_put_volume_out(struct snd_kcontrol *kcontrol, struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); int stereo = (kcontrol->private_value >> 24) & 1; - int oval, val; + unsigned int oval, val; oval = ~snd_pmac_burgundy_rcb(chip, addr) & 0xff; - val = ucontrol->value.integer.value[0]; + val = ucontrol->value.integer.value[0] & 15; if (stereo) - val |= ucontrol->value.integer.value[1] << 4; + val |= (ucontrol->value.integer.value[1] & 15) << 4; else - val |= ucontrol->value.integer.value[0] << 4; + val |= val << 4; val = ~val & 0xff; snd_pmac_burgundy_wcb(chip, addr, val); return val != oval; diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c index c5a1f0b..0c81457 100644 --- a/sound/ppc/daca.c +++ b/sound/ppc/daca.c @@ -115,7 +115,7 @@ static int daca_put_deemphasis(struct snd_kcontrol *kcontrol, return -ENODEV; change = mix->deemphasis != ucontrol->value.integer.value[0]; if (change) { - mix->deemphasis = ucontrol->value.integer.value[0]; + mix->deemphasis = !!ucontrol->value.integer.value[0]; daca_set_volume(mix); } return change; @@ -149,15 +149,20 @@ static int daca_put_volume(struct snd_kcontrol *kcontrol, { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); struct pmac_daca *mix; + unsigned int vol[2]; int change; if (! (mix = chip->mixer_data)) return -ENODEV; - change = mix->left_vol != ucontrol->value.integer.value[0] || - mix->right_vol != ucontrol->value.integer.value[1]; + vol[0] = ucontrol->value.integer.value[0]; + vol[1] = ucontrol->value.integer.value[1]; + if (vol[0] > DACA_VOL_MAX || vol[1] > DACA_VOL_MAX) + return -EINVAL; + change = mix->left_vol != vol[0] || + mix->right_vol != vol[1]; if (change) { - mix->left_vol = ucontrol->value.integer.value[0]; - mix->right_vol = ucontrol->value.integer.value[1]; + mix->left_vol = vol[0]; + mix->right_vol = vol[1]; daca_set_volume(mix); } return change; @@ -188,7 +193,7 @@ static int daca_put_amp(struct snd_kcontrol *kcontrol, return -ENODEV; change = mix->amp_on != ucontrol->value.integer.value[0]; if (change) { - mix->amp_on = ucontrol->value.integer.value[0]; + mix->amp_on = !!ucontrol->value.integer.value[0]; i2c_smbus_write_byte_data(mix->i2c.client, DACA_REG_GCFG, mix->amp_on ? 0x05 : 0x04); } diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 4f9b19c..8c47beb 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -1028,7 +1028,7 @@ static int pmac_auto_mute_put(struct snd_kcontrol *kcontrol, { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); if (ucontrol->value.integer.value[0] != chip->auto_mute) { - chip->auto_mute = ucontrol->value.integer.value[0]; + chip->auto_mute = !!ucontrol->value.integer.value[0]; if (chip->update_automute) chip->update_automute(chip, 1); return 1; diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 5821cdd..bacff3d 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -275,14 +275,20 @@ static int tumbler_put_master_volume(struct snd_kcontrol *kcontrol, { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); struct pmac_tumbler *mix = chip->mixer_data; + unsigned int vol[2]; int change; snd_assert(mix, return -ENODEV); - change = mix->master_vol[0] != ucontrol->value.integer.value[0] || - mix->master_vol[1] != ucontrol->value.integer.value[1]; + vol[0] = ucontrol->value.integer.value[0]; + vol[1] = ucontrol->value.integer.value[1]; + if (vol[0] >= ARRAY_SIZE(master_volume_table) || + vol[1] >= ARRAY_SIZE(master_volume_table)) + return -EINVAL; + change = mix->master_vol[0] != vol[0] || + mix->master_vol[1] != vol[1]; if (change) { - mix->master_vol[0] = ucontrol->value.integer.value[0]; - mix->master_vol[1] = ucontrol->value.integer.value[1]; + mix->master_vol[0] = vol[0]; + mix->master_vol[1] = vol[1]; tumbler_set_master_volume(mix); } return change; @@ -417,13 +423,22 @@ static int tumbler_put_drc_value(struct snd_kcontrol *kcontrol, { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); struct pmac_tumbler *mix; + unsigned int val; int change; if (! (mix = chip->mixer_data)) return -ENODEV; - change = mix->drc_range != ucontrol->value.integer.value[0]; + val = ucontrol->value.integer.value[0]; + if (chip->model == PMAC_TUMBLER) { + if (val > TAS3001_DRC_MAX) + return -EINVAL; + } else { + if (val > TAS3004_DRC_MAX) + return -EINVAL; + } + change = mix->drc_range != val; if (change) { - mix->drc_range = ucontrol->value.integer.value[0]; + mix->drc_range = val; if (chip->model == PMAC_TUMBLER) tumbler_set_drc(mix); else @@ -530,13 +545,17 @@ static int tumbler_put_mono(struct snd_kcontrol *kcontrol, struct tumbler_mono_vol *info = (struct tumbler_mono_vol *)kcontrol->private_value; struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); struct pmac_tumbler *mix; + unsigned int vol; int change; if (! (mix = chip->mixer_data)) return -ENODEV; - change = mix->mono_vol[info->index] != ucontrol->value.integer.value[0]; + vol = ucontrol->value.integer.value[0]; + if (vol >= info->max) + return -EINVAL; + change = mix->mono_vol[info->index] != vol; if (change) { - mix->mono_vol[info->index] = ucontrol->value.integer.value[0]; + mix->mono_vol[info->index] = vol; tumbler_set_mono_volume(mix, info); } return change; @@ -672,15 +691,21 @@ static int snapper_put_mix(struct snd_kcontrol *kcontrol, int idx = (int)kcontrol->private_value; struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); struct pmac_tumbler *mix; + unsigned int vol[2]; int change; if (! (mix = chip->mixer_data)) return -ENODEV; - change = mix->mix_vol[idx][0] != ucontrol->value.integer.value[0] || - mix->mix_vol[idx][1] != ucontrol->value.integer.value[1]; + vol[0] = ucontrol->value.integer.value[0]; + vol[1] = ucontrol->value.integer.value[1]; + if (vol[0] >= ARRAY_SIZE(mixer_volume_table) || + vol[1] >= ARRAY_SIZE(mixer_volume_table)) + return -EINVAL; + change = mix->mix_vol[idx][0] != vol[0] || + mix->mix_vol[idx][1] != vol[1]; if (change) { - mix->mix_vol[idx][0] = ucontrol->value.integer.value[0]; - mix->mix_vol[idx][1] = ucontrol->value.integer.value[1]; + mix->mix_vol[idx][0] = vol[0]; + mix->mix_vol[idx][1] = vol[1]; snapper_set_mix_vol(mix, idx); } return change; @@ -784,7 +809,7 @@ static int snapper_get_capture_source(struct snd_kcontrol *kcontrol, struct pmac_tumbler *mix = chip->mixer_data; snd_assert(mix, return -ENODEV); - ucontrol->value.integer.value[0] = mix->capture_source; + ucontrol->value.enumerated.value[0] = mix->capture_source; return 0; } @@ -796,9 +821,9 @@ static int snapper_put_capture_source(struct snd_kcontrol *kcontrol, int change; snd_assert(mix, return -ENODEV); - change = ucontrol->value.integer.value[0] != mix->capture_source; + change = ucontrol->value.enuemerated.item[0] != mix->capture_source; if (change) { - mix->capture_source = !!ucontrol->value.integer.value[0]; + mix->capture_source = !!ucontrol->value.enumerated.item[0]; snapper_set_capture_source(mix); } return change; -- cgit v0.10.2 From 02ff1324930675599694bb1285afc4341f98a7a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 16:15:29 +0100 Subject: [ALSA] ak4xxx - Check value ranges in ctl callbacks Check the value ranges in ctl put callbacks properly in ak4xxx-adda driver. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index de03f68..39bb03a 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -377,8 +377,11 @@ static int put_ak_reg(struct snd_kcontrol *kcontrol, int addr, static int snd_akm4xxx_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - return put_ak_reg(kcontrol, AK_GET_ADDR(kcontrol->private_value), - ucontrol->value.integer.value[0]); + unsigned int mask = AK_GET_MASK(kcontrol->private_value); + unsigned int val = ucontrol->value.integer.value[0]; + if (val > mask) + return -EINVAL; + return put_ak_reg(kcontrol, AK_GET_ADDR(kcontrol->private_value), val); } static int snd_akm4xxx_stereo_volume_info(struct snd_kcontrol *kcontrol, @@ -409,11 +412,16 @@ static int snd_akm4xxx_stereo_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int addr = AK_GET_ADDR(kcontrol->private_value); + unsigned int mask = AK_GET_MASK(kcontrol->private_value); + unsigned int val[2]; int change; - change = put_ak_reg(kcontrol, addr, ucontrol->value.integer.value[0]); - change |= put_ak_reg(kcontrol, addr + 1, - ucontrol->value.integer.value[1]); + val[0] = ucontrol->value.integer.value[0]; + val[1] = ucontrol->value.integer.value[1]; + if (val[0] > mask || val[1] > mask) + return -EINVAL; + change = put_ak_reg(kcontrol, addr, val[0]); + change |= put_ak_reg(kcontrol, addr + 1, val[1]); return change; } @@ -508,6 +516,18 @@ static int ak4xxx_switch_put(struct snd_kcontrol *kcontrol, #define AK5365_NUM_INPUTS 5 +static int ak4xxx_capture_num_inputs(struct snd_akm4xxx *ak, int mixer_ch) +{ + int num_names; + const char **input_names; + + input_names = ak->adc_info[mixer_ch].input_names; + num_names = 0; + while (num_names < AK5365_NUM_INPUTS && input_names[num_names]) + ++num_names; + return num_names; +} + static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -516,18 +536,16 @@ static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol, const char **input_names; int num_names, idx; - input_names = ak->adc_info[mixer_ch].input_names; - - num_names = 0; - while (num_names < AK5365_NUM_INPUTS && input_names[num_names]) - ++num_names; - + num_names = ak4xxx_capture_num_inputs(ak, mixer_ch); + if (!num_names) + return -EINVAL; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = num_names; idx = uinfo->value.enumerated.item; if (idx >= num_names) return -EINVAL; + input_names = ak->adc_info[mixer_ch].input_names; strncpy(uinfo->value.enumerated.name, input_names[idx], sizeof(uinfo->value.enumerated.name)); return 0; @@ -551,10 +569,15 @@ static int ak4xxx_capture_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol); + int mixer_ch = AK_GET_SHIFT(kcontrol->private_value); int chip = AK_GET_CHIP(kcontrol->private_value); int addr = AK_GET_ADDR(kcontrol->private_value); int mask = AK_GET_MASK(kcontrol->private_value); unsigned char oval, val; + int num_names = ak4xxx_capture_num_inputs(ak, mixer_ch); + + if (ucontrol->value.enumerated.item[0] >= num_names) + return -EINVAL; oval = snd_akm4xxx_get(ak, chip, addr); val = oval & ~mask; -- cgit v0.10.2 From 498ade1a133dffd0f3ee90952737045d56e6689a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 16:16:32 +0100 Subject: [ALSA] aoa - Check value range in ctl callbacks Check the value ranges in ctl put callbacks properly in aoa drivers. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c index 71e3f93..6a3837d 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.c +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c @@ -138,6 +138,13 @@ static int onyx_snd_vol_put(struct snd_kcontrol *kcontrol, struct onyx *onyx = snd_kcontrol_chip(kcontrol); s8 l, r; + if (ucontrol->value.integer.value[0] < -128 + VOLUME_RANGE_SHIFT || + ucontrol->value.integer.value[0] > -1 + VOLUME_RANGE_SHIFT) + return -EINVAL; + if (ucontrol->value.integer.value[1] < -128 + VOLUME_RANGE_SHIFT || + ucontrol->value.integer.value[1] > -1 + VOLUME_RANGE_SHIFT) + return -EINVAL; + mutex_lock(&onyx->mutex); onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l); onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r); @@ -206,6 +213,9 @@ static int onyx_snd_inputgain_put(struct snd_kcontrol *kcontrol, struct onyx *onyx = snd_kcontrol_chip(kcontrol); u8 v, n; + if (ucontrol->value.integer.value[0] < 3 + INPUTGAIN_RANGE_SHIFT || + ucontrol->value.integer.value[0] > 28 + INPUTGAIN_RANGE_SHIFT) + return -EINVAL; mutex_lock(&onyx->mutex); onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v); n = v; @@ -272,6 +282,8 @@ static void onyx_set_capture_source(struct onyx *onyx, int mic) static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + if (ucontrol->value.enumerated.item[0] > 1) + return -EINVAL; onyx_set_capture_source(snd_kcontrol_chip(kcontrol), ucontrol->value.enumerated.item[0]); return 1; diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c index 70c3416..7a16a33 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -248,6 +248,13 @@ static int tas_snd_vol_put(struct snd_kcontrol *kcontrol, { struct tas *tas = snd_kcontrol_chip(kcontrol); + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 177) + return -EINVAL; + if (ucontrol->value.integer.value[1] < 0 || + ucontrol->value.integer.value[1] > 177) + return -EINVAL; + mutex_lock(&tas->mtx); if (tas->cached_volume_l == ucontrol->value.integer.value[0] && tas->cached_volume_r == ucontrol->value.integer.value[1]) { @@ -401,6 +408,10 @@ static int tas_snd_drc_range_put(struct snd_kcontrol *kcontrol, { struct tas *tas = snd_kcontrol_chip(kcontrol); + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > TAS3004_DRC_MAX) + return -EINVAL; + mutex_lock(&tas->mtx); if (tas->drc_range == ucontrol->value.integer.value[0]) { mutex_unlock(&tas->mtx); @@ -447,7 +458,7 @@ static int tas_snd_drc_switch_put(struct snd_kcontrol *kcontrol, return 0; } - tas->drc_enabled = ucontrol->value.integer.value[0]; + tas->drc_enabled = !!ucontrol->value.integer.value[0]; if (tas->hw_enabled) tas3004_set_drc(tas); mutex_unlock(&tas->mtx); @@ -494,6 +505,8 @@ static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol, struct tas *tas = snd_kcontrol_chip(kcontrol); int oldacr; + if (ucontrol->value.enumerated.item[0] > 1) + return -EINVAL; mutex_lock(&tas->mtx); oldacr = tas->acr; @@ -562,6 +575,9 @@ static int tas_snd_treble_put(struct snd_kcontrol *kcontrol, { struct tas *tas = snd_kcontrol_chip(kcontrol); + if (ucontrol->value.integer.value[0] < TAS3004_TREBLE_MIN || + ucontrol->value.integer.value[0] > TAS3004_TREBLE_MAX) + return -EINVAL; mutex_lock(&tas->mtx); if (tas->treble == ucontrol->value.integer.value[0]) { mutex_unlock(&tas->mtx); @@ -610,6 +626,9 @@ static int tas_snd_bass_put(struct snd_kcontrol *kcontrol, { struct tas *tas = snd_kcontrol_chip(kcontrol); + if (ucontrol->value.integer.value[0] < TAS3004_BASS_MIN || + ucontrol->value.integer.value[0] > TAS3004_BASS_MAX) + return -EINVAL; mutex_lock(&tas->mtx); if (tas->bass == ucontrol->value.integer.value[0]) { mutex_unlock(&tas->mtx); diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c index 8b2ba99..dea7abb 100644 --- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c +++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c @@ -600,7 +600,7 @@ static int n##_control_put(struct snd_kcontrol *kcontrol, \ struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol); \ if (gpio->methods && gpio->methods->get_##n) \ gpio->methods->set_##n(gpio, \ - ucontrol->value.integer.value[0]); \ + !!ucontrol->value.integer.value[0]); \ return 1; \ } \ static struct snd_kcontrol_new n##_ctl = { \ -- cgit v0.10.2 From 3b892467786410f26dffc2c7bccd3ea445604037 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 16:17:24 +0100 Subject: [ALSA] Check value range in ctl callbacks Check the value ranges in ctl put callbacks properly (in the rest drivers). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index dcc90f9..68070cc 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -461,13 +461,14 @@ static int snd_mts64_ctl_smpte_switch_put(struct snd_kcontrol* kctl, { struct mts64 *mts = snd_kcontrol_chip(kctl); int changed = 0; + int val = !!uctl->value.integer.value[0]; spin_lock_irq(&mts->lock); - if (mts->smpte_switch == uctl->value.integer.value[0]) + if (mts->smpte_switch == val) goto __out; changed = 1; - mts->smpte_switch = uctl->value.integer.value[0]; + mts->smpte_switch = val; if (mts->smpte_switch) { mts64_smpte_start(mts->pardev->port, mts->time[0], mts->time[1], @@ -541,12 +542,13 @@ static int snd_mts64_ctl_smpte_time_put(struct snd_kcontrol *kctl, { struct mts64 *mts = snd_kcontrol_chip(kctl); int idx = kctl->private_value; + unsigned int time = uctl->value.integer.value[0] % 60; int changed = 0; spin_lock_irq(&mts->lock); - if (mts->time[idx] != uctl->value.integer.value[0]) { + if (mts->time[idx] != time) { changed = 1; - mts->time[idx] = uctl->value.integer.value[0]; + mts->time[idx] = time; } spin_unlock_irq(&mts->lock); @@ -636,6 +638,8 @@ static int snd_mts64_ctl_smpte_fps_put(struct snd_kcontrol *kctl, struct mts64 *mts = snd_kcontrol_chip(kctl); int changed = 0; + if (uctl->value.enumerated.item[0] >= 5) + return -EINVAL; spin_lock_irq(&mts->lock); if (mts->fps != uctl->value.enumerated.item[0]) { changed = 1; diff --git a/sound/i2c/other/pt2258.c b/sound/i2c/other/pt2258.c index 00c83d8..987d2c9 100644 --- a/sound/i2c/other/pt2258.c +++ b/sound/i2c/other/pt2258.c @@ -113,6 +113,8 @@ static int pt2258_stereo_volume_put(struct snd_kcontrol *kcontrol, val0 = 79 - ucontrol->value.integer.value[0]; val1 = 79 - ucontrol->value.integer.value[1]; + if (val0 < 0 || val0 > 79 || val1 < 0 || val1 > 79) + return -EINVAL; if (val0 == pt->volume[base] && val1 == pt->volume[base + 1]) return 0; diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index d295936..c2baf4c 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -483,6 +483,10 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, /* equalizer elements */ + if (left < -0x7f || left > 0x7f || + right < -0x7f || right > 0x7f) + return -EINVAL; + if (left_old > 0x80) left_old = 0x80 - left_old; if (right_old > 0x80) @@ -520,6 +524,10 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, /* non-equalizer elements */ + if (left < 0 || left > 0x20 || + right < 0 || right > 0x20) + return -EINVAL; + left_old = 0x20 - left_old; right_old = 0x20 - right_old; diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 8861d2f..12c41df 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -523,11 +523,14 @@ static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_card_aica *dreamcastcard; + unsigned int vol; dreamcastcard = kcontrol->private_data; if (unlikely(!dreamcastcard->channel)) return -ETXTBSY; - if (unlikely(dreamcastcard->channel->vol == - ucontrol->value.integer.value[0])) + vol = ucontrol->value.integer.value[0]; + if (vol > 0xff) + return -EINVAL; + if (unlikely(dreamcastcard->channel->vol == vol)) return 0; dreamcastcard->channel->vol = ucontrol->value.integer.value[0]; dreamcastcard->master_volume = ucontrol->value.integer.value[0]; diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 07962a3..b1d4315 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -859,7 +859,7 @@ static int snd_amd7930_put_volume(struct snd_kcontrol *kctl, struct snd_ctl_elem spin_lock_irqsave(&amd->lock, flags); if (*swval != ucontrol->value.integer.value[0]) { - *swval = ucontrol->value.integer.value[0]; + *swval = ucontrol->value.integer.value[0] & 0xff; __amd7930_update_map(amd); change = 1; } else diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 376b986..af1bf4b 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2279,9 +2279,20 @@ static int snd_cs4215_put_volume(struct snd_kcontrol *kcontrol, struct snd_dbri *dbri = snd_kcontrol_chip(kcontrol); struct dbri_streaminfo *info = &dbri->stream_info[kcontrol->private_value]; + unsigned int vol[2]; int changed = 0; - if (info->left_gain != ucontrol->value.integer.value[0]) { + vol[0] = ucontrol->value.integer.value[0]; + vol[1] = ucontrol->value.integer.value[1]; + if (kcontrol->private_value == DBRI_PLAY) { + if (vol[0] > DBRI_MAX_VOLUME || vol[1] > DBRI_MAX_VOLUME) + return -EINVAL; + } else { + if (vol[0] > DBRI_MAX_GAIN || vol[1] > DBRI_MAX_GAIN) + return -EINVAL; + } + + if (info->left_gain != info->left_gain = ucontrol->value.integer.value[0]; changed = 1; } -- cgit v0.10.2 From 4b6c26aff30343ac95a111b1658e72f94bf51291 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Nov 2007 16:18:14 +0100 Subject: [ALSA] at73c213 - Use common callback Use snd_ctl_boolean_mono_info callback to simplify. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index fee869b..bfe17b3 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -536,16 +536,7 @@ out: return retval; } -static int snd_at73c213_mono_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - - return 0; -} +#define snd_at73c213_mono_switch_info snd_ctl_boolean_mono_info static int snd_at73c213_mono_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v0.10.2 From 27da183402ac5ca7d5a4c4704c4a405cd1009c43 Mon Sep 17 00:00:00 2001 From: Wolke Liu Date: Fri, 16 Nov 2007 11:06:30 +0100 Subject: [ALSA] HDA-Intel - Add support for RV6xx HDMI audio This patch is to add R6xx HDMI audio support. Meanwhile, the device ID in the previous patch is changed. I have checked the patch from Herton Ronaldo Krzesinski, it's right as our spec said. :) Signed-off-by: Wolke Liu Signed-off-by: Andrea Zhang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a829c59..38c5916 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -106,6 +106,10 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{ATI, R600}," "{ATI, RV630}," "{ATI, RV610}," + "{ATI, RV670}," + "{ATI, RV635}," + "{ATI, RV620}," + "{ATI, RV770}," "{VIA, VT8251}," "{VIA, VT8237A}," "{SiS, SIS966}," @@ -1942,10 +1946,14 @@ static struct pci_device_id azx_ids[] = { { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */ { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */ { 0x1002, 0x7919, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS690 HDMI */ - { 0x1002, 0x960c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS780 HDMI */ + { 0x1002, 0x960f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS780 HDMI */ { 0x1002, 0xaa00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI R600 HDMI */ { 0x1002, 0xaa08, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV630 HDMI */ { 0x1002, 0xaa10, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV610 HDMI */ + { 0x1002, 0xaa18, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV670 HDMI */ + { 0x1002, 0xaa20, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV635 HDMI */ + { 0x1002, 0xaa28, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV620 HDMI */ + { 0x1002, 0xaa30, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV770 HDMI */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index fbb8969..78441e3 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -158,6 +158,6 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, - { .id = 0x1002aa01, .name = "ATI R600 HDMI", .patch = patch_atihdmi }, + { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi }, {} /* terminator */ }; -- cgit v0.10.2 From 9b1fffddc80949fb4aa93fdcc57544bc2108ced3 Mon Sep 17 00:00:00 2001 From: Ingo Molnar Date: Fri, 16 Nov 2007 15:20:28 +0100 Subject: [ALSA] snd hda suspend latency: shorten codec read not sleeping for every codec read/write but doing a short udelay and a conditional reschedule has cut suspend+resume latency by about 1 second on my T60. Signed-off-by: Ingo Molnar Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 38c5916..41edf85 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -561,7 +561,8 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) } if (!chip->rirb.cmds) return chip->rirb.res; /* the last value */ - schedule_timeout_uninterruptible(1); + udelay(10); + cond_resched(); } while (time_after_eq(timeout, jiffies)); if (chip->msi) { -- cgit v0.10.2 From e171613949e350966f5cc8c9b0023a5f746f7a5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 Nov 2007 17:52:39 +0100 Subject: [ALSA] hda-intel - Show more volume-knob attributes Show more attributs of volume-knob widgets. Also don't put empty lines when no connection list is found. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2bce925..0331510 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -135,6 +135,7 @@ enum { #define AC_PAR_PROC_CAP 0x10 #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 +#define AC_PAR_VOL_KNB_CAP 0x13 /* * AC_VERB_PARAMETERS results (32bit) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 000c6c4..7df1d16 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -305,6 +305,12 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-Out"); snd_iprintf(buffer, "\n"); + /* volume knob is a special widget that always have connection + * list + */ + if (wid_type == AC_WID_VOL_KNB) + wid_caps |= AC_WCAP_CONN_LIST; + if (wid_caps & AC_WCAP_CONN_LIST) conn_len = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); @@ -340,9 +346,15 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, "\n"); break; case AC_WID_VOL_KNB: - snd_iprintf(buffer, " Volume-Knob: 0x%x\n", - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_VOLUME_KNOB_CONTROL, 0)); + pinctls = snd_hda_param_read(codec, nid, + AC_PAR_VOL_KNB_CAP); + snd_iprintf(buffer, " Volume-Knob: delta=%d, " + "steps=%d, ", + (pinctls >> 7) & 1, pinctls & 0x7f); + pinctls = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); + snd_iprintf(buffer, "direct=%d, val=%d\n", + (pinctls >> 7) & 1, pinctls & 0x7f); break; case AC_WID_AUD_OUT: case AC_WID_AUD_IN: @@ -365,13 +377,15 @@ static void print_codec_info(struct snd_info_entry *entry, curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); snd_iprintf(buffer, " Connection: %d\n", conn_len); - snd_iprintf(buffer, " "); - for (c = 0; c < conn_len; c++) { - snd_iprintf(buffer, " 0x%02x", conn[c]); - if (c == curr) - snd_iprintf(buffer, "*"); + if (conn_len > 0) { + snd_iprintf(buffer, " "); + for (c = 0; c < conn_len; c++) { + snd_iprintf(buffer, " 0x%02x", conn[c]); + if (c == curr) + snd_iprintf(buffer, "*"); + } + snd_iprintf(buffer, "\n"); } - snd_iprintf(buffer, "\n"); } } snd_hda_power_down(codec); -- cgit v0.10.2 From f673dc829bc8a9805d82df14a3e08b94f3d6fd50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Nov 2007 11:56:26 +0100 Subject: [ALSA] hda-codec - Revert volume knob controls in STAC codecs Volume knob controls with STAC codecs seem to cause problems with some devices. Volumes change very slowly or silent suddenly. It's likely due to conflict between the software and the hardware volume knob setup. Since we'll have a virtual master control in future, it's safer to remove this control completely right now. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0817f42..d2996ad 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -374,42 +374,6 @@ static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, return 1; } -static int stac92xx_volknob_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 127; - return 0; -} - -static int stac92xx_volknob_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = kcontrol->private_value & 0xff; - return 0; -} - -static int stac92xx_volknob_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int oval = kcontrol->private_value & 0xff; - unsigned int val; - - val = ucontrol->value.integer.value[0] & 0xff; - if (val == oval) - return 0; - - kcontrol->private_value &= ~0xff; - kcontrol->private_value |= val; - - snd_hda_codec_write_cache(codec, kcontrol->private_value >> 16, 0, - AC_VERB_SET_VOLUME_KNOB_CONTROL, val | 0x80); - return 1; -} - static struct hda_verb stac9200_core_init[] = { /* set dac0mux for dac converter */ { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -507,17 +471,6 @@ static struct hda_verb stac9205_core_init[] = { .private_value = verb_read | (verb_write << 16), \ } -#define STAC_VOLKNOB(knob_nid) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Master Playback Volume", \ - .count = 1, \ - .info = stac92xx_volknob_info, \ - .get = stac92xx_volknob_get, \ - .put = stac92xx_volknob_put, \ - .private_value = 127 | (knob_nid << 16), \ - } - static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -531,7 +484,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), - STAC_VOLKNOB(0x28), /* hardware gain controls */ HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), -- cgit v0.10.2 From dec44dbe017f824ce46cdd66b39fa9b98a076ce1 Mon Sep 17 00:00:00 2001 From: Kamalesh Babulal Date: Tue, 20 Nov 2007 15:12:33 +0100 Subject: [ALSA] powermac - Fix typos MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The kernel build fails, with following error CC sound/ppc/tumbler.o sound/ppc/tumbler.c: In function ‘snapper_get_capture_source’: sound/ppc/tumbler.c:812: error: ‘union ’ has no member named ‘value’ sound/ppc/tumbler.c: In function ‘snapper_put_capture_source’: sound/ppc/tumbler.c:824: error: ‘union ’ has no member named ‘enuemerated’ make[2]: *** [sound/ppc/tumbler.o] Error 1 make[1]: *** [sound/ppc] Error 2 make: *** [sound] Error 2 Signed-off-by: Kamalesh Babulal Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index bacff3d..d4d22e1 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -809,7 +809,7 @@ static int snapper_get_capture_source(struct snd_kcontrol *kcontrol, struct pmac_tumbler *mix = chip->mixer_data; snd_assert(mix, return -ENODEV); - ucontrol->value.enumerated.value[0] = mix->capture_source; + ucontrol->value.enumerated.item[0] = mix->capture_source; return 0; } @@ -821,7 +821,7 @@ static int snapper_put_capture_source(struct snd_kcontrol *kcontrol, int change; snd_assert(mix, return -ENODEV); - change = ucontrol->value.enuemerated.item[0] != mix->capture_source; + change = ucontrol->value.enumerated.item[0] != mix->capture_source; if (change) { mix->capture_source = !!ucontrol->value.enumerated.item[0]; snapper_set_capture_source(mix); -- cgit v0.10.2 From 4581aa36fc8273579f6b7979adc4f4e666460a8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Nov 2007 18:31:22 +0100 Subject: [ALSA] dbri - Fix broken change for value range checks The last patch for value range checks included a broken merge result. Now fixed properly. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index af1bf4b..fc68317 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2292,12 +2292,12 @@ static int snd_cs4215_put_volume(struct snd_kcontrol *kcontrol, return -EINVAL; } - if (info->left_gain != - info->left_gain = ucontrol->value.integer.value[0]; + if (info->left_gain != vol[0]) { + info->left_gain = vol[0]; changed = 1; } - if (info->right_gain != ucontrol->value.integer.value[1]) { - info->right_gain = ucontrol->value.integer.value[1]; + if (info->right_gain != vol[1]) { + info->right_gain = vol[1]; changed = 1; } if (changed) { -- cgit v0.10.2 From ee6df2e1849403b5805018193a1454f6d39f9498 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Nov 2007 18:32:08 +0100 Subject: [ALSA] caiaq - Fix indent in Kconfig Fix indent of caiaq in Kconfig to the same level as others. Just a tidy up. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 7061438..1806f67 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -31,11 +31,11 @@ config SND_USB_USX2Y config SND_USB_CAIAQ tristate "Native Instruments USB audio devices" - depends on SND && USB - select SND_HWDEP - select SND_RAWMIDI - select SND_PCM - help + depends on SND && USB + select SND_HWDEP + select SND_RAWMIDI + select SND_PCM + help Say Y here to include support for caiaq USB audio interfaces, namely: -- cgit v0.10.2 From b18b493f27bfb0d112a54ef24a1db32e898abff9 Mon Sep 17 00:00:00 2001 From: Dmitry Torokhov Date: Wed, 21 Nov 2007 16:45:23 +0100 Subject: [ALSA] caiaq - misc input handling fixes - link input device with its parent so that it placed in proper spot in sysfs hierarchy - drivers that allow changing their keymaps should use private copy of the keymap so that one instance of a device does not affect another instance - it is preferred for drivers to properly set up input_dev->phys to help userspace locate devices - drivers should use usb_to_input_id(), or perform endianess conversion, themselves, otherwise ID is not correct on big-endian boxes - whitespace and formatting cleanup Acked-by: Daniel Mack Signed-off-by: Dmitry Torokhov Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index 79bc5be..6857612 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -7,7 +7,7 @@ #define USB_PID_RIGKONTROL2 0x1969 #define USB_PID_RIGKONTROL3 0x1940 -#define USB_PID_KORECONTROLLER 0x4711 +#define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_AK1 0x0815 #define USB_PID_AUDIO8DJ 0x1978 @@ -62,7 +62,7 @@ struct snd_usb_caiaqdev { struct urb **data_urbs_in; struct urb **data_urbs_out; struct snd_usb_caiaq_cb_info *data_cb_info; - + unsigned char ep1_in_buf[EP1_BUFSIZE]; unsigned char ep1_out_buf[EP1_BUFSIZE]; unsigned char midi_out_buf[EP1_BUFSIZE]; @@ -72,7 +72,7 @@ struct snd_usb_caiaqdev { wait_queue_head_t ep1_wait_queue; wait_queue_head_t prepare_wait_queue; int spec_received, audio_parm_answer; - + char vendor_name[CAIAQ_USB_STR_LEN]; char product_name[CAIAQ_USB_STR_LEN]; char serial[CAIAQ_USB_STR_LEN]; @@ -93,8 +93,10 @@ struct snd_usb_caiaqdev { /* Linux input */ #ifdef CONFIG_SND_USB_CAIAQ_INPUT struct input_dev *input_dev; + char phys[64]; /* physical device path */ + unsigned short keycode[10]; #endif - + /* ALSA */ struct snd_pcm *pcm; struct snd_pcm_hardware pcm_info; diff --git a/sound/usb/caiaq/caiaq-input.c b/sound/usb/caiaq/caiaq-input.c index cd536ca..03bf4c6 100644 --- a/sound/usb/caiaq/caiaq-input.c +++ b/sound/usb/caiaq/caiaq-input.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -31,18 +32,18 @@ #ifdef CONFIG_SND_USB_CAIAQ_INPUT -static unsigned char keycode_ak1[] = { KEY_C, KEY_B, KEY_A }; -static unsigned char keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4, - KEY_5, KEY_6, KEY_7 }; -static unsigned char keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4, - KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 }; +static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A }; +static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4, + KEY_5, KEY_6, KEY_7 }; +static unsigned short keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4, + KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 }; -#define DEG90 (range/2) -#define DEG180 (range) -#define DEG270 (DEG90 + DEG180) -#define DEG360 (DEG180 * 2) -#define HIGH_PEAK (268) -#define LOW_PEAK (-7) +#define DEG90 (range / 2) +#define DEG180 (range) +#define DEG270 (DEG90 + DEG180) +#define DEG360 (DEG180 * 2) +#define HIGH_PEAK (268) +#define LOW_PEAK (-7) /* some of these devices have endless rotation potentiometers * built in which use two tapers, 90 degrees phase shifted. @@ -56,8 +57,8 @@ static unsigned int decode_erp(unsigned char a, unsigned char b) int range = HIGH_PEAK - LOW_PEAK; int mid_value = (HIGH_PEAK + LOW_PEAK) / 2; - weight_b = abs(mid_value-a) - (range/2 - 100)/2; - + weight_b = abs(mid_value - a) - (range / 2 - 100) / 2; + if (weight_b < 0) weight_b = 0; @@ -93,7 +94,7 @@ static unsigned int decode_erp(unsigned char a, unsigned char b) if (ret < 0) ret += 1000; - + if (ret >= 1000) ret -= 1000; @@ -108,76 +109,80 @@ static unsigned int decode_erp(unsigned char a, unsigned char b) #undef LOW_PEAK -static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, +static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, const unsigned char *buf, unsigned int len) { - switch(dev->input_dev->id.product) { + struct input_dev *input_dev = dev->input_dev; + + switch (input_dev->id.product) { case USB_PID_RIGKONTROL2: - input_report_abs(dev->input_dev, ABS_X, (buf[4] << 8) |buf[5]); - input_report_abs(dev->input_dev, ABS_Y, (buf[0] << 8) |buf[1]); - input_report_abs(dev->input_dev, ABS_Z, (buf[2] << 8) |buf[3]); - input_sync(dev->input_dev); + input_report_abs(input_dev, ABS_X, (buf[4] << 8) | buf[5]); + input_report_abs(input_dev, ABS_Y, (buf[0] << 8) | buf[1]); + input_report_abs(input_dev, ABS_Z, (buf[2] << 8) | buf[3]); + input_sync(input_dev); break; case USB_PID_RIGKONTROL3: - input_report_abs(dev->input_dev, ABS_X, (buf[0] << 8) |buf[1]); - input_report_abs(dev->input_dev, ABS_Y, (buf[2] << 8) |buf[3]); - input_report_abs(dev->input_dev, ABS_Z, (buf[4] << 8) |buf[5]); - input_sync(dev->input_dev); + input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]); + input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]); + input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); + input_sync(input_dev); break; } } -static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, +static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, const char *buf, unsigned int len) { + struct input_dev *input_dev = dev->input_dev; int i; - switch(dev->input_dev->id.product) { + switch (input_dev->id.product) { case USB_PID_AK1: i = decode_erp(buf[0], buf[1]); - input_report_abs(dev->input_dev, ABS_X, i); - input_sync(dev->input_dev); + input_report_abs(input_dev, ABS_X, i); + input_sync(input_dev); break; } } -static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, +static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, char *buf, unsigned int len) { + struct input_dev *input_dev = dev->input_dev; + unsigned short *keycode = input_dev->keycode; int i; - unsigned char *keycode = dev->input_dev->keycode; if (!keycode) return; - if (dev->input_dev->id.product == USB_PID_RIGKONTROL2) - for (i=0; iid.product == USB_PID_RIGKONTROL2) + for (i = 0; i < len; i++) buf[i] = ~buf[i]; - for (i=0; (iinput_dev->keycodemax) && (i < len); i++) - input_report_key(dev->input_dev, keycode[i], - buf[i/8] & (1 << (i%8))); + for (i = 0; i < input_dev->keycodemax && i < len; i++) + input_report_key(input_dev, keycode[i], + buf[i / 8] & (1 << (i % 8))); - input_sync(dev->input_dev); + input_sync(input_dev); } -void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *dev, - char *buf, +void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *dev, + char *buf, unsigned int len) { - if (!dev->input_dev || (len < 1)) + if (!dev->input_dev || len < 1) return; switch (buf[0]) { case EP1_CMD_READ_ANALOG: - snd_caiaq_input_read_analog(dev, buf+1, len-1); + snd_caiaq_input_read_analog(dev, buf + 1, len - 1); break; case EP1_CMD_READ_ERP: - snd_caiaq_input_read_erp(dev, buf+1, len-1); + snd_caiaq_input_read_erp(dev, buf + 1, len - 1); break; case EP1_CMD_READ_IO: - snd_caiaq_input_read_io(dev, buf+1, len-1); + snd_caiaq_input_read_io(dev, buf + 1, len - 1); break; } } @@ -192,37 +197,34 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) if (!input) return -ENOMEM; + usb_make_path(usb_dev, dev->phys, sizeof(dev->phys)); + strlcat(dev->phys, "/input0", sizeof(dev->phys)); + input->name = dev->product_name; - input->id.bustype = BUS_USB; - input->id.vendor = usb_dev->descriptor.idVendor; - input->id.product = usb_dev->descriptor.idProduct; - input->id.version = usb_dev->descriptor.bcdDevice; + input->phys = dev->phys; + usb_to_input_id(usb_dev, &input->id); + input->dev.parent = &usb_dev->dev; switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); input->absbit[0] = BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | BIT_MASK(ABS_Z); - input->keycode = keycode_rk2; - input->keycodesize = sizeof(char); + BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_rk2)); + memcpy(dev->keycode, keycode_rk2, sizeof(keycode_rk2)); input->keycodemax = ARRAY_SIZE(keycode_rk2); - for (i=0; ikeybit); - input_set_abs_params(input, ABS_X, 0, 4096, 0, 10); input_set_abs_params(input, ABS_Y, 0, 4096, 0, 10); input_set_abs_params(input, ABS_Z, 0, 4096, 0, 10); snd_usb_caiaq_set_auto_msg(dev, 1, 10, 0); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): input->evbit[0] = BIT(EV_KEY) | BIT(EV_ABS); input->absbit[0] = BIT(ABS_X) | BIT(ABS_Y) | BIT(ABS_Z); - input->keycode = keycode_rk3; - input->keycodesize = sizeof(char); + BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_rk3)); + memcpy(dev->keycode, keycode_rk3, sizeof(keycode_rk3)); input->keycodemax = ARRAY_SIZE(keycode_rk3); - for (i=0; ikeybit); - input_set_abs_params(input, ABS_X, 0, 1024, 0, 10); input_set_abs_params(input, ABS_Y, 0, 1024, 0, 10); input_set_abs_params(input, ABS_Z, 0, 1024, 0, 10); @@ -231,12 +233,9 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); input->absbit[0] = BIT_MASK(ABS_X); - input->keycode = keycode_ak1; - input->keycodesize = sizeof(char); + BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_ak1)); + memcpy(dev->keycode, keycode_ak1, sizeof(keycode_ak1)); input->keycodemax = ARRAY_SIZE(keycode_ak1); - for (i=0; ikeybit); - input_set_abs_params(input, ABS_X, 0, 999, 0, 10); snd_usb_caiaq_set_auto_msg(dev, 1, 0, 5); break; @@ -246,6 +245,11 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) return 0; } + input->keycode = dev->keycode; + input->keycodesize = sizeof(unsigned short); + for (i = 0; i < input->keycodemax; i++) + __set_bit(dev->keycode[i], input->keybit); + ret = input_register_device(input); if (ret < 0) { input_free_device(input); -- cgit v0.10.2 From e5f73e2ae813aa216b480728548e5ffbebcc170a Mon Sep 17 00:00:00 2001 From: Dmitry Torokhov Date: Wed, 21 Nov 2007 16:47:03 +0100 Subject: [ALSA] caiaq - input device support must depend on CONFIG_INPUT Signed-off-by: Dmitry Torokhov Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 1806f67..1c07c65 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -51,6 +51,7 @@ config SND_USB_CAIAQ config SND_USB_CAIAQ_INPUT bool "enable input device for controllers" depends on SND_USB_CAIAQ + depends on INPUT=y || INPUT=SND_USB_CAIAQ help Say Y here to support input controllers like buttons, knobs, alpha dials and analog pedals on the following products: -- cgit v0.10.2 From 8e3cd08ed8e590952aa9a656758cb24d4ba898f8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 22 Nov 2007 11:40:04 +0100 Subject: [ALSA] caiaq - add control API and more input features - added support for all input controllers on Native Instrument's 'Kore controller'. - added ALSA controls to switch LEDs on 'RigKontrol 2', 'RigKontrol3', 'Audio Kontrol 1' and 'Kore controller'. - added ALSA controls to switch input mode, software lock and ground lift features on 'Audio 8 DJ'. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 1c07c65..ccbf72e 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -58,6 +58,7 @@ config SND_USB_CAIAQ_INPUT * Native Instruments RigKontrol2 * Native Instruments RigKontrol3 + * Native Instruments Kore Controller * Native Instruments Audio Kontrol 1 endmenu diff --git a/sound/usb/caiaq/Makefile b/sound/usb/caiaq/Makefile index 455c8c5..345a965 100644 --- a/sound/usb/caiaq/Makefile +++ b/sound/usb/caiaq/Makefile @@ -1,3 +1,3 @@ -snd-usb-caiaq-objs := caiaq-device.o caiaq-audio.o caiaq-midi.o caiaq-input.o +snd-usb-caiaq-objs := caiaq-device.o caiaq-audio.o caiaq-midi.o caiaq-input.o caiaq-control.o obj-$(CONFIG_SND_USB_CAIAQ) += snd-usb-caiaq.o diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index 0666908..bf551c0 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -27,9 +27,7 @@ #include #include #include -#ifdef CONFIG_SND_USB_CAIAQ_INPUT #include -#endif #include "caiaq-device.h" #include "caiaq-audio.h" diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c new file mode 100644 index 0000000..14e8e32 --- /dev/null +++ b/sound/usb/caiaq/caiaq-control.c @@ -0,0 +1,315 @@ +/* + * Copyright (c) 2007 Daniel Mack + * friendly supported by NI. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "caiaq-device.h" +#include "caiaq-control.h" + +#define CNT_INTVAL 0x10000 + +static int control_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); + struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); + int pos = kcontrol->private_value; + int is_intval = pos & CNT_INTVAL; + + uinfo->count = 1; + pos &= ~CNT_INTVAL; + + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) + && (pos == 0)) { + /* current input mode of A8DJ */ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 2; + return 0; + } + + if (is_intval) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 64; + } else { + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + } + + return 0; +} + +static int control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); + struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); + int pos = kcontrol->private_value; + + if (pos & CNT_INTVAL) + ucontrol->value.integer.value[0] + = dev->control_state[pos & ~CNT_INTVAL]; + else + ucontrol->value.integer.value[0] + = !!(dev->control_state[pos / 8] & (1 << pos % 8)); + + return 0; +} + +static int control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); + struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); + int pos = kcontrol->private_value; + + if (pos & CNT_INTVAL) { + dev->control_state[pos & ~CNT_INTVAL] + = ucontrol->value.integer.value[0]; + snd_usb_caiaq_send_command(dev, EP1_CMD_DIMM_LEDS, + dev->control_state, sizeof(dev->control_state)); + } else { + if (ucontrol->value.integer.value[0]) + dev->control_state[pos / 8] |= 1 << (pos % 8); + else + dev->control_state[pos / 8] &= ~(1 << (pos % 8)); + + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + dev->control_state, sizeof(dev->control_state)); + } + + return 1; +} + +static struct snd_kcontrol_new kcontrol_template __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = control_info, + .get = control_get, + .put = control_put, + /* name and private_value filled later */ +}; + +struct caiaq_controller { + char *name; + int index; +}; + +static struct caiaq_controller ak1_controller[] = { + { "LED left", 2 }, + { "LED middle", 1 }, + { "LED right", 0 }, + { "LED ring", 3 } +}; + +static struct caiaq_controller rk2_controller[] = { + { "LED 1", 5 }, + { "LED 2", 4 }, + { "LED 3", 3 }, + { "LED 4", 2 }, + { "LED 5", 1 }, + { "LED 6", 0 }, + { "LED pedal", 6 }, + { "LED 7seg_1b", 8 }, + { "LED 7seg_1c", 9 }, + { "LED 7seg_2a", 10 }, + { "LED 7seg_2b", 11 }, + { "LED 7seg_2c", 12 }, + { "LED 7seg_2d", 13 }, + { "LED 7seg_2e", 14 }, + { "LED 7seg_2f", 15 }, + { "LED 7seg_2g", 16 }, + { "LED 7seg_3a", 17 }, + { "LED 7seg_3b", 18 }, + { "LED 7seg_3c", 19 }, + { "LED 7seg_3d", 20 }, + { "LED 7seg_3e", 21 }, + { "LED 7seg_3f", 22 }, + { "LED 7seg_3g", 23 } +}; + +static struct caiaq_controller rk3_controller[] = { + { "LED 7seg_1a", 0 + 0 }, + { "LED 7seg_1b", 0 + 1 }, + { "LED 7seg_1c", 0 + 2 }, + { "LED 7seg_1d", 0 + 3 }, + { "LED 7seg_1e", 0 + 4 }, + { "LED 7seg_1f", 0 + 5 }, + { "LED 7seg_1g", 0 + 6 }, + { "LED 7seg_1p", 0 + 7 }, + + { "LED 7seg_2a", 8 + 0 }, + { "LED 7seg_2b", 8 + 1 }, + { "LED 7seg_2c", 8 + 2 }, + { "LED 7seg_2d", 8 + 3 }, + { "LED 7seg_2e", 8 + 4 }, + { "LED 7seg_2f", 8 + 5 }, + { "LED 7seg_2g", 8 + 6 }, + { "LED 7seg_2p", 8 + 7 }, + + { "LED 7seg_3a", 16 + 0 }, + { "LED 7seg_3b", 16 + 1 }, + { "LED 7seg_3c", 16 + 2 }, + { "LED 7seg_3d", 16 + 3 }, + { "LED 7seg_3e", 16 + 4 }, + { "LED 7seg_3f", 16 + 5 }, + { "LED 7seg_3g", 16 + 6 }, + { "LED 7seg_3p", 16 + 7 }, + + { "LED 7seg_4a", 24 + 0 }, + { "LED 7seg_4b", 24 + 1 }, + { "LED 7seg_4c", 24 + 2 }, + { "LED 7seg_4d", 24 + 3 }, + { "LED 7seg_4e", 24 + 4 }, + { "LED 7seg_4f", 24 + 5 }, + { "LED 7seg_4g", 24 + 6 }, + { "LED 7seg_4p", 24 + 7 }, + + { "LED 1", 32 + 0 }, + { "LED 2", 32 + 1 }, + { "LED 3", 32 + 2 }, + { "LED 4", 32 + 3 }, + { "LED 5", 32 + 4 }, + { "LED 6", 32 + 5 }, + { "LED 7", 32 + 6 }, + { "LED 8", 32 + 7 }, + { "LED pedal", 32 + 8 } +}; + +static struct caiaq_controller kore_controller[] = { + { "LED F1", 8 | CNT_INTVAL }, + { "LED F2", 12 | CNT_INTVAL }, + { "LED F3", 0 | CNT_INTVAL }, + { "LED F4", 4 | CNT_INTVAL }, + { "LED F5", 11 | CNT_INTVAL }, + { "LED F6", 15 | CNT_INTVAL }, + { "LED F7", 3 | CNT_INTVAL }, + { "LED F8", 7 | CNT_INTVAL }, + { "LED touch1", 10 | CNT_INTVAL }, + { "LED touch2", 14 | CNT_INTVAL }, + { "LED touch3", 2 | CNT_INTVAL }, + { "LED touch4", 6 | CNT_INTVAL }, + { "LED touch5", 9 | CNT_INTVAL }, + { "LED touch6", 13 | CNT_INTVAL }, + { "LED touch7", 1 | CNT_INTVAL }, + { "LED touch8", 5 | CNT_INTVAL }, + { "LED left", 18 | CNT_INTVAL }, + { "LED right", 22 | CNT_INTVAL }, + { "LED up", 16 | CNT_INTVAL }, + { "LED down", 20 | CNT_INTVAL }, + { "LED stop", 23 | CNT_INTVAL }, + { "LED play", 21 | CNT_INTVAL }, + { "LED record", 19 | CNT_INTVAL }, + { "LED listen", 17 | CNT_INTVAL }, + { "LED lcd", 30 | CNT_INTVAL }, + { "LED menu", 28 | CNT_INTVAL }, + { "LED sound", 31 | CNT_INTVAL }, + { "LED esc", 29 | CNT_INTVAL }, + { "LED view", 27 | CNT_INTVAL }, + { "LED enter", 24 | CNT_INTVAL }, + { "LED control", 26 | CNT_INTVAL } +}; + +static struct caiaq_controller a8dj_controller[] = { + { "Current input mode", 0 | CNT_INTVAL }, + { "GND lift for TC Vinyl mode", 24 + 0 }, + { "GND lift for TC CD/Line mode", 24 + 1 }, + { "GND lift for phono mode", 24 + 2 }, + { "GND lift for TC Vinyl mode", 24 + 3 }, + { "Software lock", 40 } +}; + +int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) +{ + int i; + struct snd_kcontrol *kc; + + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): + for (i = 0; i < ARRAY_SIZE(ak1_controller); i++) { + struct caiaq_controller *c = ak1_controller + i; + kcontrol_template.name = c->name; + kcontrol_template.private_value = c->index; + kc = snd_ctl_new1(&kcontrol_template, dev); + snd_ctl_add(dev->chip.card, kc); + } + + break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): + for (i = 0; i < ARRAY_SIZE(rk2_controller); i++) { + struct caiaq_controller *c = rk2_controller + i; + kcontrol_template.name = c->name; + kcontrol_template.private_value = c->index; + kc = snd_ctl_new1(&kcontrol_template, dev); + snd_ctl_add(dev->chip.card, kc); + } + + break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): + for (i = 0; i < ARRAY_SIZE(rk3_controller); i++) { + struct caiaq_controller *c = rk3_controller + i; + kcontrol_template.name = c->name; + kcontrol_template.private_value = c->index; + kc = snd_ctl_new1(&kcontrol_template, dev); + snd_ctl_add(dev->chip.card, kc); + } + + break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + for (i = 0; i < ARRAY_SIZE(kore_controller); i++) { + struct caiaq_controller *c = kore_controller + i; + kcontrol_template.name = c->name; + kcontrol_template.private_value = c->index; + kc = snd_ctl_new1(&kcontrol_template, dev); + snd_ctl_add(dev->chip.card, kc); + } + + break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): + for (i = 0; i < ARRAY_SIZE(a8dj_controller); i++) { + struct caiaq_controller *c = a8dj_controller + i; + kcontrol_template.name = c->name; + kcontrol_template.private_value = c->index; + kc = snd_ctl_new1(&kcontrol_template, dev); + snd_ctl_add(dev->chip.card, kc); + } + + break; + } + + return 0; +} + diff --git a/sound/usb/caiaq/caiaq-control.h b/sound/usb/caiaq/caiaq-control.h new file mode 100644 index 0000000..2e7ab1a --- /dev/null +++ b/sound/usb/caiaq/caiaq-control.h @@ -0,0 +1,6 @@ +#ifndef CAIAQ_CONTROL_H +#define CAIAQ_CONTROL_H + +int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev); + +#endif /* CAIAQ_CONTROL_H */ diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 58af814..dc2e7f7 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -31,17 +31,19 @@ #include #include #include +#include #include "caiaq-device.h" #include "caiaq-audio.h" #include "caiaq-midi.h" +#include "caiaq-control.h" #ifdef CONFIG_SND_USB_CAIAQ_INPUT #include "caiaq-input.h" #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.2.0"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.0"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -140,14 +142,21 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) case EP1_CMD_MIDI_READ: snd_usb_caiaq_midi_handle_input(dev, buf[1], buf + 3, buf[2]); break; - + case EP1_CMD_READ_IO: + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)) { + if (urb->actual_length > sizeof(dev->control_state)) + urb->actual_length = sizeof(dev->control_state); + memcpy(dev->control_state, buf + 1, urb->actual_length); + wake_up(&dev->ep1_wait_queue); + break; + } #ifdef CONFIG_SND_USB_CAIAQ_INPUT case EP1_CMD_READ_ERP: case EP1_CMD_READ_ANALOG: - case EP1_CMD_READ_IO: snd_usb_caiaq_input_dispatch(dev, buf, urb->actual_length); - break; #endif + break; } dev->ep1_in_urb.actual_length = 0; @@ -156,10 +165,10 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) log("unable to submit urb. OOM!?\n"); } -static int send_command (struct snd_usb_caiaqdev *dev, - unsigned char command, - const unsigned char *buffer, - int len) +int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, + unsigned char command, + const unsigned char *buffer, + int len) { int actual_len; struct usb_device *usb_dev = dev->chip.dev; @@ -207,7 +216,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, rate, depth, bpp); dev->audio_parm_answer = -1; - ret = send_command(dev, EP1_CMD_AUDIO_PARAMS, tmp, sizeof(tmp)); + ret = snd_usb_caiaq_send_command(dev, EP1_CMD_AUDIO_PARAMS, + tmp, sizeof(tmp)); if (ret) return ret; @@ -226,7 +236,8 @@ int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; - return send_command(dev, EP1_CMD_AUTO_MSG, tmp, sizeof(tmp)); + return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, + tmp, sizeof(tmp)); } static void setup_card(struct snd_usb_caiaqdev *dev) @@ -241,7 +252,7 @@ static void setup_card(struct snd_usb_caiaqdev *dev) val[0] = 0x00; val[1] = 0x00; val[2] = 0x01; - send_command(dev, EP1_CMD_WRITE_IO, val, 3); + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 3); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): /* RigKontrol2 - display two centered dashes ('--') */ @@ -249,12 +260,34 @@ static void setup_card(struct snd_usb_caiaqdev *dev) val[1] = 0x40; val[2] = 0x40; val[3] = 0x00; - send_command(dev, EP1_CMD_WRITE_IO, val, 4); + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 4); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): /* Audio Kontrol 1 - make USB-LED stop blinking */ val[0] = 0x00; - send_command(dev, EP1_CMD_WRITE_IO, val, 1); + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 1); + break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): + /* Audio 8 DJ - trigger read of current settings */ + dev->control_state[0] = 0xff; + snd_usb_caiaq_set_auto_msg(dev, 1, 0, 0); + snd_usb_caiaq_send_command(dev, EP1_CMD_READ_IO, NULL, 0); + + if (!wait_event_timeout(dev->ep1_wait_queue, + dev->control_state[0] != 0xff, HZ)) + return; + + /* fix up some defaults */ + if ((dev->control_state[1] != 2) || + (dev->control_state[2] != 3) || + (dev->control_state[4] != 2)) { + dev->control_state[1] = 2; + dev->control_state[2] = 3; + dev->control_state[4] = 2; + snd_usb_caiaq_send_command(dev, + EP1_CMD_WRITE_IO, dev->control_state, 6); + } + break; } @@ -278,6 +311,10 @@ static void setup_card(struct snd_usb_caiaqdev *dev) log("snd_card_register() returned %d\n", ret); snd_card_free(dev->chip.card); } + + ret = snd_usb_caiaq_control_init(dev); + if (ret < 0) + log("Unable to set up control system (ret=%d)\n", ret); } static struct snd_card* create_card(struct usb_device* usb_dev) @@ -340,7 +377,7 @@ static int init_card(struct snd_usb_caiaqdev *dev) if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; - err = send_command(dev, EP1_CMD_GET_DEVICE_INFO, NULL, 0); + err = snd_usb_caiaq_send_command(dev, EP1_CMD_GET_DEVICE_INFO, NULL, 0); if (err) return err; diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index 6857612..310b10e 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -35,6 +35,7 @@ #define EP1_CMD_MIDI_WRITE 0x7 #define EP1_CMD_AUDIO_PARAMS 0x9 #define EP1_CMD_AUTO_MSG 0xb +#define EP1_CMD_DIMM_LEDS 0xc struct caiaq_device_spec { unsigned short fw_version; @@ -90,11 +91,14 @@ struct snd_usb_caiaqdev { struct snd_pcm_substream *sub_playback[MAX_STREAMS]; struct snd_pcm_substream *sub_capture[MAX_STREAMS]; + /* Controls */ + unsigned char control_state[64]; + /* Linux input */ #ifdef CONFIG_SND_USB_CAIAQ_INPUT struct input_dev *input_dev; char phys[64]; /* physical device path */ - unsigned short keycode[10]; + unsigned short keycode[64]; #endif /* ALSA */ @@ -114,6 +118,9 @@ struct snd_usb_caiaq_cb_info { int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, int rate, int depth, int bbp); int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, int digital, int analog, int erp); - +int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, + unsigned char command, + const unsigned char *buffer, + int len); #endif /* CAIAQ_DEVICE_H */ diff --git a/sound/usb/caiaq/caiaq-input.c b/sound/usb/caiaq/caiaq-input.c index 03bf4c6..00846b1 100644 --- a/sound/usb/caiaq/caiaq-input.c +++ b/sound/usb/caiaq/caiaq-input.c @@ -38,6 +38,41 @@ static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4, static unsigned short keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4, KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 }; +static unsigned short keycode_kore[] = { + KEY_FN_F1, /* "menu" */ + KEY_FN_F7, /* "lcd backlight */ + KEY_FN_F2, /* "control" */ + KEY_FN_F3, /* "enter" */ + KEY_FN_F4, /* "view" */ + KEY_FN_F5, /* "esc" */ + KEY_FN_F6, /* "sound" */ + KEY_FN_F8, /* array spacer, never triggered. */ + KEY_RIGHT, + KEY_DOWN, + KEY_UP, + KEY_LEFT, + KEY_SOUND, /* "listen" */ + KEY_RECORD, + KEY_PLAYPAUSE, + KEY_STOP, + BTN_4, /* 8 softkeys */ + BTN_3, + BTN_2, + BTN_1, + BTN_8, + BTN_7, + BTN_6, + BTN_5, + KEY_BRL_DOT4, /* touch sensitive knobs */ + KEY_BRL_DOT3, + KEY_BRL_DOT2, + KEY_BRL_DOT1, + KEY_BRL_DOT8, + KEY_BRL_DOT7, + KEY_BRL_DOT6, + KEY_BRL_DOT5 +}; + #define DEG90 (range / 2) #define DEG180 (range) #define DEG270 (DEG90 + DEG180) @@ -115,14 +150,20 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, { struct input_dev *input_dev = dev->input_dev; - switch (input_dev->id.product) { - case USB_PID_RIGKONTROL2: + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): input_report_abs(input_dev, ABS_X, (buf[4] << 8) | buf[5]); input_report_abs(input_dev, ABS_Y, (buf[0] << 8) | buf[1]); input_report_abs(input_dev, ABS_Z, (buf[2] << 8) | buf[3]); input_sync(input_dev); break; - case USB_PID_RIGKONTROL3: + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): + input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]); + input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]); + input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); + input_sync(input_dev); + break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]); input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]); input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); @@ -137,12 +178,31 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, struct input_dev *input_dev = dev->input_dev; int i; - switch (input_dev->id.product) { - case USB_PID_AK1: + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): i = decode_erp(buf[0], buf[1]); input_report_abs(input_dev, ABS_X, i); input_sync(input_dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + i = decode_erp(buf[7], buf[5]); + input_report_abs(input_dev, ABS_HAT0X, i); + i = decode_erp(buf[12], buf[14]); + input_report_abs(input_dev, ABS_HAT0Y, i); + i = decode_erp(buf[15], buf[13]); + input_report_abs(input_dev, ABS_HAT1X, i); + i = decode_erp(buf[0], buf[2]); + input_report_abs(input_dev, ABS_HAT1Y, i); + i = decode_erp(buf[3], buf[1]); + input_report_abs(input_dev, ABS_HAT2X, i); + i = decode_erp(buf[8], buf[10]); + input_report_abs(input_dev, ABS_HAT2Y, i); + i = decode_erp(buf[11], buf[9]); + input_report_abs(input_dev, ABS_HAT3X, i); + i = decode_erp(buf[4], buf[6]); + input_report_abs(input_dev, ABS_HAT3Y, i); + input_sync(input_dev); + break; } } @@ -160,10 +220,14 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, for (i = 0; i < len; i++) buf[i] = ~buf[i]; - for (i = 0; i < input_dev->keycodemax && i < len; i++) + for (i = 0; i < input_dev->keycodemax && i < len * 8; i++) input_report_key(input_dev, keycode[i], buf[i / 8] & (1 << (i % 8))); + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER)) + input_report_abs(dev->input_dev, ABS_MISC, 255 - buf[4]); + input_sync(input_dev); } @@ -218,10 +282,10 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) input_set_abs_params(input, ABS_Z, 0, 4096, 0, 10); snd_usb_caiaq_set_auto_msg(dev, 1, 10, 0); break; - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): - input->evbit[0] = BIT(EV_KEY) | BIT(EV_ABS); - input->absbit[0] = BIT(ABS_X) | BIT(ABS_Y) | BIT(ABS_Z); + input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); + input->absbit[0] = BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | + BIT_MASK(ABS_Z); BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_rk3)); memcpy(dev->keycode, keycode_rk3, sizeof(keycode_rk3)); input->keycodemax = ARRAY_SIZE(keycode_rk3); @@ -239,6 +303,32 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) input_set_abs_params(input, ABS_X, 0, 999, 0, 10); snd_usb_caiaq_set_auto_msg(dev, 1, 0, 5); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); + input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) | + BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) | + BIT_MASK(ABS_HAT2X) | BIT_MASK(ABS_HAT2Y) | + BIT_MASK(ABS_HAT3X) | BIT_MASK(ABS_HAT3Y) | + BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | + BIT_MASK(ABS_Z); + input->absbit[BIT_WORD(ABS_MISC)] |= BIT_MASK(ABS_MISC); + BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_kore)); + memcpy(dev->keycode, keycode_kore, sizeof(keycode_kore)); + input->keycodemax = ARRAY_SIZE(keycode_kore); + input_set_abs_params(input, ABS_HAT0X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT0Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT1X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT1Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT2X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT2Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT3X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT3Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_Y, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_Z, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_MISC, 0, 255, 0, 1); + snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + break; default: /* no input methods supported on this device */ input_free_device(input); -- cgit v0.10.2 From 7f6301cdfad5469312d266bcec3f1a02a8e8b5af Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Nov 2007 11:51:54 +0100 Subject: [ALSA] caiaq - remove ifdef Remove ifdef and fix Makefile for conditional builds. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/caiaq/Makefile b/sound/usb/caiaq/Makefile index 345a965..23dadd5 100644 --- a/sound/usb/caiaq/Makefile +++ b/sound/usb/caiaq/Makefile @@ -1,3 +1,4 @@ -snd-usb-caiaq-objs := caiaq-device.o caiaq-audio.o caiaq-midi.o caiaq-input.o caiaq-control.o +snd-usb-caiaq-y := caiaq-device.o caiaq-audio.o caiaq-midi.o caiaq-control.o +snd-usb-caiaq-$(CONFIG_SND_USB_CAIAQ_INPUT) += caiaq-input.o obj-$(CONFIG_SND_USB_CAIAQ) += snd-usb-caiaq.o diff --git a/sound/usb/caiaq/caiaq-input.c b/sound/usb/caiaq/caiaq-input.c index 00846b1..ded45e5 100644 --- a/sound/usb/caiaq/caiaq-input.c +++ b/sound/usb/caiaq/caiaq-input.c @@ -30,8 +30,6 @@ #include "caiaq-device.h" #include "caiaq-input.h" -#ifdef CONFIG_SND_USB_CAIAQ_INPUT - static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A }; static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4, KEY_5, KEY_6, KEY_7 }; @@ -359,5 +357,3 @@ void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev) dev->input_dev = NULL; } -#endif /* CONFIG_SND_USB_CAIAQ_INPUT */ - -- cgit v0.10.2 From 7c7fc2d44b7a660846115e65b67772b6742a14d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Nov 2007 13:14:23 +0100 Subject: [ALSA] Fix PCM MMAP time-stamp mode When MMAP time-stamp mode is given, it's supposed to update the time-stamp only at period boundary. However, it currently updates at each status call so this is just useless. The patch fixes this misbehavior. Also it fixes the wrong check of tstamp_mode (don't use bit-and for enum). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 806f1fb..93d7ca5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -148,8 +148,6 @@ static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substre pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) return pos; /* XRUN */ - if (runtime->tstamp_mode & SNDRV_PCM_TSTAMP_MMAP) - getnstimeofday((struct timespec *)&runtime->status->tstamp); #ifdef CONFIG_SND_DEBUG if (pos >= runtime->buffer_size) { snd_printk(KERN_ERR "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size); @@ -189,6 +187,8 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt; snd_pcm_sframes_t delta; + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP) + getnstimeofday((struct timespec *)&runtime->status->tstamp); pos = snd_pcm_update_hw_ptr_pos(substream, runtime); if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index fb3dde4..6245bda 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -595,7 +595,7 @@ int snd_pcm_status(struct snd_pcm_substream *substream, status->trigger_tstamp = runtime->trigger_tstamp; if (snd_pcm_running(substream)) { snd_pcm_update_hw_ptr(substream); - if (runtime->tstamp_mode & SNDRV_PCM_TSTAMP_MMAP) + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP) status->tstamp = runtime->status->tstamp; else getnstimeofday(&status->tstamp); -- cgit v0.10.2 From df86d1149f4b7b77158755b85b85f94e7bade32f Mon Sep 17 00:00:00 2001 From: Heikki Lindholm Date: Fri, 23 Nov 2007 15:37:48 +0100 Subject: [ALSA] add number of periods constraint to snd-aoa The aoa driver is not specifying constraints on number of periods, and, it seems, it might end with a non-integer number, which it cannot deal with. Fix by adding a proper constraint. Signed-off-by: Heikki Lindholm Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c index c6b42f9..a4b0e1a 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c @@ -194,6 +194,12 @@ static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in) hw->period_bytes_max = 16384; hw->periods_min = 3; hw->periods_max = MAX_DBDMA_COMMANDS; + err = snd_pcm_hw_constraint_integer(pi->substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + result = err; + goto out_unlock; + } list_for_each_entry(cii, &sdev->codec_list, list) { if (cii->codec->open) { err = cii->codec->open(cii, pi->substream); -- cgit v0.10.2 From 3d3909ffe57174ee10a2ba2cdd75a7c7383f1983 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Nov 2007 15:41:44 +0100 Subject: [ALSA] aoa - fix compile warning Set a proper error code in the error path of i2sbus_attach_codec(). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c index a4b0e1a..e6ffea9 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c @@ -996,6 +996,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, if (dev->pcm->card != card) { printk(KERN_ERR "Can't attach same bus to different cards!\n"); + err = -EINVAL; goto out_put_ci_module; } err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1); -- cgit v0.10.2 From 682fc0a5fe1a309a9a1860a0d50a2c6977d961cd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Nov 2007 08:45:43 +0100 Subject: [ALSA] emu10k1x - Add missing snd_card_set_dev call Added the missing snd_card_set_dev() call. This will fix the incomplete sysfs entry for this card. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 1ec7eba..0030d8b8 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1583,6 +1583,8 @@ static int __devinit snd_emu10k1x_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->port, chip->irq); + snd_card_set_dev(card, &pci->dev); + if ((err = snd_card_register(card)) < 0) { snd_card_free(card); return err; -- cgit v0.10.2 From 4f1895b0b5fbbd40299d85a07a6cd1742d1d1e1a Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Mon, 26 Nov 2007 08:44:15 +0100 Subject: [ALSA] sound/isa: Add missing 'space' Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 94daf83..bc0c379 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -390,7 +390,7 @@ static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, err = sc6000_init_mss(vport, config, vmss_port, mss_config); if (err < 0) { - snd_printk(KERN_ERR "Can not initialize" + snd_printk(KERN_ERR "Can not initialize " "Microsoft Sound System mode.\n"); return -ENODEV; } -- cgit v0.10.2 From d47ac4338301c373eca9d817ec9c6e33b069ac51 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Nov 2007 08:58:48 +0100 Subject: [ALSA] drivers - Add missing snd_card_set_dev() Added the missing call of snd_card_set_dev() in drivers/* Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index c76a24e..443104d 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1312,6 +1312,8 @@ static int __devinit snd_ml403_ac97cr_probe(struct platform_device *pfdev) (unsigned long)ml403_ac97cr->port, ml403_ac97cr->irq, ml403_ac97cr->capture_irq, dev + 1); + snd_card_set_dev(card, &pfdev->dev); + err = snd_card_register(card); if (err < 0) { snd_card_free(card); diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 68070cc..e12ba3d 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1008,6 +1008,8 @@ static int __devinit snd_mts64_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); + snd_card_set_dev(card, &pdev->dev); + /* At this point card will be usable */ if ((err = snd_card_register(card)) < 0) { snd_printd("Cannot register card\n"); diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 1b83287..7e7c668 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -797,6 +797,8 @@ static int __devinit snd_portman_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); + snd_card_set_dev(card, &pdev->dev); + /* At this point card will be usable */ if ((err = snd_card_register(card)) < 0) { snd_printd("Cannot register card\n"); -- cgit v0.10.2 From 389619f1063ed21cf237e2a8081be42e66d3c9a6 Mon Sep 17 00:00:00 2001 From: Ville Syrjala Date: Mon, 26 Nov 2007 08:58:24 +0100 Subject: [ALSA] soc/wm8731: Fix stereo mixer controls Disable the simultaneous load feature for the line in and headphone out volume registers. This allows left and right volume levels to be controlled separately. Signed-off-by: Ville Syrjala Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7ca0b52..57fb95a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -562,13 +562,13 @@ static int wm8731_init(struct snd_soc_device *socdev) /* set the update bits */ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); - wm8731_write(codec, WM8731_LOUT1V, reg | 0x0100); + wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); - wm8731_write(codec, WM8731_ROUT1V, reg | 0x0100); + wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); - wm8731_write(codec, WM8731_LINVOL, reg | 0x0100); + wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); - wm8731_write(codec, WM8731_RINVOL, reg | 0x0100); + wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); wm8731_add_controls(codec); wm8731_add_widgets(codec); -- cgit v0.10.2 From 7829d0eccbddc7431cc9af662c7cd3442b5598bd Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 26 Nov 2007 09:00:56 +0100 Subject: [ALSA] usb-caiaq - add support for Kore controller 2 Added support for Native Instrument's Kore controller 2. This device has no audio but MIDI, input devices and ALSA controllers only. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index ccbf72e..9351b8a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -42,6 +42,7 @@ config SND_USB_CAIAQ * Native Instruments RigKontrol2 * Native Instruments RigKontrol3 * Native Instruments Kore Controller + * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 * Native Instruments Audio 8 DJ @@ -59,6 +60,7 @@ config SND_USB_CAIAQ_INPUT * Native Instruments RigKontrol2 * Native Instruments RigKontrol3 * Native Instruments Kore Controller + * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 endmenu diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index 14e8e32..7d25f4b 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -288,6 +288,7 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): for (i = 0; i < ARRAY_SIZE(kore_controller); i++) { struct caiaq_controller *c = kore_controller + i; kcontrol_template.name = c->name; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index dc2e7f7..48c6bed 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -43,11 +43,12 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.0"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.1"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," + "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}" "{Native Instruments, Audio 8 DJ}}"); @@ -96,6 +97,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idProduct = USB_PID_KORECONTROLLER }, { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_KORECONTROLLER2 + }, + { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AK1 @@ -291,13 +297,21 @@ static void setup_card(struct snd_usb_caiaqdev *dev) break; } - ret = snd_usb_caiaq_audio_init(dev); - if (ret < 0) - log("Unable to set up audio system (ret=%d)\n", ret); + if (dev->spec.num_analog_audio_out + + dev->spec.num_analog_audio_in + + dev->spec.num_digital_audio_out + + dev->spec.num_digital_audio_in > 0) { + ret = snd_usb_caiaq_audio_init(dev); + if (ret < 0) + log("Unable to set up audio system (ret=%d)\n", ret); + } - ret = snd_usb_caiaq_midi_init(dev); - if (ret < 0) - log("Unable to set up MIDI system (ret=%d)\n", ret); + if (dev->spec.num_midi_in + + dev->spec.num_midi_out > 0) { + ret = snd_usb_caiaq_midi_init(dev); + if (ret < 0) + log("Unable to set up MIDI system (ret=%d)\n", ret); + } #ifdef CONFIG_SND_USB_CAIAQ_INPUT ret = snd_usb_caiaq_input_init(dev); diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index 310b10e..96a4913 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -8,6 +8,7 @@ #define USB_PID_RIGKONTROL2 0x1969 #define USB_PID_RIGKONTROL3 0x1940 #define USB_PID_KORECONTROLLER 0x4711 +#define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 #define USB_PID_AUDIO8DJ 0x1978 diff --git a/sound/usb/caiaq/caiaq-input.c b/sound/usb/caiaq/caiaq-input.c index ded45e5..e6c410a 100644 --- a/sound/usb/caiaq/caiaq-input.c +++ b/sound/usb/caiaq/caiaq-input.c @@ -162,6 +162,7 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, input_sync(input_dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]); input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]); input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); @@ -183,6 +184,7 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, input_sync(input_dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): i = decode_erp(buf[7], buf[5]); input_report_abs(input_dev, ABS_HAT0X, i); i = decode_erp(buf[12], buf[14]); @@ -223,7 +225,9 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, buf[i / 8] & (1 << (i % 8))); if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER)) + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER) || + dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2)) input_report_abs(dev->input_dev, ABS_MISC, 255 - buf[4]); input_sync(input_dev); @@ -302,6 +306,7 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) snd_usb_caiaq_set_auto_msg(dev, 1, 0, 5); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) | BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) | -- cgit v0.10.2 From 1005f66fd7175916db013dc98ca6b7cec26e5f81 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Nov 2007 15:27:17 +0100 Subject: [ALSA] ice1712 - Fix word clock status control on Delta 1010LT The 'Word Clock Status' control on Delta 1010LT checks the CS8427 error register too strictly and almost always returns 1 (unlocked). It should check only the lock status bit. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 371f784..519b5d4 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -405,7 +405,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco if (snd_i2c_sendbytes(ice->cs8427, ®, 1) != 1) snd_printk(KERN_ERR "unable to send register 0x%x byte to CS8427\n", reg); snd_i2c_readbytes(ice->cs8427, ®, 1); - ucontrol->value.integer.value[0] = (reg ? 1 : 0); + ucontrol->value.integer.value[0] = (reg & CS8427_UNLOCK) ? 1 : 0; return 0; } -- cgit v0.10.2 From df1deb67532ea26f1a033a5f48bf34b30bec8e1d Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Wed, 28 Nov 2007 11:58:56 +0100 Subject: [ALSA] sound/core/memalloc.c: Add missing pci_dev_put There should be a pci_dev_put when breaking out of a loop that iterates over calls to pci_get_device and similar functions. In this case, the return under the initial if needs a pci_dev_put in the same way that the return under the subsequent for loop has a pci_dev_put. This was fixed using the following semantic patch. // @@ type T; identifier d; expression e; @@ T *d; ... while ((d = \(pci_get_device\|pci_get_device_reverse\|pci_get_subsys\|pci_get_class\)(..., d)) != NULL) {... when != pci_dev_put(d) when != e = d ( return d; | + pci_dev_put(d); ? return ...; ) ...} // Signed-off-by: Julia Lawall Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 9b4992e..920e578 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -568,6 +568,7 @@ static ssize_t snd_mem_proc_write(struct file *file, const char __user * buffer, if (pci_set_dma_mask(pci, mask) < 0 || pci_set_consistent_dma_mask(pci, mask) < 0) { printk(KERN_ERR "snd-page-alloc: cannot set DMA mask %lx for pci %04x:%04x\n", mask, vendor, device); + pci_dev_put(pci); return count; } } -- cgit v0.10.2 From 109c53f840e551d6e99ecfd8b0131a968332c89f Mon Sep 17 00:00:00 2001 From: Rene Herman Date: Fri, 30 Nov 2007 17:59:25 +0100 Subject: [ALSA] sound/isa: kill pnp_resource_change This removes the pnp_resource_change use from the ALSA ISAPnP drivers. In 2.4 these were useful in providing an easy path to setting the resources, but in 2.6 they retain function as a layering violation only. This makes for a nice cleanup (-550 lines) of ALSA but moreover, ALSA is the only remaining user of pnp_init_resource_table(), pnp_resource_change() and pnp_manual_config_dev() (and, in fact, of 'struct pnp_resource_table') in the tree outide of drivers/pnp itself meaning it makes for more cleanup potential inside the PnP layer. Thomas Renninger acked their removal from that side, you did from the ALSA side (CC list just copied from that thread). Against current alsa-kernel HG. Many more potential cleanups in there, but this _only_ removes the pnp_resource_change code. Compile tested against current alsa-kernel HG and compile- and use-tested against 2.6.23.x (few offsets). Cc: Thomas Renninger Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index f5e77c7..23f4db0 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -150,13 +150,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. - port - port # for AD1816A chip (PnP setup) - mpu_port - port # for MPU-401 UART (PnP setup) - fm_port - port # for OPL3 (PnP setup) - irq - IRQ # for AD1816A chip (PnP setup) - mpu_irq - IRQ # for MPU-401 UART (PnP setup) - dma1 - first DMA # for AD1816A chip (PnP setup) - dma2 - second DMA # for AD1816A chip (PnP setup) clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz) This module supports multiple cards, autoprobe and PnP. @@ -203,14 +196,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. - port - port # for ALS100 (SB16) chip (PnP setup) - irq - IRQ # for ALS100 (SB16) chip (PnP setup) - dma8 - 8-bit DMA # for ALS100 (SB16) chip (PnP setup) - dma16 - 16-bit DMA # for ALS100 (SB16) chip (PnP setup) - mpu_port - port # for MPU-401 UART (PnP setup) - mpu_irq - IRQ # for MPU-401 (PnP setup) - fm_port - port # for OPL3 FM (PnP setup) - This module supports multiple cards, autoprobe and PnP. The power-management is supported. @@ -304,15 +289,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). - port - port # for AZT2320 chip (PnP setup) - wss_port - port # for WSS (PnP setup) - mpu_port - port # for MPU-401 UART (PnP setup) - fm_port - FM port # for AZT2320 chip (PnP setup) - irq - IRQ # for AZT2320 (WSS) chip (PnP setup) - mpu_irq - IRQ # for MPU-401 UART (PnP setup) - dma1 - 1st DMA # for AZT2320 (WSS) chip (PnP setup) - dma2 - 2nd DMA # for AZT2320 (WSS) chip (PnP setup) - This module supports multiple cards, PnP and autoprobe. The power-management is supported. @@ -505,13 +481,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP only) - port - Port # (PnP setup) - mpu_port - Port # for MPU-401 (PnP setup) - fm_port - Port # for FM OPL-3 (PnP setup) - irq - IRQ # (PnP setup) - mpu_irq - IRQ # for MPU-401 (PnP setup) - dma8 - DMA # (PnP setup) - This module supports multiple cards. This module is enabled only with ISA PnP support. @@ -609,10 +578,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on ESS ES968 chip (PnP only). - port - port # for ES968 (SB8) chip (PnP setup) - irq - IRQ # for ES968 (SB8) chip (PnP setup) - dma1 - DMA # for ES968 (SB8) chip (PnP setup) - This module supports multiple cards, PnP and autoprobe. The power-management is supported. diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index fc88a31..994bc85 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -61,20 +61,6 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for ad1816a based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable ad1816a based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for ad1816a driver."); -module_param_array(mpu_port, long, NULL, 0444); -MODULE_PARM_DESC(mpu_port, "MPU-401 port # for ad1816a driver."); -module_param_array(fm_port, long, NULL, 0444); -MODULE_PARM_DESC(fm_port, "FM port # for ad1816a driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for ad1816a driver."); -module_param_array(mpu_irq, int, NULL, 0444); -MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for ad1816a driver."); -module_param_array(dma1, int, NULL, 0444); -MODULE_PARM_DESC(dma1, "1st DMA # for ad1816a driver."); -module_param_array(dma2, int, NULL, 0444); -MODULE_PARM_DESC(dma2, "2nd DMA # for ad1816a driver."); module_param_array(clockfreq, int, NULL, 0444); MODULE_PARM_DESC(clockfreq, "Clock frequency for ad1816a driver (default = 0)."); @@ -117,16 +103,12 @@ static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acar const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -EBUSY; - } + acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL); if (acard->devmpu == NULL) { mpu_port[dev] = -1; @@ -134,25 +116,10 @@ static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acar } pdev = acard->dev; - pnp_init_resource_table(cfg); - - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], port[dev], 16); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "AUDIO PnP configure failure\n"); - kfree(cfg); return -EBUSY; } @@ -162,20 +129,11 @@ static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acar dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - if (acard->devmpu == NULL) { - kfree(cfg); + if (acard->devmpu == NULL) return 0; - } - pdev = acard->devmpu; - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); + pdev = acard->devmpu; - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "MPU401 PnP configure failure\n"); @@ -186,7 +144,6 @@ static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acar mpu_irq[dev] = pnp_irq(pdev, 0); } - kfree(cfg); return 0; } diff --git a/sound/isa/als100.c b/sound/isa/als100.c index f2bcfb2..35e25e5 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -63,20 +63,6 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for als100 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable als100 based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for als100 driver."); -module_param_array(mpu_port, long, NULL, 0444); -MODULE_PARM_DESC(mpu_port, "MPU-401 port # for als100 driver."); -module_param_array(fm_port, long, NULL, 0444); -MODULE_PARM_DESC(fm_port, "FM port # for als100 driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for als100 driver."); -module_param_array(mpu_irq, int, NULL, 0444); -MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for als100 driver."); -module_param_array(dma8, int, NULL, 0444); -MODULE_PARM_DESC(dma8, "8-bit DMA # for als100 driver."); -module_param_array(dma16, int, NULL, 0444); -MODULE_PARM_DESC(dma16, "16-bit DMA # for als100 driver."); struct snd_card_als100 { int dev_no; @@ -111,38 +97,20 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -ENODEV; - } + acard->devmpu = pnp_request_card_device(card, id->devs[1].id, acard->dev); acard->devopl = pnp_request_card_device(card, id->devs[2].id, acard->dev); pdev = acard->dev; - pnp_init_resource_table(cfg); - - /* override resources */ - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (dma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma8[dev], 1); - if (dma16[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma16[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - kfree(cfg); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -152,13 +120,6 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, pdev = acard->devmpu; if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "MPU401 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) goto __mpu_error; @@ -176,11 +137,6 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, pdev = acard->devopl; if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], fm_port[dev], 4); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "OPL3 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) goto __fm_error; @@ -195,7 +151,6 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, fm_port[dev] = -1; } - kfree(cfg); return 0; } diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index b615538..bfe3a25 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -72,22 +72,6 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for azt2320 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable azt2320 based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for azt2320 driver."); -module_param_array(wss_port, long, NULL, 0444); -MODULE_PARM_DESC(wss_port, "WSS Port # for azt2320 driver."); -module_param_array(mpu_port, long, NULL, 0444); -MODULE_PARM_DESC(mpu_port, "MPU-401 port # for azt2320 driver."); -module_param_array(fm_port, long, NULL, 0444); -MODULE_PARM_DESC(fm_port, "FM port # for azt2320 driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for azt2320 driver."); -module_param_array(mpu_irq, int, NULL, 0444); -MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for azt2320 driver."); -module_param_array(dma1, int, NULL, 0444); -MODULE_PARM_DESC(dma1, "1st DMA # for azt2320 driver."); -module_param_array(dma2, int, NULL, 0444); -MODULE_PARM_DESC(dma2, "2nd DMA # for azt2320 driver."); struct snd_card_azt2320 { int dev_no; @@ -121,43 +105,19 @@ static int __devinit snd_card_azt2320_pnp(int dev, struct snd_card_azt2320 *acar const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -ENODEV; - } acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL); pdev = acard->dev; - pnp_init_resource_table(cfg); - - /* override resources */ - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (wss_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], wss_port[dev], 4); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - kfree(cfg); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -169,13 +129,6 @@ static int __devinit snd_card_azt2320_pnp(int dev, struct snd_card_azt2320 *acar pdev = acard->devmpu; if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "MPU401 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) goto __mpu_error; @@ -191,7 +144,6 @@ static int __devinit snd_card_azt2320_pnp(int dev, struct snd_card_azt2320 *acar mpu_port[dev] = -1; } - kfree (cfg); return 0; } diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index f471f8a..c166e13 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -286,39 +286,21 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; acard->cap = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->cap == NULL) { - kfree(cfg); + if (acard->cap == NULL) return -EBUSY; - } + acard->play = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->play == NULL) { - kfree(cfg); + if (acard->play == NULL) return -EBUSY; - } pdev = acard->cap; - pnp_init_resource_table(cfg); - /* allocate AD1848 resources */ - if (wssport[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], wssport[dev], 8); - if (wssdma[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], wssdma[dev], 1); - if (wssirq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], wssirq[dev], 1); - - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP manual resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); - kfree(cfg); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); @@ -327,23 +309,10 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, /* allocate SB16 resources */ pdev = acard->play; - pnp_init_resource_table(cfg); - if (sbport[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], sbport[dev], 16); - if (sbdma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], sbdma8[dev], 1); - if (sbdma16[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], sbdma16[dev], 1); - if (sbirq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], sbirq[dev], 1); - - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR "CMI8330/C3D (SB16) PnP manual resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "CMI8330/C3D (SB16) PnP configure failure\n"); - kfree(cfg); return -EBUSY; } sbport[dev] = pnp_port_start(pdev, 0); @@ -351,7 +320,6 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); - kfree(cfg); return 0; } #endif diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 5784b43..0a3fece 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -270,29 +270,9 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids); /* WSS initialization */ -static int __devinit snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev, - struct pnp_resource_table *cfg) +static int __devinit snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev) { - int err; - - pnp_init_resource_table(cfg); - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 4); - if (fm_port[dev] != SNDRV_AUTO_PORT && fm_port[dev] > 0) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (sb_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], sb_port[dev], 16); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev] < 0 ? 4 : dma2[dev], 1); - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR IDENT " WSS PnP manual resources are invalid, using auto config\n"); - err = pnp_activate_dev(pdev); - if (err < 0) { + if (pnp_activate_dev(pdev) < 0) { printk(KERN_ERR IDENT " WSS PnP configure failed for WSS (out of resources?)\n"); return -EBUSY; } @@ -311,19 +291,9 @@ static int __devinit snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev, } /* CTRL initialization */ -static int __devinit snd_cs423x_pnp_init_ctrl(int dev, struct pnp_dev *pdev, - struct pnp_resource_table *cfg) +static int __devinit snd_cs423x_pnp_init_ctrl(int dev, struct pnp_dev *pdev) { - int err; - - pnp_init_resource_table(cfg); - if (cport[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], cport[dev], 8); - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR IDENT " CTRL PnP manual resources are invalid, using auto config\n"); - err = pnp_activate_dev(pdev); - if (err < 0) { + if (pnp_activate_dev(pdev) < 0) { printk(KERN_ERR IDENT " CTRL PnP configure failed for WSS (out of resources?)\n"); return -EBUSY; } @@ -333,21 +303,9 @@ static int __devinit snd_cs423x_pnp_init_ctrl(int dev, struct pnp_dev *pdev, } /* MPU initialization */ -static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev, - struct pnp_resource_table *cfg) +static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) { - int err; - - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] >= 0) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR IDENT " MPU401 PnP manual resources are invalid, using auto config\n"); - err = pnp_activate_dev(pdev); - if (err < 0) { + if (pnp_activate_dev(pdev) < 0) { printk(KERN_ERR IDENT " MPU401 PnP configure failed for WSS (out of resources?)\n"); mpu_port[dev] = SNDRV_AUTO_PORT; mpu_irq[dev] = SNDRV_AUTO_IRQ; @@ -368,15 +326,8 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev, static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, struct pnp_dev *pdev) { - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); - - if (!cfg) - return -ENOMEM; - if (snd_cs423x_pnp_init_wss(dev, acard->wss, cfg) < 0) { - kfree(cfg); + if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; - } - kfree(cfg); cport[dev] = -1; return 0; } @@ -386,43 +337,33 @@ static int __devinit snd_card_cs423x_pnpc(int dev, struct snd_card_cs4236 *acard struct pnp_card_link *card, const struct pnp_card_device_id *id) { - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); - - if (!cfg) - return -ENOMEM; - acard->wss = pnp_request_card_device(card, id->devs[0].id, NULL); if (acard->wss == NULL) - goto error; + return -EBUSY; acard->ctrl = pnp_request_card_device(card, id->devs[1].id, NULL); if (acard->ctrl == NULL) - goto error; + return -EBUSY; if (id->devs[2].id[0]) { acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); if (acard->mpu == NULL) - goto error; + return -EBUSY; } /* WSS initialization */ - if (snd_cs423x_pnp_init_wss(dev, acard->wss, cfg) < 0) - goto error; + if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) + return -EBUSY; /* CTRL initialization */ if (acard->ctrl && cport[dev] > 0) { - if (snd_cs423x_pnp_init_ctrl(dev, acard->ctrl, cfg) < 0) - goto error; + if (snd_cs423x_pnp_init_ctrl(dev, acard->ctrl) < 0) + return -EBUSY; } /* MPU initialization */ if (acard->mpu && mpu_port[dev] > 0) { - if (snd_cs423x_pnp_init_mpu(dev, acard->mpu, cfg) < 0) - goto error; + if (snd_cs423x_pnp_init_mpu(dev, acard->mpu) < 0) + return -EBUSY; } - kfree(cfg); return 0; - - error: - kfree(cfg); - return -EBUSY; } #endif /* CONFIG_PNP */ diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c index ce57d52..ab689f9 100644 --- a/sound/isa/dt019x.c +++ b/sound/isa/dt019x.c @@ -56,18 +56,6 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for DT-019X based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable DT-019X based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for dt019x driver."); -module_param_array(mpu_port, long, NULL, 0444); -MODULE_PARM_DESC(mpu_port, "MPU-401 port # for dt019x driver."); -module_param_array(fm_port, long, NULL, 0444); -MODULE_PARM_DESC(fm_port, "FM port # for dt019x driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for dt019x driver."); -module_param_array(mpu_irq, int, NULL, 0444); -MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for dt019x driver."); -module_param_array(dma8, int, NULL, 0444); -MODULE_PARM_DESC(dma8, "8-bit DMA # for dt019x driver."); struct snd_card_dt019x { struct pnp_dev *dev; @@ -95,36 +83,20 @@ static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard, const struct pnp_card_device_id *pid) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; - acard->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree (cfg); + if (acard->dev == NULL) return -ENODEV; - } + acard->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); acard->devopl = pnp_request_card_device(card, pid->devs[2].id, NULL); pdev = acard->dev; - pnp_init_resource_table(cfg); - - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (dma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma8[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "DT-019X AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "DT-019X AUDIO pnp configure failure\n"); - kfree(cfg); return err; } @@ -135,15 +107,7 @@ static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard, port[dev],irq[dev],dma8[dev]); pdev = acard->devmpu; - if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "DT-019X MPU401 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { pnp_release_card_device(pdev); @@ -162,11 +126,6 @@ static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard, pdev = acard->devopl; if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], fm_port[dev], 4); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "DT-019X OPL3 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { pnp_release_card_device(pdev); @@ -181,7 +140,6 @@ static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard, fm_port[dev] = -1; } - kfree(cfg); return 0; } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index c1af28f..5d4f163 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2035,31 +2035,9 @@ static struct pnp_device_id snd_audiodrive_pnpbiosids[] = { MODULE_DEVICE_TABLE(pnp, snd_audiodrive_pnpbiosids); /* PnP main device initialization */ -static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev, - struct pnp_resource_table *cfg) +static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev) { - int err; - - pnp_init_resource_table(cfg); - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], mpu_port[dev], 2); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (pnp_device_is_isapnp(pdev)) { - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR PFX "PnP manual resources are invalid, using auto config\n"); - } - err = pnp_activate_dev(pdev); - if (err < 0) { + if (pnp_activate_dev(pdev) < 0) { snd_printk(KERN_ERR PFX "PnP configure failure (out of resources?)\n"); return -EBUSY; } @@ -2087,16 +2065,9 @@ static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev, static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard, struct pnp_dev *pdev) { - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); - - if (!cfg) - return -ENOMEM; acard->dev = pdev; - if (snd_audiodrive_pnp_init_main(dev, acard->dev, cfg) < 0) { - kfree(cfg); + if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) return -EBUSY; - } - kfree(cfg); return 0; } @@ -2125,33 +2096,24 @@ static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, struct pnp_card_link *card, const struct pnp_card_device_id *id) { - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); - - if (!cfg) - return -ENOMEM; acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -EBUSY; - } + acard->devc = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->devc == NULL) { - kfree(cfg); + if (acard->devc == NULL) return -EBUSY; - } + /* Control port initialization */ if (pnp_activate_dev(acard->devc) < 0) { - kfree(cfg); snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n"); return -EAGAIN; } snd_printdd("pnp: port=0x%llx\n", (unsigned long long)pnp_port_start(acard->devc, 0)); - if (snd_audiodrive_pnp_init_main(dev, acard->dev, cfg) < 0) { - kfree(cfg); + if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) return -EBUSY; - } - kfree(cfg); + return 0; } #endif /* CONFIG_PNP */ diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 2091c50..9381d1e 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -560,50 +560,27 @@ static int __devinit snd_interwave_pnp(int dev, struct snd_interwave *iwcard, const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; iwcard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (iwcard->dev == NULL) { - kfree(cfg); + if (iwcard->dev == NULL) return -EBUSY; - } + #ifdef SNDRV_STB iwcard->devtc = pnp_request_card_device(card, id->devs[1].id, NULL); - if (iwcard->devtc == NULL) { - kfree(cfg); + if (iwcard->devtc == NULL) return -EBUSY; - } #endif /* Synth & Codec initialization */ pdev = iwcard->dev; - pnp_init_resource_table(cfg); - if (port[dev] != SNDRV_AUTO_PORT) { - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - pnp_resource_change(&cfg->port_resource[1], port[dev] + 0x100, 12); - pnp_resource_change(&cfg->port_resource[2], port[dev] + 0x10c, 4); - } - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (dma2[dev] < 0) - pnp_resource_change(&cfg->dma_resource[1], 4, 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "InterWave - Synth - the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { - kfree(cfg); snd_printk(KERN_ERR "InterWave PnP configure failure (out of resources?)\n"); return err; } if (pnp_port_start(pdev, 0) + 0x100 != pnp_port_start(pdev, 1) || pnp_port_start(pdev, 0) + 0x10c != pnp_port_start(pdev, 2)) { - kfree(cfg); snd_printk(KERN_ERR "PnP configure failure (wrong ports)\n"); return -ENOENT; } @@ -620,21 +597,15 @@ static int __devinit snd_interwave_pnp(int dev, struct snd_interwave *iwcard, #ifdef SNDRV_STB /* Tone Control initialization */ pdev = iwcard->devtc; - pnp_init_resource_table(cfg); - if (port_tc[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port_tc[dev], 1); - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "InterWave - ToneControl - the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { - kfree(cfg); snd_printk(KERN_ERR "InterWave ToneControl PnP configure failure (out of resources?)\n"); return err; } port_tc[dev] = pnp_port_start(pdev, 0); snd_printdd("isapnp IW: tone control port=0x%lx\n", port_tc[dev]); #endif - kfree(cfg); return 0; } #endif /* CONFIG_PNP */ diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 59af9ab..125f699 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -610,39 +610,8 @@ static int snd_opl3sa2_resume(struct snd_card *card) static int __devinit snd_opl3sa2_pnp(int dev, struct snd_opl3sa2 *chip, struct pnp_dev *pdev) { - struct pnp_resource_table * cfg; - int err; - - cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); - if (!cfg) { - snd_printk(KERN_ERR PFX "cannot allocate pnp cfg\n"); - return -ENOMEM; - } - /* PnP initialization */ - pnp_init_resource_table(cfg); - if (sb_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], sb_port[dev], 16); - if (wss_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], wss_port[dev], 8); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], fm_port[dev], 4); - if (midi_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[3], midi_port[dev], 2); - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[4], port[dev], 2); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_WARNING "PnP manual resources are invalid, using auto config\n"); - err = pnp_activate_dev(pdev); - if (err < 0) { - kfree(cfg); - snd_printk(KERN_ERR "PnP configure failure (out of resources?) err = %d\n", err); + if (pnp_activate_dev(pdev) < 0) { + snd_printk(KERN_ERR "PnP configure failure (out of resources?)\n"); return -EBUSY; } sb_port[dev] = pnp_port_start(pdev, 0); @@ -657,7 +626,6 @@ static int __devinit snd_opl3sa2_pnp(int dev, struct snd_opl3sa2 *chip, pnp_device_is_pnpbios(pdev) ? "BIOS" : "ISA", sb_port[dev], wss_port[dev], fm_port[dev], midi_port[dev]); snd_printdd("%sPnP OPL3-SA: control port=0x%lx, dma1=%i, dma2=%i, irq=%i\n", pnp_device_is_pnpbios(pdev) ? "BIOS" : "ISA", port[dev], dma1[dev], dma2[dev], irq[dev]); - kfree(cfg); return 0; } #endif /* CONFIG_PNP */ diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index ee1a824..9300cf3 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -1690,53 +1690,19 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, const struct pnp_card_device_id *pid) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; chip->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (chip->dev == NULL) { - kfree(cfg); + if (chip->dev == NULL) return -EBUSY; - } + chip->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); pdev = chip->dev; - pnp_init_resource_table(cfg); -#ifdef OPTi93X - if (port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port + 4, 4); -#else - if (pid->driver_data != 0x0924 && port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], port, 4); -#endif /* OPTi93X */ - if (irq != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq, 1); - if (dma1 != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1, 1); -#if defined(CS4231) || defined(OPTi93X) - if (dma2 != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2, 1); -#else -#ifdef snd_opti9xx_fixup_dma2 - snd_opti9xx_fixup_dma2(pdev); -#endif -#endif /* CS4231 || OPTi93X */ -#ifdef OPTi93X - if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port, 4); -#else - if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], fm_port, 4); -#endif - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); - kfree(cfg); return err; } @@ -1756,15 +1722,6 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, pdev = chip->devmpu; if (pdev && mpu_port > 0) { - pnp_init_resource_table(cfg); - - if (mpu_port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port, 2); - if (mpu_irq != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq, 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure\n"); @@ -1775,7 +1732,6 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, mpu_irq = pnp_irq(pdev, 0); } } - kfree(cfg); return pid->driver_data; } #endif /* CONFIG_PNP */ diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c index d4b2187..3049692 100644 --- a/sound/isa/sb/es968.c +++ b/sound/isa/sb/es968.c @@ -49,12 +49,6 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for es968 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable es968 based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for es968 driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for es968 driver."); -module_param_array(dma8, int, NULL, 0444); -MODULE_PARM_DESC(dma8, "8-bit DMA # for es968 driver."); struct snd_card_es968 { struct pnp_dev *dev; @@ -86,40 +80,23 @@ static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard, const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; + acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -ENODEV; - } pdev = acard->dev; - pnp_init_resource_table(cfg); - - /* override resources */ - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (dma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma8[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - kfree(cfg); return err; } port[dev] = pnp_port_start(pdev, 0); dma8[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - kfree(cfg); return 0; } diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index e7f9edd..8e1aff7 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -257,44 +257,21 @@ static int __devinit snd_card_sb16_pnp(int dev, struct snd_card_sb16 *acard, const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -ENODEV; - } + #ifdef SNDRV_SBAWE_EMU8000 acard->devwt = pnp_request_card_device(card, id->devs[1].id, acard->dev); #endif /* Audio initialization */ pdev = acard->dev; - pnp_init_resource_table(cfg); - - /* override resources */ - - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], mpu_port[dev], 2); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], fm_port[dev], 4); - if (dma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma8[dev], 1); - if (dma16[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma16[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - kfree(cfg); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -311,17 +288,6 @@ static int __devinit snd_card_sb16_pnp(int dev, struct snd_card_sb16 *acard, /* WaveTable initialization */ pdev = acard->devwt; if (pdev != NULL) { - pnp_init_resource_table(cfg); - - /* override resources */ - - if (awe_port[dev] != SNDRV_AUTO_PORT) { - pnp_resource_change(&cfg->port_resource[0], awe_port[dev], 4); - pnp_resource_change(&cfg->port_resource[1], awe_port[dev] + 0x400, 4); - pnp_resource_change(&cfg->port_resource[2], awe_port[dev] + 0x800, 4); - } - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "WaveTable the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { goto __wt_error; @@ -339,7 +305,6 @@ __wt_error: awe_port[dev] = -1; } #endif - kfree(cfg); return 0; } diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 83c2fc4..15c6046 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -104,21 +104,15 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; - /* Check for each logical device. */ /* CS4232 chip (aka "windows sound system") is logical device 0 */ acard->wss = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->wss == NULL) { - kfree(cfg); + if (acard->wss == NULL) return -EBUSY; - } /* there is a game port at logical device 1, but we ignore it completely */ @@ -133,26 +127,20 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c if (use_cs4232_midi[dev]) { acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); - if (acard->mpu == NULL) { - kfree(cfg); + if (acard->mpu == NULL) return -EBUSY; - } } /* The ICS2115 synth is logical device 4 */ acard->synth = pnp_request_card_device(card, id->devs[3].id, NULL); - if (acard->synth == NULL) { - kfree(cfg); + if (acard->synth == NULL) return -EBUSY; - } /* PCM/FM initialization */ pdev = acard->wss; - pnp_init_resource_table(cfg); - /* An interesting note from the Tropez+ FAQ: Q. [Ports] Why is the base address of the WSS I/O ports off by 4? @@ -165,23 +153,9 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c */ - if (cs4232_pcm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], cs4232_pcm_port[dev], 4); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (cs4232_pcm_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], cs4232_pcm_irq[dev], 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "PnP WSS the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "PnP WSS pnp configure failure\n"); - kfree(cfg); return err; } @@ -195,22 +169,9 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c pdev = acard->synth; - pnp_init_resource_table(cfg); - - if (ics2115_port[dev] != SNDRV_AUTO_PORT) { - pnp_resource_change(&cfg->port_resource[0], ics2115_port[dev], 16); - } - - if (ics2115_port[dev] != SNDRV_AUTO_IRQ) { - pnp_resource_change(&cfg->irq_resource[0], ics2115_irq[dev], 1); - } - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "PnP ICS2115 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "PnP ICS2115 pnp configure failure\n"); - kfree(cfg); return err; } @@ -226,15 +187,6 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c pdev = acard->mpu; - pnp_init_resource_table(cfg); - - if (cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], cs4232_mpu_port[dev], 2); - if (cs4232_mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->port_resource[0], cs4232_mpu_irq[dev], 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "PnP MPU401 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "PnP MPU401 pnp configure failure\n"); @@ -258,7 +210,6 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c ics2115_port[dev], ics2115_irq[dev]); - kfree(cfg); return 0; } -- cgit v0.10.2 From acec30ffffe1eee07e3202cff03e7ca8350d250f Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Mon, 3 Dec 2007 12:37:17 +0100 Subject: [ALSA] I2C fix for ice1724 adding i2c busy wait before sending device address to prevent reading bogus data. Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 357bdbe..b7a1d93 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1958,6 +1958,7 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice, unsigned char val; mutex_lock(&ice->i2c_mutex); + wait_i2c_busy(ice); outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR)); outb(dev & ~VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR)); wait_i2c_busy(ice); -- cgit v0.10.2 From 6632d64b0b596b9588b607806ac6d36c8c2c9696 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Mon, 3 Dec 2007 12:44:28 +0100 Subject: [ALSA] switching rate in STAC9460 codec of Prodigy192 * support for switching rate in STAC9460 - using set_rate_val of the akm infrastructure * listing all STAC9460 registers in proc * disabling mpu401 device for Prodigy192 - otherwise the currently flawed mpu401 code hangs kernel when opening the midi device * removing old unused commented-out code Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h index 891cf1a..6153b91 100644 --- a/include/sound/ak4xxx-adda.h +++ b/include/sound/ak4xxx-adda.h @@ -68,7 +68,7 @@ struct snd_akm4xxx { enum { SND_AK4524, SND_AK4528, SND_AK4529, SND_AK4355, SND_AK4358, SND_AK4381, - SND_AK5365 + SND_AK5365, NON_AKM } type; /* (array) information of combined codecs */ diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 39bb03a..fefa1ae 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -293,6 +293,11 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) case SND_AK5365: /* FIXME: any init sequence? */ return; + case NON_AKM: + /* fake value for non-akm codecs using akm infrastructure + * (e.g. of ice1724) - certainly FIXME + */ + return; default: snd_BUG(); return; diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 3c3cac3..4dc576a 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -399,6 +399,8 @@ struct snd_ice1712 { } juli; struct { struct ak4114 *ak4114; + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; } prodigy192; struct { struct { diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 4b21d5c..6d81a1c 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -81,6 +81,24 @@ static inline unsigned char stac9460_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ + +/* + * idx = STAC9460 volume register number, mute: 0 = mute, 1 = unmute + */ +static int stac9460_dac_mute(struct snd_ice1712 *ice, int idx, + unsigned char mute) +{ + unsigned char new, old; + int change; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) + /*printk ("Volume register 0x%02x: 0x%02x\n", idx, new);*/ + stac9460_put(ice, idx, new); + return change; +} + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -101,20 +119,18 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - unsigned char new, old; - int idx; - int change; + int idx, change; if (kcontrol->private_value) idx = STAC946X_MASTER_VOLUME; else idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME; - old = stac9460_get(ice, idx); - new = (~ucontrol->value.integer.value[0]<< 7 & 0x80) | (old & ~0x80); - change = (new != old); - if (change) - stac9460_put(ice, idx, new); - + /* due to possible conflicts with stac9460_set_rate_val, mutexing */ + mutex_lock(&ice->spec.prodigy192.mute_mutex); + /*printk("Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx, + ucontrol->value.integer.value[0]);*/ + change = stac9460_dac_mute(ice, idx, ucontrol->value.integer.value[0]); + mutex_unlock(&ice->spec.prodigy192.mute_mutex); return change; } @@ -162,6 +178,8 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el ovol = 0x7f - (tmp & 0x7f); change = (ovol != nvol); if (change) { + ovol = (0x7f - nvol) | (tmp & 0x80); + /*printk("DAC Volume: reg 0x%02x: 0x%02x\n", idx, ovol);*/ stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80)); } return change; @@ -251,121 +269,6 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el return change; } -#if 0 -/* - * Headphone Amplifier - */ -static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable) -{ - unsigned int tmp, tmp2; - - tmp2 = tmp = snd_ice1712_gpio_read(ice); - if (enable) - tmp |= AUREON_HP_SEL; - else - tmp &= ~ AUREON_HP_SEL; - if (tmp != tmp2) { - snd_ice1712_gpio_write(ice, tmp); - return 1; - } - return 0; -} - -static int aureon_get_headphone_amp(struct snd_ice1712 *ice) -{ - unsigned int tmp = snd_ice1712_gpio_read(ice); - - return ( tmp & AUREON_HP_SEL )!= 0; -} - -#define aureon_bool_info snd_ctl_boolean_mono_info - -static int aureon_hpamp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - ucontrol->value.integer.value[0] = aureon_get_headphone_amp(ice); - return 0; -} - - -static int aureon_hpamp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - return aureon_set_headphone_amp(ice,ucontrol->value.integer.value[0]); -} - -/* - * Deemphasis - */ -static int aureon_deemp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = (wm_get(ice, WM_DAC_CTRL2) & 0xf) == 0xf; - return 0; -} - -static int aureon_deemp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int temp, temp2; - temp2 = temp = wm_get(ice, WM_DAC_CTRL2); - if (ucontrol->value.integer.value[0]) - temp |= 0xf; - else - temp &= ~0xf; - if (temp != temp2) { - wm_put(ice, WM_DAC_CTRL2, temp); - return 1; - } - return 0; -} - -/* - * ADC Oversampling - */ -static int aureon_oversampling_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[2] = { "128x", "64x" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - - return 0; -} - -static int aureon_oversampling_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = (wm_get(ice, WM_MASTER) & 0x8) == 0x8; - return 0; -} - -static int aureon_oversampling_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - int temp, temp2; - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - temp2 = temp = wm_get(ice, WM_MASTER); - - if (ucontrol->value.enumerated.item[0]) - temp |= 0x8; - else - temp &= ~0x8; - - if (temp != temp2) { - wm_put(ice, WM_MASTER, temp); - return 1; - } - return 0; -} -#endif static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -407,6 +310,56 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, stac9460_put(ice, STAC946X_GENERAL_PURPOSE, new); return change; } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) +{ + unsigned char old, new; + int idx; + unsigned char changed[7]; + struct snd_ice1712 *ice = ak->private_data[0]; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&ice->spec.prodigy192.mute_mutex); + /* we have to remember current mute status for each DAC */ + for (idx = 0; idx < 7 ; ++idx) + changed[idx] = stac9460_dac_mute(ice, + STAC946X_MASTER_VOLUME + idx, 0); + /*printk("Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + for (idx = 0; idx < 7 ; ++idx) { + if (changed[idx]) + stac9460_dac_mute(ice, STAC946X_MASTER_VOLUME + idx, 1); + } + mutex_unlock(&ice->spec.prodigy192.mute_mutex); +} + +/* using akm infrastructure for setting rate of the codec */ +static struct snd_akm4xxx akmlike_stac9460 __devinitdata = { + .type = NON_AKM, /* special value */ + .num_adcs = 6, /* not used in any way, just for completeness */ + .num_dacs = 2, + .ops = { + .set_rate_val = stac9460_set_rate_val + } +}; + static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -483,39 +436,8 @@ static struct snd_kcontrol_new stac_controls[] __devinitdata = { .put = stac9460_mic_sw_put, }, -#if 0 - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Route", - .info = wm_adc_mux_info, - .get = wm_adc_mux_get, - .put = wm_adc_mux_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Headphone Amplifier Switch", - .info = aureon_bool_info, - .get = aureon_hpamp_get, - .put = aureon_hpamp_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "DAC Deemphasis Switch", - .info = aureon_bool_info, - .get = aureon_deemp_get, - .put = aureon_deemp_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "ADC Oversampling", - .info = aureon_oversampling_info, - .get = aureon_oversampling_get, - .put = aureon_oversampling_put - }, -#endif }; - /* AK4114 - ICE1724 connections on Prodigy192 + MI/ODI/O */ /* CDTO (pin 32) -- GPIO11 pin 86 * CDTI (pin 33) -- GPIO10 pin 77 @@ -720,6 +642,27 @@ static int prodigy192_ak4114_init(struct snd_ice1712 *ice) ice, &ice->spec.prodigy192.ak4114); } +static void stac9460_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + int reg, val; + /* registers 0x0 - 0x14 */ + for (reg = 0; reg <= 0x15; reg++) { + val = stac9460_get(ice, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + + +static void stac9460_proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ice->card, "stac9460_codec", &entry)) + snd_info_set_text_ops(entry, ice, stac9460_proc_regs_read); +} + + static int __devinit prodigy192_add_controls(struct snd_ice1712 *ice) { unsigned int i; @@ -746,6 +689,7 @@ static int __devinit prodigy192_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; } + stac9460_proc_init(ice); return 0; } @@ -778,6 +722,7 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) { static const unsigned short stac_inits_prodigy[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, /* STAC946X_MASTER_VOLUME, 0, STAC946X_LF_VOLUME, 0, STAC946X_RF_VOLUME, 0, @@ -789,6 +734,7 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) }; const unsigned short *p; int err = 0; + struct snd_akm4xxx *ak; /* prodigy 192 */ ice->num_total_dacs = 6; @@ -799,6 +745,15 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) p = stac_inits_prodigy; for (; *p != (unsigned short)-1; p += 2) stac9460_put(ice, p[0], p[1]); + /* reusing the akm codecs infrastructure, + * for setting rate on stac9460 */ + ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); + if (!ak) + return -ENOMEM; + ice->akm_codecs = 1; + err = snd_ice1712_akm4xxx_init(ak, &akmlike_stac9460, NULL, ice); + if (err < 0) + return err; /* MI/ODI/O add on card with AK4114 */ if (prodigy192_miodio_exists(ice)) { @@ -812,6 +767,8 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) if (err < 0) return err; + mutex_init(&ice->spec.prodigy192.mute_mutex); + return 0; } @@ -854,6 +811,10 @@ struct snd_ice1712_card_info snd_vt1724_prodigy192_cards[] __devinitdata = { .build_controls = prodigy192_add_controls, .eeprom_size = sizeof(prodigy71_eeprom), .eeprom_data = prodigy71_eeprom, + /* the current MPU401 code loops infinitely + * when opening midi device + */ + .no_mpu401 = 1, }, { } /* terminator */ }; -- cgit v0.10.2 From e3cde64af44624030e7635881a5d1932c353a335 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 3 Dec 2007 16:50:58 +0100 Subject: [ALSA] hda-codec - Fix typo in the ALC883 initial code The attached patch should fix typo in auto initialization verbs for ALC883 codec. Signed-off-by: Andy Shevchenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 14c8c01..10749c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7098,7 +7098,7 @@ static struct hda_verb alc883_auto_init_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, { } }; -- cgit v0.10.2 From befceea90b17792cb03cc4e22f3329c89621bba3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 3 Dec 2007 17:08:40 +0100 Subject: [ALSA] echoaudio - convert from semaphore to mutex Converted from semaphore to mutex. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 499ee1a..90ec090 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -378,7 +378,7 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) DE_ACT(("pcm_digital_in_open\n")); max_channels = num_digital_busses_in(chip) - substream->number; - down(&chip->mode_mutex); + mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) err = pcm_open(substream, max_channels); else /* If the card has ADAT, subtract the 6 channels @@ -405,7 +405,7 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) chip->can_set_rate=0; din_exit: - up(&chip->mode_mutex); + mutex_unlock(&chip->mode_mutex); return err; } @@ -420,7 +420,7 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) DE_ACT(("pcm_digital_out_open\n")); max_channels = num_digital_busses_out(chip) - substream->number; - down(&chip->mode_mutex); + mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) err = pcm_open(substream, max_channels); else /* If the card has ADAT, subtract the 6 channels @@ -447,7 +447,7 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) if (atomic_read(&chip->opencount) > 1 && chip->rate_set) chip->can_set_rate=0; dout_exit: - up(&chip->mode_mutex); + mutex_unlock(&chip->mode_mutex); return err; } @@ -1420,7 +1420,7 @@ static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol, if (dmode != chip->digital_mode) { /* mode_mutex is required to make this operation atomic wrt pcm_digital_*_open() and set_input_clock() functions. */ - down(&chip->mode_mutex); + mutex_lock(&chip->mode_mutex); /* Do not allow the user to change the digital mode when a pcm device is open because it also changes the number of channels @@ -1439,7 +1439,7 @@ static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol, if (changed >= 0) changed = 1; /* No errors */ } - up(&chip->mode_mutex); + mutex_unlock(&chip->mode_mutex); } return changed; } @@ -1566,12 +1566,12 @@ static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol, return -EINVAL; dclock = chip->clock_source_list[eclock]; if (chip->input_clock != dclock) { - down(&chip->mode_mutex); + mutex_lock(&chip->mode_mutex); spin_lock_irq(&chip->lock); if ((changed = set_input_clock(chip, dclock)) == 0) changed = 1; /* no errors */ spin_unlock_irq(&chip->lock); - up(&chip->mode_mutex); + mutex_unlock(&chip->mode_mutex); } if (changed < 0) @@ -1972,7 +1972,7 @@ static __devinit int snd_echo_create(struct snd_card *card, return err; } atomic_set(&chip->opencount, 0); - init_MUTEX(&chip->mode_mutex); + mutex_init(&chip->mode_mutex); chip->can_set_rate = 1; *rchip = chip; /* Init done ! */ diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 7e88c96..1c88e05 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -361,7 +361,7 @@ struct echoaudio { spinlock_t lock; struct snd_pcm_substream *substream[DSP_MAXPIPES]; int last_period[DSP_MAXPIPES]; - struct semaphore mode_mutex; + struct mutex mode_mutex; u16 num_digital_modes, digital_mode_list[6]; u16 num_clock_sources, clock_source_list[10]; atomic_t opencount; -- cgit v0.10.2 From 69a07304d52cc1080fa650c4ba7a82eed3cd143c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Dec 2007 12:28:52 +0100 Subject: [ALSA] cmipci: disable 'Modem' control on version 39 or newer chips On version 39 or newer chips, we better remove the 'Modem' control because this register bit now mutes the front channels of the multichannel stream. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 187203e..8cb3436 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2651,10 +2651,8 @@ static struct snd_kcontrol_new snd_cmipci_extra_mixer_switches[] __devinitdata = }; /* card control switches */ -static struct snd_kcontrol_new snd_cmipci_control_switches[] __devinitdata = { - // DEFINE_CARD_SWITCH("Joystick", joystick), /* now module option */ - DEFINE_CARD_SWITCH("Modem", modem), -}; +static struct snd_kcontrol_new snd_cmipci_modem_switch __devinitdata = +DEFINE_CARD_SWITCH("Modem", modem); static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_device) @@ -2735,9 +2733,9 @@ static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_devic } /* card switches */ - sw = snd_cmipci_control_switches; - for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_control_switches); idx++, sw++) { - err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm)); + if (cm->chip_version < 39) { + err = snd_ctl_add(cm->card, + snd_ctl_new1(&snd_cmipci_modem_switch, cm)); if (err < 0) return err; } -- cgit v0.10.2 From 25543fa785a32ce22e7374ba403eb6d38854d037 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 11 Dec 2007 08:55:16 +0100 Subject: [ALSA] cmipci: document 'Modem' control version check Add a comment that explains why the 'Modem' control doesn't work with newer chips. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 8cb3436..bc0a195 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2733,6 +2733,10 @@ static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_devic } /* card switches */ + /* + * newer chips don't have the register bits to force modem link + * detection; the bit that was FLINKON now mutes CH1 + */ if (cm->chip_version < 39) { err = snd_ctl_add(cm->card, snd_ctl_new1(&snd_cmipci_modem_switch, cm)); -- cgit v0.10.2 From b751eef1fdffca5532344285f2fad0c60d2f0158 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 13 Dec 2007 10:19:42 +0100 Subject: [ALSA] Use posix clock monotonic for PCM and timer timestamps We need an accurate and continuous (monotonic) time sources to do accurate synchronization among more timing sources. This patch allows to enable monotonic timestamps for ALSA PCM devices and enables monotonic timestamps for ALSA timer devices. Signed-off-by: Jaroslav Kysela diff --git a/include/sound/asound.h b/include/sound/asound.h index 3ad5341..475eb71 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -138,7 +138,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 8) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 9) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -435,9 +435,15 @@ struct snd_xfern { }; enum { + SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, +}; + +enum { SNDRV_PCM_IOCTL_PVERSION = _IOR('A', 0x00, int), SNDRV_PCM_IOCTL_INFO = _IOR('A', 0x01, struct snd_pcm_info), - SNDRV_PCM_IOCTL_TSTAMP = _IOW('A', 0x02, int), + SNDRV_PCM_IOCTL_TTSTAMP = _IOW('A', 0x03, int), SNDRV_PCM_IOCTL_HW_REFINE = _IOWR('A', 0x10, struct snd_pcm_hw_params), SNDRV_PCM_IOCTL_HW_PARAMS = _IOWR('A', 0x11, struct snd_pcm_hw_params), SNDRV_PCM_IOCTL_HW_FREE = _IO('A', 0x12), diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 5e9cc46..65f6362 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -323,6 +323,7 @@ struct snd_pcm_runtime { /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ + int tstamp_type; /* timestamp type */ /* -- DMA -- */ unsigned char *dma_area; /* DMA area */ @@ -952,6 +953,15 @@ void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream); void snd_pcm_timer_init(struct snd_pcm_substream *substream); void snd_pcm_timer_done(struct snd_pcm_substream *substream); +static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime, + struct timespec *tv) +{ + if (runtime->tstamp_type == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC) + do_posix_clock_monotonic_gettime(tv); + else + getnstimeofday(tv); +} + /* * Memory */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 93d7ca5..db3d7e9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -188,7 +188,7 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs snd_pcm_sframes_t delta; if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP) - getnstimeofday((struct timespec *)&runtime->status->tstamp); + snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); pos = snd_pcm_update_hw_ptr_pos(substream, runtime); if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6245bda..cdeae7c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -598,9 +598,9 @@ int snd_pcm_status(struct snd_pcm_substream *substream, if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP) status->tstamp = runtime->status->tstamp; else - getnstimeofday(&status->tstamp); + snd_pcm_gettime(runtime, &status->tstamp); } else - getnstimeofday(&status->tstamp); + snd_pcm_gettime(runtime, &status->tstamp); status->appl_ptr = runtime->control->appl_ptr; status->hw_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -688,7 +688,7 @@ static void snd_pcm_trigger_tstamp(struct snd_pcm_substream *substream) if (runtime->trigger_master == NULL) return; if (runtime->trigger_master == substream) { - getnstimeofday(&runtime->trigger_tstamp); + snd_pcm_gettime(runtime, &runtime->trigger_tstamp); } else { snd_pcm_trigger_tstamp(runtime->trigger_master); runtime->trigger_tstamp = runtime->trigger_master->runtime->trigger_tstamp; @@ -2519,6 +2519,21 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, return -EFAULT; return 0; } + +static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int arg; + + if (get_user(arg, _arg)) + return -EFAULT; + if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST) + return -EINVAL; + runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY; + if (arg == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC) + runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC; + return 0; +} static int snd_pcm_common_ioctl1(struct file *file, struct snd_pcm_substream *substream, @@ -2531,8 +2546,8 @@ static int snd_pcm_common_ioctl1(struct file *file, return put_user(SNDRV_PCM_VERSION, (int __user *)arg) ? -EFAULT : 0; case SNDRV_PCM_IOCTL_INFO: return snd_pcm_info_user(substream, arg); - case SNDRV_PCM_IOCTL_TSTAMP: /* just for compatibility */ - return 0; + case SNDRV_PCM_IOCTL_TTSTAMP: + return snd_pcm_tstamp(substream, arg); case SNDRV_PCM_IOCTL_HW_REFINE: return snd_pcm_hw_refine_user(substream, arg); case SNDRV_PCM_IOCTL_HW_PARAMS: diff --git a/sound/core/timer.c b/sound/core/timer.c index e7dc56c..7e5fe2d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -44,11 +44,14 @@ #endif static int timer_limit = DEFAULT_TIMER_LIMIT; +static int timer_tstamp_monotonic = 1; MODULE_AUTHOR("Jaroslav Kysela , Takashi Iwai "); MODULE_DESCRIPTION("ALSA timer interface"); MODULE_LICENSE("GPL"); module_param(timer_limit, int, 0444); MODULE_PARM_DESC(timer_limit, "Maximum global timers in system."); +module_param(timer_tstamp_monotonic, int, 0444); +MODULE_PARM_DESC(timer_tstamp_monotonic, "Use posix monotonic clock source for timestamps (default)."); struct snd_timer_user { struct snd_timer_instance *timeri; @@ -381,7 +384,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) struct snd_timer_instance *ts; struct timespec tstamp; - getnstimeofday(&tstamp); + if (timer_tstamp_monotonic) + do_posix_clock_monotonic_gettime(&tstamp); + else + getnstimeofday(&tstamp); snd_assert(event >= SNDRV_TIMER_EVENT_START && event <= SNDRV_TIMER_EVENT_PAUSE, return); if (event == SNDRV_TIMER_EVENT_START || @@ -1182,8 +1188,12 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); return; } - if (tu->last_resolution != resolution || ticks > 0) - getnstimeofday(&tstamp); + if (tu->last_resolution != resolution || ticks > 0) { + if (timer_tstamp_monotonic) + do_posix_clock_monotonic_gettime(&tstamp); + else + getnstimeofday(&tstamp); + } if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && tu->last_resolution != resolution) { r1.event = SNDRV_TIMER_EVENT_RESOLUTION; -- cgit v0.10.2 From 67a7be7e9c1523cec8aa10c8d744fe70f778a093 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 13 Dec 2007 10:36:03 +0100 Subject: [ALSA] PCM - fixed SNDRV_PCM_FORMAT_U24_BE silence constant Reported by Timur Tabi . Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index dd9aa51..b9ae6b3 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -75,7 +75,7 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { }, [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, - .silence = { 0x80, 0x00, 0x00 }, + .silence = { 0x00, 0x80, 0x00, 0x00 }, }, [SNDRV_PCM_FORMAT_S32_LE] = { .width = 32, .phys = 32, .le = 1, .signd = 1, -- cgit v0.10.2 From bba8dee78218752c6457d0dac9f5faa17755ac95 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Dec 2007 13:13:31 +0100 Subject: [ALSA] hda-codec - Device ID for MSI L745 Added the model targa-2ch-dig for MSL L745 (ALSA bug#3641). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 10749c2..0287bcc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7173,6 +7173,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), -- cgit v0.10.2 From 3ad5afcd5fa91a00a9a19b9e39958acd9a3a25d7 Mon Sep 17 00:00:00 2001 From: Nick Piggin Date: Thu, 13 Dec 2007 16:15:00 +0100 Subject: [ALSA] alsa: nopage Convert ALSA from nopage to fault. Switch from OOM to SIGBUS if the resource is not available. Signed-off-by: Nick Piggin Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cdeae7c..51294dd 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3033,26 +3033,23 @@ static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait) /* * mmap status record */ -static struct page * snd_pcm_mmap_status_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_status_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; - struct page * page; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - page = virt_to_page(runtime->status); - get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = virt_to_page(runtime->status); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_status = { - .nopage = snd_pcm_mmap_status_nopage, + .fault = snd_pcm_mmap_status_fault, }; static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file, @@ -3076,26 +3073,23 @@ static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file /* * mmap control record */ -static struct page * snd_pcm_mmap_control_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_control_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; - struct page * page; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - page = virt_to_page(runtime->control); - get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = virt_to_page(runtime->control); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_control = { - .nopage = snd_pcm_mmap_control_nopage, + .fault = snd_pcm_mmap_control_fault, }; static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file, @@ -3132,10 +3126,10 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file #endif /* coherent mmap */ /* - * nopage callback for mmapping a RAM page + * fault callback for mmapping a RAM page */ -static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; @@ -3145,33 +3139,30 @@ static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area, size_t dma_bytes; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - offset = area->vm_pgoff << PAGE_SHIFT; - offset += address - area->vm_start; - snd_assert((offset % PAGE_SIZE) == 0, return NOPAGE_SIGBUS); + offset = vmf->pgoff << PAGE_SHIFT; dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; if (substream->ops->page) { page = substream->ops->page(substream, offset); - if (! page) - return NOPAGE_OOM; /* XXX: is this really due to OOM? */ + if (!page) + return VM_FAULT_SIGBUS; } else { vaddr = runtime->dma_area + offset; page = virt_to_page(vaddr); } get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = page; + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_data = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, - .nopage = snd_pcm_mmap_data_nopage, + .fault = snd_pcm_mmap_data_fault, }; /* -- cgit v0.10.2 From eb415b8f6d1b259ea31b3093df958054380f7f8f Mon Sep 17 00:00:00 2001 From: Nick Piggin Date: Thu, 13 Dec 2007 16:16:40 +0100 Subject: [ALSA] alsa: usx2y nopage Convert alsa usx2y driver from nopage to fault. Signed-off-by: Nick Piggin Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index b76b3dd..a1dca34 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -34,34 +34,31 @@ int usX2Y_hwdep_pcm_new(struct snd_card *card); -static struct page * snd_us428ctls_vm_nopage(struct vm_area_struct *area, unsigned long address, int *type) +static int snd_us428ctls_vm_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { unsigned long offset; struct page * page; void *vaddr; - snd_printdd("ENTER, start %lXh, ofs %lXh, pgoff %ld, addr %lXh\n", + snd_printdd("ENTER, start %lXh, pgoff %ld\n", area->vm_start, - address - area->vm_start, - (address - area->vm_start) >> PAGE_SHIFT, - address); + vmf->pgoff); - offset = area->vm_pgoff << PAGE_SHIFT; - offset += address - area->vm_start; - snd_assert((offset % PAGE_SIZE) == 0, return NOPAGE_SIGBUS); + offset = vmf->pgoff << PAGE_SHIFT; vaddr = (char*)((struct usX2Ydev *)area->vm_private_data)->us428ctls_sharedmem + offset; page = virt_to_page(vaddr); get_page(page); - snd_printdd( "vaddr=%p made us428ctls_vm_nopage() return %p; offset=%lX\n", vaddr, page, offset); + vmf->page = page; - if (type) - *type = VM_FAULT_MINOR; + snd_printdd("vaddr=%p made us428ctls_vm_fault() page %p\n", + vaddr, page); - return page; + return 0; } static struct vm_operations_struct us428ctls_vm_ops = { - .nopage = snd_us428ctls_vm_nopage, + .fault = snd_us428ctls_vm_fault, }; static int snd_us428ctls_mmap(struct snd_hwdep * hw, struct file *filp, struct vm_area_struct *area) diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index a5e7bcd..800b5ce 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -683,30 +683,24 @@ static void snd_usX2Y_hwdep_pcm_vm_close(struct vm_area_struct *area) } -static struct page * snd_usX2Y_hwdep_pcm_vm_nopage(struct vm_area_struct *area, unsigned long address, int *type) +static int snd_usX2Y_hwdep_pcm_vm_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { unsigned long offset; - struct page *page; void *vaddr; - offset = area->vm_pgoff << PAGE_SHIFT; - offset += address - area->vm_start; - snd_assert((offset % PAGE_SIZE) == 0, return NOPAGE_OOM); + offset = vmf->pgoff << PAGE_SHIFT; vaddr = (char*)((struct usX2Ydev *)area->vm_private_data)->hwdep_pcm_shm + offset; - page = virt_to_page(vaddr); - get_page(page); - - if (type) - *type = VM_FAULT_MINOR; - - return page; + vmf->page = virt_to_page(vaddr); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_usX2Y_hwdep_pcm_vm_ops = { .open = snd_usX2Y_hwdep_pcm_vm_open, .close = snd_usX2Y_hwdep_pcm_vm_close, - .nopage = snd_usX2Y_hwdep_pcm_vm_nopage, + .fault = snd_usX2Y_hwdep_pcm_vm_fault, }; -- cgit v0.10.2 From 64e0e2d987f5e803f001c853a433ef2fc51640cb Mon Sep 17 00:00:00 2001 From: Rene Herman Date: Thu, 13 Dec 2007 16:29:33 +0100 Subject: [ALSA] Update descriptions of isapnp-specific module options Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 23f4db0..969ff83 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -328,6 +328,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on C-Media CMI8330 ISA chips. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + wssport - port # for CMI8330 chip (WSS) wssirq - IRQ # for CMI8330 chip (WSS) wssdma - first DMA # for CMI8330 chip (WSS) @@ -382,6 +386,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on CS4232/CS4232A ISA chips. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + port - port # for CS4232 chip (PnP setup - 0x534) cport - control port # for CS4232 chip (PnP setup - 0x120,0x210,0xf00) mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable @@ -390,10 +398,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) dma1 - first DMA # for CS4232 chip (0,1,3) dma2 - second DMA # for Yamaha CS4232 chip (0,1,3), -1 = disable - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) This module supports multiple cards. This module does not support autoprobe - thus main port must be specified!!! Other ports are optional. + (if ISA PnP is not used) thus main port must be specified!!! Other ports are + optional. The power-management is supported. @@ -403,6 +411,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on CS4235/CS4236/CS4236B/CS4237B/ CS4238B/CS4239 ISA chips. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + port - port # for CS4236 chip (PnP setup - 0x534) cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00) mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable @@ -411,7 +423,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) dma1 - first DMA # for CS4236 chip (0,1,3) dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) This module supports multiple cards. This module does not support autoprobe (if ISA PnP is not used) thus main port and control port must be @@ -600,13 +611,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for ESS AudioDrive ES-18xx sound cards. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + port - port # for ES-18xx chip (0x220,0x240,0x260) mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) fm_port - port # for FM (optional, not used) irq - IRQ # for ES-18xx chip (5,7,9,10) dma1 - first DMA # for ES-18xx chip (0,1,3) dma2 - first DMA # for ES-18xx chip (0,1,3) - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) This module supports multiple cards, ISA PnP and autoprobe (without MPU-401 port if native ISA PnP routines are not used). @@ -1230,15 +1244,19 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32 and other sound cards based on AMD InterWave (tm) chip. - port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for InterWave chip (3,5,9,11,12,15) - dma1 - DMA # for InterWave chip (0,1,3,5,6,7) - dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) pcm_voices - reserved PCM voices for the synthesizer (default 2) effect - 1 = InterWave effects enable (default 0); requires 8 voices + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for InterWave chip (3,5,9,11,12,15) + dma1 - DMA # for InterWave chip (0,1,3,5,6,7) + dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) This module supports multiple cards, autoprobe and ISA PnP. @@ -1249,16 +1267,20 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. and other sound cards based on AMD InterWave (tm) chip with TEA6330T circuit for extended control of bass, treble and master volume. - port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) - port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380) - irq - IRQ # for InterWave chip (3,5,9,11,12,15) - dma1 - DMA # for InterWave chip (0,1,3,5,6,7) - dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) pcm_voices - reserved PCM voices for the synthesizer (default 2) effect - 1 = InterWave effects enable (default 0); requires 8 voices + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) + port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380) + irq - IRQ # for InterWave chip (3,5,9,11,12,15) + dma1 - DMA # for InterWave chip (0,1,3,5,6,7) + dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) This module supports multiple cards, autoprobe and ISA PnP. @@ -1446,6 +1468,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for Yamaha OPL3-SA2/SA3 sound cards. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + port - control port # for OPL3-SA chip (0x370) sb_port - SB port # for OPL3-SA chip (0x220,0x240) wss_port - WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604) @@ -1454,7 +1480,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. irq - IRQ # for OPL3-SA chip (5,7,9,10) dma1 - first DMA # for Yamaha OPL3-SA chip (0,1,3) dma2 - second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) This module supports multiple cards and ISA PnP. It does not support autoprobe (if ISA PnP is not used) thus all ports must be specified!!! @@ -1467,6 +1492,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips. Module works with OAK Mozart cards as well. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) fm_port - port # for OPL3 device (0x388) @@ -1481,6 +1510,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) fm_port - port # for OPL3 device (0x388) @@ -1496,6 +1529,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on OPTi 82c93x chips. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) fm_port - port # for OPL3 device (0x388) @@ -1620,6 +1657,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. SoundBlaster AWE 32 (PnP), SoundBlaster AWE 64 PnP + mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) + csp - ASP/CSP chip support - 0 = disable (default), 1 = enable + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + port - port # for SB DSP 4.x chip (0x220,0x240,0x260) mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660) @@ -1627,9 +1670,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. irq - IRQ # for SB DSP 4.x chip (5,7,9,10) dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3) dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7) - mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) - csp - ASP/CSP chip support - 0 = disable (default), 1 = enable - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) This module supports multiple cards, autoprobe and ISA PnP. @@ -1712,18 +1752,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. + use_cs4232_midi - Use CS4232 MPU-401 interface + (inaccessibly located inside your computer) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + cs4232_pcm_port - Port # for CS4232 PCM interface. cs4232_pcm_irq - IRQ # for CS4232 PCM interface (5,7,9,11,12,15). cs4232_mpu_port - Port # for CS4232 MPU-401 interface. cs4232_mpu_irq - IRQ # for CS4232 MPU-401 interface (9,11,12,15). - use_cs4232_midi - Use CS4232 MPU-401 interface - (inaccessibly located inside your computer) ics2115_port - Port # for ICS2115 ics2115_irq - IRQ # for ICS2115 fm_port - FM OPL-3 Port # dma1 - DMA1 # for CS4232 PCM interface. dma2 - DMA2 # for CS4232 PCM interface. - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) The below are options for wavefront_synth features: wf_raw - Assume that we need to boot the OS (default:no) -- cgit v0.10.2 From f7278fd0a57ea6cde4988ab820851b01be20daef Mon Sep 17 00:00:00 2001 From: Josepch Chan Date: Thu, 13 Dec 2007 16:40:40 +0100 Subject: [ALSA] hda-codec - Add support for VIA VT1708B HD audio codec This patch adds support for VIA new HD audio codec, VT1708B. Signed-off-by: Josepch Chan Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4cdf3e6..d42d8f7 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -27,6 +27,8 @@ /* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ /* 2006-08-02 Lydia Wang Add support to VT1709 codec */ /* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */ +/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ +/* 2007-09-17 Lydia Wang Add VT1708B codec support */ /* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -51,14 +53,23 @@ #define VT1708_HP_NID 0x13 #define VT1708_DIGOUT_NID 0x14 #define VT1708_DIGIN_NID 0x16 +#define VT1708_DIGIN_PIN 0x26 #define VT1709_HP_DAC_NID 0x28 #define VT1709_DIGOUT_NID 0x13 #define VT1709_DIGIN_NID 0x17 +#define VT1709_DIGIN_PIN 0x25 + +#define VT1708B_HP_NID 0x25 +#define VT1708B_DIGOUT_NID 0x12 +#define VT1708B_DIGIN_NID 0x15 +#define VT1708B_DIGIN_PIN 0x21 #define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b) #define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713) #define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717) +#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723) +#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727) enum { @@ -131,6 +142,11 @@ static hda_nid_t vt1709_adc_nids[3] = { 0x14, 0x15, 0x16 }; +static hda_nid_t vt1708B_adc_nids[2] = { + /* ADC1-2 */ + 0x13, 0x14 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -268,9 +284,13 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, 0x18, &spec->cur_mux[adc_idx]); else if ((IS_VT1709_10CH_VENDORID(vendor_id) || - IS_VT1709_6CH_VENDORID(vendor_id)) && (adc_idx == 0) ) + IS_VT1709_6CH_VENDORID(vendor_id)) && adc_idx == 0) return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, 0x19, &spec->cur_mux[adc_idx]); + else if ((IS_VT1708B_8CH_VENDORID(vendor_id) || + IS_VT1708B_4CH_VENDORID(vendor_id)) && adc_idx == 0) + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + 0x17, &spec->cur_mux[adc_idx]); else return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->adc_nids[adc_idx], @@ -309,15 +329,15 @@ static struct hda_verb vt1708_volume_init_verbs[] = { {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget */ /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* master */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* * Set up output mixers (0x19 - 0x1b) @@ -329,10 +349,9 @@ static struct hda_verb vt1708_volume_init_verbs[] = { /* Setup default input to PW4 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Set mic as default input of sw0 */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x2}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + { } }; static int via_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -544,6 +563,31 @@ static int via_init(struct hda_codec *codec) { struct via_spec *spec = codec->spec; snd_hda_sequence_write(codec, spec->init_verbs); + /* Lydia Add for EAPD enable */ + if (!spec->dig_in_nid) { /* No Digital In connection */ + if (IS_VT1708_VENDORID(codec->vendor_id)) { + snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + 0x40); + snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x02); + } else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) || + IS_VT1709_6CH_VENDORID(codec->vendor_id)) { + snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + 0x40); + snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x02); + } else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) || + IS_VT1708B_4CH_VENDORID(codec->vendor_id)) { + snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + 0x40); + snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x02); + } + } + return 0; } @@ -623,58 +667,68 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, - "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT){ /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT)); + HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT)); + HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_INPUT)); if (err < 0) return err; /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; } else { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; } @@ -899,15 +953,15 @@ static struct hda_verb vt1709_10ch_volume_init_verbs[] = { {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget */ /* Amp Indices: AOW0=0, CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* unmute master */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* * Set up output selector (0x1a, 0x1b, 0x29) @@ -925,8 +979,6 @@ static struct hda_verb vt1709_10ch_volume_init_verbs[] = { /* Set input of PW4 as AOW4 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Set mic as default input of sw0 */ - {0x19, AC_VERB_SET_CONNECT_SEL, 0x2}, /* PW9 Output enable */ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -1073,68 +1125,80 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, /* Center/LFE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT){ /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT)); + HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT)); + HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_INPUT)); if (err < 0) return err; /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_SURROUND) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_SIDE) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_OUTPUT)); if (err < 0) return err; } @@ -1351,8 +1415,6 @@ static struct hda_verb vt1709_6ch_volume_init_verbs[] = { /* Set input of PW4 as MW0 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0}, - /* Set mic as default input of sw0 */ - {0x19, AC_VERB_SET_CONNECT_SEL, 0x2}, /* PW9 Output enable */ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -1403,6 +1465,495 @@ static int patch_vt1709_6ch(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1709_loopbacks; #endif + return 0; +} + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1708B_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb vt1708B_8ch_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* + * Set up output mixers + */ + /* set vol=0 to output mixers */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Setup default input to PW4 */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* PW9 Output enable */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* PW10 Input enable */ + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + { } +}; + +static struct hda_verb vt1708B_4ch_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* + * Set up output mixers + */ + /* set vol=0 to output mixers */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Setup default input of PW4 to MW0 */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* PW9 Output enable */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* PW10 Input enable */ + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + { } +}; + +static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_pcm_prepare, + .cleanup = via_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 4, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_pcm_prepare, + .cleanup = via_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1708B_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x13, /* NID to query formats and rates */ + .ops = { + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1708B_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare + }, +}; + +static struct hda_pcm_stream vt1708B_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1708B_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 4; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x10; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0x24; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x25; + break; + case AUTO_SEQ_SIDE: + spec->multiout.dac_nids[i] = 0x11; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1708B_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + hda_nid_t nid_vols[] = {0x16, 0x27, 0x26, 0x18}; + hda_nid_t nid, nid_vol = 0; + int i, err; + + for (i = 0; i <= AUTO_SEQ_SIDE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + + nid_vol = nid_vols[i]; + + if (i == AUTO_SEQ_CENLFE) { + /* Center/LFE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + /* add control to mixer index 0 */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_INPUT)); + if (err < 0) + return err; + + /* add control to PW3 */ + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + + return 0; +} + +static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */ + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = idx; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x1a: /* Mic */ + idx = 2; + break; + + case 0x1b: /* Line In */ + idx = 3; + break; + + case 0x1e: /* Front Mic */ + idx = 4; + break; + + case 0x1f: /* CD */ + idx = 1; + break; + } + err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], + idx, 0x16); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + return 0; +} + +static int vt1708B_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = vt1708B_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1708B_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1708B_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1708B_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID; + if (spec->autocfg.dig_in_pin) + spec->dig_in_nid = VT1708B_DIGIN_NID; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->input_mux = &spec->private_imux; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1708B_loopbacks[] = { + { 0x16, HDA_INPUT, 1 }, + { 0x16, HDA_INPUT, 2 }, + { 0x16, HDA_INPUT, 3 }, + { 0x16, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1708B_8ch(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1708B_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs = vt1708B_8ch_volume_init_verbs; + + spec->stream_name_analog = "VT1708B Analog"; + spec->stream_analog_playback = &vt1708B_8ch_pcm_analog_playback; + spec->stream_analog_capture = &vt1708B_pcm_analog_capture; + + spec->stream_name_digital = "VT1708B Digital"; + spec->stream_digital_playback = &vt1708B_pcm_digital_playback; + spec->stream_digital_capture = &vt1708B_pcm_digital_capture; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1708B_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids); + spec->mixers[spec->num_mixers] = vt1708B_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1708B_loopbacks; +#endif + + return 0; +} + +static int patch_vt1708B_4ch(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1708B_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs = vt1708B_4ch_volume_init_verbs; + + spec->stream_name_analog = "VT1708B Analog"; + spec->stream_analog_playback = &vt1708B_4ch_pcm_analog_playback; + spec->stream_analog_capture = &vt1708B_pcm_analog_capture; + + spec->stream_name_digital = "VT1708B Digital"; + spec->stream_digital_playback = &vt1708B_pcm_digital_playback; + spec->stream_digital_capture = &vt1708B_pcm_digital_capture; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1708B_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids); + spec->mixers[spec->num_mixers] = vt1708B_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1708B_loopbacks; +#endif return 0; } @@ -1415,13 +1966,37 @@ struct hda_codec_preset snd_hda_preset_via[] = { { .id = 0x11061709, .name = "VIA VT1708", .patch = patch_vt1708}, { .id = 0x1106170A, .name = "VIA VT1708", .patch = patch_vt1708}, { .id = 0x1106170B, .name = "VIA VT1708", .patch = patch_vt1708}, - { .id = 0x1106E710, .name = "VIA VT1709 10-Ch", .patch = patch_vt1709_10ch}, - { .id = 0x1106E711, .name = "VIA VT1709 10-Ch", .patch = patch_vt1709_10ch}, - { .id = 0x1106E712, .name = "VIA VT1709 10-Ch", .patch = patch_vt1709_10ch}, - { .id = 0x1106E713, .name = "VIA VT1709 10-Ch", .patch = patch_vt1709_10ch}, - { .id = 0x1106E714, .name = "VIA VT1709 6-Ch", .patch = patch_vt1709_6ch}, - { .id = 0x1106E715, .name = "VIA VT1709 6-Ch", .patch = patch_vt1709_6ch}, - { .id = 0x1106E716, .name = "VIA VT1709 6-Ch", .patch = patch_vt1709_6ch}, - { .id = 0x1106E717, .name = "VIA VT1709 6-Ch", .patch = patch_vt1709_6ch}, + { .id = 0x1106E710, .name = "VIA VT1709 10-Ch", + .patch = patch_vt1709_10ch}, + { .id = 0x1106E711, .name = "VIA VT1709 10-Ch", + .patch = patch_vt1709_10ch}, + { .id = 0x1106E712, .name = "VIA VT1709 10-Ch", + .patch = patch_vt1709_10ch}, + { .id = 0x1106E713, .name = "VIA VT1709 10-Ch", + .patch = patch_vt1709_10ch}, + { .id = 0x1106E714, .name = "VIA VT1709 6-Ch", + .patch = patch_vt1709_6ch}, + { .id = 0x1106E715, .name = "VIA VT1709 6-Ch", + .patch = patch_vt1709_6ch}, + { .id = 0x1106E716, .name = "VIA VT1709 6-Ch", + .patch = patch_vt1709_6ch}, + { .id = 0x1106E717, .name = "VIA VT1709 6-Ch", + .patch = patch_vt1709_6ch}, + { .id = 0x1106E720, .name = "VIA VT1708B 8-Ch", + .patch = patch_vt1708B_8ch}, + { .id = 0x1106E721, .name = "VIA VT1708B 8-Ch", + .patch = patch_vt1708B_8ch}, + { .id = 0x1106E722, .name = "VIA VT1708B 8-Ch", + .patch = patch_vt1708B_8ch}, + { .id = 0x1106E723, .name = "VIA VT1708B 8-Ch", + .patch = patch_vt1708B_8ch}, + { .id = 0x1106E724, .name = "VIA VT1708B 4-Ch", + .patch = patch_vt1708B_4ch}, + { .id = 0x1106E725, .name = "VIA VT1708B 4-Ch", + .patch = patch_vt1708B_4ch}, + { .id = 0x1106E726, .name = "VIA VT1708B 4-Ch", + .patch = patch_vt1708B_4ch}, + { .id = 0x1106E727, .name = "VIA VT1708B 4-Ch", + .patch = patch_vt1708B_4ch}, {} /* terminator */ }; -- cgit v0.10.2 From f6a92248aef841f14a5806cc299c431e7809c733 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 13 Dec 2007 16:52:54 +0100 Subject: [ALSA] hda-codec - Add ALC889/ALC267/ALC269 support Added the support of new Realtek codecs: 1. New ALC889 Support 2. New ALC267 Support 3. New ALC269 Support Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0287bcc..eba9be4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -110,6 +110,13 @@ enum { ALC268_MODEL_LAST /* last tag */ }; +/* ALC269 models */ +enum { + ALC269_BASIC, + ALC269_AUTO, + ALC269_MODEL_LAST /* last tag */ +}; + /* ALC861 models */ enum { ALC861_3ST, @@ -9574,6 +9581,361 @@ static int patch_alc268(struct hda_codec *codec) } /* + * ALC269 channel source setting (2 channel) + */ +#define ALC269_DIGOUT_NID ALC880_DIGOUT_NID + +#define alc269_dac_nids alc260_dac_nids + +static hda_nid_t alc269_adc_nids[1] = { + /* ADC1 */ + 0x07, +}; + +#define alc269_modes alc260_modes +#define alc269_capture_source alc880_lg_lw_capture_source + +static struct snd_kcontrol_new alc269_base_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +/* capture mixer elements */ +static struct snd_kcontrol_new alc269_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc269_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the + * analog-loopback mixer widget + * Note: PASD motherboards uses the Line In 2 as the input for + * front panel mic (mic 2) + */ + /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* + * Set up output mixers (0x0c - 0x0e) + */ + /* set vol=0 to output mixers */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* set EAPD */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + +/* add playback controls from the parsed DAC table */ +static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + hda_nid_t nid; + int err; + + spec->multiout.num_dacs = 1; /* only use one dac */ + spec->multiout.dac_nids = spec->private_dac_nids; + spec->multiout.dac_nids[0] = 2; + + nid = cfg->line_out_pins[0]; + if (nid) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = add_control(spec, ALC_CTL_WIDGET_MUTE, + "Front Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + + nid = cfg->speaker_pins[0]; + if (nid) { + if (!cfg->line_out_pins[0]) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Speaker Playback Volume", + HDA_COMPOSE_AMP_VAL(0x02, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + if (nid == 0x16) { + err = add_control(spec, ALC_CTL_WIDGET_MUTE, + "Speaker Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + err = add_control(spec, ALC_CTL_WIDGET_MUTE, + "Speaker Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + nid = cfg->hp_pins[0]; + if (nid) { + /* spec->multiout.hp_nid = 2; */ + if (!cfg->line_out_pins[0] && !cfg->speaker_pins[0]) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x02, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + if (nid == 0x16) { + err = add_control(spec, ALC_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + err = add_control(spec, ALC_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +#define alc269_auto_create_analog_input_ctls \ + alc880_auto_create_analog_input_ctls + +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc269_loopbacks alc880_loopbacks +#endif + +/* pcm configuration: identiacal with ALC880 */ +#define alc269_pcm_analog_playback alc880_pcm_analog_playback +#define alc269_pcm_analog_capture alc880_pcm_analog_capture +#define alc269_pcm_digital_playback alc880_pcm_digital_playback +#define alc269_pcm_digital_capture alc880_pcm_digital_capture + +/* + * BIOS auto configuration + */ +static int alc269_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + alc269_ignore); + if (err < 0) + return err; + + err = alc269_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc269_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = ALC269_DIGOUT_NID; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->init_verbs[spec->num_init_verbs++] = alc269_init_verbs; + spec->num_mux_defs = 1; + spec->input_mux = &spec->private_imux; + + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; + + return 1; +} + +#define alc269_auto_init_multi_out alc882_auto_init_multi_out +#define alc269_auto_init_hp_out alc882_auto_init_hp_out +#define alc269_auto_init_analog_input alc882_auto_init_analog_input + + +/* init callback for auto-configuration model -- overriding the default init */ +static void alc269_auto_init(struct hda_codec *codec) +{ + alc269_auto_init_multi_out(codec); + alc269_auto_init_hp_out(codec); + alc269_auto_init_analog_input(codec); +} + +/* + * configuration and preset + */ +static const char *alc269_models[ALC269_MODEL_LAST] = { + [ALC269_BASIC] = "basic", +}; + +static struct snd_pci_quirk alc269_cfg_tbl[] = { + {} +}; + +static struct alc_config_preset alc269_presets[] = { + [ALC269_BASIC] = { + .mixers = { alc269_base_mixer }, + .init_verbs = { alc269_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_capture_source, + }, +}; + +static int patch_alc269(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, + alc269_models, + alc269_cfg_tbl); + + if (board_config < 0) { + printk(KERN_INFO "hda_codec: Unknown model for ALC269, " + "trying auto-probe from BIOS...\n"); + board_config = ALC269_AUTO; + } + + if (board_config == ALC269_AUTO) { + /* automatic parse from the BIOS config */ + err = alc269_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO + "hda_codec: Cannot set up configuration " + "from BIOS. Using base mode...\n"); + board_config = ALC269_BASIC; + } + } + + if (board_config != ALC269_AUTO) + setup_preset(spec, &alc269_presets[board_config]); + + spec->stream_name_analog = "ALC269 Analog"; + spec->stream_analog_playback = &alc269_pcm_analog_playback; + spec->stream_analog_capture = &alc269_pcm_analog_capture; + + spec->stream_name_digital = "ALC269 Digital"; + spec->stream_digital_playback = &alc269_pcm_digital_playback; + spec->stream_digital_capture = &alc269_pcm_digital_capture; + + spec->adc_nids = alc269_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); + spec->mixers[spec->num_mixers] = alc269_capture_mixer; + spec->num_mixers++; + + codec->patch_ops = alc_patch_ops; + if (board_config == ALC269_AUTO) + spec->init_hook = alc269_auto_init; +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc269_loopbacks; +#endif + + return 0; +} + +/* * ALC861 channel source setting (2/6 channel selection for 3-stack) */ @@ -12653,7 +13015,9 @@ static int patch_alc662(struct hda_codec *codec) struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, + { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 }, + { .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, @@ -12668,5 +13032,6 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, + { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, {} /* terminator */ }; -- cgit v0.10.2 From a8848bd6476102ff1d3bbe56662a18d6ada8674f Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Thu, 13 Dec 2007 17:32:26 +0100 Subject: [ALSA] hda-codec - Initial support of the Mitac 8252D (based on ALC883) The attached patch adds initial support of the Mitac 8252D (http://www.mitac-mtc.com.tw/English/products/8252Dspec.htm). Working: - Front speakers (volume + mute) - Center/LFE speakers (volume + mute) - HP out (with Front Volume) - HP individual mute switch - HP Jack sense - Front Mic and its volume Not tested: - external mic and its volume Not working while now: - Mic Jack sense Questionable: - is Mic have Jack sense? - one or two Mic volume controls? - CD/Line-in: presense in the mixer Signed-off-by: Andy Shevchenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 969ff83..aded3b4 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -861,6 +861,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. haier-w66 Haier W66 6stack-hp HP machines with 6stack (Nettle boards) 3stack-hp HP machines with 3stack (Lucknow, Samba boards) + mitac Mitac 8252D auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eba9be4..cafdeb8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -192,6 +192,7 @@ enum { ALC883_HAIER_W66, ALC888_6ST_HP, ALC888_3ST_HP, + ALC883_MITAC, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6349,6 +6350,36 @@ static struct snd_kcontrol_new alc883_base_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_mitac_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6781,6 +6812,67 @@ static struct hda_verb alc883_init_verbs[] = { { } }; +/* toggle speaker-output according to the hp-jack state */ +static void alc883_mitac_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +/* auto-toggle front mic */ +/* +static void alc883_mitac_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); +} +*/ + +static void alc883_mitac_automute(struct hda_codec *codec) +{ + alc883_mitac_hp_automute(codec); + /* alc883_mitac_mic_automute(codec); */ +} + +static void alc883_mitac_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_mitac_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + /* alc883_mitac_mic_automute(codec); */ + break; + } +} + +static struct hda_verb alc883_mitac_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Subwoofer */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, */ + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7163,6 +7255,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_HAIER_W66] = "haier-w66", [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", + [ALC883_MITAC] = "mitac", [ALC883_AUTO] = "auto", }; @@ -7198,6 +7291,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), @@ -7445,6 +7539,19 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, + [ALC883_MITAC] = { + .mixers = { alc883_mitac_mixer }, + .init_verbs = { alc883_init_verbs, alc883_mitac_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_mitac_unsol_event, + .init_hook = alc883_mitac_automute, + }, }; -- cgit v0.10.2 From e1f0d6690817d1296161094106b23a0be9ee6ca0 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 13 Dec 2007 17:47:21 +0100 Subject: [ALSA] hda: Added STAC92HD73 support Added support for new STAC92HD73 family of codecs. Additionally added features for multiple analog loopbacks, and multiple dmux mixers. Regression testing for the analog loopback changes for STAC9205 and STAC9274D completed with any issues, as well for the dmux changes. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d2996ad..0e6af53 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -62,6 +62,11 @@ enum { }; enum { + STAC_92HD73XX_REF, + STAC_92HD73XX_MODELS +}; + +enum { STAC_92HD71BXX_REF, STAC_92HD71BXX_MODELS }; @@ -118,6 +123,8 @@ struct sigmatel_spec { unsigned int gpio_mute: 1; unsigned int gpio_mask, gpio_data; + unsigned char aloopback_mask; + unsigned char aloopback_shift; /* playback */ struct hda_multi_out multiout; @@ -130,7 +137,7 @@ struct sigmatel_spec { unsigned int num_muxes; hda_nid_t *dmic_nids; unsigned int num_dmics; - hda_nid_t dmux_nid; + hda_nid_t *dmux_nids; hda_nid_t dig_in_nid; /* pin widgets */ @@ -145,7 +152,7 @@ struct sigmatel_spec { /* capture source */ struct hda_input_mux *dinput_mux; - unsigned int cur_dmux; + unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; @@ -176,6 +183,28 @@ static hda_nid_t stac9200_dac_nids[1] = { 0x02, }; +static hda_nid_t stac92hd73xx_adc_nids[2] = { + 0x1a, 0x1b +}; + +#define STAC92HD73XX_NUM_DMICS 2 +static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { + 0x13, 0x14, 0 +}; + +#define STAC92HD73_DAC_COUNT 5 +static hda_nid_t stac92hd73xx_dac_nids[STAC92HD73_DAC_COUNT] = { + 0x15, 0x16, 0x17, 0x18, 0x19, +}; + +static hda_nid_t stac92hd73xx_mux_nids[4] = { + 0x28, 0x29, 0x2a, 0x2b, +}; + +static hda_nid_t stac92hd73xx_dmux_nids[2] = { + 0x20, 0x21, +}; + static hda_nid_t stac92hd71bxx_adc_nids[2] = { 0x12, 0x13, }; @@ -184,6 +213,10 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b }; +static hda_nid_t stac92hd71bxx_dmux_nids[1] = { + 0x1c, +}; + static hda_nid_t stac92hd71bxx_dac_nids[2] = { 0x10, /*0x11, */ }; @@ -226,6 +259,10 @@ static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dmux_nids[1] = { + 0x1b, +}; + #define STAC927X_NUM_DMICS 2 static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { 0x13, 0x14, 0 @@ -239,6 +276,10 @@ static hda_nid_t stac9205_mux_nids[2] = { 0x19, 0x1a }; +static hda_nid_t stac9205_dmux_nids[1] = { + 0x1d, +}; + #define STAC9205_NUM_DMICS 2 static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { 0x17, 0x18, 0 @@ -259,6 +300,12 @@ static hda_nid_t stac922x_pin_nids[10] = { 0x0f, 0x10, 0x11, 0x15, 0x1b, }; +static hda_nid_t stac92hd73xx_pin_nids[12] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x10, 0x11, 0x12, 0x13, + 0x14, 0x22 +}; + static hda_nid_t stac92hd71bxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, @@ -289,8 +336,9 @@ static int stac92xx_dmux_enum_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - ucontrol->value.enumerated.item[0] = spec->cur_dmux; + ucontrol->value.enumerated.item[0] = spec->cur_dmux[dmux_idx]; return 0; } @@ -299,9 +347,10 @@ static int stac92xx_dmux_enum_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); return snd_hda_input_mux_put(codec, spec->dinput_mux, ucontrol, - spec->dmux_nid, &spec->cur_dmux); + spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]); } static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -337,9 +386,11 @@ static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct sigmatel_spec *spec = codec->spec; - ucontrol->value.integer.value[0] = spec->aloopback; + ucontrol->value.integer.value[0] = !!(spec->aloopback & + (spec->aloopback_mask << idx)); return 0; } @@ -348,24 +399,33 @@ static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int dac_mode; - unsigned int val; + unsigned int val, idx_val; - val = !!ucontrol->value.integer.value[0]; + idx_val = spec->aloopback_mask << idx; + if (ucontrol->value.integer.value[0]) + val = spec->aloopback | idx_val; + else + val = spec->aloopback & ~idx_val; if (spec->aloopback == val) return 0; spec->aloopback = val; + /* Only return the bits defined by the shift value of the + * first two bytes of the mask + */ dac_mode = snd_hda_codec_read(codec, codec->afg, 0, - kcontrol->private_value & 0xFFFF, 0x0); + kcontrol->private_value & 0xFFFF, 0x0); + dac_mode >>= spec->aloopback_shift; - if (spec->aloopback) { + if (spec->aloopback & idx_val) { snd_hda_power_up(codec); - dac_mode |= 0x40; + dac_mode |= idx_val; } else { snd_hda_power_down(codec); - dac_mode &= ~0x40; + dac_mode &= ~idx_val; } snd_hda_codec_write_cache(codec, codec->afg, 0, @@ -387,6 +447,86 @@ static struct hda_verb stac9200_eapd_init[] = { {} }; +static struct hda_verb stac92hd73xx_6ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb stac92hd73xx_8ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* connect hp ports to dac3 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x03}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x03}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, + {} +}; + +static struct hda_verb stac92hd73xx_10ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02 }, + /* dac3 is connected to import3 mux */ + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f}, + /* connect hp ports to dac4 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x04}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x04}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, + {} +}; + static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -460,11 +600,11 @@ static struct hda_verb stac9205_core_init[] = { .put = stac92xx_mux_enum_put, \ } -#define STAC_ANALOG_LOOPBACK(verb_read,verb_write) \ +#define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Analog Loopback", \ - .count = 1, \ + .count = cnt, \ .info = stac92xx_aloopback_info, \ .get = stac92xx_aloopback_get, \ .put = stac92xx_aloopback_put, \ @@ -481,6 +621,99 @@ static struct snd_kcontrol_new stac9200_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), @@ -513,7 +746,7 @@ static struct snd_kcontrol_new stac925x_mixer[] = { static struct snd_kcontrol_new stac9205_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -543,7 +776,7 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), @@ -854,6 +1087,27 @@ static struct snd_pci_quirk stac925x_cfg_tbl[] = { {} /* terminator */ }; +static unsigned int ref92hd73xx_pin_configs[12] = { + 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, + 0x0181302e, 0x01014010, 0x01014020, 0x01014030, + 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, +}; + +static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, +}; + +static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD73XX_REF), + {} /* terminator */ +}; + static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01114010, 0x01a19020, 0x90a000f0, @@ -2030,7 +2284,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, continue; num_cons = snd_hda_get_connections(codec, - spec->dmux_nid, + spec->dmux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); for (j = 0; j < num_cons; j++) @@ -2211,7 +2465,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; + if (!spec->dinput_mux) + spec->dinput_mux = &spec->private_dimux; return 1; } @@ -2696,6 +2951,112 @@ static int patch_stac925x(struct hda_codec *codec) return 0; } +static struct hda_input_mux stac92hd73xx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x0b }, + { "CD", 0x08 }, + { "Digital Mic 1", 0x09 }, + { "Digital Mic 2", 0x0a }, + } +}; + +static int patch_stac92hd73xx(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + hda_nid_t conn[STAC92HD73_DAC_COUNT + 2]; + int err = 0; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); + spec->pin_nids = stac92hd73xx_pin_nids; + spec->board_config = snd_hda_check_board_config(codec, + STAC_92HD73XX_MODELS, + stac92hd73xx_models, + stac92hd73xx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + " STAC92HD73XX, using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else { + spec->pin_configs = stac92hd73xx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + spec->multiout.num_dacs = snd_hda_get_connections(codec, 0x0a, + conn, STAC92HD73_DAC_COUNT + 2) - 1; + + if (spec->multiout.num_dacs < 0) { + printk(KERN_WARNING "hda_codec: Could not determine " + "number of channels defaulting to DAC count\n"); + spec->multiout.num_dacs = STAC92HD73_DAC_COUNT; + } + + switch (spec->multiout.num_dacs) { + case 0x3: /* 6 Channel */ + spec->mixer = stac92hd73xx_6ch_mixer; + spec->init = stac92hd73xx_6ch_core_init; + break; + case 0x4: /* 8 Channel */ + spec->multiout.hp_nid = 0x18; + spec->mixer = stac92hd73xx_8ch_mixer; + spec->init = stac92hd73xx_8ch_core_init; + break; + case 0x5: /* 10 Channel */ + spec->multiout.hp_nid = 0x19; + spec->mixer = stac92hd73xx_10ch_mixer; + spec->init = stac92hd73xx_10ch_core_init; + }; + + spec->multiout.dac_nids = stac92hd73xx_dac_nids; + spec->aloopback_mask = 0x01; + spec->aloopback_shift = 8; + + spec->mux_nids = stac92hd73xx_mux_nids; + spec->adc_nids = stac92hd73xx_adc_nids; + spec->dmic_nids = stac92hd73xx_dmic_nids; + spec->dmux_nids = stac92hd73xx_dmux_nids; + + spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->dinput_mux = &stac92hd73xx_dmux; + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x000001; + stac92xx_enable_gpio_mask(codec); + + err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + + if (!err) { + if (spec->board_config < 0) { + printk(KERN_WARNING "hda_codec: No auto-config is " + "available, default to model=ref\n"); + spec->board_config = STAC_92HD73XX_REF; + goto again; + } + err = -EINVAL; + } + + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +} + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -2736,7 +3097,7 @@ again: spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nid = 0x1c; + spec->dmux_nids = stac92hd71bxx_dmux_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); @@ -2931,7 +3292,7 @@ static int patch_stac927x(struct hda_codec *codec) case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1b; + spec->dmux_nids = stac927x_dmux_nids; /* Enable DMIC0 */ stac92xx_set_config_reg(codec, 0x13, 0x90a60040); @@ -2947,6 +3308,8 @@ static int patch_stac927x(struct hda_codec *codec) } spec->multiout.dac_nids = spec->dac_nids; + spec->aloopback_mask = 0x40; + spec->aloopback_shift = 0; stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); @@ -3004,11 +3367,13 @@ static int patch_stac9205(struct hda_codec *codec) spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); spec->dmic_nids = stac9205_dmic_nids; spec->num_dmics = STAC9205_NUM_DMICS; - spec->dmux_nid = 0x1d; + spec->dmux_nids = stac9205_dmux_nids; spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; + spec->aloopback_mask = 0x40; + spec->aloopback_shift = 0; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){ @@ -3337,6 +3702,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, + { .id = 0x111d7675, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d76b0, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, {} /* terminator */ }; -- cgit v0.10.2 From 695005cfe4808e300bf76d73012c664bcae90566 Mon Sep 17 00:00:00 2001 From: Johannes Stezenbach Date: Thu, 13 Dec 2007 17:51:00 +0100 Subject: [ALSA] hda-intel: Enable Analog CD Input from internal ATAPI connector on Asus M2N-SLI Enable Analog CD Input from internal ATAPI connector on Asus M2N-SLI. Signed-off-by: Johannes Stezenbach Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b2c5380..9b0ecbf 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2084,6 +2084,8 @@ static struct hda_verb ad1988_6stack_init_verbs[] = { {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Analog CD Input */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, { } }; -- cgit v0.10.2 From 71685b90d2efab9ff3c94b7b1169e83449ab3a1f Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 14 Dec 2007 12:07:31 +0100 Subject: [ALSA] hda: 92HD73 DMIC Amps Changed hardware gain mixers for the digital mic's from HDA_OUTPUT to HDA_INPUT. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0e6af53..4bca790 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -626,8 +626,8 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -657,8 +657,8 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -688,8 +688,8 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), -- cgit v0.10.2 From 541eee8768b60ffc233f5ca0796a4cef54df699b Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 14 Dec 2007 12:08:04 +0100 Subject: [ALSA] hda: Added more 92HD71 codecs Added more codecs to the 92HD71 family, as well as support for several that don't have an analog mixer. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4bca790..8598951 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -532,9 +532,24 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* unmute mono out node */ + { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, +}; + +static struct hda_verb stac92hd71bxx_analog_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* connect headphone jack to dac1 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* connect ports 0d and 0f to audio mixer */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ @@ -714,7 +729,7 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { +static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), @@ -735,6 +750,25 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), + STAC_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + { } /* end */ +}; + static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), @@ -3088,12 +3122,25 @@ again: stac92xx_set_config_regs(codec); } + switch (codec->vendor_id) { + case 0x111d76b6: /* 4 Port without Analog Mixer */ + case 0x111d76b7: + case 0x111d76b4: /* 6 Port without Analog Mixer */ + case 0x111d76b5: + spec->mixer = stac92hd71bxx_mixer; + spec->init = stac92hd71bxx_core_init; + break; + default: + spec->mixer = stac92hd71bxx_analog_mixer; + spec->init = stac92hd71bxx_analog_core_init; + } + + spec->aloopback_mask = 0x20; + spec->aloopback_shift = 0; + spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ stac92xx_enable_gpio_mask(codec); - spec->init = stac92hd71bxx_core_init; - spec->mixer = stac92hd71bxx_mixer; - spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; @@ -3702,8 +3749,17 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, + { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7675, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d76b0, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, + { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b3, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b4, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, {} /* terminator */ }; -- cgit v0.10.2 From 6b587ef9a15fa2e1859faf5a8937702ef6c12616 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Dec 2007 12:10:26 +0100 Subject: [ALSA] Fix old tstamp ioctl for compat_ioctl Replaced the old SNDRV_PCM_IOCTL_TSTAMP with the new one in PCM compat_ioctl. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 2b53979..1fb6ae7 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -483,7 +483,7 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l switch (cmd) { case SNDRV_PCM_IOCTL_PVERSION: case SNDRV_PCM_IOCTL_INFO: - case SNDRV_PCM_IOCTL_TSTAMP: + case SNDRV_PCM_IOCTL_TTSTAMP: case SNDRV_PCM_IOCTL_HWSYNC: case SNDRV_PCM_IOCTL_PREPARE: case SNDRV_PCM_IOCTL_RESET: -- cgit v0.10.2 From 5050b0921a0f838314b1b9c4d7c16d5c8ed8c1fc Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Fri, 14 Dec 2007 12:13:12 +0100 Subject: [ALSA] copy_ctl_value_from_user() warning fix sound/core/control_compat.c: In function 'copy_ctl_value_from_user': sound/core/control_compat.c:222: warning: 'count' may be used uninitialized in this function Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 9311ca3..6101259 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -219,7 +219,8 @@ static int copy_ctl_value_from_user(struct snd_card *card, struct snd_ctl_elem_value32 __user *data32, int *typep, int *countp) { - int i, type, count, size; + int i, type, size; + int uninitialized_var(count); unsigned int indirect; if (copy_from_user(&data->id, &data32->id, sizeof(data->id))) -- cgit v0.10.2 From cd0b4ac839529e0f458e83946f49f3f30b7b5be8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Dec 2007 12:18:52 +0100 Subject: [ALSA] sb16 - Suppress compile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sound/isa/sb/sb16_csp.c: In function ‘snd_sb_csp_new’: sound/isa/sb/sb16_csp.c:121: warning: ‘version’ may be used uninitialized in this function Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 3682059..5f21aec 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -118,7 +118,8 @@ static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buff int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep) { struct snd_sb_csp *p; - int version, err; + int uninitialized_var(version); + int err; struct snd_hwdep *hw; if (rhwdep) -- cgit v0.10.2 From b05e578740b5e58b729857321048c4babb24bf2b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Marcin=20=C5=9Alusarz?= Date: Fri, 14 Dec 2007 12:50:16 +0100 Subject: [ALSA] pcm_native: fix sparse warning about shadowing 'state' symbol MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit pcm_native: fix sparse warning about shadowing 'state' symbol Signed-off-by: Marcin Ślusarz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 51294dd..7fb7c92 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1395,10 +1395,10 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) } else { /* stop running stream */ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) { - int state = snd_pcm_capture_avail(runtime) > 0 ? + int new_state = snd_pcm_capture_avail(runtime) > 0 ? SNDRV_PCM_STATE_DRAINING : SNDRV_PCM_STATE_SETUP; - snd_pcm_do_stop(substream, state); - snd_pcm_post_stop(substream, state); + snd_pcm_do_stop(substream, new_state); + snd_pcm_post_stop(substream, new_state); } } return 0; -- cgit v0.10.2 From 757d5a7561842dd89c79d079a3c0935cd49afb9b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Marcin=20=C5=9Alusarz?= Date: Fri, 14 Dec 2007 12:51:24 +0100 Subject: [ALSA] via82xx: minor optimization in snd_via82xx_free MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit via82xx: minor optimization in snd_via82xx_free don't check X, when we just checked !X before goto Signed-off-by: Marcin Ślusarz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index ad58ca5..18a58e4 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2238,9 +2238,9 @@ static int snd_via82xx_free(struct via82xx *chip) for (i = 0; i < chip->num_devs; i++) snd_via82xx_channel_reset(chip, &chip->devs[i]); synchronize_irq(chip->irq); - __end_hw: if (chip->irq >= 0) free_irq(chip->irq, chip); + __end_hw: release_and_free_resource(chip->mpu_res); pci_release_regions(chip->pci); -- cgit v0.10.2 From be3e0115e3732d77d357724a394ee465e5d0b872 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Marcin=20=C5=9Alusarz?= Date: Fri, 14 Dec 2007 12:52:53 +0100 Subject: [ALSA] pcm_lib: fix sparse warning about shadowing 'n' symbol MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit pcm_lib: fix sparse warning about shadowing 'n' symbol Signed-off-by: Marcin Ślusarz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index db3d7e9..2d3e70b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1497,14 +1497,18 @@ void snd_pcm_tick_prepare(struct snd_pcm_substream *substream) avail = snd_pcm_capture_avail(runtime); } if (avail < runtime->control->avail_min) { - snd_pcm_sframes_t n = runtime->control->avail_min - avail; - if (n > 0 && frames > (snd_pcm_uframes_t)n) - frames = n; + snd_pcm_sframes_t to_avail_min = + runtime->control->avail_min - avail; + if (to_avail_min > 0 && + frames > (snd_pcm_uframes_t)to_avail_min) + frames = to_avail_min; } if (avail < runtime->buffer_size) { - snd_pcm_sframes_t n = runtime->buffer_size - avail; - if (n > 0 && frames > (snd_pcm_uframes_t)n) - frames = n; + snd_pcm_sframes_t to_buffer_size = + runtime->buffer_size - avail; + if (to_buffer_size > 0 && + frames > (snd_pcm_uframes_t)to_buffer_size) + frames = to_buffer_size; } if (frames == ULONG_MAX) { snd_pcm_tick_set(substream, 0); -- cgit v0.10.2 From 67c393172c00a710121d61bb7aff31b1e4f44b8d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Marcin=20=C5=9Alusarz?= Date: Fri, 14 Dec 2007 12:53:21 +0100 Subject: [ALSA] pcm_lib: fix sparse warning about different signedness MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit pcm_lib: fix sparse warning about different signedness Signed-off-by: Marcin Ślusarz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2d3e70b..48ffa40 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1139,7 +1139,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_step); static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - static int pow2_sizes[] = { + static unsigned int pow2_sizes[] = { 1<<0, 1<<1, 1<<2, 1<<3, 1<<4, 1<<5, 1<<6, 1<<7, 1<<8, 1<<9, 1<<10, 1<<11, 1<<12, 1<<13, 1<<14, 1<<15, 1<<16, 1<<17, 1<<18, 1<<19, 1<<20, 1<<21, 1<<22, 1<<23, -- cgit v0.10.2 From 36b9cdfea64650069d985c5999cb5c2b6f8b85fb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Marcin=20=C5=9Alusarz?= Date: Fri, 14 Dec 2007 12:58:45 +0100 Subject: [ALSA] info_oss: move prototype of snd_card_info_read_oss to info.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit info_oss: move prototype of snd_card_info_read_oss to info.h Signed-off-by: Marcin Ślusarz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/info.h b/include/sound/info.h index fecbb1f..8ae72e7 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -100,8 +100,10 @@ int snd_info_minor_unregister(void); extern struct snd_info_entry *snd_seq_root; #ifdef CONFIG_SND_OSSEMUL extern struct snd_info_entry *snd_oss_root; +void snd_card_info_read_oss(struct snd_info_buffer *buffer); #else #define snd_oss_root NULL +static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {} #endif int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) __attribute__ ((format (printf, 2, 3))); diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index 435c939..9e8b816 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -66,8 +66,6 @@ int snd_oss_info_register(int dev, int num, char *string) EXPORT_SYMBOL(snd_oss_info_register); -extern void snd_card_info_read_oss(struct snd_info_buffer *buffer); - static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int dev) { int idx, ok = -1; -- cgit v0.10.2 From 3b378e1f7e5b372a88189398f3946fd55da5923a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Marcin=20=C5=9Alusarz?= Date: Fri, 14 Dec 2007 12:59:50 +0100 Subject: [ALSA] sound/core/seq: move declarations of globally visible variables to proper headers MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sound/core/seq: move declarations of globally visible variables to proper headers Signed-off-by: Marcin Ślusarz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 69421ca..6444bd8 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -130,8 +130,6 @@ static struct snd_seq_client *clientptr(int clientid) return clienttab[clientid]; } -extern int seq_client_load[]; - struct snd_seq_client *snd_seq_client_use_ptr(int clientid) { unsigned long flags; diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 5e04e20..20f0a72 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -98,4 +98,6 @@ int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table int snd_seq_client_notify_subscription(int client, int port, struct snd_seq_port_subscribe *info, int evtype); +extern int seq_client_load[15]; + #endif diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 8716352..82a5b33 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -27,13 +27,6 @@ #include "seq_queue.h" #include "seq_info.h" -extern int seq_default_timer_class; -extern int seq_default_timer_sclass; -extern int seq_default_timer_card; -extern int seq_default_timer_device; -extern int seq_default_timer_subdevice; -extern int seq_default_timer_resolution; - /* allowed sequencer timer frequencies, in Hz */ #define MIN_FREQUENCY 10 #define MAX_FREQUENCY 6250 diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h index e9ee154..88dfb71 100644 --- a/sound/core/seq/seq_timer.h +++ b/sound/core/seq/seq_timer.h @@ -138,4 +138,11 @@ int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew, unsigne snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr); snd_seq_tick_time_t snd_seq_timer_get_cur_tick(struct snd_seq_timer *tmr); +extern int seq_default_timer_class; +extern int seq_default_timer_sclass; +extern int seq_default_timer_card; +extern int seq_default_timer_device; +extern int seq_default_timer_subdevice; +extern int seq_default_timer_resolution; + #endif -- cgit v0.10.2 From 175859bf1602c7ee38d720daa14a287072cc2b72 Mon Sep 17 00:00:00 2001 From: David Dillow Date: Fri, 14 Dec 2007 14:40:23 +0100 Subject: [ALSA] sis7019: support the SiS 7019 Audio Accelerator Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth support at the moment, but audio playback and capture with two periods per buffer has seen extensive use. Arbitrary period and buffer sizes (with multiple periods per buffer) have seen light testing, but are believed to be production ready. Signed-off-by: David Dillow Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 356bf21..45f0f6c 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -802,6 +802,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND && X86 && !X86_64 + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 09ddc82..56738da 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -23,6 +23,7 @@ snd-intel8x0m-objs := intel8x0m.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_INTEL8X0M) += snd-intel8x0m.o obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c new file mode 100644 index 0000000..2f17859 --- /dev/null +++ b/sound/pci/sis7019.c @@ -0,0 +1,1461 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "sis7019.h" + +MODULE_AUTHOR("David Dillow "); +MODULE_DESCRIPTION("SiS7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = 1; + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is one or two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the SSO on the playback + * channel to clock out virtual periods, and adjust the virtual period length + * to maintain synchronization. This algorithm came from the Trident driver. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u16 sync_period_size; + u16 sync_buffer_size; + u32 sso; + u32 vperiod; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +/* We need four pages to store our wave parameters during a suspend. If + * we're not doing power management, we still need to allocate a page + * for the silence buffer. + */ +#ifdef CONFIG_PM +#define SIS_SUSPEND_PAGES 4 +#else +#define SIS_SUSPEND_PAGES 1 +#endif + +struct sis7019 { + unsigned long ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* Protect against more than one thread hitting the AC97 + * registers (in a more polite manner than pounding the hardware + * semaphore) + */ + struct mutex ac97_mutex; + + /* voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; + + /* Allocate pages to store the internal wave state during + * suspends. When we're operating, this can be used as a silence + * buffer for a timing channel. + */ + void *suspend_state[SIS_SUSPEND_PAGES]; + + int silence_users; + dma_addr_t silence_dma_addr; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +/* The HW offset parameters (Loop End, Stop Sample, End Sample) have a + * documented range of 8-0xfff8 samples. Given that they are 0-based, + * that places our period/buffer range at 9-0xfff9 samples. That makes the + * max buffer size 0xfff9 samples * 2 channels * 2 bytes per sample, and + * max samples / min samples gives us the max periods in a buffer. + * + * We'll add a constraint upon open that limits the period and buffer sample + * size to values that are legal for the hardware. + */ +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static void sis_update_sso(struct voice *voice, u16 period) +{ + void __iomem *base = voice->ctrl_base; + + voice->sso += period; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* Enforce the documented hardware minimum offset */ + if (voice->sso < 8) + voice->sso = 8; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); +} + +static void sis_update_voice(struct voice *voice) +{ + if (voice->flags & VOICE_SSO_TIMING) { + sis_update_sso(voice, voice->period_size); + } else if (voice->flags & VOICE_SYNC_TIMING) { + int sync; + + /* If we've not hit the end of the virtual period, update + * our records and keep going. + */ + if (voice->vperiod > voice->period_size) { + voice->vperiod -= voice->period_size; + if (voice->vperiod < voice->period_size) + sis_update_sso(voice, voice->vperiod); + else + sis_update_sso(voice, voice->period_size); + return; + } + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = voice->sync_cso; + sync -= readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + if (sync > (voice->sync_buffer_size / 2)) + sync -= voice->sync_buffer_size; + + /* If sync is positive, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + */ + if (sync > 0) { + if (sync < 16) + sync = 16; + sis_update_sso(voice, sync); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * a minimum wait time, which makes us quite late here, + * eating into the user's time to refresh the buffer, esp. + * if using small periods. + * + * If we're less than 9 samples behind, we're on target. + */ + if (sync > -9) + voice->vperiod = voice->sync_period_size + 1; + else + voice->vperiod = voice->sync_period_size - 4; + + if (voice->vperiod < voice->buffer_size) { + sis_update_sso(voice, voice->vperiod); + voice->vperiod = 0; + } else + sis_update_sso(voice, voice->period_size); + + sync = voice->sync_cso + voice->sync_period_size; + if (sync >= voice->sync_buffer_size) + sync -= voice->sync_buffer_size; + voice->sync_cso = sync; + } + + snd_pcm_period_elapsed(voice->substream); +} + +static void sis_voice_irq(u32 status, struct voice *voice) +{ + int bit; + + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); + voice++; + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + do { + status = inl(io + SIS_PISR_A); + if (status) { + sis_voice_irq(status, sis->voices); + outl(status, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + sis_voice_irq(status, &sis->voices[32]); + outl(status, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void __sis_map_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + if (!sis->silence_users) + sis->silence_dma_addr = pci_map_single(sis->pci, + sis->suspend_state[0], + 4096, PCI_DMA_TODEVICE); + sis->silence_users++; +} + +static void __sis_unmap_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + sis->silence_users--; + if (!sis->silence_users) + pci_unmap_single(sis->pci, sis->silence_dma_addr, 4096, + PCI_DMA_TODEVICE); +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + __sis_unmap_silence(sis); + voice->timing->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *__sis_alloc_playback_voice(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + struct voice *voice; + int i; + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + return voice; +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + voice = __sis_alloc_playback_voice(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + unsigned int period_size, buffer_size; + unsigned long flags; + int needed; + + /* If there are one or two periods per buffer, we don't need a + * timing voice, as we can use the capture channel's interrupts + * to clock out the periods. + */ + period_size = params_period_size(hw_params); + buffer_size = params_buffer_size(hw_params); + needed = (period_size != buffer_size && + period_size != (buffer_size / 2)); + + if (needed && !voice->timing) { + spin_lock_irqsave(&sis->voice_lock, flags); + voice->timing = __sis_alloc_playback_voice(sis); + if (voice->timing) + __sis_map_silence(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + /* The baseline setup is for a single period per buffer, and + * we add bells and whistles as needed from there. + */ + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + if (runtime->period_size == (runtime->buffer_size / 2)) { + control |= SIS_PLAY_DMA_INTR_AT_MLP; + } else if (runtime->period_size != runtime->buffer_size) { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size - 1; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control &= ~SIS_PLAY_DMA_INTR_AT_LEO; + control |= SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso |= (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + /* Ok, we're ready to go, set up the channel. + */ + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + /* No locks needed, as the PCM core will hold the locks on the + * substreams, and the HW will only start/stop the indicated voices + * without changing the state of the others. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static void sis_prepare_timing_voice(struct voice *voice, + struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *timing = voice->timing; + void __iomem *play_base = timing->ctrl_base; + void __iomem *wave_base = timing->wave_base; + u16 buffer_size, period_size; + u32 format, control, sso_eso, delta; + u32 vperiod, sso, reg; + + /* Set our initial buffer and period as large as we can given a + * single page of silence. + */ + buffer_size = 4096 / runtime->channels; + buffer_size /= snd_pcm_format_size(runtime->format, 1); + period_size = buffer_size; + + /* Initially, we want to interrupt just a bit behind the end of + * the period we're clocking out. 10 samples seems to give a good + * delay. + * + * We want to spread our interrupts throughout the virtual period, + * so that we don't end up with two interrupts back to back at the + * end -- this helps minimize the effects of any jitter. Adjust our + * clocking period size so that the last period is at least a fourth + * of a full period. + * + * This is all moot if we don't need to use virtual periods. + */ + vperiod = runtime->period_size + 10; + if (vperiod > period_size) { + u16 tail = vperiod % period_size; + u16 quarter_period = period_size / 4; + + if (tail && tail < quarter_period) { + u16 loops = vperiod / period_size; + + tail = quarter_period - tail; + tail += loops - 1; + tail /= loops; + period_size -= tail; + } + + sso = period_size - 1; + } else { + /* The initial period will fit inside the buffer, so we + * don't need to use virtual periods -- disable them. + */ + period_size = runtime->period_size; + sso = vperiod - 1; + vperiod = 0; + } + + /* The interrupt handler implements the timing syncronization, so + * setup its state. + */ + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_base = voice->ctrl_base; + timing->sync_cso = runtime->period_size - 1; + timing->sync_period_size = runtime->period_size; + timing->sync_buffer_size = runtime->buffer_size; + timing->period_size = period_size; + timing->buffer_size = buffer_size; + timing->sso = sso; + timing->vperiod = vperiod; + + /* Using unsigned samples with the all-zero silence buffer + * forces the output to the lower rail, killing playback. + * So ignore unsigned vs signed -- it doesn't change the timing. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + control = timing->buffer_size - 1; + control |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso = timing->buffer_size - 1; + sso_eso |= timing->sso << 16; + + delta = sis_rate_to_delta(runtime->rate); + + /* We've done the math, now configure the channel. + */ + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(sis->silence_dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(control, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *rec_base = voice->ctrl_base; + u32 format, dma_addr, control; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_CAPTURE_DMA_LOOP; + + /* If we've got more than two periods per buffer, then we have + * use a timing voice to clock out the periods. Otherwise, we can + * use the capture channel's interrupts. + */ + if (voice->timing) { + sis_prepare_timing_voice(voice, substream); + } else { + control |= SIS_CAPTURE_DMA_INTR_AT_LEO; + if (runtime->period_size != runtime->buffer_size) + control |= SIS_CAPTURE_DMA_INTR_AT_MLP; + } + + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(control, rec_base + SIS_CAPTURE_DMA_CONTROL); + + /* Force the writes to post. */ + readl(rec_base); + + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + /* We have 64 voices, and the driver currently records from + * only one channel, though that could change in the future. + */ + rc = snd_pcm_new(sis->card, "SiS7019", 0, 64, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + + /* Get the AC97 semaphore -- software first, so we don't spin + * pounding out IO reads on the hardware semaphore... + */ + mutex_lock(&sis->ac97_mutex); + + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + mutex_unlock(&sis->ac97_mutex); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static void sis_free_suspend(struct sis7019 *sis) +{ + int i; + + for (i = 0; i < SIS_SUSPEND_PAGES; i++) + kfree(sis->suspend_state[i]); +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + sis_free_suspend(sis); + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * + * FIXME: Maximum attenuation is set for the music amp, which will + * need to change if we start using the synth engine. + */ + outl(0xffff0000, io + SIS_WEVCR); + + /* Ensure that the wave engine is in normal operating mode. + */ + outl(0, io + SIS_WECCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +#ifdef CONFIG_PM +static int sis_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(sis->pcm); + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[2]); + + /* snd_pcm_suspend_all() stopped all channels, so we're quiescent. + */ + if (sis->irq >= 0) { + synchronize_irq(sis->irq); + free_irq(sis->irq, sis); + sis->irq = -1; + } + + /* Save the internal state away + */ + for (i = 0; i < 4; i++) { + memcpy_fromio(sis->suspend_state[i], ioaddr, 4096); + ioaddr += 4096; + } + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int sis_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "sis7019: unable to re-enable device\n"); + goto error; + } + + if (sis_chip_init(sis)) { + printk(KERN_ERR "sis7019: unable to re-init controller\n"); + goto error; + } + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); + goto error; + } + + /* Restore saved state, then clear out the page we use for the + * silence buffer. + */ + for (i = 0; i < 4; i++) { + memcpy_toio(ioaddr, sis->suspend_state[i], 4096); + ioaddr += 4096; + } + + memset(sis->suspend_state[0], 0, 4096); + + sis->irq = pci->irq; + pci_set_master(pci); + + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[2]); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + +error: + snd_card_disconnect(card); + return -EIO; +} +#endif /* CONFIG_PM */ + +static int sis_alloc_suspend(struct sis7019 *sis) +{ + int i; + + /* We need 16K to store the internal wave engine state during a + * suspend, but we don't need it to be contiguous, so play nice + * with the memory system. We'll also use this area for a silence + * buffer. + */ + for (i = 0; i < SIS_SUSPEND_PAGES; i++) { + sis->suspend_state[i] = kmalloc(4096, GFP_KERNEL); + if (!sis->suspend_state[i]) + return -ENOMEM; + } + memset(sis->suspend_state[0], 0, 4096); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "30-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + mutex_init(&sis->ac97_mutex); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap_nocache(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_alloc_suspend(sis); + if (rc < 0) { + printk(KERN_ERR "sis7019: unable to allocate state storage\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + int rc; + + rc = -ENOENT; + if (!enable) + goto error_out; + + rc = -ENOMEM; + card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + strcpy(card->driver, "SiS7019"); + strcpy(card->shortname, "SiS7019"); + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + +#ifdef CONFIG_PM + .suspend = sis_suspend, + .resume = sis_resume, +#endif +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); diff --git a/sound/pci/sis7019.h b/sound/pci/sis7019.h new file mode 100644 index 0000000..013b673 --- /dev/null +++ b/sound/pci/sis7019.h @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0x00ff0000 +#define SIS_MIXER_LEFT_NO_ATTEN 0x00ff0000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */ -- cgit v0.10.2 From f85bf29c9435baf927e1817e6b43c9429b84f822 Mon Sep 17 00:00:00 2001 From: Oliver Neukum Date: Fri, 14 Dec 2007 14:42:41 +0100 Subject: [ALSA] usb audio suspend support This patch implements suspend/resume support for USB audio devices. It works with the microphone in my camera. Signed-off-by: Oliver Neukum Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 967b823..c52461d 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2077,6 +2077,8 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, static int usb_audio_probe(struct usb_interface *intf, const struct usb_device_id *id); static void usb_audio_disconnect(struct usb_interface *intf); +static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message); +static int usb_audio_resume(struct usb_interface *intf); static struct usb_device_id usb_audio_ids [] = { #include "usbquirks.h" @@ -2092,6 +2094,8 @@ static struct usb_driver usb_audio_driver = { .name = "snd-usb-audio", .probe = usb_audio_probe, .disconnect = usb_audio_disconnect, + .suspend = usb_audio_suspend, + .resume = usb_audio_resume, .id_table = usb_audio_ids, }; @@ -3654,6 +3658,43 @@ static void usb_audio_disconnect(struct usb_interface *intf) dev_get_drvdata(&intf->dev)); } +static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) +{ + struct snd_usb_audio *chip = dev_get_drvdata(&intf->dev); + struct list_head *p; + struct snd_usb_stream *as; + + if (chip == (void *)-1L) + return 0; + + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + if (!chip->num_suspended_intf++) { + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + snd_pcm_suspend_all(as->pcm); + } + } + + return 0; +} + +static int usb_audio_resume(struct usb_interface *intf) +{ + struct snd_usb_audio *chip = dev_get_drvdata(&intf->dev); + + if (chip == (void *)-1L) + return 0; + if (--chip->num_suspended_intf) + return 0; + /* + * ALSA leaves material resumption to user space + * we just notify + */ + + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); + + return 0; +} static int __init snd_usb_audio_init(void) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 2272f45..7cf18c3 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -126,6 +126,7 @@ struct snd_usb_audio { u32 usb_id; int shutdown; int num_interfaces; + int num_suspended_intf; struct list_head pcm_list; /* list of pcm streams */ int pcm_devs; -- cgit v0.10.2 From 28e9e47384d333239a8335b439a92a13d29f91d6 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 17 Dec 2007 09:02:22 +0100 Subject: [ALSA] PCM - added back TSTAMP ioctl for PCM (for old alsa-lib binaries) Signed-off-by: Jaroslav Kysela diff --git a/include/sound/asound.h b/include/sound/asound.h index 475eb71..eda5c63 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -443,6 +443,7 @@ enum { enum { SNDRV_PCM_IOCTL_PVERSION = _IOR('A', 0x00, int), SNDRV_PCM_IOCTL_INFO = _IOR('A', 0x01, struct snd_pcm_info), + SNDRV_PCM_IOCTL_TSTAMP = _IOW('A', 0x02, int), SNDRV_PCM_IOCTL_TTSTAMP = _IOW('A', 0x03, int), SNDRV_PCM_IOCTL_HW_REFINE = _IOWR('A', 0x10, struct snd_pcm_hw_params), SNDRV_PCM_IOCTL_HW_PARAMS = _IOWR('A', 0x11, struct snd_pcm_hw_params), diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7fb7c92..2e7b1e6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2546,6 +2546,8 @@ static int snd_pcm_common_ioctl1(struct file *file, return put_user(SNDRV_PCM_VERSION, (int __user *)arg) ? -EFAULT : 0; case SNDRV_PCM_IOCTL_INFO: return snd_pcm_info_user(substream, arg); + case SNDRV_PCM_IOCTL_TSTAMP: /* just for compatibility */ + return 0; case SNDRV_PCM_IOCTL_TTSTAMP: return snd_pcm_tstamp(substream, arg); case SNDRV_PCM_IOCTL_HW_REFINE: -- cgit v0.10.2 From 5a7f261921cf482e17bb9d6641bef8f9c57b7409 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Dec 2007 11:44:25 +0100 Subject: [ALSA] Add SNDRV_PCM_IOCTL_TSTAMP back to compat ioctl The replaced one should be re-added for older alsa-lib. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 1fb6ae7..49aa693 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -483,6 +483,7 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l switch (cmd) { case SNDRV_PCM_IOCTL_PVERSION: case SNDRV_PCM_IOCTL_INFO: + case SNDRV_PCM_IOCTL_TSTAMP: case SNDRV_PCM_IOCTL_TTSTAMP: case SNDRV_PCM_IOCTL_HWSYNC: case SNDRV_PCM_IOCTL_PREPARE: -- cgit v0.10.2 From 38fcaf8efcac6f89dd3dafa3df17f49fcf3403ba Mon Sep 17 00:00:00 2001 From: Andrew Paprocki Date: Mon, 17 Dec 2007 11:49:44 +0100 Subject: [ALSA] hda-codec - Fix definition of AC_KNBCAP_DELTA to match spec AC_KNBCAP_DELTA is incorrectly defined as (1<<8). According to the Intel HDA spec, this is bit 7 after AC_KNBCAP_NUM_STEPS which is a 0x7f mask. Signed-off-by: Andrew Paprocki Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0331510..851abfe 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -234,7 +234,7 @@ enum { /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) -#define AC_KNBCAP_DELTA (1<<8) +#define AC_KNBCAP_DELTA (1<<7) /* * Control Parameters -- cgit v0.10.2 From 8e9068b1c7a154246f4cee93fd68c862b81b04e1 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 17 Dec 2007 11:58:13 +0100 Subject: [ALSA] hda: STAC927x DMIC Cleanup Cleaned up STAC927x and added several subsystem id's for more laptops. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8598951..1e2d3bf 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -107,6 +107,7 @@ enum { STAC_D965_3ST, STAC_D965_5ST, STAC_DELL_3ST, + STAC_DELL_BIOS, STAC_927X_MODELS }; @@ -1408,22 +1409,24 @@ static unsigned int d965_5st_pin_configs[14] = { static unsigned int dell_3st_pin_configs[14] = { 0x02211230, 0x02a11220, 0x01a19040, 0x01114210, 0x01111212, 0x01116211, 0x01813050, 0x01112214, - 0x403003fa, 0x40000100, 0x40000100, 0x404003fb, + 0x403003fa, 0x90a60040, 0x90a60040, 0x404003fb, 0x40c003fc, 0x40000100 }; static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { - [STAC_D965_REF] = ref927x_pin_configs, - [STAC_D965_3ST] = d965_3st_pin_configs, - [STAC_D965_5ST] = d965_5st_pin_configs, - [STAC_DELL_3ST] = dell_3st_pin_configs, + [STAC_D965_REF] = ref927x_pin_configs, + [STAC_D965_3ST] = d965_3st_pin_configs, + [STAC_D965_5ST] = d965_5st_pin_configs, + [STAC_DELL_3ST] = dell_3st_pin_configs, + [STAC_DELL_BIOS] = NULL, }; static const char *stac927x_models[STAC_927X_MODELS] = { - [STAC_D965_REF] = "ref", - [STAC_D965_3ST] = "3stack", - [STAC_D965_5ST] = "5stack", - [STAC_DELL_3ST] = "dell-3stack", + [STAC_D965_REF] = "ref", + [STAC_D965_3ST] = "3stack", + [STAC_D965_5ST] = "5stack", + [STAC_DELL_3ST] = "dell-3stack", + [STAC_DELL_BIOS] = "dell-bios", }; static struct snd_pci_quirk stac927x_cfg_tbl[] = { @@ -1450,13 +1453,21 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_D965_3ST), /* Dell 3 stack systems */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_3ST), + /* Dell 3 stack systems with verb table in BIOS */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS), /* 965 based 5 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), @@ -1992,6 +2003,7 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf for (i = 0; i < codec->num_nodes; i++) { wcaps = codec->wcaps[i]; wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wtype == AC_WID_AUD_OUT && !(wcaps & AC_WCAP_DIGITAL)) num_dacs++; } @@ -2079,7 +2091,6 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, wcaps = snd_hda_param_read(codec, conn[j], AC_PAR_AUDIO_WIDGET_CAP); wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) continue; @@ -3293,72 +3304,61 @@ static int patch_stac927x(struct hda_codec *codec) stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0) { - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC927x, using BIOS defaults\n"); + if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) { + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); if (err < 0) { stac92xx_free(codec); return err; } spec->pin_configs = spec->bios_pin_configs; - } else if (stac927x_brd_tbl[spec->board_config] != NULL) { + } else { spec->pin_configs = stac927x_brd_tbl[spec->board_config]; stac92xx_set_config_regs(codec); } + spec->adc_nids = stac927x_adc_nids; + spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + spec->mux_nids = stac927x_mux_nids; + spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); + spec->multiout.dac_nids = spec->dac_nids; + switch (spec->board_config) { case STAC_D965_3ST: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->init = d965_core_init; - spec->mixer = stac927x_mixer; - break; case STAC_D965_5ST: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->num_dmics = 0; + spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; - default: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->init = stac927x_core_init; - spec->mixer = stac927x_mixer; - } - - switch (codec->subsystem_id) { - case 0x10280242: /* STAC 9228 */ - case 0x102801f3: - case 0x1028020A: - case 0x10280209: + case STAC_DELL_BIOS: + case STAC_DELL_3ST: + /* GPIO2 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000004; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nids = stac927x_dmux_nids; - /* Enable DMIC0 */ - stac92xx_set_config_reg(codec, 0x13, 0x90a60040); - - /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000004; + spec->init = d965_core_init; + spec->mixer = stac927x_mixer; + spec->dmux_nids = stac927x_dmux_nids; break; default: - spec->num_dmics = 0; - /* GPIO0 High = Enable EAPD */ spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->num_dmics = 0; + + spec->init = stac927x_core_init; + spec->mixer = stac927x_mixer; } - spec->multiout.dac_nids = spec->dac_nids; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; + stac92xx_enable_gpio_mask(codec); - err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) { -- cgit v0.10.2 From 48ecb7e879ea172a0b3fd432ad49a870d7be6a59 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Dec 2007 14:32:49 +0100 Subject: [ALSA] hda-codec - Avoid overload of PCM volume on Cx5045 codec The PCM volume of Cx5045 codec has overload that isn't useful but rather harmful. Add a hack to override the amp info to set the max level 0 dB. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 68f23b8..a6916da 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -826,6 +826,17 @@ static int patch_cxt5045(struct hda_codec *codec) spec->init_verbs[0] = cxt5045_test_init_verbs; #endif } + + /* + * Fix max PCM level to 0 dB + * (originall it has 0x2b steps with 0dB offset 0x14) + */ + snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, + (0x14 << AC_AMPCAP_OFFSET_SHIFT) | + (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + return 0; } -- cgit v0.10.2 From f78dfac904325a996b8f97b045647898cc1ccc40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Dec 2007 16:24:04 +0100 Subject: [ALSA] Add missing device link Added the missing link to struct device from the card instance. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 55c6c82..d255503 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -352,6 +352,7 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) snprintf(card->longname, sizeof(card->longname), "%s (%s)", dev->dev.driver->name, card->mixername); + snd_card_set_dev(card, &dev->dev); ret = snd_card_register(card); if (ret == 0) { platform_set_drvdata(dev, card); diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index de683b0..5f5bbea 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -129,6 +129,8 @@ static int snd_pdacf_probe(struct pcmcia_device *link) return -ENODEV; } + snd_card_set_dev(card, &handle_to_dev(link)); + pdacf->index = i; card_list[i] = card; diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 27b6189..af812dc 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -954,6 +954,7 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) snd_ps3_init_avsetting(&the_card); /* register the card */ + snd_card_set_dev(the_card.card, &dev->core); ret = snd_card_register(the_card.card); if (ret < 0) goto clean_dma_map; -- cgit v0.10.2 From ac3e37412c195f1b48fe06327eb4ad0c072a1222 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Dec 2007 17:14:18 +0100 Subject: [ALSA] hda-codec - sort pci quirk list Sort pci quirk list in the order of PCI SSID. This makes easier to find out the buggy duplicated entries. Thanks to Andy Shevchenko for providing the sort script. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9b0ecbf..70a70c5 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -833,10 +833,9 @@ static const char *ad1986a_models[AD1986A_MODELS] = { static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD), + SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD), @@ -848,13 +847,14 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), + SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), + SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), @@ -1499,14 +1499,14 @@ static const char *ad1981_models[AD1981_MODELS] = { }; static struct snd_pci_quirk ad1981_cfg_tbl[] = { + SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), /* All HP models */ SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP), - /* HP nx6320 (reversed SSID, H/W bug) */ - SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP), + SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), /* Lenovo Thinkpad T60/X60/Z6xx */ SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1981_THINKPAD), - SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), + /* HP nx6320 (reversed SSID, H/W bug) */ + SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP), {} }; @@ -2727,8 +2727,8 @@ static const char *ad1988_models[AD1988_MODEL_LAST] = { }; static struct snd_pci_quirk ad1988_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), {} }; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a6916da..4609b55 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -757,10 +757,10 @@ static const char *cxt5045_models[CXT5045_MODELS] = { }; static struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), @@ -1267,9 +1267,9 @@ static const char *cxt5047_models[CXT5047_MODELS] = { static struct snd_pci_quirk cxt5047_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30a0, "HP DV1000", CXT5047_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cafdeb8..2bbf89e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2733,23 +2733,17 @@ static const char *alc880_models[ALC880_MODEL_LAST] = { }; static struct snd_pci_quirk alc880_cfg_tbl[] = { - /* Broken BIOS configuration */ - SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), - + SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810), SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST), - SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810), SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), - SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST), - SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V), SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), @@ -2764,54 +2758,50 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), - SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), - - SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), + SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), /* default ASUS */ SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), + SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), + SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST), SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST), - SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), - SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO), - SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO), - SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810), - SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), - SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), - SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO), + SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO), SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), - + SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), + SND_PCI_QUIRK(0x1584, 0x9054, "Uniwlll", ALC880_F1734), SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), - SND_PCI_QUIRK(0x1584, 0x9054, "Uniwlll", ALC880_F1734), - + SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810), + SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), + SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), - + SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG), - SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW), - - SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG), + SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), + SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ + SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), + SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG), SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG), SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), - + SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), /* default Intel */ + SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), + SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), {} }; @@ -4754,8 +4744,8 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X), SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), - SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V), + SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL), {} }; @@ -5753,16 +5743,16 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ - SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M), + SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), + SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ + SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), {} }; @@ -7261,50 +7251,50 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), - SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), + SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), + SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), + SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), SND_PCI_QUIRK(0x1458, 0xa002, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), - SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), - SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), - SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), + SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), - SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), - SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), - SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), - SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), + SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), + SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), - SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), - SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} }; @@ -8753,36 +8743,36 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), - SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), - SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL), - SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), + SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF), + SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF), + SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF), + SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735", ALC262_HP_TC_T5735), + SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), - SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), - SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), - SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), - SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), + SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), + SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), + SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), {} }; @@ -9554,12 +9544,12 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { }; static struct snd_pci_quirk alc268_cfg_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), + SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), + SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), - SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), {} }; @@ -11015,22 +11005,23 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST), SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), - SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), + SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), /* FIXME: the entry below breaks Toshiba A100 (model=auto works!) * Any other models that need this preset? */ /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */ - SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), - SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), + SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST), + SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST), SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), + SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), + SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP), + /* FIXME: the below seems conflict */ + /* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */ SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST), SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST), - SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST), - SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST), {} }; @@ -11748,21 +11739,20 @@ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { }; static struct snd_pci_quirk alc861vd_cfg_tbl[] = { + SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), + SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), - SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), - + SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/ SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), - SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), + SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), - SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), {} }; @@ -12708,8 +12698,8 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { }; static struct snd_pci_quirk alc662_cfg_tbl[] = { - SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), + SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), {} }; -- cgit v0.10.2 From f488d9fcc84692ca0060b4e16c1c61a8d514cea8 Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Mon, 17 Dec 2007 17:30:06 +0100 Subject: [ALSA] at73c213: replace spinlock in mixer functions with a mutex This patch fixes the locking bug in the at73c213 SPI sound driver. This bug was triggered because spinlocks were wrapped around the spi_sync call which might sleep. The fix was to add a mutex to the sound driver and replace the spinlocks in the mixer functions with mutex lock/unlock. Tested on STK1000/STK1002. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index bfe17b3..5e8cf9f 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include @@ -76,8 +77,10 @@ struct snd_at73c213 { u8 spi_rbuffer[2]; /* Image of the SPI registers in AT73C213. */ u8 reg_image[18]; - /* Protect registers against concurrent access. */ + /* Protect SSC registers against concurrent access. */ spinlock_t lock; + /* Protect mixer registers against concurrent access. */ + struct mutex mixer_lock; }; #define get_chip(card) ((struct snd_at73c213 *)card->private_data) @@ -398,7 +401,7 @@ static int snd_at73c213_mono_get(struct snd_kcontrol *kcontrol, int mask = (kcontrol->private_value >> 16) & 0xff; int invert = (kcontrol->private_value >> 24) & 0xff; - spin_lock_irq(&chip->lock); + mutex_lock(&chip->mixer_lock); ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & mask; @@ -407,7 +410,7 @@ static int snd_at73c213_mono_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - spin_unlock_irq(&chip->lock); + mutex_unlock(&chip->mixer_lock); return 0; } @@ -428,13 +431,13 @@ static int snd_at73c213_mono_put(struct snd_kcontrol *kcontrol, val = mask - val; val <<= shift; - spin_lock_irq(&chip->lock); + mutex_lock(&chip->mixer_lock); val = (chip->reg_image[reg] & ~(mask << shift)) | val; change = val != chip->reg_image[reg]; retval = snd_at73c213_write_reg(chip, reg, val); - spin_unlock_irq(&chip->lock); + mutex_unlock(&chip->mixer_lock); if (retval) return retval; @@ -470,7 +473,7 @@ static int snd_at73c213_stereo_get(struct snd_kcontrol *kcontrol, int mask = (kcontrol->private_value >> 24) & 0xff; int invert = (kcontrol->private_value >> 22) & 1; - spin_lock_irq(&chip->lock); + mutex_lock(&chip->mixer_lock); ucontrol->value.integer.value[0] = (chip->reg_image[left_reg] >> shift_left) & mask; @@ -484,7 +487,7 @@ static int snd_at73c213_stereo_get(struct snd_kcontrol *kcontrol, mask - ucontrol->value.integer.value[1]; } - spin_unlock_irq(&chip->lock); + mutex_unlock(&chip->mixer_lock); return 0; } @@ -511,7 +514,7 @@ static int snd_at73c213_stereo_put(struct snd_kcontrol *kcontrol, val1 <<= shift_left; val2 <<= shift_right; - spin_lock_irq(&chip->lock); + mutex_lock(&chip->mixer_lock); val1 = (chip->reg_image[left_reg] & ~(mask << shift_left)) | val1; val2 = (chip->reg_image[right_reg] & ~(mask << shift_right)) | val2; @@ -519,16 +522,16 @@ static int snd_at73c213_stereo_put(struct snd_kcontrol *kcontrol, || val2 != chip->reg_image[right_reg]; retval = snd_at73c213_write_reg(chip, left_reg, val1); if (retval) { - spin_unlock_irq(&chip->lock); + mutex_unlock(&chip->mixer_lock); goto out; } retval = snd_at73c213_write_reg(chip, right_reg, val2); if (retval) { - spin_unlock_irq(&chip->lock); + mutex_unlock(&chip->mixer_lock); goto out; } - spin_unlock_irq(&chip->lock); + mutex_unlock(&chip->mixer_lock); return change; @@ -546,7 +549,7 @@ static int snd_at73c213_mono_switch_get(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0xff; int invert = (kcontrol->private_value >> 24) & 0xff; - spin_lock_irq(&chip->lock); + mutex_lock(&chip->mixer_lock); ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & 0x01; @@ -555,7 +558,7 @@ static int snd_at73c213_mono_switch_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = 0x01 - ucontrol->value.integer.value[0]; - spin_unlock_irq(&chip->lock); + mutex_unlock(&chip->mixer_lock); return 0; } @@ -580,14 +583,14 @@ static int snd_at73c213_mono_switch_put(struct snd_kcontrol *kcontrol, val = mask - val; val <<= shift; - spin_lock_irq(&chip->lock); + mutex_lock(&chip->mixer_lock); val |= (chip->reg_image[reg] & ~(mask << shift)); change = val != chip->reg_image[reg]; retval = snd_at73c213_write_reg(chip, reg, val); - spin_unlock_irq(&chip->lock); + mutex_unlock(&chip->mixer_lock); if (retval) return retval; @@ -884,6 +887,7 @@ static int __devinit snd_at73c213_dev_init(struct snd_card *card, return irq; spin_lock_init(&chip->lock); + mutex_init(&chip->mixer_lock); chip->card = card; chip->irq = -1; -- cgit v0.10.2 From 561de31a23a8a5d23baeb8b2ce7c6102cb649616 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Tue, 18 Dec 2007 13:13:47 +0100 Subject: [ALSA] sound/: Spelling fixes Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 65de3a7..3958dbd 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -455,7 +455,7 @@ static void snd_uart16550_do_open(struct snd_uart16550 * uart) | UART_IER_THRI /* Enable Transmitter holding register empty interrupt */ ; } - outb(byte, uart->base + UART_IER); /* Interupt enable Register */ + outb(byte, uart->base + UART_IER); /* Interrupt enable Register */ inb(uart->base + UART_LSR); /* Clear any pre-existing overrun indication */ inb(uart->base + UART_IIR); /* Clear any pre-existing transmit interrupt */ @@ -473,7 +473,7 @@ static void snd_uart16550_do_close(struct snd_uart16550 * uart) outb((0 & UART_IER_RDI) /* Disable Receiver data interrupt */ |(0 & UART_IER_THRI) /* Disable Transmitter holding register empty interrupt */ - ,uart->base + UART_IER); /* Interupt enable Register */ + ,uart->base + UART_IER); /* Interrupt enable Register */ switch (uart->adaptor) { default: @@ -653,7 +653,7 @@ static void snd_uart16550_output_write(struct snd_rawmidi_substream *substream) char first; static unsigned long lasttime = 0; - /* Interupts are disabled during the updating of the tx_buff, + /* Interrupts are disabled during the updating of the tx_buff, * since it is 'bad' to have two processes updating the same * variables (ie buff_in & buff_out) */ diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 5d4f163..ece5041 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -163,7 +163,7 @@ struct snd_audiodrive { #define ES18XX_DUPLEX_SAME 0x0010 /* Playback and record must share the same rate */ #define ES18XX_NEW_RATE 0x0020 /* More precise rate setting */ #define ES18XX_AUXB 0x0040 /* AuxB mixer control */ -#define ES18XX_HWV 0x0080 /* Has seperate hardware volume mixer controls*/ +#define ES18XX_HWV 0x0080 /* Has separate hardware volume mixer controls*/ #define ES18XX_MONO 0x0100 /* Mono_in mixer control */ #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4a336ea..333c62d 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2395,7 +2395,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) if (!(hwread(vortex->mmio, VORTEX_STAT) & 0x1)) return IRQ_NONE; - // This is the Interrrupt Enable flag we set before (consistency check). + // This is the Interrupt Enable flag we set before (consistency check). if (!(hwread(vortex->mmio, VORTEX_CTRL) & CTRL_IRQ_ENABLE)) return IRQ_NONE; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 8c44fef..28f98bd 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -8,7 +8,7 @@ * - Sometimes the SPDIF input DSP tasks get's unsynchronized * and the SPDIF get somewhat "distorcionated", or/and left right channel * are swapped. To get around this problem when it happens, mute and unmute - * the SPDIF input mixer controll. + * the SPDIF input mixer control. * - On the Hercules Game Theater XP the amplifier are sometimes turned * off on inadecuate moments which causes distorcions on sound. * diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 851abfe..2e242db 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -418,7 +418,7 @@ struct hda_bus_ops { /* free the private data */ void (*private_free)(struct hda_bus *); #ifdef CONFIG_SND_HDA_POWER_SAVE - /* notify power-up/down from codec to contoller */ + /* notify power-up/down from codec to controller */ void (*pm_notify)(struct hda_codec *codec); #endif }; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7956b24..67ec08c 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3604,7 +3604,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) /* ASSUMPTION: hdsp->lock is either held, or there is no need to hold it (e.g. during module - initalization). + initialization). */ /* set defaults: diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f1bdda6..6b1d529 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3348,7 +3348,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm) unsigned int i; /* ASSUMPTION: hdspm->lock is either held, or there is no need to - hold it (e.g. during module initalization). + hold it (e.g. during module initialization). */ /* set defaults: */ @@ -3416,7 +3416,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm) /*------------------------------------------------------------ - interupt + interrupt ------------------------------------------------------------*/ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 34f96f1..1d73be6 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -148,7 +148,7 @@ MODULE_SUPPORTED_DEVICE("{{RME,Hammerfall}," #define RME9652_start_bit (1<<0) /* start record/play */ /* bits 1-3 encode buffersize/latency */ #define RME9652_Master (1<<4) /* Clock Mode Master=1,Slave/Auto=0 */ -#define RME9652_IE (1<<5) /* Interupt Enable */ +#define RME9652_IE (1<<5) /* Interrupt Enable */ #define RME9652_freq (1<<6) /* samplerate 0=44.1/88.2, 1=48/96 kHz */ #define RME9652_freq1 (1<<7) /* if 0, 32kHz, else always 1 */ #define RME9652_DS (1<<8) /* Doule Speed 0=44.1/48, 1=88.2/96 Khz */ @@ -1826,7 +1826,7 @@ static void snd_rme9652_set_defaults(struct snd_rme9652 *rme9652) /* ASSUMPTION: rme9652->lock is either held, or there is no need to hold it (e.g. during module - initalization). + initialization). */ /* set defaults: diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 59a3195..c8d5665 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -436,7 +436,7 @@ static void snd_trident_free_synth_channel(struct snd_trident *trident, int chan Description: This routine will complete and write the 5 hardware channel registers to hardware. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure Each register field. @@ -514,7 +514,7 @@ EXPORT_SYMBOL(snd_trident_write_voice_regs); Description: This routine will write the new CSO offset register to hardware. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure CSO - new CSO value @@ -540,7 +540,7 @@ static void snd_trident_write_cso_reg(struct snd_trident * trident, Description: This routine will write the new ESO offset register to hardware. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure ESO - new ESO value @@ -566,7 +566,7 @@ static void snd_trident_write_eso_reg(struct snd_trident * trident, Description: This routine will write the new voice volume register to hardware. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure Vol - new voice volume @@ -597,7 +597,7 @@ static void snd_trident_write_vol_reg(struct snd_trident * trident, Description: This routine will write the new voice pan register to hardware. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure Pan - new pan value @@ -619,7 +619,7 @@ static void snd_trident_write_pan_reg(struct snd_trident * trident, Description: This routine will write the new reverb volume register to hardware. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure RVol - new reverb volume @@ -643,7 +643,7 @@ static void snd_trident_write_rvol_reg(struct snd_trident * trident, Description: This routine will write the new chorus volume register to hardware. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure CVol - new chorus volume @@ -666,7 +666,7 @@ static void snd_trident_write_cvol_reg(struct snd_trident * trident, Description: This routine converts rate in HZ to hardware delta value. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. rate - Real or Virtual channel number. Returns: Delta value. @@ -696,7 +696,7 @@ static unsigned int snd_trident_convert_rate(unsigned int rate) Description: This routine converts rate in HZ to hardware delta value. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. rate - Real or Virtual channel number. Returns: Delta value. @@ -726,7 +726,7 @@ static unsigned int snd_trident_convert_adc_rate(unsigned int rate) Description: This routine converts rate in HZ to spurious threshold. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. rate - Real or Virtual channel number. Returns: Delta value. @@ -748,7 +748,7 @@ static unsigned int snd_trident_spurious_threshold(unsigned int rate, Description: This routine returns a control mode for a PCM channel. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. substream - PCM substream Returns: Control value. @@ -781,7 +781,7 @@ static unsigned int snd_trident_control_mode(struct snd_pcm_substream *substream Description: Device I/O control handler for playback/capture parameters. - Paramters: substream - PCM substream class + Parameters: substream - PCM substream class cmd - what ioctl message to process arg - additional message infoarg @@ -1664,7 +1664,7 @@ static snd_pcm_uframes_t snd_trident_playback_pointer(struct snd_pcm_substream * Description: This routine return the capture position - Paramters: pcm1 - PCM device class + Parameters: pcm1 - PCM device class Returns: position of buffer @@ -2157,7 +2157,7 @@ static struct snd_pcm_ops snd_trident_spdif_7018_ops = { Description: This routine registers the 4DWave device for PCM support. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. Returns: None @@ -2215,7 +2215,7 @@ int __devinit snd_trident_pcm(struct snd_trident * trident, Description: This routine registers the 4DWave device for foldback PCM support. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. Returns: None @@ -2272,7 +2272,7 @@ int __devinit snd_trident_foldback_pcm(struct snd_trident * trident, Description: This routine registers the 4DWave-NX device for SPDIF support. - Paramters: trident - pointer to target device class for 4DWave-NX. + Parameters: trident - pointer to target device class for 4DWave-NX. Returns: None @@ -2956,7 +2956,7 @@ static int snd_trident_pcm_mixer_free(struct snd_trident *trident, struct snd_tr Description: This routine registers the 4DWave device for mixer support. - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. Returns: None @@ -3338,7 +3338,7 @@ static int snd_trident_dev_free(struct snd_device *device) Description: Allocate and set up the TLB page table on 4D NX. Each entry has 4 bytes (physical PCI address). - Paramters: trident - pointer to target device class for 4DWave. + Parameters: trident - pointer to target device class for 4DWave. Returns: 0 or negative error code @@ -3515,7 +3515,7 @@ static int snd_trident_sis_init(struct snd_trident *trident) Description: This routine will create the device specific class for the 4DWave card. It will also perform basic initialization. - Paramters: card - which card to create + Parameters: card - which card to create pci - interface to PCI bus resource info dma1ptr - playback dma buffer dma2ptr - capture dma buffer @@ -3661,7 +3661,7 @@ int __devinit snd_trident_create(struct snd_card *card, Description: This routine will free the device specific class for the 4DWave card. - Paramters: trident - device specific private data for 4DWave card + Parameters: trident - device specific private data for 4DWave card Returns: None. @@ -3699,7 +3699,7 @@ static int snd_trident_free(struct snd_trident *trident) Description: ISR for Trident 4DWave device - Paramters: trident - device specific private data for 4DWave card + Parameters: trident - device specific private data for 4DWave card Problems: It seems that Trident chips generates interrupts more than one time in special cases. The spurious interrupts are -- cgit v0.10.2 From b7d2a8035a382ad268aba8c0612797b4f2625f61 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Tue, 18 Dec 2007 13:14:21 +0100 Subject: [ALSA] include/sound/: Spelling fixes Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/ad1848.h b/include/sound/ad1848.h index d04f9e7..d9aebdf 100644 --- a/include/sound/ad1848.h +++ b/include/sound/ad1848.h @@ -48,7 +48,7 @@ #define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ #define AD1848_PIN_CTRL 0x0a /* pin control */ #define AD1848_TEST_INIT 0x0b /* test and initialization */ -#define AD1848_MISC_INFO 0x0c /* miscellaneaous information */ +#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ #define AD1848_LOOPBACK 0x0d /* loopback control */ #define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ #define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h index f149026..e8d1f3e 100644 --- a/include/sound/cs4231-regs.h +++ b/include/sound/cs4231-regs.h @@ -45,7 +45,7 @@ #define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ #define CS4231_PIN_CTRL 0x0a /* pin control */ #define CS4231_TEST_INIT 0x0b /* test and initialization */ -#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */ +#define CS4231_MISC_INFO 0x0c /* miscellaneous information */ #define CS4231_LOOPBACK 0x0d /* loopback control */ #define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */ #define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */ diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2b1ae8e..b9d5864 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -22,7 +22,7 @@ #define SND_SOC_NOPM -1 /* - * SoC dynamic audio power managment + * SoC dynamic audio power management * * We can have upto 4 power domains * 1. Codec domain - VREF, VMID -- cgit v0.10.2 From 8432395fd9124aa9408f61c94aa743878b4ddaf9 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 18 Dec 2007 15:42:53 +0100 Subject: [ALSA] cs4270: wrong sample rate when CONFIG_SND_SOC_CS4270_VD33_ERRATA is set When CONFIG_SND_SOC_CS4270_VD33_ERRATA is set, there was a mismatch between the mclk_ratios[] and cs4270_mode_ratios[] arrays. The two arrays have been merged and code has been shuffled. One side effect is that the cs4270_set_dai_sysclk() and cs4270_set_dai_fmt() functions are available only if I2C has been enabled. Signed-off-by: Timur Tabi Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index dab22cc..968eda3 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -48,12 +48,130 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ }; -/* The number of MCLK/LRCK ratios supported by the CS4270 */ -#define NUM_MCLK_RATIOS 9 +/* + * The codec isn't really big-endian or little-endian, since the I2S + * interface requires data to be sent serially with the MSbit first. + * However, to support BE and LE I2S devices, we specify both here. That + * way, ALSA will always match the bit patterns. + */ +#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#ifdef USE_I2C + +/* CS4270 registers addresses */ +#define CS4270_CHIPID 0x01 /* Chip ID */ +#define CS4270_PWRCTL 0x02 /* Power Control */ +#define CS4270_MODE 0x03 /* Mode Control */ +#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ +#define CS4270_TRANS 0x05 /* Transition Control */ +#define CS4270_MUTE 0x06 /* Mute Control */ +#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ +#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ + +#define CS4270_FIRSTREG 0x01 +#define CS4270_LASTREG 0x08 +#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) -/* The actual MCLK/LRCK ratios, in increasing numerical order */ -static unsigned int mclk_ratios[NUM_MCLK_RATIOS] = - {64, 96, 128, 192, 256, 384, 512, 768, 1024}; +/* Bit masks for the CS4270 registers */ +#define CS4270_CHIPID_ID 0xF0 +#define CS4270_CHIPID_REV 0x0F +#define CS4270_PWRCTL_FREEZE 0x80 +#define CS4270_PWRCTL_PDN_ADC 0x20 +#define CS4270_PWRCTL_PDN_DAC 0x02 +#define CS4270_PWRCTL_PDN 0x01 +#define CS4270_MODE_SPEED_MASK 0x30 +#define CS4270_MODE_1X 0x00 +#define CS4270_MODE_2X 0x10 +#define CS4270_MODE_4X 0x20 +#define CS4270_MODE_SLAVE 0x30 +#define CS4270_MODE_DIV_MASK 0x0E +#define CS4270_MODE_DIV1 0x00 +#define CS4270_MODE_DIV15 0x02 +#define CS4270_MODE_DIV2 0x04 +#define CS4270_MODE_DIV3 0x06 +#define CS4270_MODE_DIV4 0x08 +#define CS4270_MODE_POPGUARD 0x01 +#define CS4270_FORMAT_FREEZE_A 0x80 +#define CS4270_FORMAT_FREEZE_B 0x40 +#define CS4270_FORMAT_LOOPBACK 0x20 +#define CS4270_FORMAT_DAC_MASK 0x18 +#define CS4270_FORMAT_DAC_LJ 0x00 +#define CS4270_FORMAT_DAC_I2S 0x08 +#define CS4270_FORMAT_DAC_RJ16 0x18 +#define CS4270_FORMAT_DAC_RJ24 0x10 +#define CS4270_FORMAT_ADC_MASK 0x01 +#define CS4270_FORMAT_ADC_LJ 0x00 +#define CS4270_FORMAT_ADC_I2S 0x01 +#define CS4270_TRANS_ONE_VOL 0x80 +#define CS4270_TRANS_SOFT 0x40 +#define CS4270_TRANS_ZERO 0x20 +#define CS4270_TRANS_INV_ADC_A 0x08 +#define CS4270_TRANS_INV_ADC_B 0x10 +#define CS4270_TRANS_INV_DAC_A 0x02 +#define CS4270_TRANS_INV_DAC_B 0x04 +#define CS4270_TRANS_DEEMPH 0x01 +#define CS4270_MUTE_AUTO 0x20 +#define CS4270_MUTE_ADC_A 0x08 +#define CS4270_MUTE_ADC_B 0x10 +#define CS4270_MUTE_POLARITY 0x04 +#define CS4270_MUTE_DAC_A 0x01 +#define CS4270_MUTE_DAC_B 0x02 + +/* + * Clock Ratio Selection for Master Mode with I2C enabled + * + * The data for this chart is taken from Table 5 of the CS4270 reference + * manual. + * + * This table is used to determine how to program the Mode Control register. + * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + * + * Each element in this array corresponds to the ratios in mclk_ratios[]. + * These two arrays need to be in sync. + * + * 'speed_mode' is the corresponding bit pattern to be written to the + * MODE bits of the Mode Control Register + * + * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * the Mode Control Register. + * + * In situations where a single ratio is represented by multiple speed + * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick + * double-speed instead of quad-speed. However, the CS4270 errata states + * that Divide-By-1.5 can cause failures, so we avoid that mode where + * possible. + * + * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not + * work if VD = 3.3V. If this effects you, select the + * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will + * never select any sample rates that require divide-by-1.5. + */ +static struct { + unsigned int ratio; + u8 speed_mode; + u8 mclk; +} cs4270_mode_ratios[] = { + {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, +#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA + {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, +#endif + {128, CS4270_MODE_2X, CS4270_MODE_DIV1}, + {192, CS4270_MODE_4X, CS4270_MODE_DIV3}, + {256, CS4270_MODE_1X, CS4270_MODE_DIV1}, + {384, CS4270_MODE_2X, CS4270_MODE_DIV3}, + {512, CS4270_MODE_1X, CS4270_MODE_DIV2}, + {768, CS4270_MODE_1X, CS4270_MODE_DIV3}, + {1024, CS4270_MODE_1X, CS4270_MODE_DIV4} +}; + +/* The number of MCLK/LRCK ratios supported by the CS4270 */ +#define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) /* * Determine the CS4270 samples rates. @@ -97,7 +215,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, cs4270->mclk = freq; for (i = 0; i < NUM_MCLK_RATIOS; i++) { - unsigned int rate = freq / mclk_ratios[i]; + unsigned int rate = freq / cs4270_mode_ratios[i].ratio; rates |= snd_pcm_rate_to_rate_bit(rate); if (rate < rate_min) rate_min = rate; @@ -155,80 +273,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, } /* - * The codec isn't really big-endian or little-endian, since the I2S - * interface requires data to be sent serially with the MSbit first. - * However, to support BE and LE I2S devices, we specify both here. That - * way, ALSA will always match the bit patterns. - */ -#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) - -#ifdef USE_I2C - -/* CS4270 registers addresses */ -#define CS4270_CHIPID 0x01 /* Chip ID */ -#define CS4270_PWRCTL 0x02 /* Power Control */ -#define CS4270_MODE 0x03 /* Mode Control */ -#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ -#define CS4270_TRANS 0x05 /* Transition Control */ -#define CS4270_MUTE 0x06 /* Mute Control */ -#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ -#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ - -#define CS4270_FIRSTREG 0x01 -#define CS4270_LASTREG 0x08 -#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) - -/* Bit masks for the CS4270 registers */ -#define CS4270_CHIPID_ID 0xF0 -#define CS4270_CHIPID_REV 0x0F -#define CS4270_PWRCTL_FREEZE 0x80 -#define CS4270_PWRCTL_PDN_ADC 0x20 -#define CS4270_PWRCTL_PDN_DAC 0x02 -#define CS4270_PWRCTL_PDN 0x01 -#define CS4270_MODE_SPEED_MASK 0x30 -#define CS4270_MODE_1X 0x00 -#define CS4270_MODE_2X 0x10 -#define CS4270_MODE_4X 0x20 -#define CS4270_MODE_SLAVE 0x30 -#define CS4270_MODE_DIV_MASK 0x0E -#define CS4270_MODE_DIV1 0x00 -#define CS4270_MODE_DIV15 0x02 -#define CS4270_MODE_DIV2 0x04 -#define CS4270_MODE_DIV3 0x06 -#define CS4270_MODE_DIV4 0x08 -#define CS4270_MODE_POPGUARD 0x01 -#define CS4270_FORMAT_FREEZE_A 0x80 -#define CS4270_FORMAT_FREEZE_B 0x40 -#define CS4270_FORMAT_LOOPBACK 0x20 -#define CS4270_FORMAT_DAC_MASK 0x18 -#define CS4270_FORMAT_DAC_LJ 0x00 -#define CS4270_FORMAT_DAC_I2S 0x08 -#define CS4270_FORMAT_DAC_RJ16 0x18 -#define CS4270_FORMAT_DAC_RJ24 0x10 -#define CS4270_FORMAT_ADC_MASK 0x01 -#define CS4270_FORMAT_ADC_LJ 0x00 -#define CS4270_FORMAT_ADC_I2S 0x01 -#define CS4270_TRANS_ONE_VOL 0x80 -#define CS4270_TRANS_SOFT 0x40 -#define CS4270_TRANS_ZERO 0x20 -#define CS4270_TRANS_INV_ADC_A 0x08 -#define CS4270_TRANS_INV_ADC_B 0x10 -#define CS4270_TRANS_INV_DAC_A 0x02 -#define CS4270_TRANS_INV_DAC_B 0x04 -#define CS4270_TRANS_DEEMPH 0x01 -#define CS4270_MUTE_AUTO 0x20 -#define CS4270_MUTE_ADC_A 0x08 -#define CS4270_MUTE_ADC_B 0x10 -#define CS4270_MUTE_POLARITY 0x04 -#define CS4270_MUTE_DAC_A 0x01 -#define CS4270_MUTE_DAC_B 0x02 - -/* * A list of addresses on which this CS4270 could use. I2C addresses are * 7 bits. For the CS4270, the upper four bits are always 1001, and the * lower three bits are determined via the AD2, AD1, and AD0 pins @@ -315,53 +359,6 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, } /* - * Clock Ratio Selection for Master Mode with I2C enabled - * - * The data for this chart is taken from Table 5 of the CS4270 reference - * manual. - * - * This table is used to determine how to program the Mode Control register. - * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling - * rates the CS4270 currently supports. - * - * Each element in this array corresponds to the ratios in mclk_ratios[]. - * These two arrays need to be in sync. - * - * 'speed_mode' is the corresponding bit pattern to be written to the - * MODE bits of the Mode Control Register - * - * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of - * the Mode Control Register. - * - * In situations where a single ratio is represented by multiple speed - * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick - * double-speed instead of quad-speed. However, the CS4270 errata states - * that Divide-By-1.5 can cause failures, so we avoid that mode where - * possible. - * - * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not - * work if VD = 3.3V. If this effects you, select the - * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will - * never select any sample rates that require divide-by-1.5. - */ -static struct { - u8 speed_mode; - u8 mclk; -} cs4270_mode_ratios[NUM_MCLK_RATIOS] = { - {CS4270_MODE_4X, CS4270_MODE_DIV1}, /* 64 */ -#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA - {CS4270_MODE_4X, CS4270_MODE_DIV15}, /* 96 */ -#endif - {CS4270_MODE_2X, CS4270_MODE_DIV1}, /* 128 */ - {CS4270_MODE_4X, CS4270_MODE_DIV3}, /* 192 */ - {CS4270_MODE_1X, CS4270_MODE_DIV1}, /* 256 */ - {CS4270_MODE_2X, CS4270_MODE_DIV3}, /* 384 */ - {CS4270_MODE_1X, CS4270_MODE_DIV2}, /* 512 */ - {CS4270_MODE_1X, CS4270_MODE_DIV3}, /* 768 */ - {CS4270_MODE_1X, CS4270_MODE_DIV4} /* 1024 */ -}; - -/* * Program the CS4270 with the given hardware parameters. * * The .dai_ops functions are used to provide board-specific data, like @@ -388,7 +385,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ratio = cs4270->mclk / rate; /* MCLK/LRCK ratio */ for (i = 0; i < NUM_MCLK_RATIOS; i++) { - if (mclk_ratios[i] == ratio) + if (cs4270_mode_ratios[i].ratio == ratio) break; } @@ -669,7 +666,7 @@ error: return ret; } -#endif +#endif /* USE_I2C*/ struct snd_soc_codec_dai cs4270_dai = { .name = "CS4270", @@ -687,10 +684,6 @@ struct snd_soc_codec_dai cs4270_dai = { .rates = 0, .formats = CS4270_FORMATS, }, - .dai_ops = { - .set_sysclk = cs4270_set_dai_sysclk, - .set_fmt = cs4270_set_dai_fmt, - } }; EXPORT_SYMBOL_GPL(cs4270_dai); @@ -752,6 +745,8 @@ static int cs4270_probe(struct platform_device *pdev) if (codec->control_data) { /* Initialize codec ops */ cs4270_dai.ops.hw_params = cs4270_hw_params; + cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt; #ifdef CONFIG_SND_SOC_CS4270_HWMUTE cs4270_dai.dai_ops.digital_mute = cs4270_mute; #endif -- cgit v0.10.2 From 1697055e6c82ee5e99f459c15619605782eb7fcc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Dec 2007 18:05:52 +0100 Subject: [ALSA] hda-codec - Fix invalid access to non-existing dmux on STAC The digital mux on STAC codecs doesn't always exist although the driver builds dmux enum mixer elements unconditionally. Now the driver creates 'digital input source' mixer elements only when dmux is available. Also, the patch adds the missing dmux definition for STAC925x. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1e2d3bf..a074155 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -139,6 +139,7 @@ struct sigmatel_spec { hda_nid_t *dmic_nids; unsigned int num_dmics; hda_nid_t *dmux_nids; + unsigned int num_dmuxes; hda_nid_t dig_in_nid; /* pin widgets */ @@ -244,6 +245,10 @@ static hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = { 0x15, 0 }; +static hda_nid_t stac925x_dmux_nids[1] = { + 0x14, +}; + static hda_nid_t stac922x_adc_nids[2] = { 0x06, 0x07, }; @@ -278,7 +283,7 @@ static hda_nid_t stac9205_mux_nids[2] = { }; static hda_nid_t stac9205_dmux_nids[1] = { - 0x1d, + 0x1d, }; #define STAC9205_NUM_DMICS 2 @@ -596,16 +601,6 @@ static struct hda_verb stac9205_core_init[] = { {} }; -#define STAC_DIGITAL_INPUT_SOURCE(cnt) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Digital Input Source", \ - .count = cnt, \ - .info = stac92xx_dmux_enum_info, \ - .get = stac92xx_dmux_enum_get, \ - .put = stac92xx_dmux_enum_put,\ - } - #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -638,7 +633,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { }; static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { - STAC_DIGITAL_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), /* hardware gain controls */ @@ -669,7 +663,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { }; static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { - STAC_DIGITAL_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), /* hardware gain controls */ @@ -700,7 +693,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { }; static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { - STAC_DIGITAL_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), /* hardware gain controls */ @@ -731,7 +723,6 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { - STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), /* hardware gain controls */ @@ -752,7 +743,6 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { - STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), @@ -779,7 +769,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = { }; static struct snd_kcontrol_new stac9205_mixer[] = { - STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), @@ -809,7 +798,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { - STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), @@ -827,6 +815,15 @@ static struct snd_kcontrol_new stac927x_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac_dmux_mixer = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Input Source", + /* count set later */ + .info = stac92xx_dmux_enum_info, + .get = stac92xx_dmux_enum_get, + .put = stac92xx_dmux_enum_put, +}; + static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -842,6 +839,13 @@ static int stac92xx_build_controls(struct hda_codec *codec) if (err < 0) return err; } + if (spec->num_dmuxes > 0) { + stac_dmux_mixer.count = spec->num_dmuxes; + err = snd_ctl_add(codec->bus->card, + snd_ctl_new1(&stac_dmux_mixer, codec)); + if (err < 0) + return err; + } if (spec->multiout.dig_out_nid) { err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); @@ -2967,6 +2971,8 @@ static int patch_stac925x(struct hda_codec *codec) case 0x83847637: /* STAC9251D */ spec->num_dmics = STAC925X_NUM_DMICS; spec->dmic_nids = stac925x_dmic_nids; + spec->num_dmuxes = ARRAY_SIZE(stac925x_dmux_nids); + spec->dmux_nids = stac925x_dmux_nids; break; default: spec->num_dmics = 0; @@ -3075,6 +3081,7 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ spec->gpio_mask = spec->gpio_data = 0x000001; @@ -3160,6 +3167,7 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); spec->num_dmics = STAC92HD71BXX_NUM_DMICS; + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); spec->multiout.num_dacs = 2; spec->multiout.hp_nid = 0x11; @@ -3345,6 +3353,7 @@ static int patch_stac927x(struct hda_codec *codec) spec->init = d965_core_init; spec->mixer = stac927x_mixer; spec->dmux_nids = stac927x_dmux_nids; + spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); break; default: /* GPIO0 High = Enable EAPD */ @@ -3415,6 +3424,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->dmic_nids = stac9205_dmic_nids; spec->num_dmics = STAC9205_NUM_DMICS; spec->dmux_nids = stac9205_dmux_nids; + spec->num_dmuxes = ARRAY_SIZE(stac9205_dmux_nids); spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; -- cgit v0.10.2 From 3982d17e3d38850908ed3400cb5a68fdc623877d Mon Sep 17 00:00:00 2001 From: Andrew Paprocki Date: Wed, 19 Dec 2007 12:13:44 +0100 Subject: [ALSA] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 23d3bef..cd80719 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1435,7 +1435,8 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) return err; } codec->spdif_ctls = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } @@ -1482,7 +1483,7 @@ static int snd_hda_spdif_in_status_get(struct snd_kcontrol *kcontrol, unsigned short val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -1533,7 +1534,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) return err; } codec->spdif_in_enable = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0) & AC_DIG1_ENABLE; return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2e242db..20be776 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -77,12 +77,14 @@ enum { #define AC_VERB_GET_PIN_SENSE 0x0f09 #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -110,6 +112,7 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -190,6 +193,10 @@ enum { #define AC_PINCAP_OUT (1<<4) /* output capable */ #define AC_PINCAP_IN (1<<5) /* input capable */ #define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification, + * but is marked reserved in the Intel HDA specification. + */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2bbf89e..c251974 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -572,7 +572,7 @@ static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, 0x00); + AC_VERB_GET_DIGI_CONVERT_1, 0x00); *valp = (val & mask) != 0; return 0; @@ -586,7 +586,7 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long val = *ucontrol->value.integer.value; unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); /* Set/unset the masked control bit(s) as needed */ -- cgit v0.10.2 From 358ce0cf3f147802ef81aa53d5b341f633d608d1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 19 Dec 2007 14:25:24 +0100 Subject: [ALSA] usb-audio: add UR-80 PCM quirk Add a quirk entry to handle Edirol UR-80 audio I/O. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 59410f4..938dff5 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1004,11 +1004,35 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* has ID 0x0049 when not in "Advanced Driver" mode */ + USB_DEVICE(0x0582, 0x0047), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "EDIROL", */ + /* .product_name = "UR-80", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + /* in the 96 kHz modes, only interface 1 is there */ + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, +{ /* has ID 0x004a when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0048), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "EDIROL", - .product_name = "UR-80", + /* .vendor_name = "EDIROL", */ + /* .product_name = "UR-80", */ .ifnum = 0, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { -- cgit v0.10.2 From aa9673cf31c7daca42a9713c8aa604bce280e85d Mon Sep 17 00:00:00 2001 From: Harald Welte Date: Wed, 19 Dec 2007 15:37:49 +0100 Subject: [ALSA] neo1973: ASoC include pathname fix Fix s3c24xx include file path changes in asoc driver Signed-off-by: Harald Welte Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index f1f6b94..bde4cdc 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -30,13 +30,15 @@ #include #include -#include #include #include #include #include #include #include + +#include + #include "../codecs/wm8753.h" #include "lm4857.h" #include "s3c24xx-pcm.h" diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index cd89c41..fe30e0d 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -33,13 +33,14 @@ #include #include -#include #include #include #include #include #include +#include + #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -- cgit v0.10.2 From 6e7939bb4de289d4e5176b08f1d00e42e0a07d77 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 19 Dec 2007 17:49:02 +0100 Subject: [ALSA] hda-codec - Fix capture mixers of ALC662 models The commit that added support for ASUS P701 eeepc also changed the mixers of other ALC662 models, duplicating entries for the Capture items, making them to not work anymore. This fixes it by removing duplicated entries using where possible the common alc662_capture_mixer. Also alc662_capture_mixer should use alc662* functions and not alc882 (I checked /proc/asound/card0/codec* on an eepc model and it's ok). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c251974..a9d7fe9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12335,18 +12335,6 @@ static struct snd_kcontrol_new alc662_base_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT), - - /* Capture mixer control */ - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, { } /* end */ }; @@ -12364,17 +12352,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12398,17 +12375,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12422,17 +12388,6 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12598,9 +12553,9 @@ static struct snd_kcontrol_new alc662_capture_mixer[] = { /* .name = "Capture Source", */ .name = "Input Source", .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, + .info = alc662_mux_enum_info, + .get = alc662_mux_enum_get, + .put = alc662_mux_enum_put, }, { } /* end */ }; -- cgit v0.10.2 From 6b7eb1960d3fd713205277e86da72a6e3979c9ac Mon Sep 17 00:00:00 2001 From: Jiang Zhe Date: Thu, 20 Dec 2007 13:01:28 +0100 Subject: [ALSA] hda-codec - Device ID for Toshiba laptop which uses AD1986A The model laptop-eapd get rid of the high-pitched noise. (ALSA bug#3662) Signed-off-by: Jiang Zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 70a70c5..5e70d55 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -859,6 +859,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP), + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), {} }; -- cgit v0.10.2 From a9b3aa8a0a203b9b62e15c465ba7d4797a6a2c79 Mon Sep 17 00:00:00 2001 From: Jiang Zhe Date: Thu, 20 Dec 2007 13:13:13 +0100 Subject: [ALSA] hda-codec - alc268 input_mux should be a selector instead of mixer According to the [0003659], the node 0x23,0x24 is a selector. I checked the alc268 spec on the REALTEK website and it showed that they were selectors indeed. However, current code implement the alc268 input_mux in a mixer way. Signed-off-by: Jiang Zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a9d7fe9..617f3ee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9196,19 +9196,13 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1c, 14, 15, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + /* Unmute Selector 23h,24h and set the default input to mic-in */ + + {0x23, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x24, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, { } }; @@ -9253,29 +9247,14 @@ static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); static hda_nid_t capture_mixers[3] = { 0x23, 0x24 }; hda_nid_t nid = capture_mixers[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, - idx ); - } - *cur_val = idx; - return 1; + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + nid, + &spec->cur_mux[adc_idx]); } static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { -- cgit v0.10.2 From d0ce9946c52e7bdf95afb09553775cf28b752254 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 23 Dec 2007 19:50:57 +0100 Subject: [ALSA] add CMI8788 driver Add the snd-oxygen driver for the C-Media CMI8788 (Oxygen) chip, used on the Asound A-8788, AuzenTech X-Meridian, Bgears b-Enspirer, Club3D Theatron DTS, HT-Omega Claro, Razer Barracuda AC-1, Sondigo Inferno, and TempoTec HIFIER sound cards. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 45f0f6c..d3be87d 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -183,6 +183,31 @@ config SND_CMIPCI To compile this driver as a module, choose M here: the module will be called snd-cmipci. +config SND_OXYGEN_LIB + tristate + depends on SND + select SND_PCM + select SND_MPU401_UART + +config SND_OXYGEN + tristate "C-Media 8788 (Oxygen)" + depends on SND + select SND_OXYGEN_LIB + help + Say Y here to include support for sound cards based on the + C-Media CMI8788 (Oxygen HD Audio) chip: + * Asound A-8788 + * AuzenTech X-Meridian + * Bgears b-Enspirer + * Club3D Theatron DTS + * HT-Omega Claro + * Razer Barracuda AC-1 + * Sondigo Inferno + * TempoTec HIFIER + + To compile this driver as a module, choose M here: the module + will be called snd-oxygen. + config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" depends on SND diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 56738da..2d42fd2 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -68,6 +68,7 @@ obj-$(CONFIG_SND) += \ korg1212/ \ mixart/ \ nm256/ \ + oxygen/ \ pcxhr/ \ riptide/ \ rme9652/ \ diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile new file mode 100644 index 0000000..99455dd --- /dev/null +++ b/sound/pci/oxygen/Makefile @@ -0,0 +1,5 @@ +snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o +snd-oxygen-objs := oxygen.o + +obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o +obj-$(CONFIG_SND_OXYGEN) += snd-oxygen.o diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c new file mode 100644 index 0000000..bfef5ab --- /dev/null +++ b/sound/pci/oxygen/oxygen.c @@ -0,0 +1,314 @@ +/* + * C-Media CMI8788 driver for C-Media's reference design and for the X-Meridian + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +/* + * SPI 0 -> 1st AK4396 (front) + * SPI 1 -> 2nd AK4396 (side) + * SPI 2 -> 3rd AK4396 (center/LFE) + * SPI 3 -> WM8785 + * SPI 4 -> 4th AK4396 (rear) + * + * GPIO 0 -> DFS0 of AK5385 + * GPIO 1 -> DFS1 of AK5385 + */ + +#include +#include +#include +#include +#include +#include +#include +#include "oxygen.h" + +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_DESCRIPTION("C-Media CMI8788 driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "card index"); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string"); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "enable card"); + +static struct pci_device_id oxygen_ids[] __devinitdata = { + { OXYGEN_PCI_SUBID(0x10b0, 0x0216) }, + { OXYGEN_PCI_SUBID(0x10b0, 0x0218) }, + { OXYGEN_PCI_SUBID(0x10b0, 0x0219) }, + { OXYGEN_PCI_SUBID(0x13f6, 0x0001) }, + { OXYGEN_PCI_SUBID(0x13f6, 0x0010) }, + { OXYGEN_PCI_SUBID(0x13f6, 0x8788) }, + { OXYGEN_PCI_SUBID(0x147a, 0xa017) }, + { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, + { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, + { OXYGEN_PCI_SUBID(0x1a58, 0x0910) }, + { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = 1 }, + { OXYGEN_PCI_SUBID(0x7284, 0x9761) }, + { } +}; +MODULE_DEVICE_TABLE(pci, oxygen_ids); + +#define AK4396_WRITE 0x2000 + +/* register 0 */ +#define AK4396_RSTN 0x01 +#define AK4396_DIF_24_MSB 0x04 +/* register 1 */ +#define AK4396_SMUTE 0x01 +#define AK4396_DEM_OFF 0x02 +#define AK4396_DFS_MASK 0x18 +#define AK4396_DFS_NORMAL 0x00 +#define AK4396_DFS_DOUBLE 0x08 +#define AK4396_DFS_QUAD 0x10 + +/* register 0 */ +#define WM8785_OSR_SINGLE 0x000 +#define WM8785_OSR_DOUBLE 0x008 +#define WM8785_OSR_QUAD 0x010 +#define WM8785_FORMAT_LJUST 0x020 +#define WM8785_FORMAT_I2S 0x040 +/* register 1 */ +#define WM8785_WL_16 0x000 +#define WM8785_WL_20 0x001 +#define WM8785_WL_24 0x002 +#define WM8785_WL_32 0x003 + +static void ak4396_write(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + /* maps ALSA channel pair number to SPI output */ + static const u8 codec_spi_map[4] = { + 0, 4, 2, 1 + }; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER_WRITE | + OXYGEN_SPI_DATA_LENGTH_2 | + (codec_spi_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_MAGIC, + AK4396_WRITE | (reg << 8) | value); +} + +static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) +{ + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER_WRITE | + OXYGEN_SPI_DATA_LENGTH_2 | + (3 << OXYGEN_SPI_CODEC_SHIFT), + (reg << 9) | value); +} + +static void ak4396_init(struct oxygen *chip) +{ + unsigned int i; + + chip->ak4396_reg1 = AK4396_DEM_OFF | AK4396_DFS_NORMAL; + for (i = 0; i < 4; ++i) { + ak4396_write(chip, i, 0, AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write(chip, i, 1, chip->ak4396_reg1); + ak4396_write(chip, i, 2, 0); + ak4396_write(chip, i, 3, 0xff); + ak4396_write(chip, i, 4, 0xff); + } + snd_component_add(chip->card, "AK4396"); +} + +static void ak5385_init(struct oxygen *chip) +{ + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, 0x0003); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, 0x0003); + snd_component_add(chip->card, "AK5385"); +} + +static void wm8785_init(struct oxygen *chip) +{ + wm8785_write(chip, 7, 0); + wm8785_write(chip, 0, WM8785_FORMAT_LJUST | WM8785_OSR_SINGLE); + wm8785_write(chip, 1, WM8785_WL_24); + snd_component_add(chip->card, "WM8785"); +} + +static void generic_init(struct oxygen *chip) +{ + ak4396_init(chip); + wm8785_init(chip); +} + +static void meridian_init(struct oxygen *chip) +{ + ak4396_init(chip); + ak5385_init(chip); +} + +static void generic_cleanup(struct oxygen *chip) +{ +} + +static void set_ak4396_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + unsigned int i; + u8 value; + + value = chip->ak4396_reg1 & ~AK4396_DFS_MASK; + if (params_rate(params) <= 54000) + value |= AK4396_DFS_NORMAL; + else if (params_rate(params) < 120000) + value |= AK4396_DFS_DOUBLE; + else + value |= AK4396_DFS_QUAD; + chip->ak4396_reg1 = value; + for (i = 0; i < 4; ++i) { + ak4396_write(chip, i, 0, AK4396_DIF_24_MSB); + ak4396_write(chip, i, 1, value); + ak4396_write(chip, i, 0, AK4396_DIF_24_MSB | AK4396_RSTN); + } +} + +static void update_ak4396_volume(struct oxygen *chip) +{ + unsigned int i; + + for (i = 0; i < 4; ++i) { + ak4396_write(chip, i, 3, chip->dac_volume[i * 2]); + ak4396_write(chip, i, 4, chip->dac_volume[i * 2 + 1]); + } +} + +static void update_ak4396_mute(struct oxygen *chip) +{ + unsigned int i; + u8 value; + + value = chip->ak4396_reg1 & ~AK4396_SMUTE; + if (chip->dac_mute) + value |= AK4396_SMUTE; + for (i = 0; i < 4; ++i) + ak4396_write(chip, i, 1, value); +} + +static void set_wm8785_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + unsigned int value; + + wm8785_write(chip, 7, 0); + + value = WM8785_FORMAT_LJUST; + if (params_rate(params) == 96000) + value |= WM8785_OSR_DOUBLE; + else if (params_rate(params) == 192000) + value |= WM8785_OSR_QUAD; + else + value |= WM8785_OSR_SINGLE; + wm8785_write(chip, 0, value); + + if (snd_pcm_format_width(params_format(params)) <= 16) + value = WM8785_WL_16; + else + value = WM8785_WL_24; + wm8785_write(chip, 1, value); +} + +static void set_ak5385_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + unsigned int value; + + if (params_rate(params) <= 54000) + value = 0; + else if (params_rate(params) <= 108000) + value = 1; + else + value = 2; + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, value, 0x0003); +} + +static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); + +static const struct oxygen_model model_generic = { + .shortname = "C-Media CMI8788", + .longname = "C-Media Oxygen HD Audio", + .chip = "CMI8788", + .owner = THIS_MODULE, + .init = generic_init, + .cleanup = generic_cleanup, + .set_dac_params = set_ak4396_params, + .set_adc_params = set_wm8785_params, + .update_dac_volume = update_ak4396_volume, + .update_dac_mute = update_ak4396_mute, + .dac_tlv = ak4396_db_scale, +}; +static const struct oxygen_model model_meridian = { + .shortname = "C-Media CMI8788", + .longname = "C-Media Oxygen HD Audio", + .chip = "CMI8788", + .owner = THIS_MODULE, + .init = meridian_init, + .cleanup = generic_cleanup, + .set_dac_params = set_ak4396_params, + .set_adc_params = set_ak5385_params, + .update_dac_volume = update_ak4396_volume, + .update_dac_mute = update_ak4396_mute, + .dac_tlv = ak4396_db_scale, + .record_from_dma_b = 1, +}; + +static int __devinit generic_oxygen_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + const struct oxygen_model *model; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + ++dev; + return -ENOENT; + } + model = pci_id->driver_data ? &model_meridian : &model_generic; + err = oxygen_pci_probe(pci, index[dev], id[dev], model); + if (err >= 0) + ++dev; + return err; +} + +static struct pci_driver oxygen_driver = { + .name = "CMI8788", + .id_table = oxygen_ids, + .probe = generic_oxygen_probe, + .remove = __devexit_p(oxygen_pci_remove), +}; + +static int __init alsa_card_oxygen_init(void) +{ + return pci_register_driver(&oxygen_driver); +} + +static void __exit alsa_card_oxygen_exit(void) +{ + pci_unregister_driver(&oxygen_driver); +} + +module_init(alsa_card_oxygen_init) +module_exit(alsa_card_oxygen_exit) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h new file mode 100644 index 0000000..248f7ed --- /dev/null +++ b/sound/pci/oxygen/oxygen.h @@ -0,0 +1,167 @@ +#ifndef OXYGEN_H_INCLUDED +#define OXYGEN_H_INCLUDED + +#include +#include +#include +#include "oxygen_regs.h" + +/* 1 << PCM_x == OXYGEN_CHANNEL_x */ +#define PCM_A 0 +#define PCM_B 1 +#define PCM_C 2 +#define PCM_SPDIF 3 +#define PCM_MULTICH 4 +#define PCM_AC97 5 +#define PCM_COUNT 6 + +#define OXYGEN_PCI_SUBID(sv, sd) \ + .vendor = PCI_VENDOR_ID_CMEDIA, \ + .device = 0x8788, \ + .subvendor = sv, \ + .subdevice = sd + +struct pci_dev; +struct snd_card; +struct snd_pcm_substream; +struct snd_pcm_hw_params; +struct snd_rawmidi; +struct oxygen_model; + +struct oxygen { + unsigned long addr; + spinlock_t reg_lock; + struct mutex mutex; + struct snd_card *card; + struct pci_dev *pci; + struct snd_rawmidi *midi; + int irq; + const struct oxygen_model *model; + unsigned int interrupt_mask; + u8 dac_volume[8]; + u8 dac_mute; + u8 pcm_active; + u8 pcm_running; + u8 dac_routing; + u8 spdif_playback_enable; + u8 ak4396_reg1; + u8 revision; + u8 has_2nd_ac97_codec; + u32 spdif_bits; + u32 spdif_pcm_bits; + struct snd_pcm_substream *streams[PCM_COUNT]; + struct snd_kcontrol *spdif_pcm_ctl; + struct snd_kcontrol *spdif_input_bits_ctl; + struct work_struct spdif_input_bits_work; +}; + +struct oxygen_model { + const char *shortname; + const char *longname; + const char *chip; + struct module *owner; + void (*init)(struct oxygen *chip); + int (*mixer_init)(struct oxygen *chip); + void (*cleanup)(struct oxygen *chip); + void (*set_dac_params)(struct oxygen *chip, + struct snd_pcm_hw_params *params); + void (*set_adc_params)(struct oxygen *chip, + struct snd_pcm_hw_params *params); + void (*update_dac_volume)(struct oxygen *chip); + void (*update_dac_mute)(struct oxygen *chip); + const unsigned int *dac_tlv; + u8 record_from_dma_b; + u8 cd_in_from_video_in; + u8 dac_minimum_volume; +}; + +/* oxygen_lib.c */ + +int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, + const struct oxygen_model *model); +void oxygen_pci_remove(struct pci_dev *pci); + +/* oxygen_mixer.c */ + +int oxygen_mixer_init(struct oxygen *chip); +void oxygen_update_dac_routing(struct oxygen *chip); +void oxygen_update_spdif_source(struct oxygen *chip); + +/* oxygen_pcm.c */ + +int oxygen_pcm_init(struct oxygen *chip); + +/* oxygen_io.c */ + +u8 oxygen_read8(struct oxygen *chip, unsigned int reg); +u16 oxygen_read16(struct oxygen *chip, unsigned int reg); +u32 oxygen_read32(struct oxygen *chip, unsigned int reg); +void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value); +void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value); +void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value); +void oxygen_write8_masked(struct oxygen *chip, unsigned int reg, + u8 value, u8 mask); +void oxygen_write16_masked(struct oxygen *chip, unsigned int reg, + u16 value, u16 mask); +void oxygen_write32_masked(struct oxygen *chip, unsigned int reg, + u32 value, u32 mask); + +u16 oxygen_read_ac97(struct oxygen *chip, unsigned int codec, + unsigned int index); +void oxygen_write_ac97(struct oxygen *chip, unsigned int codec, + unsigned int index, u16 data); +void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, + unsigned int index, u16 data, u16 mask); + +void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data); + +static inline void oxygen_set_bits8(struct oxygen *chip, + unsigned int reg, u8 value) +{ + oxygen_write8_masked(chip, reg, value, value); +} + +static inline void oxygen_set_bits16(struct oxygen *chip, + unsigned int reg, u16 value) +{ + oxygen_write16_masked(chip, reg, value, value); +} + +static inline void oxygen_set_bits32(struct oxygen *chip, + unsigned int reg, u32 value) +{ + oxygen_write32_masked(chip, reg, value, value); +} + +static inline void oxygen_clear_bits8(struct oxygen *chip, + unsigned int reg, u8 value) +{ + oxygen_write8_masked(chip, reg, 0, value); +} + +static inline void oxygen_clear_bits16(struct oxygen *chip, + unsigned int reg, u16 value) +{ + oxygen_write16_masked(chip, reg, 0, value); +} + +static inline void oxygen_clear_bits32(struct oxygen *chip, + unsigned int reg, u32 value) +{ + oxygen_write32_masked(chip, reg, 0, value); +} + +static inline void oxygen_ac97_set_bits(struct oxygen *chip, unsigned int codec, + unsigned int index, u16 value) +{ + oxygen_write_ac97_masked(chip, codec, index, value, value); +} + +static inline void oxygen_ac97_clear_bits(struct oxygen *chip, + unsigned int codec, + unsigned int index, u16 value) +{ + oxygen_write_ac97_masked(chip, codec, index, 0, value); +} + +#endif diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c new file mode 100644 index 0000000..5f4feea --- /dev/null +++ b/sound/pci/oxygen/oxygen_io.c @@ -0,0 +1,194 @@ +/* + * C-Media CMI8788 driver - helper functions + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include "oxygen.h" + +u8 oxygen_read8(struct oxygen *chip, unsigned int reg) +{ + return inb(chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_read8); + +u16 oxygen_read16(struct oxygen *chip, unsigned int reg) +{ + return inw(chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_read16); + +u32 oxygen_read32(struct oxygen *chip, unsigned int reg) +{ + return inl(chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_read32); + +void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value) +{ + outb(value, chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_write8); + +void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value) +{ + outw(value, chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_write16); + +void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value) +{ + outl(value, chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_write32); + +void oxygen_write8_masked(struct oxygen *chip, unsigned int reg, + u8 value, u8 mask) +{ + u8 tmp = inb(chip->addr + reg); + outb((tmp & ~mask) | (value & mask), chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_write8_masked); + +void oxygen_write16_masked(struct oxygen *chip, unsigned int reg, + u16 value, u16 mask) +{ + u16 tmp = inw(chip->addr + reg); + outw((tmp & ~mask) | (value & mask), chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_write16_masked); + +void oxygen_write32_masked(struct oxygen *chip, unsigned int reg, + u32 value, u32 mask) +{ + u32 tmp = inl(chip->addr + reg); + outl((tmp & ~mask) | (value & mask), chip->addr + reg); +} +EXPORT_SYMBOL(oxygen_write32_masked); + +static int oxygen_ac97_wait(struct oxygen *chip, unsigned int mask) +{ + unsigned long timeout = jiffies + msecs_to_jiffies(1); + do { + udelay(5); + cond_resched(); + if (oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS) & mask) + return 0; + } while (time_after_eq(timeout, jiffies)); + return -EIO; +} + +/* + * About 10% of AC'97 register reads or writes fail to complete, but even those + * where the controller indicates completion aren't guaranteed to have actually + * happened. + * + * It's hard to assign blame to either the controller or the codec because both + * were made by C-Media ... + */ + +void oxygen_write_ac97(struct oxygen *chip, unsigned int codec, + unsigned int index, u16 data) +{ + unsigned int count, succeeded; + u32 reg; + + reg = data; + reg |= index << OXYGEN_AC97_REG_ADDR_SHIFT; + reg |= OXYGEN_AC97_REG_DIR_WRITE; + reg |= codec << OXYGEN_AC97_REG_CODEC_SHIFT; + succeeded = 0; + for (count = 5; count > 0; --count) { + udelay(5); + oxygen_write32(chip, OXYGEN_AC97_REGS, reg); + /* require two "completed" writes, just to be sure */ + if (oxygen_ac97_wait(chip, OXYGEN_AC97_WRITE_COMPLETE) >= 0 && + ++succeeded >= 2) + return; + } + snd_printk(KERN_ERR "AC'97 write timeout\n"); +} +EXPORT_SYMBOL(oxygen_write_ac97); + +u16 oxygen_read_ac97(struct oxygen *chip, unsigned int codec, + unsigned int index) +{ + unsigned int count; + unsigned int last_read = UINT_MAX; + u32 reg; + + reg = index << OXYGEN_AC97_REG_ADDR_SHIFT; + reg |= OXYGEN_AC97_REG_DIR_READ; + reg |= codec << OXYGEN_AC97_REG_CODEC_SHIFT; + for (count = 5; count > 0; --count) { + udelay(5); + oxygen_write32(chip, OXYGEN_AC97_REGS, reg); + udelay(10); + if (oxygen_ac97_wait(chip, OXYGEN_AC97_READ_COMPLETE) >= 0) { + u16 value = oxygen_read16(chip, OXYGEN_AC97_REGS); + /* we require two consecutive reads of the same value */ + if (value == last_read) + return value; + last_read = value; + /* + * Invert the register value bits to make sure that two + * consecutive unsuccessful reads do not return the same + * value. + */ + reg ^= 0xffff; + } + } + snd_printk(KERN_ERR "AC'97 read timeout on codec %u\n", codec); + return 0; +} +EXPORT_SYMBOL(oxygen_read_ac97); + +void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, + unsigned int index, u16 data, u16 mask) +{ + u16 value = oxygen_read_ac97(chip, codec, index); + value &= ~mask; + value |= data & mask; + oxygen_write_ac97(chip, codec, index, value); +} +EXPORT_SYMBOL(oxygen_write_ac97_masked); + +void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) +{ + unsigned int count; + + /* should not need more than 3.84 us (24 * 160 ns) */ + count = 10; + while ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & OXYGEN_SPI_BUSY) + && count > 0) { + udelay(1); + --count; + } + + spin_lock_irq(&chip->reg_lock); + oxygen_write8(chip, OXYGEN_SPI_DATA1, data); + oxygen_write8(chip, OXYGEN_SPI_DATA2, data >> 8); + if (control & OXYGEN_SPI_DATA_LENGTH_3) + oxygen_write8(chip, OXYGEN_SPI_DATA3, data >> 16); + oxygen_write8(chip, OXYGEN_SPI_CONTROL, control); + spin_unlock_irq(&chip->reg_lock); +} +EXPORT_SYMBOL(oxygen_write_spi); diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c new file mode 100644 index 0000000..6180cc8 --- /dev/null +++ b/sound/pci/oxygen/oxygen_lib.c @@ -0,0 +1,361 @@ +/* + * C-Media CMI8788 driver - main driver module + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "oxygen.h" + +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_DESCRIPTION("C-Media CMI8788 helper library"); +MODULE_LICENSE("GPL"); + + +static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) +{ + struct oxygen *chip = dev_id; + unsigned int status, clear, elapsed_streams, i; + + status = oxygen_read16(chip, OXYGEN_INTERRUPT_STATUS); + if (!status) + return IRQ_NONE; + + spin_lock(&chip->reg_lock); + + clear = status & (OXYGEN_CHANNEL_A | + OXYGEN_CHANNEL_B | + OXYGEN_CHANNEL_C | + OXYGEN_CHANNEL_SPDIF | + OXYGEN_CHANNEL_MULTICH | + OXYGEN_CHANNEL_AC97 | + OXYGEN_INT_SPDIF_IN_CHANGE | + OXYGEN_INT_GPIO); + if (clear) { + if (clear & OXYGEN_INT_SPDIF_IN_CHANGE) + chip->interrupt_mask &= ~OXYGEN_INT_SPDIF_IN_CHANGE; + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, + chip->interrupt_mask & ~clear); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, + chip->interrupt_mask); + } + + elapsed_streams = status & chip->pcm_running; + + spin_unlock(&chip->reg_lock); + + for (i = 0; i < PCM_COUNT; ++i) + if ((elapsed_streams & (1 << i)) && chip->streams[i]) + snd_pcm_period_elapsed(chip->streams[i]); + + if (status & OXYGEN_INT_SPDIF_IN_CHANGE) { + spin_lock(&chip->reg_lock); + i = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); + if (i & OXYGEN_SPDIF_IN_CHANGE) { + oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, i); + schedule_work(&chip->spdif_input_bits_work); + } + spin_unlock(&chip->reg_lock); + } + + if (status & OXYGEN_INT_GPIO) + ; + + if ((status & OXYGEN_INT_MIDI) && chip->midi) + snd_mpu401_uart_interrupt(0, chip->midi->private_data); + + return IRQ_HANDLED; +} + +static void oxygen_spdif_input_bits_changed(struct work_struct *work) +{ + struct oxygen *chip = container_of(work, struct oxygen, + spdif_input_bits_work); + + spin_lock_irq(&chip->reg_lock); + oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_IN_INVERT); + spin_unlock_irq(&chip->reg_lock); + msleep(1); + if (!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) + & OXYGEN_SPDIF_IN_VALID)) { + spin_lock_irq(&chip->reg_lock); + oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_IN_INVERT); + spin_unlock_irq(&chip->reg_lock); + msleep(1); + if (!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) + & OXYGEN_SPDIF_IN_VALID)) { + spin_lock_irq(&chip->reg_lock); + oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_IN_INVERT); + spin_unlock_irq(&chip->reg_lock); + } + } + + if (chip->spdif_input_bits_ctl) { + spin_lock_irq(&chip->reg_lock); + chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_CHANGE; + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, + chip->interrupt_mask); + spin_unlock_irq(&chip->reg_lock); + + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->spdif_input_bits_ctl->id); + } +} + +#ifdef CONFIG_PROC_FS +static void oxygen_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct oxygen *chip = entry->private_data; + int i, j; + + snd_iprintf(buffer, "CMI8788\n\n"); + for (i = 0; i < 0x100; i += 0x10) { + snd_iprintf(buffer, "%02x:", i); + for (j = 0; j < 0x10; ++j) + snd_iprintf(buffer, " %02x", oxygen_read8(chip, i + j)); + snd_iprintf(buffer, "\n"); + } + if (mutex_lock_interruptible(&chip->mutex) < 0) + return; + snd_iprintf(buffer, "\nAC97\n"); + for (i = 0; i < 0x80; i += 0x10) { + snd_iprintf(buffer, "%02x:", i); + for (j = 0; j < 0x10; j += 2) + snd_iprintf(buffer, " %04x", + oxygen_read_ac97(chip, 0, i + j)); + snd_iprintf(buffer, "\n"); + } + mutex_unlock(&chip->mutex); +} + +static void __devinit oxygen_proc_init(struct oxygen *chip) +{ + struct snd_info_entry *entry; + + if (!snd_card_proc_new(chip->card, "cmi8788", &entry)) + snd_info_set_text_ops(entry, chip, oxygen_proc_read); +} +#else +#define oxygen_proc_init(chip) +#endif + +static void __devinit oxygen_init(struct oxygen *chip) +{ + unsigned int i; + + chip->dac_routing = 1; + for (i = 0; i < 8; ++i) + chip->dac_volume[i] = 0xff; + chip->spdif_playback_enable = 1; + chip->spdif_bits = OXYGEN_SPDIF_C | OXYGEN_SPDIF_ORIGINAL | + (IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT); + chip->spdif_pcm_bits = chip->spdif_bits; + + if (oxygen_read8(chip, OXYGEN_REVISION) & OXYGEN_REVISION_2) + chip->revision = 2; + else + chip->revision = 1; + + if (chip->revision == 1) + oxygen_set_bits8(chip, OXYGEN_MISC, OXYGEN_MISC_MAGIC); + + oxygen_set_bits8(chip, OXYGEN_FUNCTION, + OXYGEN_FUNCTION_RESET_CODEC | + OXYGEN_FUNCTION_ENABLE_SPI_4_5); + oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT, 0x010a); + oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, 0x010a); + oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, 0x010a); + oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, 0x010a); + oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_MAGIC2); + oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); + oxygen_write16(chip, OXYGEN_PLAY_ROUTING, 0x6c00); + oxygen_write8(chip, OXYGEN_REC_ROUTING, 0x10); + oxygen_write8(chip, OXYGEN_ADC_MONITOR, 0x00); + oxygen_write8(chip, OXYGEN_A_MONITOR_ROUTING, 0xe4); + + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); + + oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0x00); + oxygen_clear_bits16(chip, OXYGEN_AC97_OUT_CONFIG, + OXYGEN_AC97_OUT_MAGIC3); + oxygen_set_bits16(chip, OXYGEN_AC97_IN_CONFIG, + OXYGEN_AC97_IN_MAGIC3); + oxygen_write_ac97(chip, 0, AC97_RESET, 0); + msleep(1); + oxygen_ac97_set_bits(chip, 0, 0x70, 0x0300); + oxygen_ac97_set_bits(chip, 0, 0x64, 0x8043); + oxygen_ac97_set_bits(chip, 0, 0x62, 0x180f); + oxygen_write_ac97(chip, 0, AC97_MASTER, 0x0000); + oxygen_write_ac97(chip, 0, AC97_PC_BEEP, 0x8000); + oxygen_write_ac97(chip, 0, AC97_MIC, 0x8808); + oxygen_write_ac97(chip, 0, AC97_LINE, 0x0808); + oxygen_write_ac97(chip, 0, AC97_CD, 0x8808); + oxygen_write_ac97(chip, 0, AC97_VIDEO, 0x8808); + oxygen_write_ac97(chip, 0, AC97_AUX, 0x8808); + oxygen_write_ac97(chip, 0, AC97_REC_GAIN, 0x8000); + oxygen_write_ac97(chip, 0, AC97_CENTER_LFE_MASTER, 0x8080); + oxygen_write_ac97(chip, 0, AC97_SURROUND_MASTER, 0x8080); + oxygen_ac97_clear_bits(chip, 0, 0x72, 0x0001); + /* power down unused ADCs and DACs */ + oxygen_ac97_set_bits(chip, 0, AC97_POWERDOWN, + AC97_PD_PR0 | AC97_PD_PR1); + oxygen_ac97_set_bits(chip, 0, AC97_EXTENDED_STATUS, + AC97_EA_PRI | AC97_EA_PRJ | AC97_EA_PRK); +} + +static void oxygen_card_free(struct snd_card *card) +{ + struct oxygen *chip = card->private_data; + + spin_lock_irq(&chip->reg_lock); + chip->interrupt_mask = 0; + chip->pcm_running = 0; + oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + spin_unlock_irq(&chip->reg_lock); + if (chip->irq >= 0) { + free_irq(chip->irq, chip); + synchronize_irq(chip->irq); + } + flush_scheduled_work(); + chip->model->cleanup(chip); + mutex_destroy(&chip->mutex); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); +} + +int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, + const struct oxygen_model *model) +{ + struct snd_card *card; + struct oxygen *chip; + int err; + + card = snd_card_new(index, id, model->owner, sizeof *chip); + if (!card) + return -ENOMEM; + + chip = card->private_data; + chip->card = card; + chip->pci = pci; + chip->irq = -1; + chip->model = model; + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->mutex); + INIT_WORK(&chip->spdif_input_bits_work, + oxygen_spdif_input_bits_changed); + + err = pci_enable_device(pci); + if (err < 0) + goto err_card; + + err = pci_request_regions(pci, model->chip); + if (err < 0) { + snd_printk(KERN_ERR "cannot reserve PCI resources\n"); + goto err_pci_enable; + } + + if (!(pci_resource_flags(pci, 0) & IORESOURCE_IO) || + pci_resource_len(pci, 0) < 0x100) { + snd_printk(KERN_ERR "invalid PCI I/O range\n"); + err = -ENXIO; + goto err_pci_regions; + } + chip->addr = pci_resource_start(pci, 0); + + pci_set_master(pci); + snd_card_set_dev(card, &pci->dev); + card->private_free = oxygen_card_free; + + oxygen_init(chip); + model->init(chip); + + err = request_irq(pci->irq, oxygen_interrupt, IRQF_SHARED, + model->chip, chip); + if (err < 0) { + snd_printk(KERN_ERR "cannot grab interrupt %d\n", pci->irq); + goto err_card; + } + chip->irq = pci->irq; + + strcpy(card->driver, model->chip); + strcpy(card->shortname, model->shortname); + sprintf(card->longname, "%s (rev %u) at %#lx, irq %i", + model->longname, chip->revision, chip->addr, chip->irq); + strcpy(card->mixername, model->chip); + snd_component_add(card, model->chip); + + err = oxygen_pcm_init(chip); + if (err < 0) + goto err_card; + + err = oxygen_mixer_init(chip); + if (err < 0) + goto err_card; + + if (oxygen_read8(chip, OXYGEN_MISC) & OXYGEN_MISC_MIDI) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, + chip->addr + OXYGEN_MPU401, + MPU401_INFO_INTEGRATED, 0, 0, + &chip->midi); + if (err < 0) + goto err_card; + } + + oxygen_proc_init(chip); + + spin_lock_irq(&chip->reg_lock); + chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_CHANGE; + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); + spin_unlock_irq(&chip->reg_lock); + + err = snd_card_register(card); + if (err < 0) + goto err_card; + + pci_set_drvdata(pci, card); + return 0; + +err_pci_regions: + pci_release_regions(pci); +err_pci_enable: + pci_disable_device(pci); +err_card: + snd_card_free(card); + return err; +} +EXPORT_SYMBOL(oxygen_pci_probe); + +void __devexit oxygen_pci_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} +EXPORT_SYMBOL(oxygen_pci_remove); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c new file mode 100644 index 0000000..e252aba --- /dev/null +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -0,0 +1,623 @@ +/* + * C-Media CMI8788 driver - mixer code + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include +#include "oxygen.h" + +static int dac_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + struct oxygen *chip = ctl->private_data; + + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 8; + info->value.integer.min = chip->model->dac_minimum_volume; + info->value.integer.max = 0xff; + return 0; +} + +static int dac_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + unsigned int i; + + mutex_lock(&chip->mutex); + for (i = 0; i < 8; ++i) + value->value.integer.value[i] = chip->dac_volume[i]; + mutex_unlock(&chip->mutex); + return 0; +} + +static int dac_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + unsigned int i; + int changed; + + changed = 0; + mutex_lock(&chip->mutex); + for (i = 0; i < 8; ++i) + if (value->value.integer.value[i] != chip->dac_volume[i]) { + chip->dac_volume[i] = value->value.integer.value[i]; + changed = 1; + } + if (changed) + chip->model->update_dac_volume(chip); + mutex_unlock(&chip->mutex); + return changed; +} + +static int dac_mute_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = !chip->dac_mute; + mutex_unlock(&chip->mutex); + return 0; +} + +static int dac_mute_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + int changed; + + mutex_lock(&chip->mutex); + changed = !value->value.integer.value[0] != chip->dac_mute; + if (changed) { + chip->dac_mute = !value->value.integer.value[0]; + chip->model->update_dac_mute(chip); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "Front", "Front+Rear", "Front+Rear+Side" + }; + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item > 2) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + mutex_lock(&chip->mutex); + value->value.enumerated.item[0] = chip->dac_routing; + mutex_unlock(&chip->mutex); + return 0; +} + +void oxygen_update_dac_routing(struct oxygen *chip) +{ + /* + * hardware channel order: front, side, center/lfe, rear + * ALSA channel order: front, rear, center/lfe, side + */ + static const unsigned int reg_values[3] = { + 0x6c00, 0x2c00, 0x2000 + }; + unsigned int reg_value; + + if ((oxygen_read8(chip, OXYGEN_PLAY_CHANNELS) & + OXYGEN_PLAY_CHANNELS_MASK) == OXYGEN_PLAY_CHANNELS_2) + reg_value = reg_values[chip->dac_routing]; + else + reg_value = 0x6c00; + oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value, 0xff00); +} + +static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + int changed; + + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != chip->dac_routing; + if (changed) { + chip->dac_routing = min(value->value.enumerated.item[0], 2u); + spin_lock_irq(&chip->reg_lock); + oxygen_update_dac_routing(chip); + spin_unlock_irq(&chip->reg_lock); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int spdif_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = chip->spdif_playback_enable; + mutex_unlock(&chip->mutex); + return 0; +} + +static unsigned int oxygen_spdif_rate(unsigned int oxygen_rate) +{ + switch (oxygen_rate) { + case OXYGEN_RATE_32000: + return IEC958_AES3_CON_FS_32000 << OXYGEN_SPDIF_CS_RATE_SHIFT; + case OXYGEN_RATE_44100: + return IEC958_AES3_CON_FS_44100 << OXYGEN_SPDIF_CS_RATE_SHIFT; + default: /* OXYGEN_RATE_48000 */ + return IEC958_AES3_CON_FS_48000 << OXYGEN_SPDIF_CS_RATE_SHIFT; + case OXYGEN_RATE_64000: + return 0xb << OXYGEN_SPDIF_CS_RATE_SHIFT; + case OXYGEN_RATE_88200: + return 0x8 << OXYGEN_SPDIF_CS_RATE_SHIFT; + case OXYGEN_RATE_96000: + return 0xa << OXYGEN_SPDIF_CS_RATE_SHIFT; + case OXYGEN_RATE_176400: + return 0xc << OXYGEN_SPDIF_CS_RATE_SHIFT; + case OXYGEN_RATE_192000: + return 0xe << OXYGEN_SPDIF_CS_RATE_SHIFT; + } +} + +void oxygen_update_spdif_source(struct oxygen *chip) +{ + u32 old_control, new_control; + u16 old_routing, new_routing; + unsigned int oxygen_rate; + + old_control = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); + old_routing = oxygen_read16(chip, OXYGEN_PLAY_ROUTING); + if (chip->pcm_active & (1 << PCM_SPDIF)) { + new_control = old_control | OXYGEN_SPDIF_OUT_ENABLE; + new_routing = (old_routing & ~0x00e0) | 0x0000; + oxygen_rate = (old_control >> OXYGEN_SPDIF_OUT_RATE_SHIFT) + & OXYGEN_I2S_RATE_MASK; + /* S/PDIF rate was already set by the caller */ + } else if ((chip->pcm_active & (1 << PCM_MULTICH)) && + chip->spdif_playback_enable) { + new_routing = (old_routing & ~0x00e0) | 0x0020; + oxygen_rate = oxygen_read16(chip, OXYGEN_I2S_MULTICH_FORMAT) + & OXYGEN_I2S_RATE_MASK; + new_control = (old_control & ~OXYGEN_SPDIF_OUT_RATE_MASK) | + (oxygen_rate << OXYGEN_SPDIF_OUT_RATE_SHIFT) | + OXYGEN_SPDIF_OUT_ENABLE; + } else { + new_control = old_control & ~OXYGEN_SPDIF_OUT_ENABLE; + new_routing = old_routing; + oxygen_rate = OXYGEN_RATE_44100; + } + if (old_routing != new_routing) { + oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, + new_control & ~OXYGEN_SPDIF_OUT_ENABLE); + oxygen_write16(chip, OXYGEN_PLAY_ROUTING, new_routing); + } + if (new_control & OXYGEN_SPDIF_OUT_ENABLE) + oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, + oxygen_spdif_rate(oxygen_rate) | + ((chip->pcm_active & (1 << PCM_SPDIF)) ? + chip->spdif_pcm_bits : chip->spdif_bits)); + oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, new_control); +} + +static int spdif_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + int changed; + + mutex_lock(&chip->mutex); + changed = value->value.integer.value[0] != chip->spdif_playback_enable; + if (changed) { + chip->spdif_playback_enable = !!value->value.integer.value[0]; + spin_lock_irq(&chip->reg_lock); + oxygen_update_spdif_source(chip); + spin_unlock_irq(&chip->reg_lock); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int spdif_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_IEC958; + info->count = 1; + return 0; +} + +static void oxygen_to_iec958(u32 bits, struct snd_ctl_elem_value *value) +{ + value->value.iec958.status[0] = + bits & (OXYGEN_SPDIF_NONAUDIO | OXYGEN_SPDIF_C | + OXYGEN_SPDIF_PREEMPHASIS); + value->value.iec958.status[1] = /* category and original */ + bits >> OXYGEN_SPDIF_CATEGORY_SHIFT; +} + +static u32 iec958_to_oxygen(struct snd_ctl_elem_value *value) +{ + u32 bits; + + bits = value->value.iec958.status[0] & + (OXYGEN_SPDIF_NONAUDIO | OXYGEN_SPDIF_C | + OXYGEN_SPDIF_PREEMPHASIS); + bits |= value->value.iec958.status[1] << OXYGEN_SPDIF_CATEGORY_SHIFT; + if (bits & OXYGEN_SPDIF_NONAUDIO) + bits |= OXYGEN_SPDIF_V; + return bits; +} + +static inline void write_spdif_bits(struct oxygen *chip, u32 bits) +{ + oxygen_write32_masked(chip, OXYGEN_SPDIF_OUTPUT_BITS, bits, + OXYGEN_SPDIF_NONAUDIO | + OXYGEN_SPDIF_C | + OXYGEN_SPDIF_PREEMPHASIS | + OXYGEN_SPDIF_CATEGORY_MASK | + OXYGEN_SPDIF_ORIGINAL | + OXYGEN_SPDIF_V); +} + +static int spdif_default_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + mutex_lock(&chip->mutex); + oxygen_to_iec958(chip->spdif_bits, value); + mutex_unlock(&chip->mutex); + return 0; +} + +static int spdif_default_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u32 new_bits; + int changed; + + new_bits = iec958_to_oxygen(value); + mutex_lock(&chip->mutex); + changed = new_bits != chip->spdif_bits; + if (changed) { + chip->spdif_bits = new_bits; + if (!(chip->pcm_active & (1 << PCM_SPDIF))) + write_spdif_bits(chip, new_bits); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int spdif_mask_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + value->value.iec958.status[0] = IEC958_AES0_NONAUDIO | + IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS; + value->value.iec958.status[1] = + IEC958_AES1_CON_CATEGORY | IEC958_AES1_CON_ORIGINAL; + return 0; +} + +static int spdif_pcm_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + mutex_lock(&chip->mutex); + oxygen_to_iec958(chip->spdif_pcm_bits, value); + mutex_unlock(&chip->mutex); + return 0; +} + +static int spdif_pcm_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u32 new_bits; + int changed; + + new_bits = iec958_to_oxygen(value); + mutex_lock(&chip->mutex); + changed = new_bits != chip->spdif_pcm_bits; + if (changed) { + chip->spdif_pcm_bits = new_bits; + if (chip->pcm_active & (1 << PCM_SPDIF)) + write_spdif_bits(chip, new_bits); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int spdif_input_mask_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + value->value.iec958.status[0] = 0xff; + value->value.iec958.status[1] = 0xff; + value->value.iec958.status[2] = 0xff; + value->value.iec958.status[3] = 0xff; + return 0; +} + +static int spdif_input_default_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u32 bits; + + bits = oxygen_read32(chip, OXYGEN_SPDIF_INPUT_BITS); + value->value.iec958.status[0] = bits; + value->value.iec958.status[1] = bits >> 8; + value->value.iec958.status[2] = bits >> 16; + value->value.iec958.status[3] = bits >> 24; + return 0; +} + +static int ac97_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + unsigned int index = ctl->private_value & 0xff; + unsigned int bitnr = (ctl->private_value >> 8) & 0xff; + int invert = ctl->private_value & (1 << 16); + u16 reg; + + mutex_lock(&chip->mutex); + reg = oxygen_read_ac97(chip, 0, index); + mutex_unlock(&chip->mutex); + if (!(reg & (1 << bitnr)) ^ !invert) + value->value.integer.value[0] = 1; + else + value->value.integer.value[0] = 0; + return 0; +} + +static int ac97_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + unsigned int index = ctl->private_value & 0xff; + unsigned int bitnr = (ctl->private_value >> 8) & 0xff; + int invert = ctl->private_value & (1 << 16); + u16 oldreg, newreg; + int change; + + mutex_lock(&chip->mutex); + oldreg = oxygen_read_ac97(chip, 0, index); + newreg = oldreg; + if (!value->value.integer.value[0] ^ !invert) + newreg |= 1 << bitnr; + else + newreg &= ~(1 << bitnr); + change = newreg != oldreg; + if (change) { + oxygen_write_ac97(chip, 0, index, newreg); + if (index == AC97_LINE) + oxygen_write_ac97_masked(chip, 0, 0x72, + !!(newreg & 0x8000), 0x0001); + } + mutex_unlock(&chip->mutex); + return change; +} + +static int ac97_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0; + info->value.integer.max = 0x1f; + return 0; +} + +static int ac97_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + unsigned int index = ctl->private_value; + u16 reg; + + mutex_lock(&chip->mutex); + reg = oxygen_read_ac97(chip, 0, index); + mutex_unlock(&chip->mutex); + value->value.integer.value[0] = 31 - (reg & 0x1f); + value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); + return 0; +} + +static int ac97_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + unsigned int index = ctl->private_value; + u16 oldreg, newreg; + int change; + + mutex_lock(&chip->mutex); + oldreg = oxygen_read_ac97(chip, 0, index); + newreg = oldreg; + newreg = (newreg & ~0x1f) | + (31 - (value->value.integer.value[0] & 0x1f)); + newreg = (newreg & ~0x1f00) | + ((31 - (value->value.integer.value[0] & 0x1f)) << 8); + change = newreg != oldreg; + if (change) + oxygen_write_ac97(chip, 0, index, newreg); + mutex_unlock(&chip->mutex); + return change; +} + +#define AC97_SWITCH(xname, index, bitnr, invert) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = ac97_switch_get, \ + .put = ac97_switch_put, \ + .private_value = ((invert) << 16) | ((bitnr) << 8) | (index), \ + } +#define AC97_VOLUME(xname, index) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = ac97_volume_info, \ + .get = ac97_volume_get, \ + .put = ac97_volume_put, \ + .tlv = { .p = ac97_db_scale, }, \ + .private_value = (index), \ + } + +static DECLARE_TLV_DB_SCALE(ac97_db_scale, -3450, 150, 0); + +static const struct snd_kcontrol_new controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = dac_volume_info, + .get = dac_volume_get, + .put = dac_volume_put, + .tlv = { + .p = NULL, /* set later */ + }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = dac_mute_get, + .put = dac_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Stereo Upmixing", + .info = upmix_info, + .get = upmix_get, + .put = upmix_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = spdif_switch_get, + .put = spdif_switch_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .device = 1, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = spdif_info, + .get = spdif_default_get, + .put = spdif_default_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .device = 1, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = spdif_info, + .get = spdif_mask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .device = 1, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM), + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_INACTIVE, + .info = spdif_info, + .get = spdif_pcm_get, + .put = spdif_pcm_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .device = 1, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK), + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = spdif_info, + .get = spdif_input_mask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .device = 1, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = spdif_info, + .get = spdif_input_default_get, + }, + AC97_VOLUME("Mic Capture Volume", AC97_MIC), + AC97_SWITCH("Mic Capture Switch", AC97_MIC, 15, 1), + AC97_SWITCH("Mic Boost (+20dB)", AC97_MIC, 6, 0), + AC97_SWITCH("Line Capture Switch", AC97_LINE, 15, 1), + AC97_VOLUME("CD Capture Volume", AC97_CD), + AC97_SWITCH("CD Capture Switch", AC97_CD, 15, 1), + AC97_VOLUME("Aux Capture Volume", AC97_AUX), + AC97_SWITCH("Aux Capture Switch", AC97_AUX, 15, 1), +}; + +static void oxygen_any_ctl_free(struct snd_kcontrol *ctl) +{ + struct oxygen *chip = ctl->private_data; + + /* I'm too lazy to write a function for each control :-) */ + chip->spdif_pcm_ctl = NULL; + chip->spdif_input_bits_ctl = NULL; +} + +int oxygen_mixer_init(struct oxygen *chip) +{ + unsigned int i; + struct snd_kcontrol *ctl; + int err; + + for (i = 0; i < ARRAY_SIZE(controls); ++i) { + ctl = snd_ctl_new1(&controls[i], chip); + if (!ctl) + return -ENOMEM; + if (!strcmp(ctl->id.name, "PCM Playback Volume")) + ctl->tlv.p = chip->model->dac_tlv; + else if (chip->model->cd_in_from_video_in && + !strncmp(ctl->id.name, "CD Capture ", 11)) + ctl->private_value ^= AC97_CD ^ AC97_VIDEO; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + if (!strcmp(ctl->id.name, + SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM))) { + chip->spdif_pcm_ctl = ctl; + ctl->private_free = oxygen_any_ctl_free; + } else if (!strcmp(ctl->id.name, + SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT))) { + chip->spdif_input_bits_ctl = ctl; + ctl->private_free = oxygen_any_ctl_free; + } + } + return chip->model->mixer_init ? chip->model->mixer_init(chip) : 0; +} diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c new file mode 100644 index 0000000..941399b --- /dev/null +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -0,0 +1,726 @@ +/* + * C-Media CMI8788 driver - PCM code + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include +#include "oxygen.h" + +static struct snd_pcm_hardware oxygen_hardware[PCM_COUNT] = { + [PCM_A] = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .rate_min = 44100, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 256 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 2048, + }, + [PCM_B] = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 32000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 256 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 2048, + }, + [PCM_C] = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 256 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 2048, + }, + [PCM_SPDIF] = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 32000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 256 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 2048, + }, + [PCM_MULTICH] = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 32000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = 2048 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 256 * 1024, + .periods_min = 2, + .periods_max = 16384, + }, + [PCM_AC97] = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 256 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 2048, + }, +}; + +static int oxygen_open(struct snd_pcm_substream *substream, + unsigned int channel) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + runtime->private_data = (void *)channel; + runtime->hw = oxygen_hardware[channel]; + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); + if (err < 0) + return err; + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); + if (err < 0) + return err; + if (runtime->hw.formats & SNDRV_PCM_FMTBIT_S32_LE) { + err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + if (err < 0) + return err; + } + if (runtime->hw.channels_max > 2) { + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2); + if (err < 0) + return err; + } + snd_pcm_set_sync(substream); + chip->streams[channel] = substream; + + mutex_lock(&chip->mutex); + chip->pcm_active |= 1 << channel; + if (channel == PCM_SPDIF) { + chip->spdif_pcm_bits = chip->spdif_bits; + chip->spdif_pcm_ctl->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, + &chip->spdif_pcm_ctl->id); + } + mutex_unlock(&chip->mutex); + + return 0; +} + +static int oxygen_rec_a_open(struct snd_pcm_substream *substream) +{ + return oxygen_open(substream, PCM_A); +} + +static int oxygen_rec_b_open(struct snd_pcm_substream *substream) +{ + return oxygen_open(substream, PCM_B); +} + +static int oxygen_rec_c_open(struct snd_pcm_substream *substream) +{ + return oxygen_open(substream, PCM_C); +} + +static int oxygen_spdif_open(struct snd_pcm_substream *substream) +{ + return oxygen_open(substream, PCM_SPDIF); +} + +static int oxygen_multich_open(struct snd_pcm_substream *substream) +{ + return oxygen_open(substream, PCM_MULTICH); +} + +static int oxygen_ac97_open(struct snd_pcm_substream *substream) +{ + return oxygen_open(substream, PCM_AC97); +} + +static int oxygen_close(struct snd_pcm_substream *substream) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + unsigned int channel = (unsigned int)substream->runtime->private_data; + + mutex_lock(&chip->mutex); + chip->pcm_active &= ~(1 << channel); + if (channel == PCM_SPDIF) { + chip->spdif_pcm_ctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, + &chip->spdif_pcm_ctl->id); + } + if (channel == PCM_SPDIF || channel == PCM_MULTICH) + oxygen_update_spdif_source(chip); + mutex_unlock(&chip->mutex); + + chip->streams[channel] = NULL; + return 0; +} + +static unsigned int oxygen_format(struct snd_pcm_hw_params *hw_params) +{ + if (params_format(hw_params) == SNDRV_PCM_FORMAT_S32_LE) + return OXYGEN_FORMAT_24; + else + return OXYGEN_FORMAT_16; +} + +static unsigned int oxygen_rate(struct snd_pcm_hw_params *hw_params) +{ + switch (params_rate(hw_params)) { + case 32000: + return OXYGEN_RATE_32000; + case 44100: + return OXYGEN_RATE_44100; + default: /* 48000 */ + return OXYGEN_RATE_48000; + case 64000: + return OXYGEN_RATE_64000; + case 88200: + return OXYGEN_RATE_88200; + case 96000: + return OXYGEN_RATE_96000; + case 176400: + return OXYGEN_RATE_176400; + case 192000: + return OXYGEN_RATE_192000; + } +} + +static unsigned int oxygen_i2s_magic2(struct snd_pcm_hw_params *hw_params) +{ + return params_rate(hw_params) <= 96000 ? 0x10 : 0x00; +} + +static unsigned int oxygen_i2s_format(struct snd_pcm_hw_params *hw_params) +{ + if (params_format(hw_params) == SNDRV_PCM_FORMAT_S32_LE) + return OXYGEN_I2S_FORMAT_24; + else + return OXYGEN_I2S_FORMAT_16; +} + +static unsigned int oxygen_play_channels(struct snd_pcm_hw_params *hw_params) +{ + switch (params_channels(hw_params)) { + default: /* 2 */ + return OXYGEN_PLAY_CHANNELS_2; + case 4: + return OXYGEN_PLAY_CHANNELS_4; + case 6: + return OXYGEN_PLAY_CHANNELS_6; + case 8: + return OXYGEN_PLAY_CHANNELS_8; + } +} + +static const unsigned int channel_base_registers[PCM_COUNT] = { + [PCM_A] = OXYGEN_DMA_A_ADDRESS, + [PCM_B] = OXYGEN_DMA_B_ADDRESS, + [PCM_C] = OXYGEN_DMA_C_ADDRESS, + [PCM_SPDIF] = OXYGEN_DMA_SPDIF_ADDRESS, + [PCM_MULTICH] = OXYGEN_DMA_MULTICH_ADDRESS, + [PCM_AC97] = OXYGEN_DMA_AC97_ADDRESS, +}; + +static int oxygen_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + unsigned int channel = (unsigned int)substream->runtime->private_data; + int err; + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + oxygen_write32(chip, channel_base_registers[channel], + (u32)substream->runtime->dma_addr); + if (channel == PCM_MULTICH) { + oxygen_write32(chip, OXYGEN_DMA_MULTICH_COUNT, + params_buffer_bytes(hw_params) / 4 - 1); + oxygen_write32(chip, OXYGEN_DMA_MULTICH_TCOUNT, + params_period_bytes(hw_params) / 4 - 1); + } else { + oxygen_write16(chip, channel_base_registers[channel] + 4, + params_buffer_bytes(hw_params) / 4 - 1); + oxygen_write16(chip, channel_base_registers[channel] + 6, + params_period_bytes(hw_params) / 4 - 1); + } + return 0; +} + +static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + int err; + + err = oxygen_hw_params(substream, hw_params); + if (err < 0) + return err; + + spin_lock_irq(&chip->reg_lock); + oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, + oxygen_format(hw_params) << OXYGEN_REC_FORMAT_A_SHIFT, + OXYGEN_REC_FORMAT_A_MASK); + oxygen_write8_masked(chip, OXYGEN_I2S_A_FORMAT, + oxygen_rate(hw_params) | + oxygen_i2s_magic2(hw_params) | + oxygen_i2s_format(hw_params), + OXYGEN_I2S_RATE_MASK | + OXYGEN_I2S_MAGIC2_MASK | + OXYGEN_I2S_FORMAT_MASK); + oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x08); + spin_unlock_irq(&chip->reg_lock); + + mutex_lock(&chip->mutex); + chip->model->set_adc_params(chip, hw_params); + mutex_unlock(&chip->mutex); + return 0; +} + +static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + int err; + + err = oxygen_hw_params(substream, hw_params); + if (err < 0) + return err; + + spin_lock_irq(&chip->reg_lock); + oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, + oxygen_format(hw_params) << OXYGEN_REC_FORMAT_B_SHIFT, + OXYGEN_REC_FORMAT_B_MASK); + oxygen_write8_masked(chip, OXYGEN_I2S_B_FORMAT, + oxygen_rate(hw_params) | + oxygen_i2s_magic2(hw_params) | + oxygen_i2s_format(hw_params), + OXYGEN_I2S_RATE_MASK | + OXYGEN_I2S_MAGIC2_MASK | + OXYGEN_I2S_FORMAT_MASK); + oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x10); + spin_unlock_irq(&chip->reg_lock); + + mutex_lock(&chip->mutex); + chip->model->set_adc_params(chip, hw_params); + mutex_unlock(&chip->mutex); + return 0; +} + +static int oxygen_rec_c_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + int err; + + err = oxygen_hw_params(substream, hw_params); + if (err < 0) + return err; + + spin_lock_irq(&chip->reg_lock); + oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, + oxygen_format(hw_params) << OXYGEN_REC_FORMAT_C_SHIFT, + OXYGEN_REC_FORMAT_C_MASK); + oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x20); + spin_unlock_irq(&chip->reg_lock); + return 0; +} + +static int oxygen_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + int err; + + err = oxygen_hw_params(substream, hw_params); + if (err < 0) + return err; + + spin_lock_irq(&chip->reg_lock); + oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_OUT_ENABLE); + oxygen_write8_masked(chip, OXYGEN_PLAY_FORMAT, + oxygen_format(hw_params) << OXYGEN_SPDIF_FORMAT_SHIFT, + OXYGEN_SPDIF_FORMAT_MASK); + oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, + oxygen_rate(hw_params) << OXYGEN_SPDIF_OUT_RATE_SHIFT, + OXYGEN_SPDIF_OUT_RATE_MASK); + oxygen_update_spdif_source(chip); + spin_unlock_irq(&chip->reg_lock); + return 0; +} + +static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + int err; + + err = oxygen_hw_params(substream, hw_params); + if (err < 0) + return err; + + spin_lock_irq(&chip->reg_lock); + oxygen_write8_masked(chip, OXYGEN_PLAY_CHANNELS, + oxygen_play_channels(hw_params), + OXYGEN_PLAY_CHANNELS_MASK); + oxygen_write8_masked(chip, OXYGEN_PLAY_FORMAT, + oxygen_format(hw_params) << OXYGEN_MULTICH_FORMAT_SHIFT, + OXYGEN_MULTICH_FORMAT_MASK); + oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, + oxygen_rate(hw_params) | oxygen_i2s_format(hw_params), + OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK); + oxygen_clear_bits16(chip, OXYGEN_PLAY_ROUTING, 0x001f); + oxygen_update_dac_routing(chip); + oxygen_update_spdif_source(chip); + spin_unlock_irq(&chip->reg_lock); + + mutex_lock(&chip->mutex); + chip->model->set_dac_params(chip, hw_params); + mutex_unlock(&chip->mutex); + return 0; +} + +static int oxygen_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + int err; + + err = oxygen_hw_params(substream, hw_params); + if (err < 0) + return err; + + spin_lock_irq(&chip->reg_lock); + oxygen_write8_masked(chip, OXYGEN_PLAY_FORMAT, + oxygen_format(hw_params) << OXYGEN_AC97_FORMAT_SHIFT, + OXYGEN_AC97_FORMAT_MASK); + spin_unlock_irq(&chip->reg_lock); + return 0; +} + +static int oxygen_hw_free(struct snd_pcm_substream *substream) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + unsigned int channel = (unsigned int)substream->runtime->private_data; + + spin_lock_irq(&chip->reg_lock); + chip->interrupt_mask &= ~(1 << channel); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); + spin_unlock_irq(&chip->reg_lock); + + return snd_pcm_lib_free_pages(substream); +} + +static int oxygen_spdif_hw_free(struct snd_pcm_substream *substream) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + + spin_lock_irq(&chip->reg_lock); + oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_OUT_ENABLE); + spin_unlock_irq(&chip->reg_lock); + return oxygen_hw_free(substream); +} + +static int oxygen_prepare(struct snd_pcm_substream *substream) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + unsigned int channel = (unsigned int)substream->runtime->private_data; + unsigned int channel_mask = 1 << channel; + + spin_lock_irq(&chip->reg_lock); + oxygen_set_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); + oxygen_clear_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); + + chip->interrupt_mask |= channel_mask; + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); + spin_unlock_irq(&chip->reg_lock); + return 0; +} + +static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_substream *s; + unsigned int mask = 0; + int running; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + running = 0; + break; + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + running = 1; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) == chip) { + mask |= 1 << (unsigned int)s->runtime->private_data; + snd_pcm_trigger_done(s, substream); + } + } + + spin_lock(&chip->reg_lock); + if (running) + chip->pcm_running |= mask; + else + chip->pcm_running &= ~mask; + oxygen_write8(chip, OXYGEN_DMA_STATUS, chip->pcm_running); + spin_unlock(&chip->reg_lock); + return 0; +} + +static snd_pcm_uframes_t oxygen_pointer(struct snd_pcm_substream *substream) +{ + struct oxygen *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int channel = (unsigned int)runtime->private_data; + u32 curr_addr; + + /* no spinlock, this read should be atomic */ + curr_addr = oxygen_read32(chip, channel_base_registers[channel]); + return bytes_to_frames(runtime, curr_addr - (u32)runtime->dma_addr); +} + +static struct snd_pcm_ops oxygen_rec_a_ops = { + .open = oxygen_rec_a_open, + .close = oxygen_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = oxygen_rec_a_hw_params, + .hw_free = oxygen_hw_free, + .prepare = oxygen_prepare, + .trigger = oxygen_trigger, + .pointer = oxygen_pointer, +}; + +static struct snd_pcm_ops oxygen_rec_b_ops = { + .open = oxygen_rec_b_open, + .close = oxygen_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = oxygen_rec_b_hw_params, + .hw_free = oxygen_hw_free, + .prepare = oxygen_prepare, + .trigger = oxygen_trigger, + .pointer = oxygen_pointer, +}; + +static struct snd_pcm_ops oxygen_rec_c_ops = { + .open = oxygen_rec_c_open, + .close = oxygen_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = oxygen_rec_c_hw_params, + .hw_free = oxygen_hw_free, + .prepare = oxygen_prepare, + .trigger = oxygen_trigger, + .pointer = oxygen_pointer, +}; + +static struct snd_pcm_ops oxygen_spdif_ops = { + .open = oxygen_spdif_open, + .close = oxygen_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = oxygen_spdif_hw_params, + .hw_free = oxygen_spdif_hw_free, + .prepare = oxygen_prepare, + .trigger = oxygen_trigger, + .pointer = oxygen_pointer, +}; + +static struct snd_pcm_ops oxygen_multich_ops = { + .open = oxygen_multich_open, + .close = oxygen_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = oxygen_multich_hw_params, + .hw_free = oxygen_hw_free, + .prepare = oxygen_prepare, + .trigger = oxygen_trigger, + .pointer = oxygen_pointer, +}; + +static struct snd_pcm_ops oxygen_ac97_ops = { + .open = oxygen_ac97_open, + .close = oxygen_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = oxygen_ac97_hw_params, + .hw_free = oxygen_hw_free, + .prepare = oxygen_prepare, + .trigger = oxygen_trigger, + .pointer = oxygen_pointer, +}; + +static void oxygen_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +int __devinit oxygen_pcm_init(struct oxygen *chip) +{ + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "Analog", 0, 1, 1, &pcm); + if (err < 0) + return err; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &oxygen_multich_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + chip->model->record_from_dma_b ? + &oxygen_rec_b_ops : &oxygen_rec_a_ops); + pcm->private_data = chip; + pcm->private_free = oxygen_pcm_free; + strcpy(pcm->name, "Analog"); + snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 512 * 1024, 2048 * 1024); + snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 128 * 1024, 256 * 1024); + + err = snd_pcm_new(chip->card, "Digital", 1, 1, 1, &pcm); + if (err < 0) + return err; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &oxygen_spdif_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &oxygen_rec_c_ops); + pcm->private_data = chip; + pcm->private_free = oxygen_pcm_free; + strcpy(pcm->name, "Digital"); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 128 * 1024, 256 * 1024); + + if (chip->has_2nd_ac97_codec) { + err = snd_pcm_new(chip->card, "AC97", 2, 1, 0, &pcm); + if (err < 0) + return err; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &oxygen_ac97_ops); + pcm->private_data = chip; + pcm->private_free = oxygen_pcm_free; + strcpy(pcm->name, "Front Panel"); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 128 * 1024, 256 * 1024); + } + return 0; +} diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h new file mode 100644 index 0000000..7a4726d --- /dev/null +++ b/sound/pci/oxygen/oxygen_regs.h @@ -0,0 +1,246 @@ +#ifndef OXYGEN_REGS_H_INCLUDED +#define OXYGEN_REGS_H_INCLUDED + +/* recording channel A */ +#define OXYGEN_DMA_A_ADDRESS 0x00 /* 32-bit base address */ +#define OXYGEN_DMA_A_COUNT 0x04 /* buffer counter (dwords) */ +#define OXYGEN_DMA_A_TCOUNT 0x06 /* interrupt counter (dwords) */ + +/* recording channel B */ +#define OXYGEN_DMA_B_ADDRESS 0x08 +#define OXYGEN_DMA_B_COUNT 0x0c +#define OXYGEN_DMA_B_TCOUNT 0x0e + +/* recording channel C */ +#define OXYGEN_DMA_C_ADDRESS 0x10 +#define OXYGEN_DMA_C_COUNT 0x14 +#define OXYGEN_DMA_C_TCOUNT 0x16 + +/* SPDIF playback channel */ +#define OXYGEN_DMA_SPDIF_ADDRESS 0x18 +#define OXYGEN_DMA_SPDIF_COUNT 0x1c +#define OXYGEN_DMA_SPDIF_TCOUNT 0x1e + +/* multichannel playback channel */ +#define OXYGEN_DMA_MULTICH_ADDRESS 0x20 +#define OXYGEN_DMA_MULTICH_COUNT 0x24 /* 32 bits */ +#define OXYGEN_DMA_MULTICH_TCOUNT 0x28 /* 32 bits */ + +/* AC'97 (front panel) playback channel */ +#define OXYGEN_DMA_AC97_ADDRESS 0x30 +#define OXYGEN_DMA_AC97_COUNT 0x34 +#define OXYGEN_DMA_AC97_TCOUNT 0x36 + +/* all registers 0x00..0x36 return current position on read */ + +#define OXYGEN_DMA_STATUS 0x40 /* 1 = running, 0 = stop */ +#define OXYGEN_CHANNEL_A 0x01 +#define OXYGEN_CHANNEL_B 0x02 +#define OXYGEN_CHANNEL_C 0x04 +#define OXYGEN_CHANNEL_SPDIF 0x08 +#define OXYGEN_CHANNEL_MULTICH 0x10 +#define OXYGEN_CHANNEL_AC97 0x20 + +#define OXYGEN_DMA_RESET 0x42 +/* OXYGEN_CHANNEL_* */ + +#define OXYGEN_PLAY_CHANNELS 0x43 +#define OXYGEN_PLAY_CHANNELS_MASK 0x03 +#define OXYGEN_PLAY_CHANNELS_2 0x00 +#define OXYGEN_PLAY_CHANNELS_4 0x01 +#define OXYGEN_PLAY_CHANNELS_6 0x02 +#define OXYGEN_PLAY_CHANNELS_8 0x03 + +#define OXYGEN_INTERRUPT_MASK 0x44 +/* OXYGEN_CHANNEL_* */ +#define OXYGEN_INT_SPDIF_IN_CHANGE 0x0100 +#define OXYGEN_INT_GPIO 0x0800 + +#define OXYGEN_INTERRUPT_STATUS 0x46 +/* OXYGEN_CHANNEL_* amd OXYGEN_INT_* */ +#define OXYGEN_INT_MIDI 0x1000 + +#define OXYGEN_MISC 0x48 +#define OXYGEN_MISC_MAGIC 0x20 +#define OXYGEN_MISC_MIDI 0x40 + +#define OXYGEN_REC_FORMAT 0x4a +#define OXYGEN_REC_FORMAT_A_MASK 0x03 +#define OXYGEN_REC_FORMAT_A_SHIFT 0 +#define OXYGEN_REC_FORMAT_B_MASK 0x0c +#define OXYGEN_REC_FORMAT_B_SHIFT 2 +#define OXYGEN_REC_FORMAT_C_MASK 0x30 +#define OXYGEN_REC_FORMAT_C_SHIFT 4 +#define OXYGEN_FORMAT_16 0x00 +#define OXYGEN_FORMAT_24 0x01 +#define OXYGEN_FORMAT_32 0x02 + +#define OXYGEN_PLAY_FORMAT 0x4b +#define OXYGEN_SPDIF_FORMAT_MASK 0x03 +#define OXYGEN_SPDIF_FORMAT_SHIFT 0 +#define OXYGEN_MULTICH_FORMAT_MASK 0x0c +#define OXYGEN_MULTICH_FORMAT_SHIFT 2 +#define OXYGEN_AC97_FORMAT_MASK 0x30 +#define OXYGEN_AC97_FORMAT_SHIFT 4 +/* OXYGEN_FORMAT_* */ + +#define OXYGEN_REC_CHANNELS 0x4c +#define OXYGEN_REC_A_CHANNELS_MASK 0x07 +#define OXYGEN_REC_CHANNELS_2 0x00 +#define OXYGEN_REC_CHANNELS_4 0x01 +#define OXYGEN_REC_CHANNELS_6 0x03 /* or 0x02 */ +#define OXYGEN_REC_CHANNELS_8 0x04 + +#define OXYGEN_FUNCTION 0x50 +#define OXYGEN_FUNCTION_RESET_CODEC 0x02 +#define OXYGEN_FUNCTION_ENABLE_SPI_4_5 0x80 + +#define OXYGEN_I2S_MULTICH_FORMAT 0x60 +#define OXYGEN_I2S_RATE_MASK 0x0007 +#define OXYGEN_RATE_32000 0x0000 +#define OXYGEN_RATE_44100 0x0001 +#define OXYGEN_RATE_48000 0x0002 +#define OXYGEN_RATE_64000 0x0003 +#define OXYGEN_RATE_88200 0x0004 +#define OXYGEN_RATE_96000 0x0005 +#define OXYGEN_RATE_176400 0x0006 +#define OXYGEN_RATE_192000 0x0007 +#define OXYGEN_I2S_MAGIC1_MASK 0x0008 +#define OXYGEN_I2S_MAGIC2_MASK 0x0030 +#define OXYGEN_I2S_FORMAT_MASK 0x00c0 +#define OXYGEN_I2S_FORMAT_16 0x0000 +#define OXYGEN_I2S_FORMAT_20 0x0040 +#define OXYGEN_I2S_FORMAT_24 0x0080 +#define OXYGEN_I2S_FORMAT_32 0x00c0 + +#define OXYGEN_I2S_A_FORMAT 0x62 +#define OXYGEN_I2S_B_FORMAT 0x64 +#define OXYGEN_I2S_C_FORMAT 0x66 +/* OXYGEN_I2S_RATE_* and OXYGEN_I2S_FORMAT_* */ + +#define OXYGEN_SPDIF_CONTROL 0x70 +#define OXYGEN_SPDIF_OUT_ENABLE 0x00000002 +#define OXYGEN_SPDIF_LOOPBACK 0x00000004 +#define OXYGEN_SPDIF_MAGIC2 0x00000020 +#define OXYGEN_SPDIF_MAGIC3 0x00000040 +#define OXYGEN_SPDIF_IN_VALID 0x00001000 +#define OXYGEN_SPDIF_IN_CHANGE 0x00008000 /* r/wc */ +#define OXYGEN_SPDIF_IN_INVERT 0x00010000 /* ? */ +#define OXYGEN_SPDIF_OUT_RATE_MASK 0x07000000 +#define OXYGEN_SPDIF_OUT_RATE_SHIFT 24 +/* OXYGEN_RATE_* << OXYGEN_SPDIF_OUT_RATE_SHIFT */ + +#define OXYGEN_SPDIF_OUTPUT_BITS 0x74 +#define OXYGEN_SPDIF_NONAUDIO 0x00000002 +#define OXYGEN_SPDIF_C 0x00000004 +#define OXYGEN_SPDIF_PREEMPHASIS 0x00000008 +#define OXYGEN_SPDIF_CATEGORY_MASK 0x000007f0 +#define OXYGEN_SPDIF_CATEGORY_SHIFT 4 +#define OXYGEN_SPDIF_ORIGINAL 0x00000800 +#define OXYGEN_SPDIF_CS_RATE_MASK 0x0000f000 +#define OXYGEN_SPDIF_CS_RATE_SHIFT 12 +#define OXYGEN_SPDIF_V 0x00010000 /* 0 = valid */ + +#define OXYGEN_SPDIF_INPUT_BITS 0x78 +/* 32 bits, IEC958_AES_* */ + +#define OXYGEN_2WIRE_CONTROL 0x90 +#define OXYGEN_2WIRE_DIR_MASK 0x01 +#define OXYGEN_2WIRE_DIR_WRITE 0x00 /* ? */ +#define OXYGEN_2WIRE_DIR_READ 0x01 /* ? */ +#define OXYGEN_2WIRE_ADDRESS_MASK 0xfe /* slave device address */ +#define OXYGEN_2WIRE_ADDRESS_SHIFT 1 + +#define OXYGEN_2WIRE_MAP 0x91 /* address, 8 bits */ +#define OXYGEN_2WIRE_DATA 0x92 /* data, 16 bits */ + +#define OXYGEN_2WIRE_BUS_STATUS 0x94 +#define OXYGEN_2WIRE_BUSY 0x01 + +#define OXYGEN_SPI_CONTROL 0x98 +#define OXYGEN_SPI_BUSY 0x01 /* read */ +#define OXYGEN_SPI_TRIGGER_WRITE 0x01 /* write */ +#define OXYGEN_SPI_DATA_LENGTH_MASK 0x02 +#define OXYGEN_SPI_DATA_LENGTH_2 0x00 +#define OXYGEN_SPI_DATA_LENGTH_3 0x02 +#define OXYGEN_SPI_CODEC_MASK 0x70 /* 0..5 */ +#define OXYGEN_SPI_CODEC_SHIFT 4 +#define OXYGEN_SPI_MAGIC 0x80 + +#define OXYGEN_SPI_DATA1 0x99 +#define OXYGEN_SPI_DATA2 0x9a +#define OXYGEN_SPI_DATA3 0x9b + +#define OXYGEN_MPU401 0xa0 + +#define OXYGEN_GPI_DATA 0xa4 + +#define OXYGEN_GPI_INTERRUPT_MASK 0xa5 + +#define OXYGEN_GPIO_DATA 0xa6 + +#define OXYGEN_GPIO_CONTROL 0xa8 +/* 0: input, 1: output */ + +#define OXYGEN_GPIO_INTERRUPT_MASK 0xaa + +#define OXYGEN_DEVICE_SENSE 0xac /* ? */ + +#define OXYGEN_PLAY_ROUTING 0xc0 +#define OXYGEN_PLAY_DAC0_SOURCE_MASK 0x0300 +#define OXYGEN_PLAY_DAC1_SOURCE_MASK 0x0700 +#define OXYGEN_PLAY_DAC2_SOURCE_MASK 0x3000 +#define OXYGEN_PLAY_DAC3_SOURCE_MASK 0x7000 + +#define OXYGEN_REC_ROUTING 0xc2 + +#define OXYGEN_ADC_MONITOR 0xc3 +#define OXYGEN_ADC_MONITOR_MULTICH 0x01 +#define OXYGEN_ADC_MONITOR_AC97 0x04 +#define OXYGEN_ADC_MONITOR_SPDIF 0x10 + +#define OXYGEN_A_MONITOR_ROUTING 0xc4 + +#define OXYGEN_AC97_CONTROL 0xd0 +#define OXYGEN_AC97_RESET1 0x0001 +#define OXYGEN_AC97_RESET1_BUSY 0x0002 +#define OXYGEN_AC97_RESET2 0x0008 +#define OXYGEN_AC97_CODEC_0 0x0010 +#define OXYGEN_AC97_CODEC_1 0x0020 + +#define OXYGEN_AC97_INTERRUPT_MASK 0xd2 + +#define OXYGEN_AC97_INTERRUPT_STATUS 0xd3 +#define OXYGEN_AC97_READ_COMPLETE 0x01 +#define OXYGEN_AC97_WRITE_COMPLETE 0x02 + +#define OXYGEN_AC97_OUT_CONFIG 0xd4 +#define OXYGEN_AC97_OUT_MAGIC1 0x00000011 +#define OXYGEN_AC97_OUT_MAGIC2 0x00000033 +#define OXYGEN_AC97_OUT_MAGIC3 0x0000ff00 + +#define OXYGEN_AC97_IN_CONFIG 0xd8 +#define OXYGEN_AC97_IN_MAGIC1 0x00000011 +#define OXYGEN_AC97_IN_MAGIC2 0x00000033 +#define OXYGEN_AC97_IN_MAGIC3 0x00000300 + +#define OXYGEN_AC97_REGS 0xdc +#define OXYGEN_AC97_REG_DATA_MASK 0x0000ffff +#define OXYGEN_AC97_REG_ADDR_MASK 0x007f0000 +#define OXYGEN_AC97_REG_ADDR_SHIFT 16 +#define OXYGEN_AC97_REG_DIR_MASK 0x00800000 +#define OXYGEN_AC97_REG_DIR_WRITE 0x00000000 +#define OXYGEN_AC97_REG_DIR_READ 0x00800000 +#define OXYGEN_AC97_REG_CODEC_MASK 0x01000000 +#define OXYGEN_AC97_REG_CODEC_SHIFT 24 + +#define OXYGEN_DMA_FLUSH 0xe1 +/* OXYGEN_CHANNEL_* */ + +#define OXYGEN_CODEC_VERSION 0xe4 + +#define OXYGEN_REVISION 0xe6 +#define OXYGEN_REVISION_2 0x08 /* bit flag */ +#define OXYGEN_REVISION_8787 0x14 /* all 8 bits */ + +#endif -- cgit v0.10.2 From 1b8ff22fa8d724e7f4367ec220c2c44ae38743fc Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 23 Dec 2007 19:52:08 +0100 Subject: [ALSA] add Asus Xonar driver Add the snd-virtuoso driver for the Asus Virtuoso 200 chip used on the PCI and PCI-E models of the Xonar sound card. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index d3be87d..5baee6b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -885,6 +885,17 @@ config SND_VIA82XX_MODEM To compile this driver as a module, choose M here: the module will be called snd-via82xx-modem. +config SND_VIRTUOSO + tristate "Asus Virtuoso 200 (Xonar)" + depends on SND + select SND_OXYGEN_LIB + help + Say Y here to include support for sound cards based on the + Asus AV200 chip, i.e., Xonar D2 and Xonar D2X. + + To compile this driver as a module, choose M here: the module + will be called snd-virtuoso. + config SND_VX222 tristate "Digigram VX222" depends on SND diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index 99455dd..98d5bc0 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -1,5 +1,7 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o snd-oxygen-objs := oxygen.o +snd-virtuoso-objs := virtuoso.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o obj-$(CONFIG_SND_OXYGEN) += snd-oxygen.o +obj-$(CONFIG_SND_VIRTUOSO) += snd-virtuoso.o diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c new file mode 100644 index 0000000..5678dc3 --- /dev/null +++ b/sound/pci/oxygen/virtuoso.c @@ -0,0 +1,269 @@ +/* + * C-Media CMI8788 driver for Asus Xonar cards + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +/* + * SPI 0 -> 1st PCM1796 (front) + * SPI 1 -> 2nd PCM1796 (side) + * SPI 2 -> 3rd PCM1796 (center/LFE) + * SPI 4 -> 4th PCM1796 (rear) + * + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 5 <- ? (D2X only) + * GPIO 7 -> ALT + * GPIO 8 -> ? (amps enable?) + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "oxygen.h" + +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_DESCRIPTION("Asus AV200 driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Asus,AV200}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "card index"); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string"); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "enable card"); + +static struct pci_device_id xonar_ids[] __devinitdata = { + { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, /* Asus Xonar D2 */ + { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, /* Asus Xonar D2X */ + { } +}; +MODULE_DEVICE_TABLE(pci, xonar_ids); + +/* register 0x12 */ +#define PCM1796_MUTE 0x01 +#define PCM1796_FMT_24_MSB 0x30 +#define PCM1796_ATLD 0x80 +/* register 0x14 */ +#define PCM1796_OS_64 0x00 +#define PCM1796_OS_32 0x01 +#define PCM1796_OS_128 0x02 + +static void pcm1796_write(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + /* maps ALSA channel pair number to SPI output */ + static const u8 codec_map[4] = { + 0, 4, 2, 1 + }; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER_WRITE | + OXYGEN_SPI_DATA_LENGTH_2 | + (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_MAGIC, + (reg << 8) | value); +} + +static void xonar_init(struct oxygen *chip) +{ + unsigned int i; + + for (i = 0; i < 4; ++i) { + pcm1796_write(chip, i, 0x12, PCM1796_FMT_24_MSB | PCM1796_ATLD); + pcm1796_write(chip, i, 0x13, 0); + pcm1796_write(chip, i, 0x14, PCM1796_OS_64); + pcm1796_write(chip, i, 0x15, 0); + pcm1796_write(chip, i, 0x10, 0xff); + pcm1796_write(chip, i, 0x11, 0xff); + } + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, 0x8c); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, 0x00, 0x8c); +#if 0 + oxygen_clear_bits16(chip, OXYGEN_I2S_MULTICH_FORMAT, + OXYGEN_I2S_MAGIC1_MASK); +#endif + oxygen_ac97_set_bits(chip, 0, 0x62, 0x0080); + msleep(300); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, 0x100); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, 0x100); + + snd_component_add(chip->card, "PCM1796"); + snd_component_add(chip->card, "CS5381"); +} + +static void xonar_cleanup(struct oxygen *chip) +{ + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, 0x100); +} + +static void set_pcm1796_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ +#if 0 + unsigned int i; + u8 value; + + value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; + for (i = 0; i < 4; ++i) + pcm1796_write(chip, i, 0x14, value); +#endif +} + +static void update_pcm1796_volume(struct oxygen *chip) +{ + unsigned int i; + + for (i = 0; i < 4; ++i) { + pcm1796_write(chip, i, 0x10, chip->dac_volume[i * 2]); + pcm1796_write(chip, i, 0x11, chip->dac_volume[i * 2 + 1]); + } +} + +static void update_pcm1796_mute(struct oxygen *chip) +{ + unsigned int i; + u8 value; + + value = PCM1796_FMT_24_MSB | PCM1796_ATLD; + if (chip->dac_mute) + value |= PCM1796_MUTE; + for (i = 0; i < 4; ++i) + pcm1796_write(chip, i, 0x12, value); +} + +static void set_cs5381_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + unsigned int value; + + if (params_rate(params) <= 54000) + value = 0; + else if (params_rate(params) <= 108000) + value = 4; + else + value = 8; + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, value, 0x000c); +} + +static int alt_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + value->value.integer.value[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & 0x80); + return 0; +} + +static int alt_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 old_bits, new_bits; + int changed; + + spin_lock_irq(&chip->reg_lock); + old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (value->value.integer.value[0]) + new_bits = old_bits | 0x80; + else + new_bits = old_bits & ~0x80; + changed = new_bits != old_bits; + if (changed) + oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits); + spin_unlock_irq(&chip->reg_lock); + return changed; +} + +static const struct snd_kcontrol_new alt_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Loopback Switch", + .info = snd_ctl_boolean_mono_info, + .get = alt_switch_get, + .put = alt_switch_put, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0); + +static int xonar_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); +} + +static const struct oxygen_model model_xonar = { + .shortname = "Asus AV200", + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_init, + .mixer_init = xonar_mixer_init, + .cleanup = xonar_cleanup, + .set_dac_params = set_pcm1796_params, + .set_adc_params = set_cs5381_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .dac_tlv = pcm1796_db_scale, + .record_from_dma_b = 1, + .cd_in_from_video_in = 1, + .dac_minimum_volume = 15, +}; + +static int __devinit xonar_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + ++dev; + return -ENOENT; + } + err = oxygen_pci_probe(pci, index[dev], id[dev], &model_xonar); + if (err >= 0) + ++dev; + return err; +} + +static struct pci_driver xonar_driver = { + .name = "AV200", + .id_table = xonar_ids, + .probe = xonar_probe, + .remove = __devexit_p(oxygen_pci_remove), +}; + +static int __init alsa_card_xonar_init(void) +{ + return pci_register_driver(&xonar_driver); +} + +static void __exit alsa_card_xonar_exit(void) +{ + pci_unregister_driver(&xonar_driver); +} + +module_init(alsa_card_xonar_init) +module_exit(alsa_card_xonar_exit) -- cgit v0.10.2 From 7db756f2b1aab15b2d4834ebb373bb5dc07d95dc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 24 Dec 2007 14:36:09 +0100 Subject: [ALSA] hda-codec - Sort ad1986a cfg table Sort the ad1986a config table by PCI SSID (the last toshiba entry was added wrongly). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 5e70d55..0fa5f00 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -848,6 +848,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_LAPTOP_EAPD), @@ -859,7 +860,6 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), {} }; -- cgit v0.10.2 From 93521d274b7fb4e6da5772768683e4984783d9e7 Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Mon, 24 Dec 2007 14:40:56 +0100 Subject: [ALSA] sound/usb/usbaudio.c: fix build with CONFIG_PM=n sound/usb/usbaudio.c: In function 'usb_audio_suspend': sound/usb/usbaudio.c:3674: error: implicit declaration of function 'snd_pcm_sus\pend_all' Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c52461d..c6d628a 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2077,8 +2077,14 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, static int usb_audio_probe(struct usb_interface *intf, const struct usb_device_id *id); static void usb_audio_disconnect(struct usb_interface *intf); + +#ifdef CONFIG_PM static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message); static int usb_audio_resume(struct usb_interface *intf); +#else +#define usb_audio_suspend NULL +#define usb_audio_resume NULL +#endif static struct usb_device_id usb_audio_ids [] = { #include "usbquirks.h" @@ -3658,6 +3664,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) dev_get_drvdata(&intf->dev)); } +#ifdef CONFIG_PM static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) { struct snd_usb_audio *chip = dev_get_drvdata(&intf->dev); @@ -3695,6 +3702,7 @@ static int usb_audio_resume(struct usb_interface *intf) return 0; } +#endif /* CONFIG_PM */ static int __init snd_usb_audio_init(void) { -- cgit v0.10.2 From 740eb8358f2bf337fd4ec62abb5b8cca7c652e5a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 4 Jan 2008 09:22:20 +0100 Subject: [ALSA] oxygen: use uintptr_t in pointer casts When we store the DMA channel number in the substream's private_data pointer, use uintptr_t as an intermediate step when casting from/to unsigned int to prevent the compiler from whining when the pointer size is different. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 941399b..ff46ba5 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -168,6 +168,12 @@ static struct snd_pcm_hardware oxygen_hardware[PCM_COUNT] = { }, }; +static inline unsigned int +oxygen_substream_channel(struct snd_pcm_substream *substream) +{ + return (unsigned int)(uintptr_t)substream->runtime->private_data; +} + static int oxygen_open(struct snd_pcm_substream *substream, unsigned int channel) { @@ -175,7 +181,7 @@ static int oxygen_open(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; int err; - runtime->private_data = (void *)channel; + runtime->private_data = (void *)(uintptr_t)channel; runtime->hw = oxygen_hardware[channel]; err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); @@ -248,7 +254,7 @@ static int oxygen_ac97_open(struct snd_pcm_substream *substream) static int oxygen_close(struct snd_pcm_substream *substream) { struct oxygen *chip = snd_pcm_substream_chip(substream); - unsigned int channel = (unsigned int)substream->runtime->private_data; + unsigned int channel = oxygen_substream_channel(substream); mutex_lock(&chip->mutex); chip->pcm_active &= ~(1 << channel); @@ -337,7 +343,7 @@ static int oxygen_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct oxygen *chip = snd_pcm_substream_chip(substream); - unsigned int channel = (unsigned int)substream->runtime->private_data; + unsigned int channel = oxygen_substream_channel(substream); int err; err = snd_pcm_lib_malloc_pages(substream, @@ -516,7 +522,7 @@ static int oxygen_ac97_hw_params(struct snd_pcm_substream *substream, static int oxygen_hw_free(struct snd_pcm_substream *substream) { struct oxygen *chip = snd_pcm_substream_chip(substream); - unsigned int channel = (unsigned int)substream->runtime->private_data; + unsigned int channel = oxygen_substream_channel(substream); spin_lock_irq(&chip->reg_lock); chip->interrupt_mask &= ~(1 << channel); @@ -540,7 +546,7 @@ static int oxygen_spdif_hw_free(struct snd_pcm_substream *substream) static int oxygen_prepare(struct snd_pcm_substream *substream) { struct oxygen *chip = snd_pcm_substream_chip(substream); - unsigned int channel = (unsigned int)substream->runtime->private_data; + unsigned int channel = oxygen_substream_channel(substream); unsigned int channel_mask = 1 << channel; spin_lock_irq(&chip->reg_lock); @@ -575,7 +581,7 @@ static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd) snd_pcm_group_for_each_entry(s, substream) { if (snd_pcm_substream_chip(s) == chip) { - mask |= 1 << (unsigned int)s->runtime->private_data; + mask |= 1 << oxygen_substream_channel(s); snd_pcm_trigger_done(s, substream); } } @@ -594,7 +600,7 @@ static snd_pcm_uframes_t oxygen_pointer(struct snd_pcm_substream *substream) { struct oxygen *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int channel = (unsigned int)runtime->private_data; + unsigned int channel = oxygen_substream_channel(substream); u32 curr_addr; /* no spinlock, this read should be atomic */ -- cgit v0.10.2 From c481fca3e68155459e9805dc8c303e012ccbee54 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 7 Jan 2008 12:18:28 +0100 Subject: [ALSA] hda: STAC927x VREF fix Some laptops incorrectly assume the front input jack as a line in instead of a microphone in. Which in turn disables the voltage reference, in which non-amplified input is not possible. This patch enables VREF80 for the input jack. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a074155..4e71bba 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3344,6 +3344,9 @@ static int patch_stac927x(struct hda_codec *codec) spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + /* correct the front input jack as a mic */ + stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); + /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ spec->gpio_mask = spec->gpio_data = 0x00000004; -- cgit v0.10.2 From ebf00c54a6e1be0ae25d41236a063747c74ed5bf Mon Sep 17 00:00:00 2001 From: Jerone Young Date: Mon, 7 Jan 2008 12:22:18 +0100 Subject: [ALSA] hda-codec - Add IEC958 digital out support for Lenovo Thinkpads T61/X61 This patch adds IEC958 digital out support for the AD1984 sound card. This card can be found in Lenovo Thinkapds T61/X61. The digital out is not located on the Thinkpad, but optional docking station (it's coxial digital out). I've add this support as it is done the exact same way for the AD1983 & AD1884. I have tested this patch with my Lenovo Thinkpad T61 hooked up to a docking station (that has the digital coxial) and then run to my home theater reciever. Works like a charm :-) Signed-off-by: Jerone Young Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0fa5f00..67144dc 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3095,6 +3095,16 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { .get = ad198x_mux_enum_get, .put = ad198x_mux_enum_put, }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, { } /* end */ }; @@ -3197,7 +3207,7 @@ static int patch_ad1984(struct hda_codec *codec) codec->patch_ops.build_pcms = ad1984_build_pcms; break; case AD1984_THINKPAD: - spec->multiout.dig_out_nid = 0; + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; spec->input_mux = &ad1984_thinkpad_capture_source; spec->mixers[0] = ad1984_thinkpad_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; -- cgit v0.10.2 From 1bc9eed379399484d3f5d5a0834674983969bc1e Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 7 Jan 2008 12:24:45 +0100 Subject: [ALSA] es18xx: Enable wavetable input from ESS chips This patch enables wavetable chips ES689/ES69X connected to ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal from the wavetable is active. It has no effect if there is no ESS wavetable chip present. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index ece5041..91cb478 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1442,6 +1442,8 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) snd_es18xx_write(chip, 0xB2, 0x50); /* Enable MPU and hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x42); + /* Enable ESS wavetable input */ + snd_es18xx_mixer_bits(chip, 0x48, 0x10, 0x10); } else { int irqmask, dma1mask, dma2mask; -- cgit v0.10.2 From f11b799282201fbd8c88b51815176a902b1f15b8 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Mon, 7 Jan 2008 13:33:45 +0100 Subject: [ALSA] sound: Use time_before, time_before_eq, etc. The functions time_before, time_before_eq, time_after, and time_after_eq are more robust for comparing jiffies against other values. A simplified version of the semantic patch making this change is as follows: (http://www.emn.fr/x-info/coccinelle/) // @ change_compare_np @ expression E; @@ ( - jiffies <= E + time_before_eq(jiffies,E) | - jiffies >= E + time_after_eq(jiffies,E) | - jiffies < E + time_before(jiffies,E) | - jiffies > E + time_after(jiffies,E) ) @ include depends on change_compare_np @ @@ #include @ no_include depends on !include && change_compare_np @ @@ #include + #include // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 3958dbd..f415601 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -43,6 +43,7 @@ #include #include +#include #include @@ -694,7 +695,7 @@ static void snd_uart16550_output_write(struct snd_rawmidi_substream *substream) (uart->adaptor == SNDRV_SERIAL_SOUNDCANVAS || uart->adaptor == SNDRV_SERIAL_GENERIC) && (uart->prev_out != substream->number || - jiffies-lasttime > 3*HZ)) { + time_after(jiffies, lasttime + 3*HZ))) { if (snd_uart16550_buffer_can_write(uart, 3)) { /* Roland Soundcanvas part selection */ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index fe30e0d..9d8af3a 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include @@ -185,7 +186,7 @@ static int s3c24xx_snd_lrsync(void) if (iiscon & S3C2410_IISCON_LRINDEX) break; - if (timeout < jiffies) + if (time_after(jiffies, timeout)) return -ETIMEDOUT; } -- cgit v0.10.2 From 5aba4f8ec72b2b880c694b5ce58e67be508f7b7a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Jan 2008 15:16:37 +0100 Subject: [ALSA] hda-intel - Support multiple devices It turned out that there can be multiple HD-audio devices on a single machine (e.g. on-board audio and HDMI on graphic cards), so we need to support multiple devices with snd-hda-intel driver. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index aded3b4..afdb6ff 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -744,9 +744,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. VIA VT8251/VT8237A, SIS966, ULI M5461 + [Multiple options for each card instance] model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) + + [Single (global) options] single_cmd - Use single immediate commands to communicate with codecs (for debugging only) enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) @@ -755,8 +758,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. power_save_controller - Reset HD-audio controller in power-saving mode (default = on) - This module supports one card and autoprobe. - + This module supports multiple cards and autoprobe. + Each codec may have a model table for different configurations. If your machine isn't listed there, the default (usually minimal) configuration is set up. You can pass "model=" option to diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 41edf85..a37e894 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -50,29 +50,32 @@ #include "hda_codec.h" -static int index = SNDRV_DEFAULT_IDX1; -static char *id = SNDRV_DEFAULT_STR1; -static char *model; -static int position_fix; -static int probe_mask = -1; +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static char *model[SNDRV_CARDS]; +static int position_fix[SNDRV_CARDS]; +static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int single_cmd; static int enable_msi; -module_param(index, int, 0444); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); -module_param(id, charp, 0444); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Intel HD audio interface."); -module_param(model, charp, 0444); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Intel HD audio interface."); +module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); -module_param(position_fix, int, 0444); +module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "Fix DMA pointer " "(0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)."); -module_param(probe_mask, int, 0444); +module_param_array(probe_mask, int, NULL, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); module_param(single_cmd, bool, 0444); MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); -module_param(enable_msi, int, 0); +module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -87,10 +90,6 @@ module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif -/* just for backward compatibility */ -static int enable; -module_param(enable, bool, 0444); - MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH6M}," @@ -1038,7 +1037,8 @@ static unsigned int azx_max_codecs[] __devinitdata = { [AZX_DRIVER_NVIDIA] = 3, /* FIXME: correct? */ }; -static int __devinit azx_codec_create(struct azx *chip, const char *model) +static int __devinit azx_codec_create(struct azx *chip, const char *model, + unsigned int codec_probe_mask) { struct hda_bus_template bus_temp; int c, codecs, audio_codecs, err; @@ -1059,7 +1059,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) codecs = audio_codecs = 0; for (c = 0; c < AZX_MAX_CODECS; c++) { - if ((chip->codec_mask & (1 << c)) & probe_mask) { + if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { struct hda_codec *codec; err = snd_hda_codec_new(chip->bus, c, &codec); if (err < 0) @@ -1072,7 +1072,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) if (!audio_codecs) { /* probe additional slots if no codec is found */ for (; c < azx_max_codecs[chip->driver_type]; c++) { - if ((chip->codec_mask & (1 << c)) & probe_mask) { + if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { err = snd_hda_codec_new(chip->bus, c, NULL); if (err < 0) continue; @@ -1683,18 +1683,18 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { {} }; -static void __devinit check_probe_mask(struct azx *chip) +static void __devinit check_probe_mask(struct azx *chip, int dev) { const struct snd_pci_quirk *q; - if (probe_mask == -1) { + if (probe_mask[dev] == -1) { q = snd_pci_quirk_lookup(chip->pci, probe_mask_list); if (q) { printk(KERN_INFO "hda_intel: probe_mask set to 0x%x " "for device %04x:%04x\n", q->value, q->subvendor, q->subdevice); - probe_mask = q->value; + probe_mask[dev] = q->value; } } } @@ -1704,7 +1704,7 @@ static void __devinit check_probe_mask(struct azx *chip) * constructor */ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, - int driver_type, + int dev, int driver_type, struct azx **rchip) { struct azx *chip; @@ -1734,8 +1734,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->driver_type = driver_type; chip->msi = enable_msi; - chip->position_fix = check_position_fix(chip, position_fix); - check_probe_mask(chip); + chip->position_fix = check_position_fix(chip, position_fix[dev]); + check_probe_mask(chip, dev); chip->single_cmd = single_cmd; @@ -1876,17 +1876,25 @@ static void power_down_all_codecs(struct azx *chip) static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { + static int dev; struct snd_card *card; struct azx *chip; int err; - card = snd_card_new(index, id, THIS_MODULE, 0); + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (!card) { snd_printk(KERN_ERR SFX "Error creating card!\n"); return -ENOMEM; } - err = azx_create(card, pci, pci_id->driver_data, &chip); + err = azx_create(card, pci, dev, pci_id->driver_data, &chip); if (err < 0) { snd_card_free(card); return err; @@ -1894,7 +1902,7 @@ static int __devinit azx_probe(struct pci_dev *pci, card->private_data = chip; /* create codec instances */ - err = azx_codec_create(chip, model); + err = azx_codec_create(chip, model[dev], probe_mask[dev]); if (err < 0) { snd_card_free(card); return err; -- cgit v0.10.2 From 0678accd2da33873455ef8d41d847bd550727159 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 8 Jan 2008 12:10:50 +0100 Subject: [ALSA] hda: Dynamically create digital gain mixers Dynamically create digital gain mixers for dmics that have out-amp support. Also some 92HD73xx's codecs don't have DMIC gains, so this also prevents creating dead mixers. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4e71bba..299e02a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -463,9 +463,6 @@ static struct hda_verb stac92hd73xx_6ch_core_init[] = { /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -490,9 +487,6 @@ static struct hda_verb stac92hd73xx_8ch_core_init[] = { /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -519,9 +513,6 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -635,10 +626,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -665,10 +652,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -695,10 +678,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -725,10 +704,6 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), @@ -746,10 +721,6 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), @@ -2313,15 +2284,18 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *dimux = &spec->private_dimux; hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - int i, j; + int err, i, j; + char name[32]; dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0]; dimux->items[dimux->num_items].index = 0; dimux->num_items++; for (i = 0; i < spec->num_dmics; i++) { + hda_nid_t nid; int index; int num_cons; + unsigned int wcaps; unsigned int def_conf; def_conf = snd_hda_codec_read(codec, @@ -2332,17 +2306,32 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; + nid = spec->dmic_nids[i]; num_cons = snd_hda_get_connections(codec, spec->dmux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); for (j = 0; j < num_cons; j++) - if (con_lst[j] == spec->dmic_nids[i]) { + if (con_lst[j] == nid) { index = j; goto found; } continue; found: + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + sprintf(name, "%s Capture Volume", + stac92xx_dmic_labels[dimux->num_items]); + + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_VOL, + name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + dimux->items[dimux->num_items].label = stac92xx_dmic_labels[dimux->num_items]; dimux->items[dimux->num_items].index = index; -- cgit v0.10.2 From a713b5834731f32757b30de038dcb995afac2ad1 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 8 Jan 2008 12:24:01 +0100 Subject: [ALSA] PCM core - remove SNDRV_PCM_TSTAMP_MMAP condition in snd_pcm_status() The condition caused that the returned ring buffer position does not match with timestamp when SNDRV_PCM_TSTAMP_MMAP mode was enabled. Removing condition makes unified behaviour and interrupt based timestamp can be accessed via PCM_IOCTL_SYNC_PTR or mmaped status area. Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 2e7b1e6..3fc33de 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -593,14 +593,9 @@ int snd_pcm_status(struct snd_pcm_substream *substream, if (status->state == SNDRV_PCM_STATE_OPEN) goto _end; status->trigger_tstamp = runtime->trigger_tstamp; - if (snd_pcm_running(substream)) { + if (snd_pcm_running(substream)) snd_pcm_update_hw_ptr(substream); - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP) - status->tstamp = runtime->status->tstamp; - else - snd_pcm_gettime(runtime, &status->tstamp); - } else - snd_pcm_gettime(runtime, &status->tstamp); + snd_pcm_gettime(runtime, &status->tstamp); status->appl_ptr = runtime->control->appl_ptr; status->hw_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- cgit v0.10.2 From f8225f6d1f68c3d0a0fe844dc40a11cd432a853b Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 8 Jan 2008 12:16:54 +0100 Subject: [ALSA] hda-codec - Add EAPD controls for ALC260 test model This implements a switch control for the EAPD signal output by the ALC26x chips. Since some laptops may utilise this to activate useful things it is handy to have a control for this in the ALC26x test models. The patch includes the control in the ALC260 test model. Signed-off-by: Jonathan Woithe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 617f3ee..0a64d24 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -608,6 +608,59 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, .private_value = nid | (mask<<16) } #endif /* CONFIG_SND_DEBUG */ +/* A switch control to allow the enabling EAPD digital outputs on the ALC26x. + * Again, this is only used in the ALC26x test models to help identify when + * the EAPD line must be asserted for features to work. + */ +#ifdef CONFIG_SND_DEBUG +#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info + +static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_EAPD_BTLENABLE, 0x00); + + *valp = (val & mask) != 0; + return 0; +} + +static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_EAPD_BTLENABLE, + 0x00); + + /* Set/unset the masked control bit(s) as needed */ + change = (!val ? 0 : mask) != (ctrl_data & mask); + if (!val) + ctrl_data &= ~mask; + else + ctrl_data |= mask; + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, + ctrl_data); + + return change; +} + +#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .info = alc_eapd_ctrl_info, \ + .get = alc_eapd_ctrl_get, \ + .put = alc_eapd_ctrl_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ + /* * set up from the preset table */ @@ -4332,6 +4385,12 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), + /* A switch allowing EAPD to be enabled. Some laptops seem to use + * this output to turn on an external amplifier. + */ + ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02), + ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02), + { } /* end */ }; static struct hda_verb alc260_test_init_verbs[] = { -- cgit v0.10.2 From 86c53bd26b431767438c65f89863471be177f0f7 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 8 Jan 2008 12:33:19 +0100 Subject: [ALSA] hda-codec - Add test model for ALC268 This implements a test model for the ALC268. It depends on the feature added by alc260-test-eapd-0.2.diff. This patch also adds a mention of the ALC260 test model to ALSA-Configuration.txt since this seems to have been missed. Signed-off-by: Jonathan Woithe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index afdb6ff..22aee1a 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -801,6 +801,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. will Will laptops (PB V7900) replacer Replacer 672V basic fixed pin assignment (old default model) + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y auto auto-config reading BIOS (default) ALC262 @@ -821,6 +824,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y auto auto-config reading BIOS (default) ALC662 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0a64d24..1b2ad52 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -106,6 +106,9 @@ enum { ALC268_3ST, ALC268_TOSHIBA, ALC268_ACER, +#ifdef CONFIG_SND_DEBUG + ALC268_TEST, +#endif ALC268_AUTO, ALC268_MODEL_LAST /* last tag */ }; @@ -9366,6 +9369,60 @@ static struct hda_input_mux alc268_capture_source = { }, }; +#ifdef CONFIG_SND_DEBUG +static struct snd_kcontrol_new alc268_test_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + + /* Volume widgets */ + HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT), + HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT), + HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT), + HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT), + HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT), + HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT), + HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT), + HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT), + + /* Modes for retasking pin widgets */ + ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT), + + /* Controls for GPIO pins, assuming they are configured as outputs */ + ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01), + ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02), + ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), + ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), + + /* Switches to allow the digital SPDIF output pin to be enabled. + * The ALC268 does not have an SPDIF input. + */ + ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01), + + /* A switch allowing EAPD to be enabled. Some laptops seem to use + * this output to turn on an external amplifier. + */ + ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02), + ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02), + + { } /* end */ +}; +#endif + /* create input playback/capture controls for the given pin */ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, const char *ctlname, int idx) @@ -9578,6 +9635,9 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", +#ifdef CONFIG_SND_DEBUG + [ALC268_TEST] = "test", +#endif [ALC268_AUTO] = "auto", }; @@ -9636,6 +9696,22 @@ static struct alc_config_preset alc268_presets[] = { .unsol_event = alc268_acer_unsol_event, .init_hook = alc268_acer_init_hook, }, +#ifdef CONFIG_SND_DEBUG + [ALC268_TEST] = { + .mixers = { alc268_test_mixer, alc268_capture_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc268_volume_init_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .dig_out_nid = ALC268_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + }, +#endif }; static int patch_alc268(struct hda_codec *codec) -- cgit v0.10.2 From 38f8b397e15033b716f0b550f0ec780ad3d890e6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Jan 2008 17:19:22 +0100 Subject: [ALSA] hda-codec - Add model=laptop for HP 350 laptop Added the proper model=laptop for HP 350 laptop with Cxt5045 codec. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4609b55..e960189 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -762,6 +762,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), -- cgit v0.10.2 From 8ace4f3c9d83fd60e7539526a3a70bf5730db8c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Jan 2008 17:57:26 +0100 Subject: [ALSA] Remove indirect control access This patch removes the indirect control access to the control elements. The indirect access has never been used and is even broken on 32bit ioctl wrapper. Let's clean it up. The pointers still remain in snd_ctl_elem_* structs just to make sure that the struct size won't change. Once after checking the size consistency, we can get rid of them, too. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/asound.h b/include/sound/asound.h index eda5c63..ee753d7 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -696,7 +696,7 @@ struct snd_timer_tread { * * ****************************************************************************/ -#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 4) +#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 5) struct snd_ctl_card_info { int card; /* card number */ @@ -745,8 +745,7 @@ typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_ACCESS_OWNER (1<<10) /* write lock owner */ #define SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK (1<<28) /* kernel use a TLV callback */ #define SNDRV_CTL_ELEM_ACCESS_USER (1<<29) /* user space element */ -#define SNDRV_CTL_ELEM_ACCESS_DINDIRECT (1<<30) /* indirect access for matrix dimensions in the info structure */ -#define SNDRV_CTL_ELEM_ACCESS_INDIRECT (1<<31) /* indirect access for element value in the value structure */ +/* bits 30 and 31 are obsoleted (for indirect access) */ /* for further details see the ACPI and PCI power management specification */ #define SNDRV_CTL_POWER_D0 0x0000 /* full On */ @@ -800,30 +799,30 @@ struct snd_ctl_elem_info { } value; union { unsigned short d[4]; /* dimensions */ - unsigned short *d_ptr; /* indirect */ + unsigned short *d_ptr; /* indirect - obsoleted */ } dimen; unsigned char reserved[64-4*sizeof(unsigned short)]; }; struct snd_ctl_elem_value { struct snd_ctl_elem_id id; /* W: element ID */ - unsigned int indirect: 1; /* W: use indirect pointer (xxx_ptr member) */ + unsigned int indirect: 1; /* W: indirect access - obsoleted */ union { union { long value[128]; - long *value_ptr; + long *value_ptr; /* obsoleted */ } integer; union { long long value[64]; - long long *value_ptr; + long long *value_ptr; /* obsoleted */ } integer64; union { unsigned int item[128]; - unsigned int *item_ptr; + unsigned int *item_ptr; /* obsoleted */ } enumerated; union { unsigned char data[512]; - unsigned char *data_ptr; + unsigned char *data_ptr; /* obsoleted */ } bytes; struct snd_aes_iec958 iec958; } value; /* RO */ diff --git a/sound/core/control.c b/sound/core/control.c index df0774c..c89ca0d 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -232,8 +232,6 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_DINDIRECT| - SNDRV_CTL_ELEM_ACCESS_INDIRECT| SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); kctl.info = ncontrol->info; @@ -692,7 +690,7 @@ int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control) struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; unsigned int index_offset; - int result, indirect; + int result; down_read(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &control->id); @@ -701,17 +699,12 @@ int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control) } else { index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; - indirect = vd->access & SNDRV_CTL_ELEM_ACCESS_INDIRECT ? 1 : 0; - if (control->indirect != indirect) { - result = -EACCES; - } else { - if ((vd->access & SNDRV_CTL_ELEM_ACCESS_READ) && kctl->get != NULL) { - snd_ctl_build_ioff(&control->id, kctl, index_offset); - result = kctl->get(kctl, control); - } else { - result = -EPERM; - } - } + if ((vd->access & SNDRV_CTL_ELEM_ACCESS_READ) && + kctl->get != NULL) { + snd_ctl_build_ioff(&control->id, kctl, index_offset); + result = kctl->get(kctl, control); + } else + result = -EPERM; } up_read(&card->controls_rwsem); return result; @@ -748,7 +741,7 @@ int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; unsigned int index_offset; - int result, indirect; + int result; down_read(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &control->id); @@ -757,23 +750,19 @@ int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, } else { index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; - indirect = vd->access & SNDRV_CTL_ELEM_ACCESS_INDIRECT ? 1 : 0; - if (control->indirect != indirect) { - result = -EACCES; + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) || + kctl->put == NULL || + (file && vd->owner && vd->owner != file)) { + result = -EPERM; } else { - if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) || - kctl->put == NULL || - (file && vd->owner != NULL && vd->owner != file)) { - result = -EPERM; - } else { - snd_ctl_build_ioff(&control->id, kctl, index_offset); - result = kctl->put(kctl, control); - } - if (result > 0) { - up_read(&card->controls_rwsem); - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &control->id); - return 0; - } + snd_ctl_build_ioff(&control->id, kctl, index_offset); + result = kctl->put(kctl, control); + } + if (result > 0) { + up_read(&card->controls_rwsem); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &control->id); + return 0; } } up_read(&card->controls_rwsem); -- cgit v0.10.2 From fa5717f2099aadb2083d5df4d19af8f9685fa03e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Jan 2008 18:00:04 +0100 Subject: [ALSA] Fix PCM write blocking The snd_pcm_lib_write1() may block in some weird condition: - the stream isn't started - avail_min is big (e.g. period size) - partial write up to buffer_size - avail_min The patch fixes this invalid blocking problem. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 48ffa40..f9f9b3f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1655,8 +1655,11 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (runtime->sleep_min == 0 && runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); avail = snd_pcm_playback_avail(runtime); - if (((avail < runtime->control->avail_min && size > avail) || - (size >= runtime->xfer_align && avail < runtime->xfer_align))) { + if (!avail || + (snd_pcm_running(substream) && + ((avail < runtime->control->avail_min && size > avail) || + (size >= runtime->xfer_align && + avail < runtime->xfer_align)))) { wait_queue_t wait; enum { READY, SIGNALED, ERROR, SUSPENDED, EXPIRED, DROPPED } state; long tout; -- cgit v0.10.2 From d948035a928400ae127c873fbf771389bee18949 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Jan 2008 18:05:26 +0100 Subject: [ALSA] Remove PCM xfer_align sw params The xfer_align sw_params parameter has never been used in a sane manner, and no one understands what this does exactly. The current implementation looks also buggy because it allows write of shorter size than xfer_align. So, if you do partial writes, the write isn't actually aligned at all. Removing this parameter will make some pcm_lib_* code more readable (and less buggy). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 48e4053..835e320 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -2188,7 +2188,6 @@ struct _snd_pcm_runtime { struct timespec tstamp_mode; /* mmap timestamp is updated */ unsigned int period_step; unsigned int sleep_min; /* min ticks to sleep */ - snd_pcm_uframes_t xfer_align; /* xfer size need to be a multiple */ snd_pcm_uframes_t start_threshold; snd_pcm_uframes_t stop_threshold; snd_pcm_uframes_t silence_threshold; /* Silence filling happens when diff --git a/include/sound/asound.h b/include/sound/asound.h index ee753d7..069d912 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -363,7 +363,7 @@ struct snd_pcm_sw_params { unsigned int period_step; unsigned int sleep_min; /* min ticks to sleep */ snd_pcm_uframes_t avail_min; /* min avail frames for wakeup */ - snd_pcm_uframes_t xfer_align; /* xfer size need to be a multiple */ + snd_pcm_uframes_t xfer_align; /* obsolete: xfer size need to be a multiple */ snd_pcm_uframes_t start_threshold; /* min hw_avail frames for automatic start */ snd_pcm_uframes_t stop_threshold; /* min avail frames for automatic stop */ snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 65f6362..1270cbc 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -287,7 +287,6 @@ struct snd_pcm_runtime { int tstamp_mode; /* mmap timestamp is updated */ unsigned int period_step; unsigned int sleep_min; /* min ticks to sleep */ - snd_pcm_uframes_t xfer_align; /* xfer size need to be a multiple */ snd_pcm_uframes_t start_threshold; snd_pcm_uframes_t stop_threshold; snd_pcm_uframes_t silence_threshold; /* Silence filling happens when diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d0c4ceb..be089cc 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -988,7 +988,6 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) sw_params->sleep_min = 0; sw_params->avail_min = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : runtime->period_size; - sw_params->xfer_align = 1; if (atomic_read(&substream->mmap_count) || substream->oss.setup.nosilence) { sw_params->silence_threshold = 0; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cf9b949..97cb681 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -389,7 +389,6 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "period_step: %u\n", runtime->period_step); snd_iprintf(buffer, "sleep_min: %u\n", runtime->sleep_min); snd_iprintf(buffer, "avail_min: %lu\n", runtime->control->avail_min); - snd_iprintf(buffer, "xfer_align: %lu\n", runtime->xfer_align); snd_iprintf(buffer, "start_threshold: %lu\n", runtime->start_threshold); snd_iprintf(buffer, "stop_threshold: %lu\n", runtime->stop_threshold); snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f9f9b3f..c1c1556 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1628,8 +1628,6 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (size == 0) return 0; - if (size > runtime->xfer_align) - size -= size % runtime->xfer_align; snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { @@ -1657,9 +1655,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); if (!avail || (snd_pcm_running(substream) && - ((avail < runtime->control->avail_min && size > avail) || - (size >= runtime->xfer_align && - avail < runtime->xfer_align)))) { + (avail < runtime->control->avail_min && size > avail))) { wait_queue_t wait; enum { READY, SIGNALED, ERROR, SUSPENDED, EXPIRED, DROPPED } state; long tout; @@ -1731,8 +1727,6 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, break; } } - if (avail > runtime->xfer_align) - avail -= avail % runtime->xfer_align; frames = size > avail ? avail : size; cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size; if (frames > cont) @@ -1900,8 +1894,6 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (size == 0) return 0; - if (size > runtime->xfer_align) - size -= size % runtime->xfer_align; snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { @@ -1936,12 +1928,12 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, __draining: avail = snd_pcm_capture_avail(runtime); if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { - if (avail < runtime->xfer_align) { + if (!avail) { err = -EPIPE; goto _end_unlock; } - } else if ((avail < runtime->control->avail_min && size > avail) || - (size >= runtime->xfer_align && avail < runtime->xfer_align)) { + } else if (avail < runtime->control->avail_min && + size > avail) { wait_queue_t wait; enum { READY, SIGNALED, ERROR, SUSPENDED, EXPIRED, DROPPED } state; long tout; @@ -2014,8 +2006,6 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, break; } } - if (avail > runtime->xfer_align) - avail -= avail % runtime->xfer_align; frames = size > avail ? avail : size; cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size; if (frames > cont) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3fc33de..3c22d78 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -435,7 +435,6 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->period_step = 1; runtime->sleep_min = 0; runtime->control->avail_min = runtime->period_size; - runtime->xfer_align = runtime->period_size; runtime->start_threshold = 1; runtime->stop_threshold = runtime->buffer_size; runtime->silence_threshold = 0; @@ -532,9 +531,6 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, return -EINVAL; if (params->avail_min == 0) return -EINVAL; - if (params->xfer_align == 0 || - params->xfer_align % runtime->min_align != 0) - return -EINVAL; if (params->silence_size >= runtime->boundary) { if (params->silence_threshold != 0) return -EINVAL; @@ -553,7 +549,6 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, runtime->stop_threshold = params->stop_threshold; runtime->silence_threshold = params->silence_threshold; runtime->silence_size = params->silence_size; - runtime->xfer_align = params->xfer_align; params->boundary = runtime->boundary; if (snd_pcm_running(substream)) { if (runtime->sleep_min) @@ -2239,8 +2234,6 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst } if (frames > (snd_pcm_uframes_t)hw_avail) frames = hw_avail; - else - frames -= frames % runtime->xfer_align; appl_ptr = runtime->control->appl_ptr - frames; if (appl_ptr < 0) appl_ptr += runtime->boundary; @@ -2289,8 +2282,6 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr } if (frames > (snd_pcm_uframes_t)hw_avail) frames = hw_avail; - else - frames -= frames % runtime->xfer_align; appl_ptr = runtime->control->appl_ptr - frames; if (appl_ptr < 0) appl_ptr += runtime->boundary; @@ -2340,8 +2331,6 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs } if (frames > (snd_pcm_uframes_t)avail) frames = avail; - else - frames -= frames % runtime->xfer_align; appl_ptr = runtime->control->appl_ptr + frames; if (appl_ptr >= (snd_pcm_sframes_t)runtime->boundary) appl_ptr -= runtime->boundary; @@ -2391,8 +2380,6 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst } if (frames > (snd_pcm_uframes_t)avail) frames = avail; - else - frames -= frames % runtime->xfer_align; appl_ptr = runtime->control->appl_ptr + frames; if (appl_ptr >= (snd_pcm_sframes_t)runtime->boundary) appl_ptr -= runtime->boundary; -- cgit v0.10.2 From 130755108ba03461f69da990e54e02a254accd23 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Jan 2008 18:08:14 +0100 Subject: [ALSA] PCM - clean up snd_pcm_lib_read/write Introduce a common helper function for snd_pcm_lib_read and snd_pcm_lib_write for cleaning up the code. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index c1c1556..b406630 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1591,6 +1591,71 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) EXPORT_SYMBOL(snd_pcm_period_elapsed); +/* + * Wait until avail_min data becomes available + * Returns a negative error code if any error occurs during operation. + * The available space is stored on availp. When err = 0 and avail = 0 + * on the capture stream, it indicates the stream is in DRAINING state. + */ +static int wait_for_avail_min(struct snd_pcm_substream *substream, + snd_pcm_uframes_t *availp) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + wait_queue_t wait; + int err = 0; + snd_pcm_uframes_t avail = 0; + long tout; + + init_waitqueue_entry(&wait, current); + add_wait_queue(&runtime->sleep, &wait); + for (;;) { + if (signal_pending(current)) { + err = -ERESTARTSYS; + break; + } + set_current_state(TASK_INTERRUPTIBLE); + snd_pcm_stream_unlock_irq(substream); + tout = schedule_timeout(msecs_to_jiffies(10000)); + snd_pcm_stream_lock_irq(substream); + switch (runtime->status->state) { + case SNDRV_PCM_STATE_SUSPENDED: + err = -ESTRPIPE; + goto _endloop; + case SNDRV_PCM_STATE_XRUN: + err = -EPIPE; + goto _endloop; + case SNDRV_PCM_STATE_DRAINING: + if (is_playback) + err = -EPIPE; + else + avail = 0; /* indicate draining */ + goto _endloop; + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_DISCONNECTED: + err = -EBADFD; + goto _endloop; + } + if (!tout) { + snd_printd("%s write error (DMA or IRQ trouble?)\n", + is_playback ? "playback" : "capture"); + err = -EIO; + break; + } + if (is_playback) + avail = snd_pcm_playback_avail(runtime); + else + avail = snd_pcm_capture_avail(runtime); + if (avail >= runtime->control->avail_min) + break; + } + _endloop: + remove_wait_queue(&runtime->sleep, &wait); + *availp = avail; + return err; +} + static int snd_pcm_lib_write_transfer(struct snd_pcm_substream *substream, unsigned int hwoff, unsigned long data, unsigned int off, @@ -1653,79 +1718,14 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (runtime->sleep_min == 0 && runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); avail = snd_pcm_playback_avail(runtime); - if (!avail || - (snd_pcm_running(substream) && - (avail < runtime->control->avail_min && size > avail))) { - wait_queue_t wait; - enum { READY, SIGNALED, ERROR, SUSPENDED, EXPIRED, DROPPED } state; - long tout; - + if (!avail) { if (nonblock) { err = -EAGAIN; goto _end_unlock; } - - init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->sleep, &wait); - while (1) { - if (signal_pending(current)) { - state = SIGNALED; - break; - } - set_current_state(TASK_INTERRUPTIBLE); - snd_pcm_stream_unlock_irq(substream); - tout = schedule_timeout(10 * HZ); - snd_pcm_stream_lock_irq(substream); - if (tout == 0) { - if (runtime->status->state != SNDRV_PCM_STATE_PREPARED && - runtime->status->state != SNDRV_PCM_STATE_PAUSED) { - state = runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED; - break; - } - } - switch (runtime->status->state) { - case SNDRV_PCM_STATE_XRUN: - case SNDRV_PCM_STATE_DRAINING: - state = ERROR; - goto _end_loop; - case SNDRV_PCM_STATE_SUSPENDED: - state = SUSPENDED; - goto _end_loop; - case SNDRV_PCM_STATE_SETUP: - state = DROPPED; - goto _end_loop; - default: - break; - } - avail = snd_pcm_playback_avail(runtime); - if (avail >= runtime->control->avail_min) { - state = READY; - break; - } - } - _end_loop: - remove_wait_queue(&runtime->sleep, &wait); - - switch (state) { - case ERROR: - err = -EPIPE; - goto _end_unlock; - case SUSPENDED: - err = -ESTRPIPE; - goto _end_unlock; - case SIGNALED: - err = -ERESTARTSYS; - goto _end_unlock; - case EXPIRED: - snd_printd("playback write error (DMA or IRQ trouble?)\n"); - err = -EIO; - goto _end_unlock; - case DROPPED: - err = -EBADFD; + err = wait_for_avail_min(substream, &avail); + if (err < 0) goto _end_unlock; - default: - break; - } } frames = size > avail ? avail : size; cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size; @@ -1925,86 +1925,22 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, snd_pcm_uframes_t cont; if (runtime->sleep_min == 0 && runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); - __draining: avail = snd_pcm_capture_avail(runtime); - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { - if (!avail) { - err = -EPIPE; + if (!avail) { + if (runtime->status->state == + SNDRV_PCM_STATE_DRAINING) { + snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); goto _end_unlock; } - } else if (avail < runtime->control->avail_min && - size > avail) { - wait_queue_t wait; - enum { READY, SIGNALED, ERROR, SUSPENDED, EXPIRED, DROPPED } state; - long tout; - if (nonblock) { err = -EAGAIN; goto _end_unlock; } - - init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->sleep, &wait); - while (1) { - if (signal_pending(current)) { - state = SIGNALED; - break; - } - set_current_state(TASK_INTERRUPTIBLE); - snd_pcm_stream_unlock_irq(substream); - tout = schedule_timeout(10 * HZ); - snd_pcm_stream_lock_irq(substream); - if (tout == 0) { - if (runtime->status->state != SNDRV_PCM_STATE_PREPARED && - runtime->status->state != SNDRV_PCM_STATE_PAUSED) { - state = runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED; - break; - } - } - switch (runtime->status->state) { - case SNDRV_PCM_STATE_XRUN: - state = ERROR; - goto _end_loop; - case SNDRV_PCM_STATE_SUSPENDED: - state = SUSPENDED; - goto _end_loop; - case SNDRV_PCM_STATE_DRAINING: - goto __draining; - case SNDRV_PCM_STATE_SETUP: - state = DROPPED; - goto _end_loop; - default: - break; - } - avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) { - state = READY; - break; - } - } - _end_loop: - remove_wait_queue(&runtime->sleep, &wait); - - switch (state) { - case ERROR: - err = -EPIPE; - goto _end_unlock; - case SUSPENDED: - err = -ESTRPIPE; - goto _end_unlock; - case SIGNALED: - err = -ERESTARTSYS; - goto _end_unlock; - case EXPIRED: - snd_printd("capture read error (DMA or IRQ trouble?)\n"); - err = -EIO; - goto _end_unlock; - case DROPPED: - err = -EBADFD; + err = wait_for_avail_min(substream, &avail); + if (err < 0) goto _end_unlock; - default: - break; - } + if (!avail) + continue; /* draining */ } frames = size > avail ? avail : size; cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size; -- cgit v0.10.2 From 31e8960b35975ed235d283d6fb95d0e28dffded0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Jan 2008 18:09:57 +0100 Subject: [ALSA] Remove PCM sleep_min and tick The 'tick' in PCM is set (again) via sw_params. And, nobody uses this feature at all except for a command line option of aplay. (This is literally 'nobody', as I checked alsa-lib API calls in all programs in major distros.) Above all, if we need finer wake-ups for the position update, it's basically an issue that the driver should solve, not tuned by each application. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1270cbc..51d58cc 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -274,7 +274,6 @@ struct snd_pcm_runtime { snd_pcm_uframes_t period_size; /* period size */ unsigned int periods; /* periods */ snd_pcm_uframes_t buffer_size; /* buffer size */ - unsigned int tick_time; /* tick time */ snd_pcm_uframes_t min_align; /* Min alignment for the format */ size_t byte_align; unsigned int frame_bits; @@ -286,7 +285,6 @@ struct snd_pcm_runtime { /* -- SW params -- */ int tstamp_mode; /* mmap timestamp is updated */ unsigned int period_step; - unsigned int sleep_min; /* min ticks to sleep */ snd_pcm_uframes_t start_threshold; snd_pcm_uframes_t stop_threshold; snd_pcm_uframes_t silence_threshold; /* Silence filling happens when @@ -305,7 +303,6 @@ struct snd_pcm_runtime { /* -- locking / scheduling -- */ wait_queue_head_t sleep; - struct timer_list tick_timer; struct fasync_struct *fasync; /* -- private section -- */ @@ -810,7 +807,6 @@ static inline const struct snd_interval *hw_param_interval_c(const struct snd_pc #define params_periods(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_PERIODS)->min #define params_buffer_size(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min #define params_buffer_bytes(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min -#define params_tick_time(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_TICK_TIME)->min int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v); @@ -908,9 +904,6 @@ int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream); int snd_pcm_playback_xrun_asap(struct snd_pcm_substream *substream); int snd_pcm_capture_xrun_asap(struct snd_pcm_substream *substream); void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr); -void snd_pcm_tick_prepare(struct snd_pcm_substream *substream); -void snd_pcm_tick_set(struct snd_pcm_substream *substream, unsigned long ticks); -void snd_pcm_tick_elapsed(struct snd_pcm_substream *substream); void snd_pcm_period_elapsed(struct snd_pcm_substream *substream); snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream, const void __user *buf, diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index be089cc..f7f15e3 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -985,7 +985,6 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) sw_params->stop_threshold = runtime->buffer_size; sw_params->tstamp_mode = SNDRV_PCM_TSTAMP_NONE; sw_params->period_step = 1; - sw_params->sleep_min = 0; sw_params->avail_min = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : runtime->period_size; if (atomic_read(&substream->mmap_count) || diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 97cb681..cf3af39 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -359,7 +359,6 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den); snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size); snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size); - snd_iprintf(buffer, "tick_time: %u\n", runtime->tick_time); #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) if (substream->oss.oss) { snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format)); @@ -387,7 +386,6 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, } snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode)); snd_iprintf(buffer, "period_step: %u\n", runtime->period_step); - snd_iprintf(buffer, "sleep_min: %u\n", runtime->sleep_min); snd_iprintf(buffer, "avail_min: %lu\n", runtime->control->avail_min); snd_iprintf(buffer, "start_threshold: %lu\n", runtime->start_threshold); snd_iprintf(buffer, "stop_threshold: %lu\n", runtime->stop_threshold); @@ -764,12 +762,6 @@ static int snd_pcm_dev_free(struct snd_device *device) return snd_pcm_free(pcm); } -static void snd_pcm_tick_timer_func(unsigned long data) -{ - struct snd_pcm_substream *substream = (struct snd_pcm_substream *) data; - snd_pcm_tick_elapsed(substream); -} - int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, struct file *file, struct snd_pcm_substream **rsubstream) @@ -876,9 +868,6 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, memset((void*)runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); - init_timer(&runtime->tick_timer); - runtime->tick_timer.function = snd_pcm_tick_timer_func; - runtime->tick_timer.data = (unsigned long) substream; runtime->status->state = SNDRV_PCM_STATE_OPEN; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b406630..f00758c 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1451,112 +1451,13 @@ int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_ioctl); -/* - * Conditions - */ - -static void snd_pcm_system_tick_set(struct snd_pcm_substream *substream, - unsigned long ticks) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - if (ticks == 0) - del_timer(&runtime->tick_timer); - else { - ticks += (1000000 / HZ) - 1; - ticks /= (1000000 / HZ); - mod_timer(&runtime->tick_timer, jiffies + ticks); - } -} - -/* Temporary alias */ -void snd_pcm_tick_set(struct snd_pcm_substream *substream, unsigned long ticks) -{ - snd_pcm_system_tick_set(substream, ticks); -} - -void snd_pcm_tick_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t frames = ULONG_MAX; - snd_pcm_uframes_t avail, dist; - unsigned int ticks; - u_int64_t n; - u_int32_t r; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (runtime->silence_size >= runtime->boundary) { - frames = 1; - } else if (runtime->silence_size > 0 && - runtime->silence_filled < runtime->buffer_size) { - snd_pcm_sframes_t noise_dist; - noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled; - if (noise_dist > (snd_pcm_sframes_t)runtime->silence_threshold) - frames = noise_dist - runtime->silence_threshold; - } - avail = snd_pcm_playback_avail(runtime); - } else { - avail = snd_pcm_capture_avail(runtime); - } - if (avail < runtime->control->avail_min) { - snd_pcm_sframes_t to_avail_min = - runtime->control->avail_min - avail; - if (to_avail_min > 0 && - frames > (snd_pcm_uframes_t)to_avail_min) - frames = to_avail_min; - } - if (avail < runtime->buffer_size) { - snd_pcm_sframes_t to_buffer_size = - runtime->buffer_size - avail; - if (to_buffer_size > 0 && - frames > (snd_pcm_uframes_t)to_buffer_size) - frames = to_buffer_size; - } - if (frames == ULONG_MAX) { - snd_pcm_tick_set(substream, 0); - return; - } - dist = runtime->status->hw_ptr - runtime->hw_ptr_base; - /* Distance to next interrupt */ - dist = runtime->period_size - dist % runtime->period_size; - if (dist <= frames) { - snd_pcm_tick_set(substream, 0); - return; - } - /* the base time is us */ - n = frames; - n *= 1000000; - div64_32(&n, runtime->tick_time * runtime->rate, &r); - ticks = n + (r > 0 ? 1 : 0); - if (ticks < runtime->sleep_min) - ticks = runtime->sleep_min; - snd_pcm_tick_set(substream, (unsigned long) ticks); -} - -void snd_pcm_tick_elapsed(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime; - unsigned long flags; - - snd_assert(substream != NULL, return); - runtime = substream->runtime; - snd_assert(runtime != NULL, return); - - snd_pcm_stream_lock_irqsave(substream, flags); - if (!snd_pcm_running(substream) || - snd_pcm_update_hw_ptr(substream) < 0) - goto _end; - if (runtime->sleep_min) - snd_pcm_tick_prepare(substream); - _end: - snd_pcm_stream_unlock_irqrestore(substream, flags); -} - /** * snd_pcm_period_elapsed - update the pcm status for the next period * @substream: the pcm substream instance * * This function is called from the interrupt handler when the * PCM has processed the period size. It will update the current - * pointer, set up the tick, wake up sleepers, etc. + * pointer, wake up sleepers, etc. * * Even if more than one periods have elapsed since the last call, you * have to call this only once. @@ -1580,8 +1481,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) if (substream->timer_running) snd_timer_interrupt(substream->timer, 1); - if (runtime->sleep_min) - snd_pcm_tick_prepare(substream); _end: snd_pcm_stream_unlock_irqrestore(substream, flags); if (runtime->transfer_ack_end) @@ -1715,7 +1614,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; snd_pcm_uframes_t cont; - if (runtime->sleep_min == 0 && runtime->status->state == SNDRV_PCM_STATE_RUNNING) + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); avail = snd_pcm_playback_avail(runtime); if (!avail) { @@ -1764,9 +1663,6 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (err < 0) goto _end_unlock; } - if (runtime->sleep_min && - runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_tick_prepare(substream); } _end_unlock: snd_pcm_stream_unlock_irq(substream); @@ -1923,7 +1819,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; snd_pcm_uframes_t cont; - if (runtime->sleep_min == 0 && runtime->status->state == SNDRV_PCM_STATE_RUNNING) + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); avail = snd_pcm_capture_avail(runtime); if (!avail) { @@ -1973,9 +1869,6 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, offset += frames; size -= frames; xfer += frames; - if (runtime->sleep_min && - runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_tick_prepare(substream); } _end_unlock: snd_pcm_stream_unlock_irq(substream); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3c22d78..d6b4e6b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -413,7 +413,6 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->period_size = params_period_size(params); runtime->periods = params_periods(params); runtime->buffer_size = params_buffer_size(params); - runtime->tick_time = params_tick_time(params); runtime->info = params->info; runtime->rate_num = params->rate_num; runtime->rate_den = params->rate_den; @@ -433,7 +432,6 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, /* Default sw params */ runtime->tstamp_mode = SNDRV_PCM_TSTAMP_NONE; runtime->period_step = 1; - runtime->sleep_min = 0; runtime->control->avail_min = runtime->period_size; runtime->start_threshold = 1; runtime->stop_threshold = runtime->buffer_size; @@ -542,7 +540,6 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, } snd_pcm_stream_lock_irq(substream); runtime->tstamp_mode = params->tstamp_mode; - runtime->sleep_min = params->sleep_min; runtime->period_step = params->period_step; runtime->control->avail_min = params->avail_min; runtime->start_threshold = params->start_threshold; @@ -551,10 +548,6 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, runtime->silence_size = params->silence_size; params->boundary = runtime->boundary; if (snd_pcm_running(substream)) { - if (runtime->sleep_min) - snd_pcm_tick_prepare(substream); - else - snd_pcm_tick_set(substream, 0); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); @@ -865,8 +858,6 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - if (runtime->sleep_min) - snd_pcm_tick_prepare(substream); if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART, &runtime->trigger_tstamp); @@ -920,7 +911,6 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, &runtime->trigger_tstamp); runtime->status->state = state; - snd_pcm_tick_set(substream, 0); } wake_up(&runtime->sleep); } @@ -1004,12 +994,9 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MPAUSE, &runtime->trigger_tstamp); - snd_pcm_tick_set(substream, 0); wake_up(&runtime->sleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; - if (runtime->sleep_min) - snd_pcm_tick_prepare(substream); if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MCONTINUE, @@ -1064,7 +1051,6 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) &runtime->trigger_tstamp); runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; - snd_pcm_tick_set(substream, 0); wake_up(&runtime->sleep); } @@ -1167,8 +1153,6 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, &runtime->trigger_tstamp); runtime->status->state = runtime->status->suspended_state; - if (runtime->sleep_min) - snd_pcm_tick_prepare(substream); } static struct action_ops snd_pcm_action_resume = { @@ -1997,8 +1981,6 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) } /* FIXME: this belong to lowlevel */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_TICK_TIME, - 1000000 / HZ, 1000000 / HZ); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); return 0; @@ -2238,9 +2220,6 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst if (appl_ptr < 0) appl_ptr += runtime->boundary; runtime->control->appl_ptr = appl_ptr; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING && - runtime->sleep_min) - snd_pcm_tick_prepare(substream); ret = frames; __end: snd_pcm_stream_unlock_irq(substream); @@ -2286,9 +2265,6 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr if (appl_ptr < 0) appl_ptr += runtime->boundary; runtime->control->appl_ptr = appl_ptr; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING && - runtime->sleep_min) - snd_pcm_tick_prepare(substream); ret = frames; __end: snd_pcm_stream_unlock_irq(substream); @@ -2335,9 +2311,6 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs if (appl_ptr >= (snd_pcm_sframes_t)runtime->boundary) appl_ptr -= runtime->boundary; runtime->control->appl_ptr = appl_ptr; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING && - runtime->sleep_min) - snd_pcm_tick_prepare(substream); ret = frames; __end: snd_pcm_stream_unlock_irq(substream); @@ -2384,9 +2357,6 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst if (appl_ptr >= (snd_pcm_sframes_t)runtime->boundary) appl_ptr -= runtime->boundary; runtime->control->appl_ptr = appl_ptr; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING && - runtime->sleep_min) - snd_pcm_tick_prepare(substream); ret = frames; __end: snd_pcm_stream_unlock_irq(substream); -- cgit v0.10.2 From 9004acc70e8c49c50c4c7b652f906f1e0ed5709d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Jan 2008 18:13:27 +0100 Subject: [ALSA] Remove sound/driver.h This header file exists only for some hacks to adapt alsa-driver tree. It's useless for building in the kernel. Let's move a few lines in it to sound/core.h and remove it. With this patch, sound/driver.h isn't removed but has just a single compile warning to include it. This should be really killed in future. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 835e320..b03df4d4 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -394,7 +394,6 @@ Basic Flow for PCI Drivers - Example #include #include #include @@ -751,7 +750,6 @@ #include #include #include diff --git a/drivers/input/touchscreen/ucb1400_ts.c b/drivers/input/touchscreen/ucb1400_ts.c index 7549939..986a836 100644 --- a/drivers/input/touchscreen/ucb1400_ts.c +++ b/drivers/input/touchscreen/ucb1400_ts.c @@ -27,7 +27,6 @@ #include #include -#include #include #include diff --git a/drivers/media/video/cx88/cx88-alsa.c b/drivers/media/video/cx88/cx88-alsa.c index 8735227..316b106 100644 --- a/drivers/media/video/cx88/cx88-alsa.c +++ b/drivers/media/video/cx88/cx88-alsa.c @@ -33,7 +33,6 @@ #include #include -#include #include #include #include diff --git a/drivers/media/video/saa7134/saa7134-alsa.c b/drivers/media/video/saa7134/saa7134-alsa.c index ba25310..047add8 100644 --- a/drivers/media/video/saa7134/saa7134-alsa.c +++ b/drivers/media/video/saa7134/saa7134-alsa.c @@ -21,7 +21,6 @@ #include #include #include -#include #include #include #include diff --git a/drivers/media/video/saa7134/saa7134.h b/drivers/media/video/saa7134/saa7134.h index ce45030..b88ca99 100644 --- a/drivers/media/video/saa7134/saa7134.h +++ b/drivers/media/video/saa7134/saa7134.h @@ -38,7 +38,6 @@ #include #include #include -#include #include #include #if defined(CONFIG_VIDEO_SAA7134_DVB) || defined(CONFIG_VIDEO_SAA7134_DVB_MODULE) diff --git a/drivers/usb/gadget/gmidi.c b/drivers/usb/gadget/gmidi.c index 0689189..7da7fcb 100644 --- a/drivers/usb/gadget/gmidi.c +++ b/drivers/usb/gadget/gmidi.c @@ -24,7 +24,6 @@ #include #include -#include #include #include #include diff --git a/include/asm-arm/arch-omap/eac.h b/include/asm-arm/arch-omap/eac.h index 6662cb0..ccee3b0 100644 --- a/include/asm-arm/arch-omap/eac.h +++ b/include/asm-arm/arch-omap/eac.h @@ -31,7 +31,6 @@ #include #include -#include #include /* master codec clock source */ diff --git a/include/asm-arm/arch-omap/omap-alsa.h b/include/asm-arm/arch-omap/omap-alsa.h index fcaf44c..faa0ed2 100644 --- a/include/asm-arm/arch-omap/omap-alsa.h +++ b/include/asm-arm/arch-omap/omap-alsa.h @@ -40,7 +40,6 @@ #ifndef __OMAP_ALSA_H #define __OMAP_ALSA_H -#include #include #include #include diff --git a/include/asm-arm/arch-pxa/audio.h b/include/asm-arm/arch-pxa/audio.h index 17eccd7..52bbe3b 100644 --- a/include/asm-arm/arch-pxa/audio.h +++ b/include/asm-arm/arch-pxa/audio.h @@ -1,7 +1,6 @@ #ifndef __ASM_ARCH_AUDIO_H__ #define __ASM_ARCH_AUDIO_H__ -#include #include #include diff --git a/include/sound/core.h b/include/sound/core.h index 6954836..4fc0235 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -22,12 +22,22 @@ * */ +#include #include /* wake_up() */ #include /* struct mutex */ #include /* struct rw_semaphore */ #include /* pm_message_t */ #include +/* number of supported soundcards */ +#ifdef CONFIG_SND_DYNAMIC_MINORS +#define SNDRV_CARDS 32 +#else +#define SNDRV_CARDS 8 /* don't change - minor numbers */ +#endif + +#define CONFIG_SND_MAJOR 116 /* standard configuration */ + /* forward declarations */ #ifdef CONFIG_PCI struct pci_dev; diff --git a/include/sound/driver.h b/include/sound/driver.h index 1889929..f035943 100644 --- a/include/sound/driver.h +++ b/include/sound/driver.h @@ -1,47 +1 @@ -#ifndef __SOUND_DRIVER_H -#define __SOUND_DRIVER_H - -/* - * Main header file for the ALSA driver - * Copyright (c) 1994-2000 by Jaroslav Kysela - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#ifdef ALSA_BUILD -#include "config.h" -#endif - - -/* number of supported soundcards */ -#ifdef CONFIG_SND_DYNAMIC_MINORS -#define SNDRV_CARDS 32 -#else -#define SNDRV_CARDS 8 /* don't change - minor numbers */ -#endif - -#ifndef CONFIG_SND_MAJOR /* standard configuration */ -#define CONFIG_SND_MAJOR 116 -#endif - -#ifdef ALSA_BUILD -#include "adriver.h" -#endif - -#include - -#endif /* __SOUND_DRIVER_H */ +#warning "This file is deprecated" diff --git a/include/sound/opl3.h b/include/sound/opl3.h index d7e33ce..a0c5feb 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -51,19 +51,16 @@ * */ -#include "driver.h" -#include -#include -#include "core.h" -#include "hwdep.h" -#include "timer.h" -#include "seq_midi_emul.h" +#include +#include +#include +#include #ifdef CONFIG_SND_SEQUENCER_OSS -#include "seq_oss.h" -#include "seq_oss_legacy.h" +#include +#include #endif -#include "seq_device.h" -#include "asound_fm.h" +#include +#include /* * Register numbers for the global registers diff --git a/include/sound/soc.h b/include/sound/soc.h index f47ef1f..c22c656 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -16,7 +16,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/aoa/aoa.h b/sound/aoa/aoa.h index 541b908..e087894 100644 --- a/sound/aoa/aoa.h +++ b/sound/aoa/aoa.h @@ -10,8 +10,6 @@ #define __AOA_H #include #include -/* So apparently there's a reason for requiring driver.h to be included first! */ -#include #include #include #include diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c index efb9441..e6beb92 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c @@ -11,7 +11,6 @@ #include #include -#include #include #include diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c index e6ffea9..59bacd3 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c @@ -8,9 +8,6 @@ #include #include -/* So apparently there's a reason for requiring driver.h - * to be included first, even if I don't know it... */ -#include #include #include #include diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 3b73ba7..b0a4744 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -23,7 +23,6 @@ #include #include -#include #include #include #include diff --git a/sound/arm/devdma.c b/sound/arm/devdma.c index ca3bf4e..9d1e666 100644 --- a/sound/arm/devdma.c +++ b/sound/arm/devdma.c @@ -12,7 +12,6 @@ #include #include -#include #include #include diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index d255503..5d86e68 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -18,7 +18,6 @@ #include #include -#include #include #include #include diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index e8cf904..0ede9e4 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -16,7 +16,6 @@ #include #include -#include #include #include #include diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index 81c64b0..0eff33c 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -59,7 +59,6 @@ * ***************************************************************************************************/ -#include #include #include #include diff --git a/sound/core/control.c b/sound/core/control.c index c89ca0d..01a1a5a 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/device.c b/sound/core/device.c index ea1a062..202dac0 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index bfd9d18..6d6589f 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/info.c b/sound/core/info.c index 1ffd29b..9977ec2 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index 9e8b816..e35789a 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/init.c b/sound/core/init.c index 48d38a7..dc06e79 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/isadma.c b/sound/core/isadma.c index eb173ce..79f0f16 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -26,7 +26,6 @@ #undef HAVE_REALLY_SLOW_DMA_CONTROLLER -#include #include #include diff --git a/sound/core/memory.c b/sound/core/memory.c index 25b0f05..1161158 100644 --- a/sound/core/memory.c +++ b/sound/core/memory.c @@ -20,9 +20,9 @@ * */ -#include #include #include +#include /** * copy_to_user_fromio - copy data from mmio-space to user-space diff --git a/sound/core/misc.c b/sound/core/misc.c index 6cabab8..102d1c3 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/oss/copy.c b/sound/core/oss/copy.c index d6a04c2..9ded30d 100644 --- a/sound/core/oss/copy.c +++ b/sound/core/oss/copy.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/oss/io.c b/sound/core/oss/io.c index 3ece39f..f874f6c 100644 --- a/sound/core/oss/io.c +++ b/sound/core/oss/io.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c index 06f96a3..da3dbd4 100644 --- a/sound/core/oss/linear.c +++ b/sound/core/oss/linear.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index c5a5ab9..75daed2 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 848db82..77f9619 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f7f15e3..092c2d8 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -26,7 +26,6 @@ #define OSS_DEBUG #endif -#include #include #include #include diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 14095a9..bec9413 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -24,7 +24,6 @@ #define PLUGIN_DEBUG #endif -#include #include #include #include diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 9eb2679..14dfb31 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index de3ffde..da7ab7a 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cf3af39..1502acd 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f00758c..ed0223c 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index a13e38c..ff07b4a 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index b9ae6b3..89b7f54 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d6b4e6b..e6e4aa8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index 23aa9a2..033a024 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b8e700b..f94694c 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c index 7cd5e8f..97b30fb 100644 --- a/sound/core/rtctimer.c +++ b/sound/core/rtctimer.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index bc09923..777796e 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index 9a8567c..bf8d2b4 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -21,7 +21,6 @@ #ifndef __SEQ_OSS_DEVICE_H #define __SEQ_OSS_DEVICE_H -#include #include #include #include diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 1878208..ee0f840 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 6444bd8..f97c1ba 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 37852cd..155dc7d 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -36,7 +36,6 @@ * */ -#include #include #include #include diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index e55488d..f3bdc54 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -18,7 +18,6 @@ * */ -#include #include #include #include diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 6b055ae..3a94ed0 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include "seq_fifo.h" diff --git a/sound/core/seq/seq_info.c b/sound/core/seq/seq_info.c index 8a7fe5c..201f810 100644 --- a/sound/core/seq/seq_info.c +++ b/sound/core/seq/seq_info.c @@ -19,7 +19,6 @@ * */ -#include #include #include diff --git a/sound/core/seq/seq_lock.c b/sound/core/seq/seq_lock.c index 1a34941..54f921e 100644 --- a/sound/core/seq/seq_lock.c +++ b/sound/core/seq/seq_lock.c @@ -19,7 +19,6 @@ * */ -#include #include #include "seq_lock.h" diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index a72a194..0cf6ac4 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 5929aaf..99b3536 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -26,7 +26,6 @@ Possible options for midisynth module: */ -#include #include #include #include diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 6645fc5..07c6631 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -29,7 +29,6 @@ * code in here. If there is it should be reported as a bug. */ -#include #include #include #include diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index b6820a5..8284f17 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -19,7 +19,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index b6e23ad..1c32a53 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include "seq_system.h" diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index 0744186..85969db 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 9b87bb0..4a48c6e 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -35,7 +35,6 @@ * - Addition of experimental sync support. */ -#include #include #include #include diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c index b201b76..77884e6 100644 --- a/sound/core/seq/seq_system.c +++ b/sound/core/seq/seq_system.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include "seq_system.h" diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 82a5b33..d8fcd62 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include "seq_timer.h" diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index e11790f..86e7739 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -35,7 +35,6 @@ * */ -#include #include #include #include diff --git a/sound/core/sound.c b/sound/core/sound.c index 7b486c4..00cca4d 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index dc73313..7be5154 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -19,8 +19,6 @@ * */ -#include - #ifdef CONFIG_SND_OSSEMUL #if !defined(CONFIG_SOUND) && !(defined(MODULE) && defined(CONFIG_SOUND_MODULE)) diff --git a/sound/core/timer.c b/sound/core/timer.c index 7e5fe2d..aece465 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index e008f3c..a240eae 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -18,7 +18,6 @@ * */ -#include #include #include #include diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 443104d..05a871a 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -33,7 +33,6 @@ * But there might still be some ... */ -#include #include #include diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 1fc95da..5b996f3 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index b57f2d5..5993864a 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -28,7 +28,6 @@ * */ -#include #include #include #include diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 40eb026..b5e1a71 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -50,7 +50,6 @@ * */ -#include #include #include #include diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index e12ba3d..f057d92 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -18,7 +18,6 @@ * */ -#include #include #include #include diff --git a/sound/drivers/pcm-indirect2.c b/sound/drivers/pcm-indirect2.c index 660157d..3c93c23 100644 --- a/sound/drivers/pcm-indirect2.c +++ b/sound/drivers/pcm-indirect2.c @@ -25,8 +25,6 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ -/* #dependency of sound/core.h# */ -#include /* snd_printk/d() */ #include /* struct snd_pcm_substream, struct snd_pcm_runtime, snd_pcm_uframes_t diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 7e7c668..b1c047e 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -37,7 +37,6 @@ * - ported from alsa 0.5 to 1.0 */ -#include #include #include #include diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index f415601..d8aab9d 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -30,7 +30,6 @@ * More documentation can be found in serial-u16550.txt. */ -#include #include #include #include diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 915c867..f79e361 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -41,7 +41,6 @@ * - Run application using a midi device (eg. /dev/snd/midiC1D0) */ -#include #include #include #include diff --git a/sound/drivers/vx/vx_cmd.c b/sound/drivers/vx/vx_cmd.c index 7a22134..9529e3b 100644 --- a/sound/drivers/vx/vx_cmd.c +++ b/sound/drivers/vx/vx_cmd.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index ed19bc1..9953886 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 9a8154c..1dfe694 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c index a37f0a8..5a34732 100644 --- a/sound/drivers/vx/vx_mixer.c +++ b/sound/drivers/vx/vx_mixer.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 7e65a10..fdbf865 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -45,7 +45,6 @@ * - scheduled action on the stream. */ -#include #include #include #include diff --git a/sound/drivers/vx/vx_uer.c b/sound/drivers/vx/vx_uer.c index 7400306..fb8932a 100644 --- a/sound/drivers/vx/vx_uer.c +++ b/sound/drivers/vx/vx_uer.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 744366b..e57e9cb 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index 1e58a96..b1e74e4 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c index b074fdd..bfa5d2c 100644 --- a/sound/i2c/l3/uda1341.c +++ b/sound/i2c/l3/uda1341.c @@ -19,7 +19,6 @@ /* $Id: uda1341.c,v 1.18 2005/11/17 14:17:21 tiwai Exp $ */ -#include #include #include #include diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index facde46..15061bd 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index ee1585a..f350835 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index fefa1ae..35fbbf2 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/i2c/other/pt2258.c b/sound/i2c/other/pt2258.c index 987d2c9..797d3a6 100644 --- a/sound/i2c/other/pt2258.c +++ b/sound/i2c/other/pt2258.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 37c47fb..28a4af7 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index 9bab744..0e3a9f2 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 994bc85..68f1260 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -18,7 +18,6 @@ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index cf18fe4..4b8dfe2 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -17,7 +17,6 @@ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index a4710b5..5f5271e 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 9a64035..630c90f 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -20,7 +20,6 @@ */ #define SNDRV_MAIN_OBJECT_FILE -#include #include #include #include diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index d687207..efa8c80 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -2,7 +2,6 @@ * AdLib FM card driver. */ -#include #include #include #include diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 35e25e5..f1ce30f 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -20,7 +20,6 @@ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index bfe3a25..154e728 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -29,7 +29,6 @@ activation method (full-duplex audio!). */ -#include #include #include #include diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index c166e13..4d198ec 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -43,7 +43,6 @@ * full control over both mixers. */ -#include #include #include #include diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index 13db684..e9462b9 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index a5eb965..1cd3fe3 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -24,7 +24,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 0a3fece..dbe63db4 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 6bd0644..de71910 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -79,7 +79,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c index ab689f9..a0242c3 100644 --- a/sound/isa/dt019x.c +++ b/sound/isa/dt019x.c @@ -21,7 +21,6 @@ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 74bbc92..f88639e 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 5c26d49..1e1e575 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 91cb478..90498e4 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -77,7 +77,6 @@ * needed for ZV, so maybe the datasheet is entirely wrong here. */ -#include #include #include #include diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index fc90514..f45f611 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_dram.c b/sound/isa/gus/gus_dram.c index 9eaa932..fd2e2e2 100644 --- a/sound/isa/gus/gus_dram.c +++ b/sound/isa/gus/gus_dram.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_instr.c b/sound/isa/gus/gus_instr.c index bf137ea..4dc9caf 100644 --- a/sound/isa/gus/gus_instr.c +++ b/sound/isa/gus/gus_instr.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c index 3d4f899..ca79878 100644 --- a/sound/isa/gus/gus_io.c +++ b/sound/isa/gus/gus_io.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c index cd9a6f1c..041894d 100644 --- a/sound/isa/gus/gus_irq.c +++ b/sound/isa/gus/gus_irq.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index e4453e5..cccc16c 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index bcf4656..661205c 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_mem_proc.c b/sound/isa/gus/gus_mem_proc.c index f69a447..2803e22 100644 --- a/sound/isa/gus/gus_mem_proc.c +++ b/sound/isa/gus/gus_mem_proc.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_mixer.c b/sound/isa/gus/gus_mixer.c index a96253e..ebdb334 100644 --- a/sound/isa/gus/gus_mixer.c +++ b/sound/isa/gus/gus_mixer.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index a7971f5..99731dc 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -25,7 +25,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c index 20cfdb8..3d1fed0c 100644 --- a/sound/isa/gus/gus_reset.c +++ b/sound/isa/gus/gus_reset.c @@ -18,7 +18,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_timer.c b/sound/isa/gus/gus_timer.c index 99eac57..c537271 100644 --- a/sound/isa/gus/gus_timer.c +++ b/sound/isa/gus/gus_timer.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c index e6fd9b0..f0af3f7 100644 --- a/sound/isa/gus/gus_uart.c +++ b/sound/isa/gus/gus_uart.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gus_volume.c b/sound/isa/gus/gus_volume.c index 71a6774..c3c028a 100644 --- a/sound/isa/gus/gus_volume.c +++ b/sound/isa/gus/gus_volume.c @@ -18,7 +18,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 29e422b..8f914b3 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index fc59536..da13185 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index 4922f5d..f87c623 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 9381d1e..ca0d7ac 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -22,7 +22,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 125f699..854a9f7 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index c2baf4c..b18d14f 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -22,7 +22,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 9300cf3..1f9c557 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -23,7 +23,6 @@ */ -#include #include #include #include diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 4eea84c..b35be7d 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/sb/emu8000_local.h b/sound/isa/sb/emu8000_local.h index 2ac77f1..7e87c34 100644 --- a/sound/isa/sb/emu8000_local.h +++ b/sound/isa/sb/emu8000_local.h @@ -21,7 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c index 3049692..c8c8e21 100644 --- a/sound/isa/sb/es968.c +++ b/sound/isa/sb/es968.c @@ -20,7 +20,6 @@ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 8e1aff7..2c201f7 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 5f21aec..bed29ca 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -23,7 +23,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index c06754f..f7e8192 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -33,7 +33,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index f933aef..336a342 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index bee894b..6304c3a 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -30,7 +30,6 @@ * Cleaned up and rewrote lowlevel routines. */ -#include #include #include #include diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c index e56e563..988a8b7 100644 --- a/sound/isa/sb/sb8_midi.c +++ b/sound/isa/sb/sb8_midi.c @@ -26,7 +26,6 @@ * Added full duplex UART mode for DSP version 2.0 and later. */ -#include #include #include #include diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 176193c..d63c1af 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 03241cd..91d1422 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index bc0c379..da3d152 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -23,7 +23,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c index 922519d..a07274e 100644 --- a/sound/isa/sgalaxy.c +++ b/sound/isa/sgalaxy.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1cb921d..06ad786 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -21,7 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 15c6046..3a6c6fe 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -19,7 +19,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index fc95a87..2efaa7f 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -16,7 +16,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/isa/wavefront/wavefront_midi.c b/sound/isa/wavefront/wavefront_midi.c index cb34600..a33384a 100644 --- a/sound/isa/wavefront/wavefront_midi.c +++ b/sound/isa/wavefront/wavefront_midi.c @@ -47,7 +47,6 @@ * */ -#include #include #include #include diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index a1ebb7c..95eeca1 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/last.c b/sound/last.c index 282b0cd..bdd0857 100644 --- a/sound/last.c +++ b/sound/last.c @@ -20,7 +20,6 @@ */ #define SNDRV_MAIN_OBJECT_FILE -#include #include #include diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 24460a5..ee0741f 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -36,7 +36,6 @@ #include #include -#include #include #include #include diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index ff705c6..99f5483 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -45,7 +45,6 @@ #include #include -#include #include #include #include diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6a9966d..45fd290 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -22,7 +22,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 8cbc033..3674f35 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -23,7 +23,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index fed4a2c..060ea59 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -22,7 +22,6 @@ * */ -#include #include #include diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c index 722de45..c0c16339 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ac97/ak4531_codec.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 98970d4..be9f1a2 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -40,7 +40,6 @@ #include #include -#include #include #include #include diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 4c2bd7a..6a905ed 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -25,7 +25,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 48cc39b..0e990a7 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -30,7 +30,6 @@ * to keep track of what period we are in. */ -#include #include #include #include diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 1190ef3..27ce613 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -63,7 +63,6 @@ * - power management? (card can do voice wakeup according to datasheet!!) */ -#include #include #include #include diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 89184a4..e9c87f5 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index ce752f8..a67a869 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 5ccf0b1..4aad35b 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -18,7 +18,6 @@ #define __SOUND_AU88X0_H #ifdef __KERNEL__ -#include #include #include #include diff --git a/sound/pci/au88x0/au88x0_game.c b/sound/pci/au88x0/au88x0_game.c index a07d1de..bc212f4 100644 --- a/sound/pci/au88x0/au88x0_game.c +++ b/sound/pci/au88x0/au88x0_game.c @@ -30,7 +30,6 @@ * driver. (email: mjander@embedded.cl). */ -#include #include #include #include diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c index c96da1d..c92f493 100644 --- a/sound/pci/au88x0/au88x0_mixer.c +++ b/sound/pci/au88x0/au88x0_mixer.c @@ -5,7 +5,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 8db3d3e..0dc8d25 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 7b5baa1..526c6c5 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -21,7 +21,6 @@ * It remains stuck,and DMA transfers do not happen. */ #include -#include #include #include #include diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 36d3666..4e71a55 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -115,7 +115,6 @@ * code (but I'm not too optimistic that doing this is possible at all) */ -#include #include #include #include diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 2dba752..c9a2421 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -21,7 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 31d8db9..176e0f0 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -135,7 +135,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ -#include #include #include #include diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 3f9b5c5..af73686 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -60,7 +60,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ -#include #include #include #include diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 61f2718..c62b7d1 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -60,7 +60,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ -#include #include #include #include diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c index ad32eff..893ee4f 100644 --- a/sound/pci/ca0106/ca_midi.c +++ b/sound/pci/ca0106/ca_midi.c @@ -27,7 +27,6 @@ */ #include -#include #include #include diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index bc0a195..135f308 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -20,7 +20,6 @@ /* Does not work. Warning may block system in capture mode */ /* #define USE_VAR48KRATE */ -#include #include #include #include diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 9a55f4a..7556fd9 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 2699cb6..e876b32 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -25,7 +25,6 @@ reloading the module may solve this. */ -#include #include #include #include diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 28f98bd..87ddffc 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -45,7 +45,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 590b35d..ccc8bed 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -20,7 +20,6 @@ */ -#include #include #include #include diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index eded4df..2873cfe 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -21,7 +21,6 @@ */ -#include #include #include #include diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 240a0a4..7ff8b68 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -36,7 +36,6 @@ * same manner. */ -#include #include #include #include diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 2b35889..a0c54f0 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 21df063..499e279 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c index 838708f..564c33b 100644 --- a/sound/pci/cs5535audio/cs5535audio_pm.c +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index 87078d3..8c6db3a 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -36,7 +36,6 @@ #define BX_NUM 10 -#include #include #include #include diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 42b48f9..04cbf3e 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -40,7 +40,6 @@ #define BX_NUM 10 -#include #include #include #include diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index 8dbb7ac..4022e43 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -47,7 +47,6 @@ #define BX_NUM chip->bx_num -#include #include #include #include diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index fee2d48..c0e64b8 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -40,7 +40,6 @@ #define BX_NUM 14 -#include #include #include #include diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index d5eae47..c36a78d 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -46,7 +46,6 @@ #define BX_NUM 26 -#include #include #include #include diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index 40f601c..0a58a7c 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -38,7 +38,6 @@ #define BX_NUM 2 -#include #include #include #include diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index 771c538..2db24d2 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -38,7 +38,6 @@ #define BX_NUM 4 -#include #include #include #include diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index 49c550d..a60c0a0 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -39,7 +39,6 @@ #define BX_NUM 4 -#include #include #include #include diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 8f5483a..5061946 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -45,7 +45,6 @@ #define BX_NUM 22 -#include #include #include #include diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index 0524667..e09e3ea 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -47,7 +47,6 @@ #define BX_NUM 32 -#include #include #include #include diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index 893c7c2..f3b9b45 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -45,7 +45,6 @@ #define BX_NUM 8 -#include #include #include #include diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index 3a5d5b0..b05bad9 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -44,7 +44,6 @@ #define BX_NUM 26 -#include #include #include #include diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 9680caf..8354c1a 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -23,7 +23,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 97c41d7..69f342c 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -33,7 +33,6 @@ #include #include -#include #include #include #include diff --git a/sound/pci/emu10k1/emu10k1_synth_local.h b/sound/pci/emu10k1/emu10k1_synth_local.h index 308ddc8..25f328f 100644 --- a/sound/pci/emu10k1/emu10k1_synth_local.h +++ b/sound/pci/emu10k1/emu10k1_synth_local.h @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 0030d8b8..5512abd 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -29,7 +29,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 9bf1cd5..c35d9e1 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -28,7 +28,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index ccacd7b..5a38fe7 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -30,7 +30,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c index 04c7cf7..c4d76d1 100644 --- a/sound/pci/emu10k1/emumpu401.c +++ b/sound/pci/emu10k1/emumpu401.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 5ce5bef..cf4e3ec 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -26,7 +26,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index c3fb10e..bd8a471 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -28,7 +28,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 6702c15..2862e17 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -25,7 +25,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index 3c114b4..276d08c 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -25,7 +25,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 48097c6..916c1db 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 9fd3135..749a21b 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -87,7 +87,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/timer.c b/sound/pci/emu10k1/timer.c index 6295b2d..72321e9 100644 --- a/sound/pci/emu10k1/timer.c +++ b/sound/pci/emu10k1/timer.c @@ -25,7 +25,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index 04fa849..958cb2a 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -28,7 +28,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index b958f86..72d85a5 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -26,7 +26,6 @@ * by Kurt J. Bosch */ -#include #include #include #include diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fb25abe..fbe3da7 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -47,7 +47,6 @@ */ -#include #include #include #include diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index d69b11d..25ccfce 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -94,7 +94,6 @@ * places. */ -#include #include #include #include diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 9939109..25c1087 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cd80719..a2b40dc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -19,7 +19,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index c957eb5..f9de7c4 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index bafb7b0..2177d9a 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a37e894..3ae4b4c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -34,7 +34,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7df1d16..4b8d644 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -21,7 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include "hda_codec.h" diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 67144dc..6664a06 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -19,7 +19,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index 78441e3..9a8bb4c 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -21,7 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 6c54793..02e31e4 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -21,7 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e960189..8b8de96 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b2ad52..5e36462 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -23,7 +23,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 2a4b960..f5e43e8 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -22,7 +22,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 299e02a..61114cd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -24,7 +24,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d42d8f7..423b499 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -33,7 +33,6 @@ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ -#include #include #include #include diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c index a1aba0d..dab31b2 100644 --- a/sound/pci/ice1712/ak4xxx.c +++ b/sound/pci/ice1712/ak4xxx.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 6e13d75..3756430 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index f83ec2f..3374891 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -47,7 +47,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 519b5d4..efd180b 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -22,7 +22,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 75e4e5e..6f65da4 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -22,7 +22,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index abcfd1d..b2cfba16 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 052fc3c..47d7737 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -47,7 +47,6 @@ */ -#include #include #include #include diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index b7a1d93..3147cbc 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -22,7 +22,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 1fbe3ef..1a435df 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index c81efc2..718e935 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -33,7 +33,6 @@ * CDTI may be completely blocked by 74HCT125's gate #1 controlled by GPIO 3 */ -#include #include #include #include diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index faefd52..4945c81 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 6d81a1c..7339378 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -54,7 +54,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index d18a31e..05a751c 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/se.c b/sound/pci/ice1712/se.c index 3c8b518..6661f65 100644 --- a/sound/pci/ice1712/se.c +++ b/sound/pci/ice1712/se.c @@ -22,7 +22,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/vt1720_mobo.c b/sound/pci/ice1712/vt1720_mobo.c index 2395241..7f9674b 100644 --- a/sound/pci/ice1712/vt1720_mobo.c +++ b/sound/pci/ice1712/vt1720_mobo.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 41a153d..a08d17c 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -25,7 +25,6 @@ -#include #include #include #include diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 312373c..061072c 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -26,7 +26,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index fad806e..cadda8d 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -23,7 +23,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 6586abf..10c713d 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 93dfedc..04fa0a6 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -31,7 +31,6 @@ #define CARD_NAME "ESS Maestro3/Allegro/Canyon3D-2" #define DRIVER_NAME "Maestro3" -#include #include #include #include diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 880b824..3dd0c79 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -21,7 +21,6 @@ */ -#include #include #include #include diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index d544573..785085e 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 170781a..122c28e 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 5b3c224..6fdda1f 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 276c576..7ac654e 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -24,7 +24,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index bfef5ab..ba7a2a8 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -28,7 +28,6 @@ * GPIO 1 -> DFS1 of AK5385 */ -#include #include #include #include diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 5f4feea..616087c 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -17,7 +17,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 6180cc8..2418ceb 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -17,7 +17,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index e252aba..7208b0f 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -17,7 +17,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index ff46ba5..d4a1d73 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -17,7 +17,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 5678dc3..6603a68 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -30,7 +30,6 @@ * GPIO 8 -> ? (amps enable?) */ -#include #include #include #include diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 2d618bd..9d5bb76 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -21,7 +21,6 @@ */ -#include #include #include #include diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 0ff8dc3..c4e415d 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index d55d8bc..e6a4bfb 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index 4d86545..aabc7bc 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -21,7 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 8e54104..9408b1e 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -88,7 +88,6 @@ Adopted for Windows NT driver 01/20/98 CNL */ -#include #include #include #include diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 1475912..df184aa 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -69,7 +69,6 @@ */ -#include #include #include #include diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index aff05bd..fb0a4ee 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -23,7 +23,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 67ec08c..4ba9e01 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6b1d529..9a19ae6 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -23,7 +23,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ -#include #include #include #include diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 1d73be6..a123f0e 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -20,7 +20,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 2f17859..dcd7cd0 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -21,7 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 44a7f5f..0d3d305 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -22,7 +22,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 6193c7e..d94b16f 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -21,7 +21,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index c8d5665..71138ff 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -27,7 +27,6 @@ * SiS7018 S/PDIF support by Thomas Winischhofer */ -#include #include #include #include diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index 847b8c6..df9b487 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -23,7 +23,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 18a58e4..a756be6 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -46,7 +46,6 @@ * - Optimize position calculation for the 823x chips. */ -#include #include #include #include diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 57fb9ae..f5df1c7 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -31,7 +31,6 @@ * modems. */ -#include #include #include #include diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index 474eac9..acc352f 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index f4f0427..b4bfc1a 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 5c4256a..2631a55 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index c0789a5..42c1eb7d 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -18,7 +18,6 @@ * */ -#include #include #include #include diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 5f5bbea..819aaaa 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index 484c8f9..dfa40b0 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c index 5454336..fa4b113 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include "pdaudiocf.h" #include diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 10afcb2..01066c9 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pcmcia/vx/vxp_mixer.c b/sound/pcmcia/vx/vxp_mixer.c index bf9d3b3..a4a6642 100644 --- a/sound/pcmcia/vx/vxp_mixer.c +++ b/sound/pcmcia/vx/vxp_mixer.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 1ee0918..157b0b5 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index c57e127..706602a 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -19,7 +19,6 @@ */ -#include #include #include #include diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index b15bfb6..8441e78 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -20,7 +20,6 @@ */ -#include #include #include #include diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c index 465dd04..baa2a72 100644 --- a/sound/ppc/beep.c +++ b/sound/ppc/beep.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index fec74e8..1a545ac 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -19,7 +19,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c index 0c81457..8432c16 100644 --- a/sound/ppc/daca.c +++ b/sound/ppc/daca.c @@ -19,7 +19,6 @@ */ -#include #include #include #include diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index bb7d744..6ff99ed 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -19,7 +19,6 @@ */ -#include #include #include #include diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 8c47beb..aada1d7 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -20,7 +20,6 @@ */ -#include #include #include #include diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 2264574..c936225 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index af812dc..d8d0b4b 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index d4d22e1..71a7a97 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -24,7 +24,6 @@ */ -#include #include #include #include diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 12c41df..d49417b 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -35,7 +35,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index b39b95a..67c88e3 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -23,7 +23,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index 3d4e32c..f642d2d 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -22,7 +22,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 820a676..ad3ad9d 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -28,7 +28,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 0b8a6f8..242130c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 968eda3..bf2ab72 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -28,7 +28,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c075a28..710e028 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -39,7 +39,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 57fb95a..9c33fe8 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index f8797de..77a857b 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index efced93..e6a9b91 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -41,7 +41,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 427cb61..9ef8e5c 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5ee51a9..3f34e53 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 0915cf7..5ae59bd 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 60e6f46..815c153 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,7 +17,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 50c5c83..692b900 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 35e8fa3..daeaa4c 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -16,7 +16,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 4dd8f35..d56709e 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 5504e30..e4d40b5 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -25,7 +25,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index ec0d1a2..9ed8f2e 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -18,7 +18,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index bde4cdc..6ee115c 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 96605b7..1c1ddbf 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -23,7 +23,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 9d8af3a..ff99b76 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 4107a87..290dcf1 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -24,7 +24,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index d46cd81..b4a5630 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -17,7 +17,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index cdee374..7a3ce80 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -16,7 +16,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 34b77b9..b7b676b 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -21,7 +21,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 5563f14..2f91de8 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -9,7 +9,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index b72bc31..3388bc3 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -30,7 +30,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e6a67b5..9407525 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -32,7 +32,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 29a546f..3b8f94a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -43,7 +43,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index b1d4315..0c63e05 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -36,7 +36,6 @@ #include #include -#include #include #include #include diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index f8c7a12..1c4797b 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -19,7 +19,6 @@ #include -#include #include #include #include diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index fc68317..3d00e07 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -53,7 +53,6 @@ * other DBRI low-level stuff */ -#include #include #include #include diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 5e8cf9f..89d6e9c 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -22,7 +22,6 @@ #include #include -#include #include #include #include diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index ebcac13..c89d2ea 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index 9b63814..0a53914 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index 3436816..f60a98e 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -22,7 +22,6 @@ * midi emulation. */ -#include #ifdef CONFIG_SND_SEQUENCER_OSS diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c index 680f2b7..687e6a1 100644 --- a/sound/synth/emux/emux_proc.c +++ b/sound/synth/emux/emux_proc.c @@ -18,7 +18,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/synth/emux/emux_voice.h b/sound/synth/emux/emux_voice.h index 0a56ca18..09711f8 100644 --- a/sound/synth/emux/emux_voice.h +++ b/sound/synth/emux/emux_voice.h @@ -22,7 +22,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 455e535..36d53bd 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -25,7 +25,6 @@ * of doing things so that the old sfxload utility can be used. * Everything may change when there is an alsa way of doing things. */ -#include #include #include #include diff --git a/sound/synth/util_mem.c b/sound/synth/util_mem.c index 6fc3d2b..deabe5f 100644 --- a/sound/synth/util_mem.c +++ b/sound/synth/util_mem.c @@ -19,7 +19,6 @@ */ #include -#include #include #include #include diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index bf551c0..f7e5284 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -16,7 +16,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index 7d25f4b..798ca12 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -17,7 +17,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 48c6bed..5cd92ae 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/usb/caiaq/caiaq-input.c b/sound/usb/caiaq/caiaq-input.c index e6c410a..f743847 100644 --- a/sound/usb/caiaq/caiaq-input.c +++ b/sound/usb/caiaq/caiaq-input.c @@ -23,7 +23,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/usb/caiaq/caiaq-midi.c b/sound/usb/caiaq/caiaq-midi.c index 793ca20..5427032 100644 --- a/sound/usb/caiaq/caiaq-midi.c +++ b/sound/usb/caiaq/caiaq-midi.c @@ -23,7 +23,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c6d628a..8fa9356 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -38,7 +38,6 @@ */ -#include #include #include #include diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6330788..750e929 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -35,7 +35,6 @@ * SUCH DAMAGE. */ -#include #include #include #include diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 1f1e91c..89c63d0 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -26,7 +26,6 @@ * */ -#include #include #include #include diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index a1dca34..6495534e 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index e011fca..e5981a6 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -130,7 +130,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 48e9aa3..9a608fa 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -31,7 +31,6 @@ */ -#include #include #include #include -- cgit v0.10.2 From a780c0aeb39e9251c1b48166380f4455871bc067 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 9 Jan 2008 12:30:20 +0100 Subject: [ALSA] hda: 92HD71BXX Mono Mute Support Added a mono output mute mixer for the 92hd71bxx family of codecs, this also removes the need for the mono out node to explicitly unmuted in the core init. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61114cd..c4a85d2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -533,8 +533,6 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* unmute mono out node */ - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; static struct hda_verb stac92hd71bxx_analog_core_init[] = { @@ -552,8 +550,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* unmute mono out node */ - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -713,6 +709,8 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; @@ -727,6 +725,8 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; -- cgit v0.10.2 From 17596a80d3b57763f6d111fa95416559bad9c8dc Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Marcin=20=C5=9Alusarz?= Date: Wed, 9 Jan 2008 17:56:07 +0100 Subject: [ALSA] rawmidi: let sparse know what is going on _for real_ MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_rawmidi_kernel_read1/write1 weren't annotated but used copy_to_user/copy_from_user when one of parameters (kernel) was equal to 0 remove it and add properly annotated parameter Signed-off-by: Marcin Ślusarz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index f94694c..f7ea728 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -911,7 +911,8 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, } static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, - unsigned char *buf, long count, int kernel) + unsigned char __user *userbuf, + unsigned char *kernelbuf, long count) { unsigned long flags; long result = 0, count1; @@ -924,11 +925,11 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, spin_lock_irqsave(&runtime->lock, flags); if (count1 > (int)runtime->avail) count1 = runtime->avail; - if (kernel) { - memcpy(buf + result, runtime->buffer + runtime->appl_ptr, count1); - } else { + if (kernelbuf) + memcpy(kernelbuf + result, runtime->buffer + runtime->appl_ptr, count1); + if (userbuf) { spin_unlock_irqrestore(&runtime->lock, flags); - if (copy_to_user((char __user *)buf + result, + if (copy_to_user(userbuf + result, runtime->buffer + runtime->appl_ptr, count1)) { return result > 0 ? result : -EFAULT; } @@ -948,7 +949,7 @@ long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream, unsigned char *buf, long count) { snd_rawmidi_input_trigger(substream, 1); - return snd_rawmidi_kernel_read1(substream, buf, count, 1); + return snd_rawmidi_kernel_read1(substream, NULL/*userbuf*/, buf, count); } static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t count, @@ -989,8 +990,9 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, - (unsigned char __force *)buf, - count, 0); + (unsigned char __user *)buf, + NULL/*kernelbuf*/, + count); if (count1 < 0) return result > 0 ? result : count1; result += count1; @@ -1131,13 +1133,15 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, } static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, - const unsigned char *buf, long count, int kernel) + const unsigned char __user *userbuf, + const unsigned char *kernelbuf, + long count) { unsigned long flags; long count1, result; struct snd_rawmidi_runtime *runtime = substream->runtime; - snd_assert(buf != NULL, return -EINVAL); + snd_assert(kernelbuf != NULL || userbuf != NULL, return -EINVAL); snd_assert(runtime->buffer != NULL, return -EINVAL); result = 0; @@ -1154,12 +1158,13 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, count1 = count; if (count1 > (long)runtime->avail) count1 = runtime->avail; - if (kernel) { - memcpy(runtime->buffer + runtime->appl_ptr, buf, count1); - } else { + if (kernelbuf) + memcpy(runtime->buffer + runtime->appl_ptr, + kernelbuf + result, count1); + else if (userbuf) { spin_unlock_irqrestore(&runtime->lock, flags); if (copy_from_user(runtime->buffer + runtime->appl_ptr, - (char __user *)buf, count1)) { + userbuf + result, count1)) { spin_lock_irqsave(&runtime->lock, flags); result = result > 0 ? result : -EFAULT; goto __end; @@ -1170,7 +1175,6 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, runtime->appl_ptr %= runtime->buffer_size; runtime->avail -= count1; result += count1; - buf += count1; count -= count1; } __end: @@ -1184,7 +1188,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, long snd_rawmidi_kernel_write(struct snd_rawmidi_substream *substream, const unsigned char *buf, long count) { - return snd_rawmidi_kernel_write1(substream, buf, count, 1); + return snd_rawmidi_kernel_write1(substream, NULL, buf, count); } static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, @@ -1224,9 +1228,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, spin_lock_irq(&runtime->lock); } spin_unlock_irq(&runtime->lock); - count1 = snd_rawmidi_kernel_write1(substream, - (unsigned char __force *)buf, - count, 0); + count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); if (count1 < 0) return result > 0 ? result : count1; result += count1; -- cgit v0.10.2 From 8c427226ed549af67396794e86246bf2d361ff8f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 10 Jan 2008 13:03:59 +0100 Subject: [ALSA] hda-codec - Update realtek codec support 1. Support HP rp5700 2. Fixed alc_subsystem_id function (Bug fixed and support Desktop) 3. Support ASUS EP20 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 22aee1a..959c80d 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -811,6 +811,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. hp-bpc HP xw4400/6400/8400/9400 laptops hp-bpc-d7000 HP BPC D7000 hp-tc-t5735 HP Thin Client T5735 + hp-rp5700 HP RP5700 benq Benq ED8 benq-t31 Benq T31 hippo Hippo (ATI) with jack detection, Sony UX-90s @@ -835,7 +836,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch-dig 3-stack (6-channel) with SPDIF 6stack-dig 6-stack with SPDIF lenovo-101e Lenovo laptop - eeepc-p701 ASUS Eeepc + eeepc-p701 ASUS Eeepc P701 + eeepc-ep20 ASUS Eeepc EP20 auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e36462..1778495 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -92,6 +92,7 @@ enum { ALC262_HP_BPC_D7000_WL, ALC262_HP_BPC_D7000_WF, ALC262_HP_TC_T5735, + ALC262_HP_RP5700, ALC262_BENQ_ED8, ALC262_SONY_ASSAMD, ALC262_BENQ_T31, @@ -155,6 +156,7 @@ enum { ALC662_5ST_DIG, ALC662_LENOVO_101E, ALC662_ASUS_EEEPC_P701, + ALC662_ASUS_EEEPC_EP20, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -804,7 +806,7 @@ static void alc_subsystem_id(struct hda_codec *codec, /* check sum */ tmp = 0; for (i = 1; i < 16; i++) { - if ((ass >> i) && 1) + if ((ass >> i) & 1) tmp++; } if (((ass >> 16) & 0xf) != tmp) @@ -893,10 +895,10 @@ do_sku: break; } - /* is laptop and enable the function "Mute internal speaker + /* is laptop or Desktop and enable the function "Mute internal speaker * when the external headphone out jack is plugged" */ - if (!(ass & 0x4) || !(ass & 0x8000)) + if (!(ass & 0x8000)) return; /* * 10~8 : Jack location @@ -906,9 +908,9 @@ do_sku: * when the external headphone out jack is plugged" */ if (!spec->autocfg.speaker_pins[0]) { - if (spec->multiout.dac_nids[0]) + if (spec->autocfg.line_out_pins[0]) spec->autocfg.speaker_pins[0] = - spec->multiout.dac_nids[0]; + spec->autocfg.line_out_pins[0]; else return; } @@ -7952,6 +7954,57 @@ static struct hda_verb alc262_hp_t5735_verbs[] = { { } }; +static struct hda_bind_ctls alc262_hp_rp5700_bind_front_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x0e, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc262_hp_rp5700_bind_front_sw = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = { + HDA_BIND_VOL("PCM Playback Volume", &alc262_hp_rp5700_bind_front_vol), + HDA_BIND_SW("PCM Playback Switch", &alc262_hp_rp5700_bind_front_sw), + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_hp_rp5700_verbs[] = { + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))}, + {} +}; + +static struct hda_input_mux alc262_hp_rp5700_capture_source = { + .num_items = 1, + .items = { + { "Line", 0x1 }, + }, +}; + /* bind hp and internal speaker mute (with plug check) */ static int alc262_sony_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -8794,6 +8847,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_HP_BPC] = "hp-bpc", [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", [ALC262_HP_TC_T5735] = "hp-tc-t5735", + [ALC262_HP_RP5700] = "hp-rp5700", [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", [ALC262_SONY_ASSAMD] = "sony-assamd", @@ -8824,6 +8878,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735", ALC262_HP_TC_T5735), + SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), @@ -8929,6 +8984,15 @@ static struct alc_config_preset alc262_presets[] = { .input_mux = &alc262_capture_source, .unsol_event = alc262_hp_t5735_unsol_event, .init_hook = alc262_hp_t5735_init_hook, + }, + [ALC262_HP_RP5700] = { + .mixers = { alc262_hp_rp5700_mixer }, + .init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_hp_rp5700_capture_source, }, [ALC262_BENQ_ED8] = { .mixers = { alc262_base_mixer }, @@ -12520,6 +12584,24 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { + HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("LineOut Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc662_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -12605,6 +12687,13 @@ static struct hda_verb alc662_eeepc_sue_init_verbs[] = { {} }; +/* Set Unsolicited Event*/ +static struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = { + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -12741,6 +12830,40 @@ static void alc662_eeepc_inithook(struct hda_codec *codec) alc662_eeepc_mic_automute(codec); } +static void alc662_eeepc_ep20_automute(struct hda_codec *codec) +{ + unsigned int mute; + unsigned int present; + + snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { + /* mute internal speaker */ + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +/* unsolicited event for HP jack sensing */ +static void alc662_eeepc_ep20_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc662_eeepc_ep20_automute(codec); +} + +static void alc662_eeepc_ep20_inithook(struct hda_codec *codec) +{ + alc662_eeepc_ep20_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc662_loopbacks alc880_loopbacks #endif @@ -12762,11 +12885,13 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_5ST_DIG] = "6stack-dig", [ALC662_LENOVO_101E] = "lenovo-101e", [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", + [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20", [ALC662_AUTO] = "auto", }; static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), + SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), {} }; @@ -12854,6 +12979,21 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc662_eeepc_unsol_event, .init_hook = alc662_eeepc_inithook, }, + [ALC662_ASUS_EEEPC_EP20] = { + .mixers = { alc662_eeepc_ep20_mixer, alc662_capture_mixer, + alc662_chmode_mixer }, + .init_verbs = { alc662_init_verbs, + alc662_eeepc_ep20_sue_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), + .adc_nids = alc662_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), + .channel_mode = alc662_3ST_6ch_modes, + .input_mux = &alc662_lenovo_101e_capture_source, + .unsol_event = alc662_eeepc_ep20_unsol_event, + .init_hook = alc662_eeepc_ep20_inithook, + }, }; @@ -13013,6 +13153,7 @@ static void alc662_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; + alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); -- cgit v0.10.2 From 3d5fa2e59630a852f2dda7096885b64ffa03427d Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Thu, 10 Jan 2008 13:05:47 +0100 Subject: [ALSA] hda-codec - Device ID for Macbook sound card Please refer to the [0003680] on ALSA bugtracking system. The user found that 'model=mbp3' works and provided the ID. From: Jiang zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1778495..9184586 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6124,6 +6124,7 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; + case 0x106b00a1: /* Macbook */ case 0x106b2c00: /* Macbook Pro rev3 */ board_config = ALC885_MBP3; break; -- cgit v0.10.2 From 2f32d909f3a85a96074801f6cf20f93324f87646 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 10 Jan 2008 13:06:26 +0100 Subject: [ALSA] hda: STAC9228 VT fixes Moved 2 systems PCI_QUIRK values to STAC_DELL_BIOS. Also the second front HP jack is incorrect defined in the BIOS VT's for some laptops, this patch corrects this. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c4a85d2..a0af868 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1428,13 +1428,13 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), /* Dell 3 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_3ST), /* Dell 3 stack systems with verb table in BIOS */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), @@ -3332,6 +3332,8 @@ static int patch_stac927x(struct hda_codec *codec) spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + /* correct the front output jack as a hp out */ + stac92xx_set_config_reg(codec, 0x0f, 0x02270110); /* correct the front input jack as a mic */ stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); /* fallthru */ -- cgit v0.10.2 From 2cad0ad6781a78a2b3911440b58e8de28318798c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Jan 2008 14:33:07 +0100 Subject: [ALSA] Update MAINTAINERS for ALSA SoC Add myself as a point of contact for the ALSA SoC subsystem and add a reference to the development GIT tree. Signed-off-by: Mark Brown Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/MAINTAINERS b/MAINTAINERS index 2d5ff3e..093cf04 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3571,6 +3571,9 @@ S: Maintained SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT P: Liam Girdwood M: liam.girdwood@wolfsonmicro.com +P: Mark Brown +M: broonie@opensource.wolfsonmicro.com +T: git opensource.wolfsonmicro.com/linux-2.6-asoc L: alsa-devel@alsa-project.org (subscribers-only) S: Supported -- cgit v0.10.2 From 7570f29a545c7fdf29e913860320e0c09e60e45b Mon Sep 17 00:00:00 2001 From: Joe Sauer Date: Thu, 10 Jan 2008 14:34:56 +0100 Subject: [ALSA] Fix inverted Phone volume WM9712 mixer control Signed-off-by: Joe Sauer Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9ef8e5c..590baea 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -131,7 +131,7 @@ SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1), SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1), SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1), -SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 0), +SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1), SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1), SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0), -- cgit v0.10.2 From 0b4d221b8d56deefca4984d01b3a010107ae1f72 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 10 Jan 2008 14:36:20 +0100 Subject: [ALSA] soc - Add device level DAPM event Added a device level dapm event so that both the machine and codec are informed when dapm events occur. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index b9d5864..4158cba 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -199,6 +199,7 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev); /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/include/sound/soc.h b/include/sound/soc.h index c22c656..add5f94 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -425,6 +425,9 @@ struct snd_soc_machine { int (*resume_pre)(struct platform_device *pdev); int (*resume_post)(struct platform_device *pdev); + /* callbacks */ + int (*dapm_event)(struct snd_soc_machine *, int event); + /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9407525..bd656db 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -288,15 +288,16 @@ static void close_delayed_work(struct work_struct *work) if (codec_dai->pop_wait == 1) { codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, + snd_soc_dapm_stream_event(codec, + codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_STOP); /* power down the codec power domain if no longer active */ if (codec->active == 0) { dbg("pop wq D3 %s %s\n", codec->name, codec_dai->playback.stream_name); - if (codec->dapm_event) - codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_device_event(socdev, + SNDRV_CTL_POWER_D3hot); } } } @@ -352,12 +353,12 @@ static int soc_codec_close(struct snd_pcm_substream *substream) } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(codec, - codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_STOP); + codec_dai->capture.stream_name, + SND_SOC_DAPM_STREAM_STOP); - if (codec->active == 0 && codec_dai->pop_wait == 0){ - if (codec->dapm_event) - codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot); - } + if (codec->active == 0 && codec_dai->pop_wait == 0) + snd_soc_dapm_device_event(socdev, + SNDRV_CTL_POWER_D3hot); } mutex_unlock(&pcm_mutex); @@ -432,8 +433,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) /* no delayed work - do we need to power up codec */ if (codec->dapm_state != SNDRV_CTL_POWER_D0) { - if (codec->dapm_event) - codec->dapm_event(codec, SNDRV_CTL_POWER_D1); + snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dapm_stream_event(codec, @@ -444,8 +444,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - if (codec->dapm_event) - codec->dapm_event(codec, SNDRV_CTL_POWER_D0); + snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D0); if (codec_dai->dai_ops.digital_mute) codec_dai->dai_ops.digital_mute(codec_dai, 0); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3b8f94a..16ebb60 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1279,6 +1279,29 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** + * snd_soc_dapm_device_event - send a device event to the dapm core + * @socdev: audio device + * @event: device event + * + * Sends a device event to the dapm core. The core then makes any + * necessary machine or codec power changes.. + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event) +{ + struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_machine *machine = socdev->machine; + + if (machine->dapm_event) + machine->dapm_event(machine, event); + if (codec->dapm_event) + codec->dapm_event(codec, event); + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event); + +/** * snd_soc_dapm_set_endpoint - set audio endpoint status * @codec: audio codec * @endpoint: audio signal endpoint (or start point) -- cgit v0.10.2 From a7a4ac86b4754f44eb06221f3087debb4775d588 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 10 Jan 2008 14:37:42 +0100 Subject: [ALSA] ASoC TLV support Add TLV support to ASoC. Signed-off-by: Philipp Zabel Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 4158cba..491f81e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -131,18 +131,34 @@ .shift = wshift, .invert = winvert} /* dapm kcontrol types */ -#define SOC_DAPM_SINGLE(xname, reg, shift, mask, invert) \ +#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } -#define SOC_DAPM_DOUBLE(xname, reg, shift_left, shift_right, mask, invert, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_DAPM_DOUBLE(xname, reg, shift_left, shift_right, max, invert, \ power) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\ - ((mask) << 16) | ((invert) << 24) } + ((max) << 16) | ((invert) << 24) } +#define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_DAPM_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, \ + power, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\ + ((max) << 16) | ((invert) << 24) } #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/include/sound/soc.h b/include/sound/soc.h index add5f94..0afcdfe 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -26,27 +26,53 @@ /* * Convenience kcontrol builders */ -#define SOC_SINGLE_VALUE(reg,shift,mask,invert) ((reg) | ((shift) << 8) |\ - ((shift) << 12) | ((mask) << 16) | ((invert) << 24)) -#define SOC_SINGLE_VALUE_EXT(reg,mask,invert) ((reg) | ((mask) << 16) |\ +#define SOC_SINGLE_VALUE(reg, shift, max, invert) ((reg) | ((shift) << 8) |\ + ((shift) << 12) | ((max) << 16) | ((invert) << 24)) +#define SOC_SINGLE_VALUE_EXT(reg, max, invert) ((reg) | ((max) << 16) |\ ((invert) << 31)) -#define SOC_SINGLE(xname, reg, shift, mask, invert) \ +#define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } -#define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ + .put = snd_soc_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = (reg) | ((shift_left) << 8) | \ - ((shift_right) << 12) | ((mask) << 16) | ((invert) << 24) } -#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) \ + ((shift_right) << 12) | ((max) << 16) | ((invert) << 24) } +#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (reg_left) | ((shift) << 8) | \ - ((mask) << 12) | ((invert) << 20) | ((reg_right) << 24) } + ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } +#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ + .put = snd_soc_put_volsw, \ + .private_value = (reg) | ((shift_left) << 8) | \ + ((shift_right) << 12) | ((max) << 16) | ((invert) << 24) } +#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ + .private_value = (reg_left) | ((shift) << 8) | \ + ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -104,9 +130,21 @@ #define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated when not Tx/Rx */ /* + * DAI Sync + * Synchronous LR (Left Right) clocks and Frame signals. + */ +#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */ +#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */ + +/* + * TDM + */ +#define SND_SOC_DAIFMT_TDM (1 << 6) + +/* * DAI hardware signal inversions */ -#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ +#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */ #define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ #define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ #define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bd656db..e748b00 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1213,7 +1213,6 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, memcpy(&template, _template, sizeof(template)); if (long_name) template.name = long_name; - template.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; template.index = 0; return snd_ctl_new1(&template, data); @@ -1348,13 +1347,16 @@ EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int mask = kcontrol->private_value; + int max = kcontrol->private_value; + + if (max == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->type = - mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; + uinfo->value.integer.max = max; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); @@ -1371,15 +1373,18 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int mask = (kcontrol->private_value >> 16) & 0xff; + int max = (kcontrol->private_value >> 16) & 0xff; int shift = (kcontrol->private_value >> 8) & 0x0f; int rshift = (kcontrol->private_value >> 12) & 0x0f; - uinfo->type = - mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + if (max == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = shift == rshift ? 1 : 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; + uinfo->value.integer.max = max; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); @@ -1400,7 +1405,8 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0x0f; int rshift = (kcontrol->private_value >> 12) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; + int max = (kcontrol->private_value >> 16) & 0xff; + int mask = (1 << fls(max)) - 1; int invert = (kcontrol->private_value >> 24) & 0x01; ucontrol->value.integer.value[0] = @@ -1410,10 +1416,10 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, (snd_soc_read(codec, reg) >> rshift) & mask; if (invert) { ucontrol->value.integer.value[0] = - mask - ucontrol->value.integer.value[0]; + max - ucontrol->value.integer.value[0]; if (shift != rshift) ucontrol->value.integer.value[1] = - mask - ucontrol->value.integer.value[1]; + max - ucontrol->value.integer.value[1]; } return 0; @@ -1436,25 +1442,24 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0x0f; int rshift = (kcontrol->private_value >> 12) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; + int max = (kcontrol->private_value >> 16) & 0xff; + int mask = (1 << fls(max)) - 1; int invert = (kcontrol->private_value >> 24) & 0x01; - int err; unsigned short val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) - val = mask - val; + val = max - val; val_mask = mask << shift; val = val << shift; if (shift != rshift) { val2 = (ucontrol->value.integer.value[1] & mask); if (invert) - val2 = mask - val2; + val2 = max - val2; val_mask |= mask << rshift; val |= val2 << rshift; } - err = snd_soc_update_bits(codec, reg, val_mask, val); - return err; + return snd_soc_update_bits(codec, reg, val_mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw); @@ -1471,13 +1476,16 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw); int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int mask = (kcontrol->private_value >> 12) & 0xff; + int max = (kcontrol->private_value >> 12) & 0xff; + + if (max == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->type = - mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; + uinfo->value.integer.max = max; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); @@ -1498,7 +1506,8 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, int reg = kcontrol->private_value & 0xff; int reg2 = (kcontrol->private_value >> 24) & 0xff; int shift = (kcontrol->private_value >> 8) & 0x0f; - int mask = (kcontrol->private_value >> 12) & 0xff; + int max = (kcontrol->private_value >> 12) & 0xff; + int mask = (1<private_value >> 20) & 0x01; ucontrol->value.integer.value[0] = @@ -1507,9 +1516,9 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, (snd_soc_read(codec, reg2) >> shift) & mask; if (invert) { ucontrol->value.integer.value[0] = - mask - ucontrol->value.integer.value[0]; + max - ucontrol->value.integer.value[0]; ucontrol->value.integer.value[1] = - mask - ucontrol->value.integer.value[1]; + max - ucontrol->value.integer.value[1]; } return 0; @@ -1532,7 +1541,8 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, int reg = kcontrol->private_value & 0xff; int reg2 = (kcontrol->private_value >> 24) & 0xff; int shift = (kcontrol->private_value >> 8) & 0x0f; - int mask = (kcontrol->private_value >> 12) & 0xff; + int max = (kcontrol->private_value >> 12) & 0xff; + int mask = (1 << fls(max)) - 1; int invert = (kcontrol->private_value >> 20) & 0x01; int err; unsigned short val, val2, val_mask; @@ -1542,8 +1552,8 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, val2 = (ucontrol->value.integer.value[1] & mask); if (invert) { - val = mask - val; - val2 = mask - val2; + val = max - val; + val2 = max - val2; } val = val << shift; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 16ebb60..7eb6c5c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1018,8 +1018,9 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0x0f; int rshift = (kcontrol->private_value >> 12) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; + int max = (kcontrol->private_value >> 16) & 0xff; int invert = (kcontrol->private_value >> 24) & 0x01; + int mask = (1 << fls(max)) - 1; /* return the saved value if we are powered down */ if (widget->id == snd_soc_dapm_pga && !widget->power) { @@ -1034,10 +1035,10 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, (snd_soc_read(widget->codec, reg) >> rshift) & mask; if (invert) { ucontrol->value.integer.value[0] = - mask - ucontrol->value.integer.value[0]; + max - ucontrol->value.integer.value[0]; if (shift != rshift) ucontrol->value.integer.value[1] = - mask - ucontrol->value.integer.value[1]; + max - ucontrol->value.integer.value[1]; } return 0; @@ -1060,7 +1061,8 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0x0f; int rshift = (kcontrol->private_value >> 12) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; + int max = (kcontrol->private_value >> 16) & 0xff; + int mask = (1 << fls(max)) - 1; int invert = (kcontrol->private_value >> 24) & 0x01; unsigned short val, val2, val_mask; int ret; @@ -1068,13 +1070,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, val = (ucontrol->value.integer.value[0] & mask); if (invert) - val = mask - val; + val = max - val; val_mask = mask << shift; val = val << shift; if (shift != rshift) { val2 = (ucontrol->value.integer.value[1] & mask); if (invert) - val2 = mask - val2; + val2 = max - val2; val_mask |= mask << rshift; val |= val2 << rshift; } -- cgit v0.10.2 From 3c1c47e0eb92264d555003ee2003f836e432615b Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 10 Jan 2008 14:38:24 +0100 Subject: [ALSA] soc - Add D1 power event to power down event sequence Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e748b00..7d51be8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -287,6 +287,14 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { + /* power down the codec to D1 if no longer active */ + if (codec->active == 0) { + dbg("pop wq D1 %s %s\n", codec->name, + codec_dai->playback.stream_name); + snd_soc_dapm_device_event(socdev, + SNDRV_CTL_POWER_D1); + } + codec_dai->pop_wait = 0; snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, -- cgit v0.10.2 From 4ccab3e72e211c40d868045c7d3216948999bcf7 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 10 Jan 2008 14:39:01 +0100 Subject: [ALSA] soc - Ensure PCMs are suspended This fixes a bug whereby PCMs were not being suspended when the rest of the audio subsystem was suspended. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/soc.h b/include/sound/soc.h index 0afcdfe..6b51ff2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -447,6 +447,9 @@ struct snd_soc_dai_link { /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_codec *codec); + + /* DAI pcm */ + struct snd_pcm *pcm; }; /* SoC machine */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7d51be8..5f86e03 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -645,6 +645,10 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) dai->dai_ops.digital_mute(dai, 1); } + /* suspend all pcms */ + for (i = 0; i < machine->num_links; i++) + snd_pcm_suspend_all(machine->dai_link[i].pcm); + if (machine->suspend_pre) machine->suspend_pre(pdev, state); @@ -879,6 +883,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, return ret; } + dai_link->pcm = pcm; pcm->private_data = rtd; soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap; soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer; -- cgit v0.10.2 From 1b075e3f7323f85f7e9cc7b6ae7a36c33d9ce76e Mon Sep 17 00:00:00 2001 From: Milan plzik Date: Thu, 10 Jan 2008 14:39:46 +0100 Subject: [ALSA] soc - Fix power switching support for DAPM_SWITCH widgets Signed-off-by: Milan plzik Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7eb6c5c..91d58b5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -691,7 +691,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; } -/* test and update the power status of a mixer widget */ +/* test and update the power status of a mixer or switch widget */ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int reg, int val_mask, int val, int invert) @@ -699,7 +699,8 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_soc_dapm_path *path; int found = 0; - if (widget->id != snd_soc_dapm_mixer) + if (widget->id != snd_soc_dapm_mixer && + widget->id != snd_soc_dapm_switch) return -ENODEV; if (!snd_soc_test_bits(widget->codec, reg, val_mask, val)) -- cgit v0.10.2 From 6c59e6c4129cd80834837b58a42216f0432a4f58 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 10 Jan 2008 14:40:16 +0100 Subject: [ALSA] soc - Clean up tabs Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 91d58b5..d033e73 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1297,9 +1297,9 @@ int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event) struct snd_soc_machine *machine = socdev->machine; if (machine->dapm_event) - machine->dapm_event(machine, event); + machine->dapm_event(machine, event); if (codec->dapm_event) - codec->dapm_event(codec, event); + codec->dapm_event(codec, event); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event); -- cgit v0.10.2 From 9af6d9562414568ecadf96aaef5b88e7e8b19821 Mon Sep 17 00:00:00 2001 From: Laim Girdwood Date: Thu, 10 Jan 2008 14:41:02 +0100 Subject: [ALSA] soc - Add support for passing kcontrols with events Signed-off-by: Laim Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 491f81e..a105b01 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -289,7 +289,7 @@ struct snd_soc_dapm_widget { /* external events */ unsigned short event_flags; /* flags to specify event types */ - int (*event)(struct snd_soc_dapm_widget*, int); + int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int); /* kcontrols that relate to this widget */ int num_kcontrols; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d033e73..8b61db2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -523,11 +523,13 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, SND_SOC_DAPM_PRE_PMU); + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMU); if (ret < 0) return ret; } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, SND_SOC_DAPM_PRE_PMD); + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMD); if (ret < 0) return ret; } @@ -538,11 +540,13 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, SND_SOC_DAPM_POST_PMU); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); if (ret < 0) return ret; } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, SND_SOC_DAPM_POST_PMD); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMD); if (ret < 0) return ret; } @@ -566,26 +570,30 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) if (power) { /* power up event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, SND_SOC_DAPM_PRE_PMU); + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMU); if (ret < 0) return ret; } dapm_update_bits(w); if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, SND_SOC_DAPM_POST_PMU); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); if (ret < 0) return ret; } } else { /* power down event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, SND_SOC_DAPM_PRE_PMD); + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMD); if (ret < 0) return ret; } dapm_update_bits(w); if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, SND_SOC_DAPM_POST_PMD); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMD); if (ret < 0) return ret; } @@ -1095,13 +1103,17 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, SND_SOC_DAPM_PRE_REG); - if (ret < 0) + ret = widget->event(widget, kcontrol, + SND_SOC_DAPM_PRE_REG); + if (ret < 0) { + ret = 1; goto out; + } } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, SND_SOC_DAPM_POST_REG); + ret = widget->event(widget, kcontrol, + SND_SOC_DAPM_POST_REG); } else ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); @@ -1176,13 +1188,15 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, dapm_mux_update_power(widget, kcontrol, mask, mux, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, SND_SOC_DAPM_PRE_REG); + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); if (ret < 0) goto out; } ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, SND_SOC_DAPM_POST_REG); + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); } else ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); -- cgit v0.10.2 From 4b67d35504d005e94328558ee2c8f661a46aaaf4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Jan 2008 14:41:46 +0100 Subject: [ALSA] soc - Don't lock the codec list in snd_soc_dapm_new_widgets() snd_soc_dapm_new_widgets() takes the codec lock when adding new widgets, causing lockdep warnings when applications later call down through ALSA to adjust controls. Since widgets are only added during probe this lock should be unneeded so don't take it. Thanks to Dmitry Baryshkov for reporting this issue. Cc: Dmitry Baryshkov Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8b61db2..620d7ea 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -971,7 +971,6 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) { struct snd_soc_dapm_widget *w; - mutex_lock(&codec->mutex); list_for_each_entry(w, &codec->dapm_widgets, list) { if (w->new) @@ -1006,7 +1005,6 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) } dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); - mutex_unlock(&codec->mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); -- cgit v0.10.2 From 0a22b87de0467bbc2ce863d54b55c3a7b649ba74 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Jan 2008 14:53:48 +0100 Subject: [ALSA] Bump ASoC core version number Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/soc.h b/include/sound/soc.h index 6b51ff2..e6ea6f7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -21,7 +21,7 @@ #include #include -#define SND_SOC_VERSION "0.13.1" +#define SND_SOC_VERSION "0.13.2" /* * Convenience kcontrol builders -- cgit v0.10.2 From 2d6a4ac9eeb8e8eb0343a2c761b2c132957d2b71 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 10 Jan 2008 14:43:48 +0100 Subject: [ALSA] soc - Initial WM8753 TLV support for capture mixer Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e6a9b91..ddd9c71 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -47,6 +47,7 @@ #include #include #include +#include #include #include "wm8753.h" @@ -257,6 +258,8 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, return 1; } +static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); + static const struct snd_kcontrol_new wm8753_snd_controls[] = { SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0), @@ -286,8 +289,8 @@ SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1), SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1), SOC_ENUM("Treble Cut-off", wm8753_enum[2]), -SOC_DOUBLE("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1), -SOC_SINGLE("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1), +SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, rec_mix_tlv), +SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, rec_mix_tlv), SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0), SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), -- cgit v0.10.2 From 96d90e19307ce590097295026a2e1b36618ddd01 Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Thu, 10 Jan 2008 14:44:24 +0100 Subject: [ALSA] soc - Reinitialise DMA on every resume This one changes the DMA initialisation as it turns out the DMA driver in s3c24xx doesnt store registers between suspend/resume so you have to re-initialise the channels on every resume. Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 290dcf1..29a6c82f 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -48,7 +48,9 @@ static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | @@ -175,28 +177,6 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, } } - /* channel needs configuring for mem=>device, increment memory addr, - * sync to pclk, half-word transfers to the IIS-FIFO. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC | - S3C2410_DISRCC_APB, prtd->params->dma_addr); - - s3c2410_dma_config(prtd->params->channel, - prtd->params->dma_size, - S3C2410_DCON_SYNC_PCLK | - S3C2410_DCON_HANDSHAKE); - } else { - s3c2410_dma_config(prtd->params->channel, - prtd->params->dma_size, - S3C2410_DCON_HANDSHAKE | - S3C2410_DCON_SYNC_PCLK); - - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_HW, 0x3, - prtd->params->dma_addr); - } - s3c2410_dma_set_buffdone_fn(prtd->params->channel, s3c24xx_audio_buffdone); @@ -245,6 +225,28 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) if (!prtd->params) return 0; + /* channel needs configuring for mem=>device, increment memory addr, + * sync to pclk, half-word transfers to the IIS-FIFO. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC | + S3C2410_DISRCC_APB, prtd->params->dma_addr); + + s3c2410_dma_config(prtd->params->channel, + prtd->params->dma_size, + S3C2410_DCON_SYNC_PCLK | + S3C2410_DCON_HANDSHAKE); + } else { + s3c2410_dma_config(prtd->params->channel, + prtd->params->dma_size, + S3C2410_DCON_HANDSHAKE | + S3C2410_DCON_SYNC_PCLK); + + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_HW, 0x3, + prtd->params->dma_addr); + } + /* flush the DMA channel */ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); prtd->dma_loaded = 0; -- cgit v0.10.2 From 5cd919a2144f1beee7fb2e18ef6b8bee7105f554 Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Thu, 10 Jan 2008 14:44:58 +0100 Subject: [ALSA] soc - Support suspend and resume of the I2S interface on s3c24xx Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ff99b76..0a3c630 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -76,6 +76,10 @@ static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = { struct s3c24xx_i2s_info { void __iomem *regs; struct clk *iis_clk; + u32 iiscon; + u32 iismod; + u32 iisfcon; + u32 iispsr; }; static struct s3c24xx_i2s_info s3c24xx_i2s; @@ -406,6 +410,38 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +int s3c24xx_i2s_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); + + clk_disable(s3c24xx_i2s.iis_clk); + + return 0; +} + +int s3c24xx_i2s_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + clk_enable(s3c24xx_i2s.iis_clk); + + writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); + + return 0; +} +#else +#define s3c24xx_i2s_suspend NULL +#define s3c24xx_i2s_resume NULL +#endif + + #define S3C24XX_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -416,6 +452,8 @@ struct snd_soc_cpu_dai s3c24xx_i2s_dai = { .id = 0, .type = SND_SOC_DAI_I2S, .probe = s3c24xx_i2s_probe, + .suspend = s3c24xx_i2s_suspend, + .resume = s3c24xx_i2s_resume, .playback = { .channels_min = 2, .channels_max = 2, -- cgit v0.10.2 From 49646dfa2ae81e770da1c12c4fce227062ce4612 Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Thu, 10 Jan 2008 14:47:21 +0100 Subject: [ALSA] ASoC: S3C2412 IIS driver S3C2412 SoC IIS support for ALSA/ASoC Signed-off-by: Ben Dooks Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 14def2e..1f6dbfc 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -10,6 +10,9 @@ config SND_S3C24XX_SOC config SND_S3C24XX_SOC_I2S tristate +config SND_S3C2412_SOC_I2S + tristate + config SND_S3C2443_SOC_AC97 tristate select AC97_BUS diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 9471919..0aa5fb0 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -1,11 +1,13 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := s3c24xx-pcm.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o +snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o +obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o # S3C24XX Machine Support snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c new file mode 100644 index 0000000..b10f7ff --- /dev/null +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -0,0 +1,681 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory@wolfsonmicro.com + * linux@wolfsonmicro.com + * + * Copyright (c) 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include +#include + +#include + +#include +#include +#include + +#include "s3c24xx-pcm.h" +#include "s3c2412-i2s.h" + +#define S3C2412_I2S_DEBUG 0 +#define S3C2412_I2S_DEBUG_CON 0 + +#if S3C2412_I2S_DEBUG +#define DBG(x...) printk(KERN_INFO x) +#else +#define DBG(x...) do { } while (0) +#endif + +static struct s3c2410_dma_client s3c2412_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c2412_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = { + .client = &s3c2412_dma_client_out, + .channel = DMACH_I2S_OUT, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .dma_size = 4, +}; + +static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { + .client = &s3c2412_dma_client_in, + .channel = DMACH_I2S_IN, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .dma_size = 4, +}; + +struct s3c2412_i2s_info { + struct device *dev; + void __iomem *regs; + struct clk *iis_clk; + struct clk *iis_pclk; + struct clk *iis_cclk; +}; + +static struct s3c2412_i2s_info s3c2412_i2s; + +#define bit_set(v, b) (((v) & (b)) ? 1 : 0) + +#if S3C2412_I2S_DEBUG_CON +static void dbg_showcon(const char *fn, u32 con) +{ + printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, + bit_set(con, S3C2412_IISCON_LRINDEX), + bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_TXFIFO_FULL), + bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); + + printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", + fn, + bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_TXCH_PAUSE), + bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); + printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, + bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); +} +#else +static inline void dbg_showcon(const char *fn, u32 con) +{ +} +#endif + +/* Turn on or off the transmission path. */ +static void s3c2412_snd_txctrl(int on) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + DBG("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_TXDMA_PAUSE; + con &= ~S3C2412_IISCON_TXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXONLY: + case S3C2412_IISMOD_MODE_TXRX: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_RXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } else { + /* Note, we do not have any indication that the FIFO problems + * tha the S3C2410/2440 had apply here, so we should be able + * to disable the DMA and TX without resetting the FIFOS. + */ + + con |= S3C2412_IISCON_TXDMA_PAUSE; + con |= S3C2412_IISCON_TXCH_PAUSE; + con &= ~S3C2412_IISCON_TXDMA_ACTIVE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_RXONLY; + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + con &= ~S3C2412_IISCON_IIS_ACTIVE; + break; + + default: + dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } + + fic = readl(regs + S3C2412_IISFIC); + dbg_showcon(__func__, con); + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + +static void s3c2412_snd_rxctrl(int on) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + DBG("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_RXDMA_PAUSE; + con &= ~S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + case S3C2412_IISMOD_MODE_RXONLY: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } else { + /* See txctrl notes on FIFOs. */ + + con &= ~S3C2412_IISCON_RXDMA_ACTIVE; + con |= S3C2412_IISCON_RXDMA_PAUSE; + con |= S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_RXONLY: + con &= ~S3C2412_IISCON_IIS_ACTIVE; + mod &= ~S3C2412_IISMOD_MODE_MASK; + break; + + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXONLY; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } + + fic = readl(regs + S3C2412_IISFIC); + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c2412_snd_lrsync(void) +{ + u32 iiscon; + unsigned long timeout = jiffies + msecs_to_jiffies(5); + + DBG("Entered %s\n", __func__); + + while (1) { + iiscon = readl(s3c2412_i2s.regs + S3C2412_IISCON); + if (iiscon & S3C2412_IISCON_LRINDEX) + break; + + if (timeout < jiffies) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; + } + } + + return 0; +} + +/* + * Check whether CPU is the master or slave + */ +static inline int s3c2412_snd_is_clkmaster(void) +{ + u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + + DBG("Entered %s\n", __func__); + + iismod &= S3C2412_IISMOD_MASTER_MASK; + return !(iismod == S3C2412_IISMOD_SLAVE); +} + +/* + * Set S3C2412 I2S DAI format + */ +static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + u32 iismod; + + + DBG("Entered %s\n", __func__); + + iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + DBG("hw_params r: IISMOD: %x \n", iismod); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iismod &= ~S3C2412_IISMOD_MASTER_MASK; + iismod |= S3C2412_IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iismod &= ~S3C2412_IISMOD_MASTER_MASK; + iismod |= S3C2412_IISMOD_MASTER_INTERNAL; + break; + default: + DBG("unknwon master/slave format\n"); + return -EINVAL; + } + + iismod &= ~S3C2412_IISMOD_SDF_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_SDF_MSB; + break; + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_SDF_LSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod |= S3C2412_IISMOD_SDF_IIS; + break; + default: + DBG("Unknown data format\n"); + return -EINVAL; + } + + writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); + DBG("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + u32 iismod; + + DBG("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_out; + else + rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_in; + + /* Working copies of register */ + iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + DBG("%s: r: IISMOD: %x\n", __func__, iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C2412_IISMOD_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod &= ~S3C2412_IISMOD_8BIT; + break; + } + + writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); + DBG("%s: w: IISMOD: %x\n", __func__, iismod); + return 0; +} + +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + unsigned long irqs; + int ret = 0; + + DBG("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* On start, ensure that the FIFOs are cleared and reset. */ + + writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, + s3c2412_i2s.regs + S3C2412_IISFIC); + + /* clear again, just in case */ + writel(0x0, s3c2412_i2s.regs + S3C2412_IISFIC); + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!s3c2412_snd_is_clkmaster()) { + ret = s3c2412_snd_lrsync(); + if (ret) + goto exit_err; + } + + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(1); + else + s3c2412_snd_txctrl(1); + + local_irq_restore(irqs); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(0); + else + s3c2412_snd_txctrl(0); + + local_irq_restore(irqs); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* default table of all avaialable root fs divisors */ +static unsigned int s3c2412_iis_fs[] = { 256, 512, 384, 768, 0 }; + +int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) +{ + unsigned long clkrate = clk_get_rate(clk); + unsigned int div; + unsigned int fsclk; + unsigned int actual; + unsigned int fs; + unsigned int fsdiv; + signed int deviation = 0; + unsigned int best_fs = 0; + unsigned int best_div = 0; + unsigned int best_rate = 0; + unsigned int best_deviation = INT_MAX; + + + if (fstab == NULL) + fstab = s3c2412_iis_fs; + + for (fs = 0;; fs++) { + fsdiv = s3c2412_iis_fs[fs]; + + if (fsdiv == 0) + break; + + fsclk = clkrate / fsdiv; + div = fsclk / rate; + + if ((fsclk % rate) > (rate / 2)) + div++; + + if (div <= 1) + continue; + + actual = clkrate / (fsdiv * div); + deviation = actual - rate; + + printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", + fsdiv, div, actual, deviation); + + deviation = abs(deviation); + + if (deviation < best_deviation) { + best_fs = fsdiv; + best_div = div; + best_rate = actual; + best_deviation = deviation; + } + + if (deviation == 0) + break; + } + + printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", + best_fs, best_div, best_rate); + + info->fs_div = best_fs; + info->clk_div = best_div; + + return 0; +} +EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); + +/* + * Set S3C2412 Clock source + */ +static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + + DBG("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id, + freq, dir); + + switch (clk_id) { + case S3C2412_CLKSRC_PCLK: + iismod &= ~S3C2412_IISMOD_MASTER_MASK; + iismod |= S3C2412_IISMOD_MASTER_INTERNAL; + break; + case S3C2412_CLKSRC_I2SCLK: + iismod &= ~S3C2412_IISMOD_MASTER_MASK; + iismod |= S3C2412_IISMOD_MASTER_EXTERNAL; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); + return 0; +} + +/* + * Set S3C2412 Clock dividers + */ +static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + u32 reg; + + DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); + + switch (div_id) { + case S3C2412_DIV_BCLK: + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_BCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + + DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C2412_DIV_RCLK: + if (div > 3) { + /* convert value to bit field */ + + switch (div) { + case 256: + div = S3C2412_IISMOD_RCLK_256FS; + break; + + case 384: + div = S3C2412_IISMOD_RCLK_384FS; + break; + + case 512: + div = S3C2412_IISMOD_RCLK_512FS; + break; + + case 768: + div = S3C2412_IISMOD_RCLK_768FS; + break; + + default: + return -EINVAL; + } + } + + reg = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_RCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C2412_DIV_PRESCALER: + if (div >= 0) { + writel((div << 8) | S3C2412_IISPSR_PSREN, + i2s->regs + S3C2412_IISPSR); + } else { + writel(0x0, i2s->regs + S3C2412_IISPSR); + } + DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +struct clk *s3c2412_get_iisclk(void) +{ + return s3c2412_i2s.iis_clk; +} +EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); + + +static int s3c2412_i2s_probe(struct platform_device *pdev) +{ + DBG("Entered %s\n", __func__); + + s3c2412_i2s.dev = &pdev->dev; + + s3c2412_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); + if (s3c2412_i2s.regs == NULL) + return -ENXIO; + + s3c2412_i2s.iis_pclk = clk_get(&pdev->dev, "iis"); + if (s3c2412_i2s.iis_pclk == NULL) { + DBG("failed to get iis_clock\n"); + iounmap(s3c2412_i2s.regs); + return -ENODEV; + } + + s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); + if (s3c2412_i2s.iis_cclk == NULL) { + DBG("failed to get i2sclk clock\n"); + iounmap(s3c2412_i2s.regs); + return -ENODEV; + } + + clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); + + clk_enable(s3c2412_i2s.iis_pclk); + clk_enable(s3c2412_i2s.iis_cclk); + + s3c2412_i2s.iis_clk = s3c2412_i2s.iis_pclk; + + /* Configure the I2S pins in correct mode */ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + + s3c2412_snd_txctrl(0); + s3c2412_snd_rxctrl(0); + + return 0; +} + +#define S3C2412_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +struct snd_soc_cpu_dai s3c2412_i2s_dai = { + .name = "s3c2412-i2s", + .id = 0, + .type = SND_SOC_DAI_I2S, + .probe = s3c2412_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .trigger = s3c2412_i2s_trigger, + .hw_params = s3c2412_i2s_hw_params, + }, + .dai_ops = { + .set_fmt = s3c2412_i2s_set_fmt, + .set_clkdiv = s3c2412_i2s_set_clkdiv, + .set_sysclk = s3c2412_i2s_set_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, "); +MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h new file mode 100644 index 0000000..27f48e1 --- /dev/null +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -0,0 +1,38 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. +*/ + +#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H +#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ + +#define S3C2412_DIV_BCLK (1) +#define S3C2412_DIV_RCLK (2) +#define S3C2412_DIV_PRESCALER (3) + +#define S3C2412_CLKSRC_PCLK (0) +#define S3C2412_CLKSRC_I2SCLK (1) + +extern struct clk *s3c2412_get_iisclk(void); + +extern struct snd_soc_cpu_dai s3c2412_i2s_dai; + +struct s3c2412_rate_calc { + unsigned int clk_div; /* for prescaler */ + unsigned int fs_div; /* for root frame clock */ +}; + +extern int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk); + +#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ -- cgit v0.10.2 From 6cab2d3ddd54f2ef4872a4fca3d44655377737dd Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Thu, 10 Jan 2008 14:48:37 +0100 Subject: [ALSA] S3C2412: suspend and resume support Support for suspend/resume for the S3C2412 ASoC IIS core driver. Signed-off-by: Ben Dooks Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index b10f7ff..c4a46dd 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -79,6 +79,10 @@ struct s3c2412_i2s_info { struct clk *iis_clk; struct clk *iis_pclk; struct clk *iis_cclk; + + u32 suspend_iismod; + u32 suspend_iiscon; + u32 suspend_iispsr; }; static struct s3c2412_i2s_info s3c2412_i2s; @@ -641,6 +645,63 @@ static int s3c2412_i2s_probe(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct platform_device *dev, + struct snd_soc_cpu_dai *dai) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + u32 iismod; + + if (dai->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + dev_warn(&dev->dev, "%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + + dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n", + dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (dai->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif /* CONFIG_PM */ + #define S3C2412_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -651,6 +712,8 @@ struct snd_soc_cpu_dai s3c2412_i2s_dai = { .id = 0, .type = SND_SOC_DAI_I2S, .probe = s3c2412_i2s_probe, + .suspend = s3c2412_i2s_suspend, + .resume = s3c2412_i2s_resume, .playback = { .channels_min = 2, .channels_max = 2, -- cgit v0.10.2 From ab40d4f12cda366ed1f308d2a041480769f9a77e Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Thu, 10 Jan 2008 14:50:34 +0100 Subject: [ALSA] soc - Preliminary ac97 drivers for Toshiba e800 PDAs Currently only the AUX channel is used (touchscreen) Signed-off-by: Ian Molton Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index a83e229..484f883 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -53,3 +53,12 @@ config SND_PXA2XX_SOC_TOSA help Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). + +config SND_PXA2XX_SOC_E800 + tristate "SoC AC97 Audio support for e800" + depends on SND_PXA2XX_SOC && MACH_E800 + select SND_SOC_WM9712 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + Toshiba e800 PDA diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 78e0d6b..04e5646 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -11,10 +11,12 @@ obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o +snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o +obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c new file mode 100644 index 0000000..06e8afb --- /dev/null +++ b/sound/soc/pxa/e800_wm9712.c @@ -0,0 +1,89 @@ +/* + * e800-wm9712.c -- SoC audio for e800 + * + * Based on tosa.c + * + * Copyright 2007 (c) Ian Molton + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 ONLY. + * + */ + +#include +#include +#include + +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "../codecs/wm9712.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static struct snd_soc_machine e800; + +static struct snd_soc_dai_link e800_dai[] = { +{ + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], +}, +}; + +static struct snd_soc_machine e800 = { + .name = "Toshiba e800", + .dai_link = e800_dai, + .num_links = ARRAY_SIZE(e800_dai), +}; + +static struct snd_soc_device e800_snd_devdata = { + .machine = &e800, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *e800_snd_device; + +static int __init e800_init(void) +{ + int ret; + + if (!machine_is_e800()) + return -ENODEV; + + e800_snd_device = platform_device_alloc("soc-audio", -1); + if (!e800_snd_device) + return -ENOMEM; + + platform_set_drvdata(e800_snd_device, &e800_snd_devdata); + e800_snd_devdata.dev = &e800_snd_device->dev; + ret = platform_device_add(e800_snd_device); + + if (ret) + platform_device_put(e800_snd_device); + + return ret; +} + +static void __exit e800_exit(void) +{ + platform_device_unregister(e800_snd_device); +} + +module_init(e800_init); +module_exit(e800_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton "); +MODULE_DESCRIPTION("ALSA SoC driver for e800"); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From 3b0a5f22d4649433a5842ffc7313803292e95718 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Jan 2008 16:52:42 +0100 Subject: [ALSA] Add virtual master control helpers Added helper functions to implement virtual master volume controls. The virtual master control is a control element that has multiple slave controls. The value of master element is equally added to slave elements. The functions are written for general purpose, but it's put in the HD-audio directory as now, since HD-audio driver is the only user. It should be moved to the common place once after other drivers use vmaster. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index ab0c726..9e0d8a1 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -2,7 +2,7 @@ snd-hda-intel-y := hda_intel.o # since snd-hda-intel is the only driver using hda-codec, # merge it into a single module although it was originally # designed to be individual modules -snd-hda-intel-y += hda_codec.o +snd-hda-intel-y += hda_codec.o vmaster.o snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 8c56c9c..e09f41b 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -398,4 +398,11 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, hda_nid_t nid); #endif /* CONFIG_SND_HDA_POWER_SAVE */ +/* + * virtual master control + */ +struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, + const unsigned int *tlv); +int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave); + #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/vmaster.c b/sound/pci/hda/vmaster.c new file mode 100644 index 0000000..2da49d2 --- /dev/null +++ b/sound/pci/hda/vmaster.c @@ -0,0 +1,364 @@ +/* + * Virtual master and slave controls + * + * Copyright (c) 2008 by Takashi Iwai + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation, version 2. + * + */ + +#include +#include +#include + +/* + * a subset of information returned via ctl info callback + */ +struct link_ctl_info { + int type; /* value type */ + int count; /* item count */ + int min_val, max_val; /* min, max values */ +}; + +/* + * link master - this contains a list of slave controls that are + * identical types, i.e. info returns the same value type and value + * ranges, but may have different number of counts. + * + * The master control is so far only mono volume/switch for simplicity. + * The same value will be applied to all slaves. + */ +struct link_master { + struct list_head slaves; + struct link_ctl_info info; + int val; /* the master value */ +}; + +/* + * link slave - this contains a slave control element + * + * It fakes the control callbacsk with additional attenuation by the + * master control. A slave may have either one or two channels. + */ + +struct link_slave { + struct list_head list; + struct link_master *master; + struct link_ctl_info info; + int vals[2]; /* current values */ + struct snd_kcontrol slave; /* the copy of original control entry */ +}; + +/* get the slave ctl info and save the initial values */ +static int slave_init(struct link_slave *slave) +{ + struct snd_ctl_elem_info *uinfo; + struct snd_ctl_elem_value *uctl; + int err, ch; + + if (slave->info.count) + return 0; /* already initialized */ + + uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL); + if (!uinfo) + return -ENOMEM; + uinfo->id = slave->slave.id; + err = slave->slave.info(&slave->slave, uinfo); + if (err < 0) { + kfree(uinfo); + return err; + } + slave->info.type = uinfo->type; + slave->info.count = uinfo->count; + if (slave->info.count > 2 || + (slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER && + slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) { + snd_printk(KERN_ERR "invalid slave element\n"); + kfree(uinfo); + return -EINVAL; + } + slave->info.min_val = uinfo->value.integer.min; + slave->info.max_val = uinfo->value.integer.max; + kfree(uinfo); + + uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); + if (!uctl) + return -ENOMEM; + uctl->id = slave->slave.id; + err = slave->slave.get(&slave->slave, uctl); + for (ch = 0; ch < slave->info.count; ch++) + slave->vals[ch] = uctl->value.integer.value[ch]; + kfree(uctl); + return 0; +} + +/* initialize master volume */ +static int master_init(struct link_master *master) +{ + struct link_slave *slave; + + if (master->info.count) + return 0; /* already initialized */ + + list_for_each_entry(slave, &master->slaves, list) { + int err = slave_init(slave); + if (err < 0) + return err; + master->info = slave->info; + master->info.count = 1; /* always mono */ + /* set full volume as default (= no attenuation) */ + master->val = master->info.max_val; + return 0; + } + return -ENOENT; +} + +static int slave_get_val(struct link_slave *slave, + struct snd_ctl_elem_value *ucontrol) +{ + int err, ch; + + err = slave_init(slave); + if (err < 0) + return err; + for (ch = 0; ch < slave->info.count; ch++) + ucontrol->value.integer.value[ch] = slave->vals[ch]; + return 0; +} + +static int slave_put_val(struct link_slave *slave, + struct snd_ctl_elem_value *ucontrol) +{ + int err, ch, vol; + + err = master_init(slave->master); + if (err < 0) + return err; + + switch (slave->info.type) { + case SNDRV_CTL_ELEM_TYPE_BOOLEAN: + for (ch = 0; ch < slave->info.count; ch++) + ucontrol->value.integer.value[ch] &= + !!slave->master->val; + break; + case SNDRV_CTL_ELEM_TYPE_INTEGER: + for (ch = 0; ch < slave->info.count; ch++) { + /* max master volume is supposed to be 0 dB */ + vol = ucontrol->value.integer.value[ch]; + vol += slave->master->val - slave->master->info.max_val; + if (vol < slave->info.min_val) + vol = slave->info.min_val; + else if (vol > slave->info.max_val) + vol = slave->info.max_val; + ucontrol->value.integer.value[ch] = vol; + } + break; + } + return slave->slave.put(&slave->slave, ucontrol); +} + +/* + * ctl callbacks for slaves + */ +static int slave_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + return slave->slave.info(&slave->slave, uinfo); +} + +static int slave_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + return slave_get_val(slave, ucontrol); +} + +static int slave_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + int err, ch, changed = 0; + + err = slave_init(slave); + if (err < 0) + return err; + for (ch = 0; ch < slave->info.count; ch++) { + if (slave->vals[ch] != ucontrol->value.integer.value[ch]) { + changed = 1; + slave->vals[ch] = ucontrol->value.integer.value[ch]; + } + } + if (!changed) + return 0; + return slave_put_val(slave, ucontrol); +} + +static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + /* FIXME: this assumes that the max volume is 0 dB */ + return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv); +} + +static void slave_free(struct snd_kcontrol *kcontrol) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + if (slave->slave.private_free) + slave->slave.private_free(&slave->slave); + if (slave->master) + list_del(&slave->list); + kfree(slave); +} + +/* + * Add a slave control to the group with the given master control + * + * All slaves must be the same type (returning the same information + * via info callback). The fucntion doesn't check it, so it's your + * responsibility. + * + * Also, some additional limitations: + * - at most two channels + * - logarithmic volume control (dB level), no linear volume + * - master can only attenuate the volume, no gain + */ +int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) +{ + struct link_master *master_link = snd_kcontrol_chip(master); + struct link_slave *srec; + + srec = kzalloc(sizeof(*srec) + + slave->count * sizeof(*slave->vd), GFP_KERNEL); + if (!srec) + return -ENOMEM; + srec->slave = *slave; + memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd)); + srec->master = master_link; + + /* override callbacks */ + slave->info = slave_info; + slave->get = slave_get; + slave->put = slave_put; + if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) + slave->tlv.c = slave_tlv_cmd; + slave->private_data = srec; + slave->private_free = slave_free; + + list_add_tail(&srec->list, &master_link->slaves); + return 0; +} + +/* + * ctl callbacks for master controls + */ +static int master_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + int ret; + + ret = master_init(master); + if (ret < 0) + return ret; + uinfo->type = master->info.type; + uinfo->count = master->info.count; + uinfo->value.integer.min = master->info.min_val; + uinfo->value.integer.max = master->info.max_val; + return 0; +} + +static int master_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + int err = master_init(master); + if (err < 0) + return err; + ucontrol->value.integer.value[0] = master->val; + return 0; +} + +static int master_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + struct link_slave *slave; + struct snd_ctl_elem_value *uval; + int err, old_val; + + err = master_init(master); + if (err < 0) + return err; + old_val = master->val; + if (ucontrol->value.integer.value[0] == old_val) + return 0; + + uval = kmalloc(sizeof(*uval), GFP_KERNEL); + if (!uval) + return -ENOMEM; + list_for_each_entry(slave, &master->slaves, list) { + master->val = old_val; + uval->id = slave->slave.id; + slave_get_val(slave, uval); + master->val = ucontrol->value.integer.value[0]; + slave_put_val(slave, uval); + } + kfree(uval); + return 1; +} + +static void master_free(struct snd_kcontrol *kcontrol) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + struct link_slave *slave; + + list_for_each_entry(slave, &master->slaves, list) + slave->master = NULL; + kfree(master); +} + + +/* + * Create a virtual master control with the given name + */ +struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, + const unsigned int *tlv) +{ + struct link_master *master; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new knew; + + memset(&knew, 0, sizeof(knew)); + knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + knew.name = name; + knew.info = master_info; + + master = kzalloc(sizeof(*master), GFP_KERNEL); + if (!master) + return NULL; + INIT_LIST_HEAD(&master->slaves); + + kctl = snd_ctl_new1(&knew, master); + if (!kctl) { + kfree(master); + return NULL; + } + /* override some callbacks */ + kctl->info = master_info; + kctl->get = master_get; + kctl->put = master_put; + kctl->private_free = master_free; + + /* additional (constant) TLV read */ + if (tlv) { + /* FIXME: this assumes that the max volume is 0 dB */ + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + kctl->tlv.p = tlv; + } + return kctl; +} -- cgit v0.10.2 From 2134ea4f37d36addbe86d4901f6c67a22a5db006 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Jan 2008 16:53:55 +0100 Subject: [ALSA] hda-codec - Add virtual master controls Add master controls using vmaster to codecs that have no real hardware master volume registers. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a2b40dc..caacc58 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1012,6 +1012,66 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, return 0; } +/* + * set (static) TLV for virtual master volume; recalculated as max 0dB + */ +void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, + unsigned int *tlv) +{ + u32 caps; + int nums, step; + + caps = query_amp_caps(codec, nid, dir); + nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + step = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT; + step = (step + 1) * 25; + tlv[0] = SNDRV_CTL_TLVT_DB_SCALE; + tlv[1] = 2 * sizeof(unsigned int); + tlv[2] = -nums * step; + tlv[3] = step; +} + +/* find a mixer control element with the given name */ +struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, + const char *name) +{ + struct snd_ctl_elem_id id; + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, name); + return snd_ctl_find_id(codec->bus->card, &id); +} + +/* create a virtual master control and add slaves */ +int snd_hda_add_vmaster(struct hda_codec *codec, char *name, + unsigned int *tlv, const char **slaves) +{ + struct snd_kcontrol *kctl; + const char **s; + int err; + + kctl = snd_ctl_make_virtual_master(name, tlv); + if (!kctl) + return -ENOMEM; + err = snd_ctl_add(codec->bus->card, kctl); + if (err < 0) + return err; + + for (s = slaves; *s; s++) { + struct snd_kcontrol *sctl; + + sctl = snd_hda_find_mixer_ctl(codec, *s); + if (!sctl) { + snd_printdd("Cannot find slave %s, skipped\n", *s); + continue; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + } + return 0; +} + /* switch */ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e09f41b..ddc61a1 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -90,6 +90,13 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, void snd_hda_codec_resume_amp(struct hda_codec *codec); #endif +void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, + unsigned int *tlv); +struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, + const char *name); +int snd_hda_add_vmaster(struct hda_codec *codec, char *name, + unsigned int *tlv, const char **slaves); + /* amp value bits */ #define HDA_AMP_MUTE 0x80 #define HDA_AMP_UNMUTE 0x00 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 6664a06..b075540 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -78,6 +78,11 @@ struct ad198x_spec { #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif + /* for virtual master */ + hda_nid_t vmaster_nid; + u32 vmaster_tlv[4]; + const char **slave_vols; + const char **slave_sws; }; /* @@ -125,6 +130,28 @@ static int ad198x_init(struct hda_codec *codec) return 0; } +static const char *ad_slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "Headphone Playback Volume", + "Mono Playback Volume", + NULL +}; + +static const char *ad_slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "Headphone Playback Switch", + "Mono Playback Switch", + NULL +}; + static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -146,6 +173,27 @@ static int ad198x_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* if we have no master control, let's create it */ + if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, + HDA_OUTPUT, spec->vmaster_tlv); + err = snd_hda_add_vmaster(codec, "Master Playback Volume", + spec->vmaster_tlv, + (spec->slave_vols ? + spec->slave_vols : ad_slave_vols)); + if (err < 0) + return err; + } + if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, + (spec->slave_sws ? + spec->slave_sws : ad_slave_sws)); + if (err < 0) + return err; + } + return 0; } @@ -899,6 +947,7 @@ static int patch_ad1986a(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = ad1986a_loopbacks; #endif + spec->vmaster_nid = 0x1b; codec->patch_ops = ad198x_patch_ops; @@ -1141,6 +1190,7 @@ static int patch_ad1983(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = ad1983_loopbacks; #endif + spec->vmaster_nid = 0x05; codec->patch_ops = ad198x_patch_ops; @@ -1537,6 +1587,7 @@ static int patch_ad1981(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = ad1981_loopbacks; #endif + spec->vmaster_nid = 0x05; codec->patch_ops = ad198x_patch_ops; @@ -2850,6 +2901,7 @@ static int patch_ad1988(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = ad1988_loopbacks; #endif + spec->vmaster_nid = 0x04; return 0; } @@ -3016,6 +3068,19 @@ static struct hda_amp_list ad1884_loopbacks[] = { }; #endif +static const char *ad1884_slave_vols[] = { + "PCM Playback Volume", + "Mic Playback Volume", + "Mono Playback Volume", + "Front Mic Playback Volume", + "Mic Playback Volume", + "CD Playback Volume", + "Internal Mic Playback Volume", + "Docking Mic Playback Volume" + "Beep Playback Volume", + NULL +}; + static int patch_ad1884(struct hda_codec *codec) { struct ad198x_spec *spec; @@ -3043,6 +3108,9 @@ static int patch_ad1884(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = ad1884_loopbacks; #endif + spec->vmaster_nid = 0x04; + /* we need to cover all playback volumes */ + spec->slave_vols = ad1884_slave_vols; codec->patch_ops = ad198x_patch_ops; @@ -3485,6 +3553,7 @@ static int patch_ad1882(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = ad1882_loopbacks; #endif + spec->vmaster_nid = 0x04; codec->patch_ops = ad198x_patch_ops; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9184586..4bc7f3d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -262,6 +262,9 @@ struct alc_spec { unsigned int sense_updated: 1; unsigned int jack_present: 1; + /* for virtual master */ + hda_nid_t vmaster_nid; + u32 vmaster_tlv[4]; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -1309,8 +1312,8 @@ static hda_nid_t alc880_f1734_dac_nids[1] = { static struct snd_kcontrol_new alc880_f1734_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -1408,10 +1411,10 @@ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { /* Uniwill */ static struct snd_kcontrol_new alc880_uniwill_mixer[] = { - HDA_CODEC_VOLUME("HPhone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("HPhone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), @@ -1451,16 +1454,50 @@ static struct snd_kcontrol_new alc880_fujitsu_mixer[] = { }; static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { - HDA_CODEC_VOLUME("HPhone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("HPhone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; /* + * virtual master controls + */ + +/* + * slave controls for virtual master + */ +static const char *alc_slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "Headphone Playback Volume", + "Speaker Playback Volume", + "Mono Playback Volume", + "iSpeaker Playback Volume", + "Line-Out Playback Volume", + NULL, +}; + +static const char *alc_slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "Headphone Playback Switch", + "Speaker Playback Switch", + "Mono Playback Switch", + "iSpeaker Playback Switch", + NULL, +}; + +/* * build control elements */ static int alc_build_controls(struct hda_codec *codec) @@ -1486,6 +1523,23 @@ static int alc_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* if we have no master control, let's create it */ + if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, + HDA_OUTPUT, spec->vmaster_tlv); + err = snd_hda_add_vmaster(codec, "Master Playback Volume", + spec->vmaster_tlv, alc_slave_vols); + if (err < 0) + return err; + } + if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, alc_slave_sws); + if (err < 0) + return err; + } + return 0; } @@ -2034,8 +2088,8 @@ static struct hda_channel_mode alc880_lg_ch_modes[3] = { static struct snd_kcontrol_new alc880_lg_mixer[] = { /* FIXME: it's not really "master" but front channels */ - HDA_CODEC_VOLUME("Master Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), @@ -3592,6 +3646,8 @@ static int patch_alc880(struct hda_codec *codec) } } + spec->vmaster_nid = 0x0c; + codec->patch_ops = alc_patch_ops; if (board_config == ALC880_AUTO) spec->init_hook = alc880_auto_init; @@ -4969,6 +5025,8 @@ static int patch_alc260(struct hda_codec *codec) spec->stream_digital_playback = &alc260_pcm_digital_playback; spec->stream_digital_capture = &alc260_pcm_digital_capture; + spec->vmaster_nid = 0x08; + codec->patch_ops = alc_patch_ops; if (board_config == ALC260_AUTO) spec->init_hook = alc260_auto_init; @@ -5169,15 +5227,15 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { }; static struct snd_kcontrol_new alc885_mbp3_mixer[] = { - HDA_CODEC_VOLUME("Master Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Master Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Speaker Switch", 0x14, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Out Volume", 0x0d,0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line In Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Line In Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), { } /* end */ }; @@ -6181,6 +6239,8 @@ static int patch_alc882(struct hda_codec *codec) } } + spec->vmaster_nid = 0x0c; + codec->patch_ops = alc_patch_ops; if (board_config == ALC882_AUTO) spec->init_hook = alc882_auto_init; @@ -7763,6 +7823,8 @@ static int patch_alc883(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); } + spec->vmaster_nid = 0x0c; + codec->patch_ops = alc_patch_ops; if (board_config == ALC883_AUTO) spec->init_hook = alc883_auto_init; @@ -9123,6 +9185,8 @@ static int patch_alc262(struct hda_codec *codec) } } + spec->vmaster_nid = 0x0c; + codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) spec->init_hook = alc262_auto_init; @@ -9848,6 +9912,9 @@ static int patch_alc268(struct hda_codec *codec) } } } + + spec->vmaster_nid = 0x02; + codec->patch_ops = alc_patch_ops; if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; @@ -11358,6 +11425,8 @@ static int patch_alc861(struct hda_codec *codec) spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; + spec->vmaster_nid = 0x03; + codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) spec->init_hook = alc861_auto_init; @@ -12334,6 +12403,8 @@ static int patch_alc861vd(struct hda_codec *codec) spec->mixers[spec->num_mixers] = alc861vd_capture_mixer; spec->num_mixers++; + spec->vmaster_nid = 0x02; + codec->patch_ops = alc_patch_ops; if (board_config == ALC861VD_AUTO) @@ -13305,6 +13376,8 @@ static int patch_alc662(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); } + spec->vmaster_nid = 0x02; + codec->patch_ops = alc_patch_ops; if (board_config == ALC662_AUTO) spec->init_hook = alc662_auto_init; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a0af868..190e112 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -170,6 +170,9 @@ struct sigmatel_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_dimux; struct hda_input_mux private_imux; + + /* virtual master */ + unsigned int vmaster_tlv[4]; }; static hda_nid_t stac9200_adc_nids[1] = { @@ -794,6 +797,34 @@ static struct snd_kcontrol_new stac_dmux_mixer = { .put = stac92xx_dmux_enum_put, }; +static const char *slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "Headphone Playback Volume", + "Headphone Playback Volume", + "Speaker Playback Volume", + "External Speaker Playback Volume", + "Speaker2 Playback Volume", + NULL +}; + +static const char *slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "Headphone Playback Switch", + "Headphone Playback Switch", + "Speaker Playback Switch", + "External Speaker Playback Switch", + "Speaker2 Playback Switch", + NULL +}; + static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -827,6 +858,23 @@ static int stac92xx_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* if we have no master control, let's create it */ + if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], + HDA_OUTPUT, spec->vmaster_tlv); + err = snd_hda_add_vmaster(codec, "Master Playback Volume", + spec->vmaster_tlv, slave_vols); + if (err < 0) + return err; + } + if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, slave_sws); + if (err < 0) + return err; + } + return 0; } -- cgit v0.10.2 From a64135a2d880183a2aff149f42dab7779a37a67f Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 10 Jan 2008 16:55:06 +0100 Subject: [ALSA] hda: 92HD7XXX power management support Added support for advanced power management support for output ports on 92HD7xxx family of codecs. Inactive output ports are powered down when the pin sense doesn't detect a connection, and powered back up when a connection is sensed. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 190e112..0e85e47 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -34,7 +34,8 @@ #include "hda_local.h" #define NUM_CONTROL_ALLOC 32 -#define STAC_HP_EVENT 0x37 +#define STAC_PWR_EVENT 0x20 +#define STAC_HP_EVENT 0x30 enum { STAC_REF, @@ -126,6 +127,10 @@ struct sigmatel_spec { unsigned char aloopback_mask; unsigned char aloopback_shift; + /* power management */ + unsigned int num_pwrs; + hda_nid_t *pwr_nids; + /* playback */ struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -187,6 +192,11 @@ static hda_nid_t stac9200_dac_nids[1] = { 0x02, }; +static hda_nid_t stac92hd73xx_pwr_nids[8] = { + 0x0a, 0x0b, 0x0c, 0xd, 0x0e, + 0x0f, 0x10, 0x11 +}; + static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; @@ -209,6 +219,10 @@ static hda_nid_t stac92hd73xx_dmux_nids[2] = { 0x20, 0x21, }; +static hda_nid_t stac92hd71bxx_pwr_nids[3] = { + 0x0a, 0x0d, 0x0f +}; + static hda_nid_t stac92hd71bxx_adc_nids[2] = { 0x12, 0x13, }; @@ -546,7 +560,7 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* connect ports 0d and 0f to audio mixer */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ @@ -2704,6 +2718,16 @@ static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, (AC_USRSP_EN | event)); } +static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) +{ + int i; + for (i = 0; i < cfg->hp_outs; i++) + if (cfg->hp_pins[i] == nid) + return 1; /* nid is a HP-Out */ + + return 0; /* nid is not a HP-Out */ +}; + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -2739,10 +2763,23 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, nid, pinctl); } } - if (spec->num_dmics > 0) - for (i = 0; i < spec->num_dmics; i++) - stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], - AC_PINCTL_IN_EN); + for (i = 0; i < spec->num_dmics; i++) + stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], + AC_PINCTL_IN_EN); + for (i = 0; i < spec->num_pwrs; i++) { + int event = is_nid_hp_pin(cfg, spec->pwr_nids[i]) + ? STAC_HP_EVENT : STAC_PWR_EVENT; + int pinctl = snd_hda_codec_read(codec, spec->pwr_nids[i], + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + /* outputs are only ports capable of power management + * any attempts on powering down a input port cause the + * referenced VREF to act quirky. + */ + if (pinctl & AC_PINCTL_IN_EN) + continue; + enable_pin_detect(codec, spec->pwr_nids[i], event | i); + codec->patch_ops.unsol_event(codec, (event | i) << 26); + } if (cfg->dig_out_pin) stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, @@ -2869,12 +2906,37 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) } } +static void stac92xx_pin_sense(struct hda_codec *codec, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->pwr_nids[idx]; + int presence, val; + val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) + & 0x000000ff; + presence = get_hp_pin_presence(codec, nid); + idx = 1 << idx; + + if (presence) + val &= ~idx; + else + val |= idx; + + /* power down unused output ports */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val); +}; + static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) { - switch (res >> 26) { + struct sigmatel_spec *spec = codec->spec; + int idx = res >> 26 & 0x0f; + + switch ((res >> 26) & 0x30) { case STAC_HP_EVENT: stac92xx_hp_detect(codec, res); - break; + /* fallthru */ + case STAC_PWR_EVENT: + if (spec->num_pwrs > 0) + stac92xx_pin_sense(codec, idx); } } @@ -2945,6 +3007,7 @@ static int patch_stac9200(struct hda_codec *codec) spec->num_muxes = 1; spec->num_dmics = 0; spec->num_adcs = 1; + spec->num_pwrs = 0; if (spec->board_config == STAC_9200_GATEWAY) spec->init = stac9200_eapd_init; @@ -3000,6 +3063,7 @@ static int patch_stac925x(struct hda_codec *codec) spec->mux_nids = stac925x_mux_nids; spec->num_muxes = 1; spec->num_adcs = 1; + spec->num_pwrs = 0; switch (codec->vendor_id) { case 0x83847632: /* STAC9202 */ case 0x83847633: /* STAC9202D */ @@ -3123,6 +3187,9 @@ again: spec->gpio_mask = spec->gpio_data = 0x000001; stac92xx_enable_gpio_mask(codec); + spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); + spec->pwr_nids = stac92hd73xx_pwr_nids; + err = stac92xx_parse_auto_config(codec, 0x22, 0x24); if (!err) { @@ -3205,6 +3272,9 @@ again: spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); + spec->pwr_nids = stac92hd71bxx_pwr_nids; + spec->multiout.num_dacs = 2; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; @@ -3299,6 +3369,7 @@ static int patch_stac922x(struct hda_codec *codec) spec->num_muxes = ARRAY_SIZE(stac922x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac922x_adc_nids); spec->num_dmics = 0; + spec->num_pwrs = 0; spec->init = stac922x_core_init; spec->mixer = stac922x_mixer; @@ -3405,6 +3476,7 @@ static int patch_stac927x(struct hda_codec *codec) spec->mixer = stac927x_mixer; } + spec->num_pwrs = 0; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; @@ -3466,6 +3538,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->num_dmics = STAC9205_NUM_DMICS; spec->dmux_nids = stac9205_dmux_nids; spec->num_dmuxes = ARRAY_SIZE(stac9205_dmux_nids); + spec->num_pwrs = 0; spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; @@ -3728,6 +3801,7 @@ static int patch_stac9872(struct hda_codec *codec) spec->multiout.hp_nid = VAIO_HP_DAC; spec->num_adcs = ARRAY_SIZE(vaio_adcs); spec->adc_nids = vaio_adcs; + spec->num_pwrs = 0; spec->input_mux = &vaio_mux; spec->mux_nids = vaio_mux_nids; codec->patch_ops = stac9872_vaio_patch_ops; @@ -3741,6 +3815,7 @@ static int patch_stac9872(struct hda_codec *codec) spec->multiout.dac_nids = vaio_dacs; spec->multiout.hp_nid = VAIO_HP_DAC; spec->num_adcs = ARRAY_SIZE(vaio_adcs); + spec->num_pwrs = 0; spec->adc_nids = vaio_adcs; spec->input_mux = &vaio_mux; spec->mux_nids = vaio_mux_nids; -- cgit v0.10.2 From 8c12158687fc78091730d5456336b7efbf6f2250 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 11 Jan 2008 08:45:08 +0100 Subject: [ALSA] PCM interface - rename SNDRV_PCM_TSTAMP_MMAP to SNDRV_PCM_TSTAMP_ENABLE Change semantics for SNDRV_PCM_TSTAMP_MMAP. Doing timestamping only in the interrupt handler might cause that hw_ptr is not related to actual timestamp. With this change, grab timestamp at every hw_ptr update to have always valid timestamp + ring buffer position pair. With this change, SNDRV_PCM_TSTAMP_MMAP was renamed to SNDRV_PCM_TSTAMP_ENABLE. It's no regression (I think). Signed-off-by: Jaroslav Kysela diff --git a/include/sound/asound.h b/include/sound/asound.h index 069d912..3eaf155 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -354,8 +354,8 @@ struct snd_pcm_hw_params { enum { SNDRV_PCM_TSTAMP_NONE = 0, - SNDRV_PCM_TSTAMP_MMAP, - SNDRV_PCM_TSTAMP_LAST = SNDRV_PCM_TSTAMP_MMAP, + SNDRV_PCM_TSTAMP_ENABLE, + SNDRV_PCM_TSTAMP_LAST = SNDRV_PCM_TSTAMP_ENABLE, }; struct snd_pcm_sw_params { diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 1502acd..9dd9bc7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -227,7 +227,7 @@ static char *snd_pcm_subformat_names[] = { static char *snd_pcm_tstamp_mode_names[] = { TSTAMP(NONE), - TSTAMP(MMAP), + TSTAMP(ENABLE), }; static const char *snd_pcm_stream_name(int stream) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ed0223c..1533f03 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -144,6 +144,8 @@ static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substre { snd_pcm_uframes_t pos; + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) return pos; /* XRUN */ @@ -186,8 +188,6 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt; snd_pcm_sframes_t delta; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP) - snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); pos = snd_pcm_update_hw_ptr_pos(substream, runtime); if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e6e4aa8..6244911 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -580,9 +580,15 @@ int snd_pcm_status(struct snd_pcm_substream *substream, if (status->state == SNDRV_PCM_STATE_OPEN) goto _end; status->trigger_tstamp = runtime->trigger_tstamp; - if (snd_pcm_running(substream)) + if (snd_pcm_running(substream)) { snd_pcm_update_hw_ptr(substream); + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { + status->tstamp = runtime->status->tstamp; + goto _tstamp_end; + } + } snd_pcm_gettime(runtime, &status->tstamp); + _tstamp_end: status->appl_ptr = runtime->control->appl_ptr; status->hw_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- cgit v0.10.2 From 3fa2ef7468e86777b5d9d2462b950e3a6b6b6283 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 11 Jan 2008 11:39:06 +0100 Subject: [ALSA] hda: Add new STAC9205 PCI_QUIRK Added a new STAC 9205 quirk for Vostro 1500. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0e85e47..c3496de 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1530,6 +1530,7 @@ static unsigned int ref9205_pin_configs[12] = { 102801FD 10280204 1028021F + 10280228 (Dell Vostro 1500) */ static unsigned int dell_9205_m42_pin_configs[12] = { 0x0321101F, 0x03A11020, 0x400003FA, 0x90170310, @@ -1613,6 +1614,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, "Dell Inspiron", STAC_9205_DELL_M44), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, + "Dell Vostro 1500", STAC_9205_DELL_M42), {} /* terminator */ }; -- cgit v0.10.2 From 9c8f2abdc5723b454ef4bfe23ec33ae2a46f62fc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 11 Jan 2008 16:12:23 +0100 Subject: [ALSA] hda-codec - print control name in error messages Print the name of the defect controls in error messages in amp info callback. This will make debugging easier. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index caacc58..b5e69b1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -933,7 +933,8 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; if (!caps) { printk(KERN_WARNING "hda_codec: " - "num_steps = 0 for NID=0x%x\n", nid); + "num_steps = 0 for NID=0x%x (ctl = %s)\n", nid, + kcontrol->id.name); return -EINVAL; } uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; -- cgit v0.10.2 From ce22e03e62fd37fb2612abb7af1c66cc17038606 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 11 Jan 2008 17:38:35 +0100 Subject: [ALSA] hda-codec - Don't build boost controls for digital mics The ALC auto-probe creates mic boost controls automatically for the probed pins, but it assumes that they are analog mics. The digital mics have no boost controls and must be skipped. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4bc7f3d..11bd68b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6111,7 +6111,7 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) hda_nid_t nid; nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; - if (nid) { + if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { err = add_control(spec, ALC_CTL_WIDGET_VOL, "Mic Boost", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); @@ -6119,7 +6119,7 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) return err; } nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; - if (nid) { + if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { err = add_control(spec, ALC_CTL_WIDGET_VOL, "Front Mic Boost", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); -- cgit v0.10.2 From 17467f23395f05ba7b361f7b504fe0f1095d5bb7 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 11 Jan 2008 18:15:26 +0100 Subject: [ALSA] Add ASoC drivers for the Freescale MPC8610 SoC Add the ASoC drivers for the Freescale MPC8610 SoC and the MPC8610 HPCD reference board. Signed-off-by: Timur Tabi Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 97b2552..2765852 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -28,6 +28,7 @@ source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/fsl/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 3041403..4869c9a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig new file mode 100644 index 0000000..257101f --- /dev/null +++ b/sound/soc/fsl/Kconfig @@ -0,0 +1,20 @@ +menu "ALSA SoC audio for Freescale SOCs" + +config SND_SOC_MPC8610 + bool "ALSA SoC support for the MPC8610 SOC" + depends on SND_SOC && MPC8610_HPCD + default y if MPC8610 + help + Say Y if you want to add support for codecs attached to the SSI + device on an MPC8610. + +config SND_SOC_MPC8610_HPCD + bool "ALSA SoC support for the Freescale MPC8610 HPCD board" + depends on SND_SOC_MPC8610 + select SND_SOC_CS4270 + select SND_SOC_CS4270_VD33_ERRATA + default y if MPC8610_HPCD + help + Say Y if you want to enable audio on the Freescale MPC8610 HPCD. + +endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile new file mode 100644 index 0000000..62f680a --- /dev/null +++ b/sound/soc/fsl/Makefile @@ -0,0 +1,6 @@ +# MPC8610 HPCD Machine Support +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o + +# MPC8610 Platform Support +obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o + diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c new file mode 100644 index 0000000..2173203 --- /dev/null +++ b/sound/soc/fsl/fsl_dma.c @@ -0,0 +1,839 @@ +/* + * Freescale DMA ALSA SoC PCM driver + * + * Author: Timur Tabi + * + * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. + * + * This driver implements ASoC support for the Elo DMA controller, which is + * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, + * the PCM driver is what handles the DMA buffer. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include + +#include "fsl_dma.h" + +/* + * The formats that the DMA controller supports, which is anything + * that is 8, 16, or 32 bits. + */ +#define FSLDMA_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_U24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_S32_BE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_U32_BE) + +#define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS) + +/* DMA global data. This structure is used by fsl_dma_open() to determine + * which DMA channels to assign to a substream. Unfortunately, ASoC V1 does + * not allow the machine driver to provide this information to the PCM + * driver in advance, and there's no way to differentiate between the two + * DMA controllers. So for now, this driver only supports one SSI device + * using two DMA channels. We cannot support multiple DMA devices. + * + * ssi_stx_phys: bus address of SSI STX register + * ssi_srx_phys: bus address of SSI SRX register + * dma_channel: pointer to the DMA channel's registers + * irq: IRQ for this DMA channel + * assigned: set to 1 if that DMA channel is assigned to a substream + */ +static struct { + dma_addr_t ssi_stx_phys; + dma_addr_t ssi_srx_phys; + struct ccsr_dma_channel __iomem *dma_channel[2]; + unsigned int irq[2]; + unsigned int assigned[2]; +} dma_global_data; + +/* + * The number of DMA links to use. Two is the bare minimum, but if you + * have really small links you might need more. + */ +#define NUM_DMA_LINKS 2 + +/** fsl_dma_private: p-substream DMA data + * + * Each substream has a 1-to-1 association with a DMA channel. + * + * The link[] array is first because it needs to be aligned on a 32-byte + * boundary, so putting it first will ensure alignment without padding the + * structure. + * + * @link[]: array of link descriptors + * @controller_id: which DMA controller (0, 1, ...) + * @channel_id: which DMA channel on the controller (0, 1, 2, ...) + * @dma_channel: pointer to the DMA channel's registers + * @irq: IRQ for this DMA channel + * @substream: pointer to the substream object, needed by the ISR + * @ssi_sxx_phys: bus address of the STX or SRX register to use + * @ld_buf_phys: physical address of the LD buffer + * @current_link: index into link[] of the link currently being processed + * @dma_buf_phys: physical address of the DMA buffer + * @dma_buf_next: physical address of the next period to process + * @dma_buf_end: physical address of the byte after the end of the DMA + * @buffer period_size: the size of a single period + * @num_periods: the number of periods in the DMA buffer + */ +struct fsl_dma_private { + struct fsl_dma_link_descriptor link[NUM_DMA_LINKS]; + unsigned int controller_id; + unsigned int channel_id; + struct ccsr_dma_channel __iomem *dma_channel; + unsigned int irq; + struct snd_pcm_substream *substream; + dma_addr_t ssi_sxx_phys; + dma_addr_t ld_buf_phys; + unsigned int current_link; + dma_addr_t dma_buf_phys; + dma_addr_t dma_buf_next; + dma_addr_t dma_buf_end; + size_t period_size; + unsigned int num_periods; +}; + +/** + * fsl_dma_hardare: define characteristics of the PCM hardware. + * + * The PCM hardware is the Freescale DMA controller. This structure defines + * the capabilities of that hardware. + * + * Since the sampling rate and data format are not controlled by the DMA + * controller, we specify no limits for those values. The only exception is + * period_bytes_min, which is set to a reasonably low value to prevent the + * DMA controller from generating too many interrupts per second. + * + * Since each link descriptor has a 32-bit byte count field, we set + * period_bytes_max to the largest 32-bit number. We also have no maximum + * number of periods. + */ +static const struct snd_pcm_hardware fsl_dma_hardware = { + + .info = SNDRV_PCM_INFO_INTERLEAVED, + .formats = FSLDMA_PCM_FORMATS, + .rates = FSLDMA_PCM_RATES, + .rate_min = 5512, + .rate_max = 192000, + .period_bytes_min = 512, /* A reasonable limit */ + .period_bytes_max = (u32) -1, + .periods_min = NUM_DMA_LINKS, + .periods_max = (unsigned int) -1, + .buffer_bytes_max = 128 * 1024, /* A reasonable limit */ +}; + +/** + * fsl_dma_abort_stream: tell ALSA that the DMA transfer has aborted + * + * This function should be called by the ISR whenever the DMA controller + * halts data transfer. + */ +static void fsl_dma_abort_stream(struct snd_pcm_substream *substream) +{ + unsigned long flags; + + snd_pcm_stream_lock_irqsave(substream, flags); + + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + + snd_pcm_stream_unlock_irqrestore(substream, flags); +} + +/** + * fsl_dma_update_pointers - update LD pointers to point to the next period + * + * As each period is completed, this function changes the the link + * descriptor pointers for that period to point to the next period. + */ +static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) +{ + struct fsl_dma_link_descriptor *link = + &dma_private->link[dma_private->current_link]; + + /* Update our link descriptors to point to the next period */ + if (dma_private->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->source_addr = + cpu_to_be32(dma_private->dma_buf_next); + else + link->dest_addr = + cpu_to_be32(dma_private->dma_buf_next); + + /* Update our variables for next time */ + dma_private->dma_buf_next += dma_private->period_size; + + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + dma_private->dma_buf_next = dma_private->dma_buf_phys; + + if (++dma_private->current_link >= NUM_DMA_LINKS) + dma_private->current_link = 0; +} + +/** + * fsl_dma_isr: interrupt handler for the DMA controller + * + * @irq: IRQ of the DMA channel + * @dev_id: pointer to the dma_private structure for this DMA channel + */ +static irqreturn_t fsl_dma_isr(int irq, void *dev_id) +{ + struct fsl_dma_private *dma_private = dev_id; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + irqreturn_t ret = IRQ_NONE; + u32 sr, sr2 = 0; + + /* We got an interrupt, so read the status register to see what we + were interrupted for. + */ + sr = in_be32(&dma_channel->sr); + + if (sr & CCSR_DMA_SR_TE) { + dev_err(dma_private->substream->pcm->card->dev, + "DMA transmit error (controller=%u channel=%u irq=%u\n", + dma_private->controller_id, + dma_private->channel_id, irq); + fsl_dma_abort_stream(dma_private->substream); + sr2 |= CCSR_DMA_SR_TE; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_CH) + ret = IRQ_HANDLED; + + if (sr & CCSR_DMA_SR_PE) { + dev_err(dma_private->substream->pcm->card->dev, + "DMA%u programming error (channel=%u irq=%u)\n", + dma_private->controller_id, + dma_private->channel_id, irq); + fsl_dma_abort_stream(dma_private->substream); + sr2 |= CCSR_DMA_SR_PE; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_EOLNI) { + sr2 |= CCSR_DMA_SR_EOLNI; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_CB) + ret = IRQ_HANDLED; + + if (sr & CCSR_DMA_SR_EOSI) { + struct snd_pcm_substream *substream = dma_private->substream; + + /* Tell ALSA we completed a period. */ + snd_pcm_period_elapsed(substream); + + /* + * Update our link descriptors to point to the next period. We + * only need to do this if the number of periods is not equal to + * the number of links. + */ + if (dma_private->num_periods != NUM_DMA_LINKS) + fsl_dma_update_pointers(dma_private); + + sr2 |= CCSR_DMA_SR_EOSI; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_EOLSI) { + sr2 |= CCSR_DMA_SR_EOLSI; + ret = IRQ_HANDLED; + } + + /* Clear the bits that we set */ + if (sr2) + out_be32(&dma_channel->sr, sr2); + + return ret; +} + +/** + * fsl_dma_new: initialize this PCM driver. + * + * This function is called when the codec driver calls snd_soc_new_pcms(), + * once for each .dai_link in the machine driver's snd_soc_machine + * structure. + */ +static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + static u64 fsl_dma_dmamask = DMA_BIT_MASK(32); + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &fsl_dma_dmamask; + + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = fsl_dma_dmamask; + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[0].substream->dma_buffer); + if (ret) { + dev_err(card->dev, + "Can't allocate playback DMA buffer (size=%u)\n", + fsl_dma_hardware.buffer_bytes_max); + return -ENOMEM; + } + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[1].substream->dma_buffer); + if (ret) { + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + dev_err(card->dev, + "Can't allocate capture DMA buffer (size=%u)\n", + fsl_dma_hardware.buffer_bytes_max); + return -ENOMEM; + } + + return 0; +} + +/** + * fsl_dma_open: open a new substream. + * + * Each substream has its own DMA buffer. + */ +static int fsl_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private; + dma_addr_t ld_buf_phys; + unsigned int channel; + int ret = 0; + + /* + * Reject any DMA buffer whose size is not a multiple of the period + * size. We need to make sure that the DMA buffer can be evenly divided + * into periods. + */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(substream->pcm->card->dev, "invalid buffer size\n"); + return ret; + } + + channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + if (dma_global_data.assigned[channel]) { + dev_err(substream->pcm->card->dev, + "DMA channel already assigned\n"); + return -EBUSY; + } + + dma_private = dma_alloc_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); + if (!dma_private) { + dev_err(substream->pcm->card->dev, + "can't allocate DMA private data\n"); + return -ENOMEM; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_private->ssi_sxx_phys = dma_global_data.ssi_stx_phys; + else + dma_private->ssi_sxx_phys = dma_global_data.ssi_srx_phys; + + dma_private->dma_channel = dma_global_data.dma_channel[channel]; + dma_private->irq = dma_global_data.irq[channel]; + dma_private->substream = substream; + dma_private->ld_buf_phys = ld_buf_phys; + dma_private->dma_buf_phys = substream->dma_buffer.addr; + + /* We only support one DMA controller for now */ + dma_private->controller_id = 0; + dma_private->channel_id = channel; + + ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); + if (ret) { + dev_err(substream->pcm->card->dev, + "can't register ISR for IRQ %u (ret=%i)\n", + dma_private->irq, ret); + dma_free_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), + dma_private, dma_private->ld_buf_phys); + return ret; + } + + dma_global_data.assigned[channel] = 1; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); + runtime->private_data = dma_private; + + return 0; +} + +/** + * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors. + * + * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link + * descriptors that ping-pong from one period to the next. For example, if + * there are six periods and two link descriptors, this is how they look + * before playback starts: + * + * The last link descriptor + * ____________ points back to the first + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * | | + * V V + * _________________________________________ + * | | | | | | | The DMA buffer is + * | | | | | | | divided into 6 parts + * |______|______|______|______|______|______| + * + * and here's how they look after the first period is finished playing: + * + * ____________ + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * |______________ + * | | + * V V + * _________________________________________ + * | | | | | | | + * | | | | | | | + * |______|______|______|______|______|______| + * + * The first link descriptor now points to the third period. The DMA + * controller is currently playing the second period. When it finishes, it + * will jump back to the first descriptor and play the third period. + * + * There are four reasons we do this: + * + * 1. The only way to get the DMA controller to automatically restart the + * transfer when it gets to the end of the buffer is to use chaining + * mode. Basic direct mode doesn't offer that feature. + * 2. We need to receive an interrupt at the end of every period. The DMA + * controller can generate an interrupt at the end of every link transfer + * (aka segment). Making each period into a DMA segment will give us the + * interrupts we need. + * 3. By creating only two link descriptors, regardless of the number of + * periods, we do not need to reallocate the link descriptors if the + * number of periods changes. + * 4. All of the audio data is still stored in a single, contiguous DMA + * buffer, which is what ALSA expects. We're just dividing it into + * contiguous parts, and creating a link descriptor for each one. + * + * Note that due to a quirk of the SSI's STX register, the target address + * for the DMA operations depends on the sample size. So we don't program + * the dest_addr (for playback -- source_addr for capture) fields in the + * link descriptors here. We do that in fsl_dma_prepare() + */ +static int fsl_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + + dma_addr_t temp_addr; /* Pointer to next period */ + u64 temp_link; /* Pointer to next link descriptor */ + u32 mr; /* Temporary variable for MR register */ + + unsigned int i; + + /* Get all the parameters we need */ + size_t buffer_size = params_buffer_bytes(hw_params); + size_t period_size = params_period_bytes(hw_params); + + /* Initialize our DMA tracking variables */ + dma_private->period_size = period_size; + dma_private->num_periods = params_periods(hw_params); + dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; + dma_private->dma_buf_next = dma_private->dma_buf_phys + + (NUM_DMA_LINKS * period_size); + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + dma_private->dma_buf_next = dma_private->dma_buf_phys; + + /* + * Initialize each link descriptor. + * + * The actual address in STX0 (destination for playback, source for + * capture) is based on the sample size, but we don't know the sample + * size in this function, so we'll have to adjust that later. See + * comments in fsl_dma_prepare(). + * + * The DMA controller does not have a cache, so the CPU does not + * need to tell it to flush its cache. However, the DMA + * controller does need to tell the CPU to flush its cache. + * That's what the SNOOP bit does. + * + * Also, even though the DMA controller supports 36-bit addressing, for + * simplicity we currently support only 32-bit addresses for the audio + * buffer itself. + */ + temp_addr = substream->dma_buffer.addr; + temp_link = dma_private->ld_buf_phys + + sizeof(struct fsl_dma_link_descriptor); + + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + link->count = cpu_to_be32(period_size); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->next = cpu_to_be64(temp_link); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->source_addr = cpu_to_be32(temp_addr); + else + link->dest_addr = cpu_to_be32(temp_addr); + + temp_addr += period_size; + temp_link += sizeof(struct fsl_dma_link_descriptor); + } + /* The last link descriptor points to the first */ + dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + + /* Tell the DMA controller where the first link descriptor is */ + out_be32(&dma_channel->clndar, + CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); + out_be32(&dma_channel->eclndar, + CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); + + /* The manual says the BCR must be clear before enabling EMP */ + out_be32(&dma_channel->bcr, 0); + + /* + * Program the mode register for interrupts, external master control, + * and source/destination hold. Also clear the Channel Abort bit. + */ + mr = in_be32(&dma_channel->mr) & + ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); + + /* + * We want External Master Start and External Master Pause enabled, + * because the SSI is controlling the DMA controller. We want the DMA + * controller to be set up in advance, and then we signal only the SSI + * to start transfering. + * + * We want End-Of-Segment Interrupts enabled, because this will generate + * an interrupt at the end of each segment (each link descriptor + * represents one segment). Each DMA segment is the same thing as an + * ALSA period, so this is how we get an interrupt at the end of every + * period. + * + * We want Error Interrupt enabled, so that we can get an error if + * the DMA controller is mis-programmed somehow. + */ + mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + CCSR_DMA_MR_EMS_EN; + + /* For playback, we want the destination address to be held. For + capture, set the source address to be held. */ + mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; + + out_be32(&dma_channel->mr, mr); + + return 0; +} + +/** + * fsl_dma_prepare - prepare the DMA registers for playback. + * + * This function is called after the specifics of the audio data are known, + * i.e. snd_pcm_runtime is initialized. + * + * In this function, we finish programming the registers of the DMA + * controller that are dependent on the sample size. + * + * One of the drawbacks with big-endian is that when copying integers of + * different sizes to a fixed-sized register, the address to which the + * integer must be copied is dependent on the size of the integer. + * + * For example, if P is the address of a 32-bit register, and X is a 32-bit + * integer, then X should be copied to address P. However, if X is a 16-bit + * integer, then it should be copied to P+2. If X is an 8-bit register, + * then it should be copied to P+3. + * + * So for playback of 8-bit samples, the DMA controller must transfer single + * bytes from the DMA buffer to the last byte of the STX0 register, i.e. + * offset by 3 bytes. For 16-bit samples, the offset is two bytes. + * + * For 24-bit samples, the offset is 1 byte. However, the DMA controller + * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4, + * and 8 bytes at a time). So we do not support packed 24-bit samples. + * 24-bit data must be padded to 32 bits. + */ +static int fsl_dma_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + u32 mr; + unsigned int i; + dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ + unsigned int frame_size; /* Number of bytes per frame */ + + ssi_sxx_phys = dma_private->ssi_sxx_phys; + + mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | + CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); + + switch (runtime->sample_bits) { + case 8: + mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; + ssi_sxx_phys += 3; + break; + case 16: + mr |= CCSR_DMA_MR_DAHTS_2 | CCSR_DMA_MR_SAHTS_2; + ssi_sxx_phys += 2; + break; + case 32: + mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4; + break; + default: + dev_err(substream->pcm->card->dev, + "unsupported sample size %u\n", runtime->sample_bits); + return -EINVAL; + } + + frame_size = runtime->frame_bits / 8; + /* + * BWC should always be a multiple of the frame size. BWC determines + * how many bytes are sent/received before the DMA controller checks the + * SSI to see if it needs to stop. For playback, the transmit FIFO can + * hold three frames, so we want to send two frames at a time. For + * capture, the receive FIFO is triggered when it contains one frame, so + * we want to receive one frame at a time. + */ + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mr |= CCSR_DMA_MR_BWC(2 * frame_size); + else + mr |= CCSR_DMA_MR_BWC(frame_size); + + out_be32(&dma_channel->mr, mr); + + /* + * Program the address of the DMA transfer to/from the SSI. + */ + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->dest_addr = cpu_to_be32(ssi_sxx_phys); + else + link->source_addr = cpu_to_be32(ssi_sxx_phys); + } + + return 0; +} + +/** + * fsl_dma_pointer: determine the current position of the DMA transfer + * + * This function is called by ALSA when ALSA wants to know where in the + * stream buffer the hardware currently is. + * + * For playback, the SAR register contains the physical address of the most + * recent DMA transfer. For capture, the value is in the DAR register. + * + * The base address of the buffer is stored in the source_addr field of the + * first link descriptor. + */ +static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + dma_addr_t position; + snd_pcm_uframes_t frames; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + position = in_be32(&dma_channel->sar); + else + position = in_be32(&dma_channel->dar); + + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); + + /* + * If the current address is just past the end of the buffer, wrap it + * around. + */ + if (frames == runtime->buffer_size) + frames = 0; + + return frames; +} + +/** + * fsl_dma_hw_free: release resources allocated in fsl_dma_hw_params() + * + * Release the resources allocated in fsl_dma_hw_params() and de-program the + * registers. + * + * This function can be called multiple times. + */ +static int fsl_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + + if (dma_private) { + struct ccsr_dma_channel __iomem *dma_channel; + + dma_channel = dma_private->dma_channel; + + /* Stop the DMA */ + out_be32(&dma_channel->mr, CCSR_DMA_MR_CA); + out_be32(&dma_channel->mr, 0); + + /* Reset all the other registers */ + out_be32(&dma_channel->sr, -1); + out_be32(&dma_channel->clndar, 0); + out_be32(&dma_channel->eclndar, 0); + out_be32(&dma_channel->satr, 0); + out_be32(&dma_channel->sar, 0); + out_be32(&dma_channel->datr, 0); + out_be32(&dma_channel->dar, 0); + out_be32(&dma_channel->bcr, 0); + out_be32(&dma_channel->nlndar, 0); + out_be32(&dma_channel->enlndar, 0); + } + + return 0; +} + +/** + * fsl_dma_close: close the stream. + */ +static int fsl_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + if (dma_private) { + if (dma_private->irq) + free_irq(dma_private->irq, dma_private); + + if (dma_private->ld_buf_phys) { + dma_unmap_single(substream->pcm->dev, + dma_private->ld_buf_phys, + sizeof(dma_private->link), DMA_TO_DEVICE); + } + + /* Deallocate the fsl_dma_private structure */ + dma_free_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), + dma_private, dma_private->ld_buf_phys); + substream->runtime->private_data = NULL; + } + + dma_global_data.assigned[dir] = 0; + + return 0; +} + +/* + * Remove this PCM driver. + */ +static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) { + substream = pcm->streams[i].substream; + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + } +} + +static struct snd_pcm_ops fsl_dma_ops = { + .open = fsl_dma_open, + .close = fsl_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsl_dma_hw_params, + .hw_free = fsl_dma_hw_free, + .prepare = fsl_dma_prepare, + .pointer = fsl_dma_pointer, +}; + +struct snd_soc_platform fsl_soc_platform = { + .name = "fsl-dma", + .pcm_ops = &fsl_dma_ops, + .pcm_new = fsl_dma_new, + .pcm_free = fsl_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(fsl_soc_platform); + +/** + * fsl_dma_configure: store the DMA parameters from the fabric driver. + * + * This function is called by the ASoC fabric driver to give us the DMA and + * SSI channel information. + * + * Unfortunately, ASoC V1 does make it possible to determine the DMA/SSI + * data when a substream is created, so for now we need to store this data + * into a global variable. This means that we can only support one DMA + * controller, and hence only one SSI. + */ +int fsl_dma_configure(struct fsl_dma_info *dma_info) +{ + static int initialized; + + /* We only support one DMA controller for now */ + if (initialized) + return 0; + + dma_global_data.ssi_stx_phys = dma_info->ssi_stx_phys; + dma_global_data.ssi_srx_phys = dma_info->ssi_srx_phys; + dma_global_data.dma_channel[0] = dma_info->dma_channel[0]; + dma_global_data.dma_channel[1] = dma_info->dma_channel[1]; + dma_global_data.irq[0] = dma_info->dma_irq[0]; + dma_global_data.irq[1] = dma_info->dma_irq[1]; + dma_global_data.assigned[0] = 0; + dma_global_data.assigned[1] = 0; + + initialized = 1; + return 1; +} +EXPORT_SYMBOL_GPL(fsl_dma_configure); + +MODULE_AUTHOR("Timur Tabi "); +MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h new file mode 100644 index 0000000..430a6ce --- /dev/null +++ b/sound/soc/fsl/fsl_dma.h @@ -0,0 +1,149 @@ +/* + * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MPC8610_PCM_H +#define _MPC8610_PCM_H + +struct ccsr_dma { + u8 res0[0x100]; + struct ccsr_dma_channel { + __be32 mr; /* Mode register */ + __be32 sr; /* Status register */ + __be32 eclndar; /* Current link descriptor extended addr reg */ + __be32 clndar; /* Current link descriptor address register */ + __be32 satr; /* Source attributes register */ + __be32 sar; /* Source address register */ + __be32 datr; /* Destination attributes register */ + __be32 dar; /* Destination address register */ + __be32 bcr; /* Byte count register */ + __be32 enlndar; /* Next link descriptor extended address reg */ + __be32 nlndar; /* Next link descriptor address register */ + u8 res1[4]; + __be32 eclsdar; /* Current list descriptor extended addr reg */ + __be32 clsdar; /* Current list descriptor address register */ + __be32 enlsdar; /* Next list descriptor extended address reg */ + __be32 nlsdar; /* Next list descriptor address register */ + __be32 ssr; /* Source stride register */ + __be32 dsr; /* Destination stride register */ + u8 res2[0x38]; + } channel[4]; + __be32 dgsr; +}; + +#define CCSR_DMA_MR_BWC_DISABLED 0x0F000000 +#define CCSR_DMA_MR_BWC_SHIFT 24 +#define CCSR_DMA_MR_BWC_MASK 0x0F000000 +#define CCSR_DMA_MR_BWC(x) \ + ((ilog2(x) << CCSR_DMA_MR_BWC_SHIFT) & CCSR_DMA_MR_BWC_MASK) +#define CCSR_DMA_MR_EMP_EN 0x00200000 +#define CCSR_DMA_MR_EMS_EN 0x00040000 +#define CCSR_DMA_MR_DAHTS_MASK 0x00030000 +#define CCSR_DMA_MR_DAHTS_1 0x00000000 +#define CCSR_DMA_MR_DAHTS_2 0x00010000 +#define CCSR_DMA_MR_DAHTS_4 0x00020000 +#define CCSR_DMA_MR_DAHTS_8 0x00030000 +#define CCSR_DMA_MR_SAHTS_MASK 0x0000C000 +#define CCSR_DMA_MR_SAHTS_1 0x00000000 +#define CCSR_DMA_MR_SAHTS_2 0x00004000 +#define CCSR_DMA_MR_SAHTS_4 0x00008000 +#define CCSR_DMA_MR_SAHTS_8 0x0000C000 +#define CCSR_DMA_MR_DAHE 0x00002000 +#define CCSR_DMA_MR_SAHE 0x00001000 +#define CCSR_DMA_MR_SRW 0x00000400 +#define CCSR_DMA_MR_EOSIE 0x00000200 +#define CCSR_DMA_MR_EOLNIE 0x00000100 +#define CCSR_DMA_MR_EOLSIE 0x00000080 +#define CCSR_DMA_MR_EIE 0x00000040 +#define CCSR_DMA_MR_XFE 0x00000020 +#define CCSR_DMA_MR_CDSM_SWSM 0x00000010 +#define CCSR_DMA_MR_CA 0x00000008 +#define CCSR_DMA_MR_CTM 0x00000004 +#define CCSR_DMA_MR_CC 0x00000002 +#define CCSR_DMA_MR_CS 0x00000001 + +#define CCSR_DMA_SR_TE 0x00000080 +#define CCSR_DMA_SR_CH 0x00000020 +#define CCSR_DMA_SR_PE 0x00000010 +#define CCSR_DMA_SR_EOLNI 0x00000008 +#define CCSR_DMA_SR_CB 0x00000004 +#define CCSR_DMA_SR_EOSI 0x00000002 +#define CCSR_DMA_SR_EOLSI 0x00000001 + +/* ECLNDAR takes bits 32-36 of the CLNDAR register */ +static inline u32 CCSR_DMA_ECLNDAR_ADDR(u64 x) +{ + return (x >> 32) & 0xf; +} + +#define CCSR_DMA_CLNDAR_ADDR(x) ((x) & 0xFFFFFFFE) +#define CCSR_DMA_CLNDAR_EOSIE 0x00000008 + +/* SATR and DATR, combined */ +#define CCSR_DMA_ATR_PBATMU 0x20000000 +#define CCSR_DMA_ATR_TFLOWLVL_0 0x00000000 +#define CCSR_DMA_ATR_TFLOWLVL_1 0x06000000 +#define CCSR_DMA_ATR_TFLOWLVL_2 0x08000000 +#define CCSR_DMA_ATR_TFLOWLVL_3 0x0C000000 +#define CCSR_DMA_ATR_PCIORDER 0x02000000 +#define CCSR_DMA_ATR_SME 0x01000000 +#define CCSR_DMA_ATR_NOSNOOP 0x00040000 +#define CCSR_DMA_ATR_SNOOP 0x00050000 +#define CCSR_DMA_ATR_ESAD_MASK 0x0000000F + +/** + * List Descriptor for extended chaining mode DMA operations. + * + * The CLSDAR register points to the first (in a linked-list) List + * Descriptor. Each object must be aligned on a 32-byte boundary. Each + * list descriptor points to a linked-list of link Descriptors. + */ +struct fsl_dma_list_descriptor { + __be64 next; /* Address of next list descriptor */ + __be64 first_link; /* Address of first link descriptor */ + __be32 source; /* Source stride */ + __be32 dest; /* Destination stride */ + u8 res[8]; /* Reserved */ +} __attribute__ ((aligned(32), packed)); + +/** + * Link Descriptor for basic and extended chaining mode DMA operations. + * + * A Link Descriptor points to a single DMA buffer. Each link descriptor + * must be aligned on a 32-byte boundary. + */ +struct fsl_dma_link_descriptor { + __be32 source_attr; /* Programmed into SATR register */ + __be32 source_addr; /* Programmed into SAR register */ + __be32 dest_attr; /* Programmed into DATR register */ + __be32 dest_addr; /* Programmed into DAR register */ + __be64 next; /* Address of next link descriptor */ + __be32 count; /* Byte count */ + u8 res[4]; /* Reserved */ +} __attribute__ ((aligned(32), packed)); + +/* DMA information needed to create a snd_soc_cpu_dai object + * + * ssi_stx_phys: bus address of SSI STX register to use + * ssi_srx_phys: bus address of SSI SRX register to use + * dma[0]: points to the DMA channel to use for playback + * dma[1]: points to the DMA channel to use for capture + * dma_irq[0]: IRQ of the DMA channel to use for playback + * dma_irq[1]: IRQ of the DMA channel to use for capture + */ +struct fsl_dma_info { + dma_addr_t ssi_stx_phys; + dma_addr_t ssi_srx_phys; + struct ccsr_dma_channel __iomem *dma_channel[2]; + unsigned int dma_irq[2]; +}; + +extern struct snd_soc_platform fsl_soc_platform; + +int fsl_dma_configure(struct fsl_dma_info *dma_info); + +#endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c new file mode 100644 index 0000000..145ad13 --- /dev/null +++ b/sound/soc/fsl/fsl_ssi.c @@ -0,0 +1,644 @@ +/* + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * + * Author: Timur Tabi + * + * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include + +#include "fsl_ssi.h" + +/** + * FSLSSI_I2S_RATES: sample rates supported by the I2S + * + * This driver currently only supports the SSI running in I2S slave mode, + * which means the codec determines the sample rate. Therefore, we tell + * ALSA that we support all rates and let the codec driver decide what rates + * are really supported. + */ +#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS) + +/** + * FSLSSI_I2S_FORMATS: audio formats supported by the SSI + * + * This driver currently only supports the SSI running in I2S slave mode. + * + * The SSI has a limitation in that the samples must be in the same byte + * order as the host CPU. This is because when multiple bytes are written + * to the STX register, the bytes and bits must be written in the same + * order. The STX is a shift register, so all the bits need to be aligned + * (bit-endianness must match byte-endianness). Processors typically write + * the bits within a byte in the same order that the bytes of a word are + * written in. So if the host CPU is big-endian, then only big-endian + * samples will be written to STX properly. + */ +#ifdef __BIG_ENDIAN +#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S24_BE) +#else +#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) +#endif + +/** + * fsl_ssi_private: per-SSI private data + * + * @name: short name for this device ("SSI0", "SSI1", etc) + * @ssi: pointer to the SSI's registers + * @ssi_phys: physical address of the SSI registers + * @irq: IRQ of this SSI + * @dev: struct device pointer + * @playback: the number of playback streams opened + * @capture: the number of capture streams opened + * @cpu_dai: the CPU DAI for this device + * @dev_attr: the sysfs device attribute structure + * @stats: SSI statistics + */ +struct fsl_ssi_private { + char name[8]; + struct ccsr_ssi __iomem *ssi; + dma_addr_t ssi_phys; + unsigned int irq; + struct device *dev; + unsigned int playback; + unsigned int capture; + struct snd_soc_cpu_dai cpu_dai; + struct device_attribute dev_attr; + + struct { + unsigned int rfrc; + unsigned int tfrc; + unsigned int cmdau; + unsigned int cmddu; + unsigned int rxt; + unsigned int rdr1; + unsigned int rdr0; + unsigned int tde1; + unsigned int tde0; + unsigned int roe1; + unsigned int roe0; + unsigned int tue1; + unsigned int tue0; + unsigned int tfs; + unsigned int rfs; + unsigned int tls; + unsigned int rls; + unsigned int rff1; + unsigned int rff0; + unsigned int tfe1; + unsigned int tfe0; + } stats; +}; + +/** + * fsl_ssi_isr: SSI interrupt handler + * + * Although it's possible to use the interrupt handler to send and receive + * data to/from the SSI, we use the DMA instead. Programming is more + * complicated, but the performance is much better. + * + * This interrupt handler is used only to gather statistics. + * + * @irq: IRQ of the SSI device + * @dev_id: pointer to the ssi_private structure for this SSI device + */ +static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) +{ + struct fsl_ssi_private *ssi_private = dev_id; + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + irqreturn_t ret = IRQ_NONE; + __be32 sisr; + __be32 sisr2 = 0; + + /* We got an interrupt, so read the status register to see what we + were interrupted for. We mask it with the Interrupt Enable register + so that we only check for events that we're interested in. + */ + sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier); + + if (sisr & CCSR_SSI_SISR_RFRC) { + ssi_private->stats.rfrc++; + sisr2 |= CCSR_SSI_SISR_RFRC; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFRC) { + ssi_private->stats.tfrc++; + sisr2 |= CCSR_SSI_SISR_TFRC; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_CMDAU) { + ssi_private->stats.cmdau++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_CMDDU) { + ssi_private->stats.cmddu++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RXT) { + ssi_private->stats.rxt++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RDR1) { + ssi_private->stats.rdr1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RDR0) { + ssi_private->stats.rdr0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TDE1) { + ssi_private->stats.tde1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TDE0) { + ssi_private->stats.tde0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_ROE1) { + ssi_private->stats.roe1++; + sisr2 |= CCSR_SSI_SISR_ROE1; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_ROE0) { + ssi_private->stats.roe0++; + sisr2 |= CCSR_SSI_SISR_ROE0; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TUE1) { + ssi_private->stats.tue1++; + sisr2 |= CCSR_SSI_SISR_TUE1; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TUE0) { + ssi_private->stats.tue0++; + sisr2 |= CCSR_SSI_SISR_TUE0; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFS) { + ssi_private->stats.tfs++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFS) { + ssi_private->stats.rfs++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TLS) { + ssi_private->stats.tls++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RLS) { + ssi_private->stats.rls++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFF1) { + ssi_private->stats.rff1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFF0) { + ssi_private->stats.rff0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFE1) { + ssi_private->stats.tfe1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFE0) { + ssi_private->stats.tfe0++; + ret = IRQ_HANDLED; + } + + /* Clear the bits that we set */ + if (sisr2) + out_be32(&ssi->sisr, sisr2); + + return ret; +} + +/** + * fsl_ssi_startup: create a new substream + * + * This is the first function called when a stream is opened. + * + * If this is the first stream open, then grab the IRQ and program most of + * the SSI registers. + */ +static int fsl_ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + /* + * If this is the first stream opened, then request the IRQ + * and initialize the SSI registers. + */ + if (!ssi_private->playback && !ssi_private->capture) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int ret; + + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, + ssi_private->name, ssi_private); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not claim irq %u\n", ssi_private->irq); + return ret; + } + + /* + * Section 16.5 of the MPC8610 reference manual says that the + * SSI needs to be disabled before updating the registers we set + * here. + */ + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + /* + * Program the SSI into I2S Slave Non-Network Synchronous mode. + * Also enable the transmit and receive FIFO. + * + * FIXME: Little-endian samples require a different shift dir + */ + clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, + CCSR_SSI_SCR_TFR_CLK_DIS | + CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); + + out_be32(&ssi->stcr, + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | + CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | + CCSR_SSI_STCR_TSCKP); + + out_be32(&ssi->srcr, + CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | + CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | + CCSR_SSI_SRCR_RSCKP); + + /* + * The DC and PM bits are only used if the SSI is the clock + * master. + */ + + /* 4. Enable the interrupts and DMA requests */ + out_be32(&ssi->sier, + CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | + CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | + CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | + CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN); + + /* + * Set the watermark for transmit FIFI 0 and receive FIFO 0. We + * don't use FIFO 1. Since the SSI only supports stereo, the + * watermark should never be an odd number. + */ + out_be32(&ssi->sfcsr, + CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2)); + + /* + * We keep the SSI disabled because if we enable it, then the + * DMA controller will start. It's not supposed to start until + * the SCR.TE (or SCR.RE) bit is set, but it does anyway. The + * DMA controller will transfer one "BWC" of data (i.e. the + * amount of data that the MR.BWC bits are set to). The reason + * this is bad is because at this point, the PCM driver has not + * finished initializing the DMA controller. + */ + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssi_private->playback++; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ssi_private->capture++; + + return 0; +} + +/** + * fsl_ssi_prepare: prepare the SSI. + * + * Most of the SSI registers have been programmed in the startup function, + * but the word length must be programmed here. Unfortunately, programming + * the SxCCR.WL bits requires the SSI to be temporarily disabled. This can + * cause a problem with supporting simultaneous playback and capture. If + * the SSI is already playing a stream, then that stream may be temporarily + * stopped when you start capture. + * + * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the + * clock master. + */ +static int fsl_ssi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 wl; + + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + + setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + return 0; +} + +/** + * fsl_ssi_trigger: start and stop the DMA transfer. + * + * This function is called by ALSA to start, stop, pause, and resume the DMA + * transfer of data. + * + * The DMA channel is in external master start and pause mode, which + * means the SSI completely controls the flow of data. + */ +static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + } else { + setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + + /* + * I think we need this delay to allow time for the SSI + * to put data into its FIFO. Without it, ALSA starts + * to complain about overruns. + */ + msleep(1); + } + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); + else + clrbits32(&ssi->scr, CCSR_SSI_SCR_RE); + break; + + default: + return -EINVAL; + } + + return 0; +} + +/** + * fsl_ssi_shutdown: shutdown the SSI + * + * Shutdown the SSI if there are no other substreams open. + */ +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssi_private->playback--; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ssi_private->capture--; + + /* + * If this is the last active substream, disable the SSI and release + * the IRQ. + */ + if (!ssi_private->playback && !ssi_private->capture) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + free_irq(ssi_private->irq, ssi_private); + } +} + +/** + * fsl_ssi_set_sysclk: set the clock frequency and direction + * + * This function is called by the machine driver to tell us what the clock + * frequency and direction are. + * + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN), + * and we don't care about the frequency. Return an error if the direction + * is not SND_SOC_CLOCK_IN. + * + * @clk_id: reserved, should be zero + * @freq: the frequency of the given clock ID, currently ignored + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) + */ +static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + + return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; +} + +/** + * fsl_ssi_set_fmt: set the serial format. + * + * This function is called by the machine driver to tell us what serial + * format to use. + * + * Currently, we only support I2S mode. Return an error if the format is + * not SND_SOC_DAIFMT_I2S. + * + * @format: one of SND_SOC_DAIFMT_xxx + */ +static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +{ + return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; +} + +/** + * fsl_ssi_dai_template: template CPU DAI for the SSI + */ +static struct snd_soc_cpu_dai fsl_ssi_dai_template = { + .playback = { + /* The SSI does not support monaural audio. */ + .channels_min = 2, + .channels_max = 2, + .rates = FSLSSI_I2S_RATES, + .formats = FSLSSI_I2S_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = FSLSSI_I2S_RATES, + .formats = FSLSSI_I2S_FORMATS, + }, + .ops = { + .startup = fsl_ssi_startup, + .prepare = fsl_ssi_prepare, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_trigger, + }, + .dai_ops = { + .set_sysclk = fsl_ssi_set_sysclk, + .set_fmt = fsl_ssi_set_fmt, + }, +}; + +/** + * fsl_sysfs_ssi_show: display SSI statistics + * + * Display the statistics for the current SSI device. + */ +static ssize_t fsl_sysfs_ssi_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct fsl_ssi_private *ssi_private = + container_of(attr, struct fsl_ssi_private, dev_attr); + ssize_t length; + + length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc); + length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc); + length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau); + length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu); + length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt); + length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1); + length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0); + length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1); + length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0); + length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1); + length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0); + length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1); + length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0); + length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs); + length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs); + length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls); + length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls); + length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1); + length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0); + length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1); + length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0); + + return length; +} + +/** + * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure + * + * This function is called by the machine driver to create a snd_soc_cpu_dai + * structure. The function creates an ssi_private object, which contains + * the snd_soc_cpu_dai. It also creates the sysfs statistics device. + */ +struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +{ + struct snd_soc_cpu_dai *fsl_ssi_dai; + struct fsl_ssi_private *ssi_private; + int ret = 0; + struct device_attribute *dev_attr; + + ssi_private = kzalloc(sizeof(struct fsl_ssi_private), GFP_KERNEL); + if (!ssi_private) { + dev_err(ssi_info->dev, "could not allocate DAI object\n"); + return NULL; + } + memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, + sizeof(struct snd_soc_cpu_dai)); + + fsl_ssi_dai = &ssi_private->cpu_dai; + dev_attr = &ssi_private->dev_attr; + + sprintf(ssi_private->name, "ssi%u", (u8) ssi_info->id); + ssi_private->ssi = ssi_info->ssi; + ssi_private->ssi_phys = ssi_info->ssi_phys; + ssi_private->irq = ssi_info->irq; + ssi_private->dev = ssi_info->dev; + + ssi_private->dev->driver_data = fsl_ssi_dai; + + /* Initialize the the device_attribute structure */ + dev_attr->attr.name = "ssi-stats"; + dev_attr->attr.mode = S_IRUGO; + dev_attr->show = fsl_sysfs_ssi_show; + + ret = device_create_file(ssi_private->dev, dev_attr); + if (ret) { + dev_err(ssi_info->dev, "could not create sysfs %s file\n", + ssi_private->dev_attr.attr.name); + kfree(fsl_ssi_dai); + return NULL; + } + + fsl_ssi_dai->private_data = ssi_private; + fsl_ssi_dai->name = ssi_private->name; + fsl_ssi_dai->id = ssi_info->id; + + return fsl_ssi_dai; +} +EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); + +/** + * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object + * + * This function undoes the operations of fsl_ssi_create_dai() + */ +void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +{ + struct fsl_ssi_private *ssi_private = + container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); + + device_remove_file(ssi_private->dev, &ssi_private->dev_attr); + + kfree(ssi_private); +} +EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai); + +MODULE_AUTHOR("Timur Tabi "); +MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h new file mode 100644 index 0000000..c5ce88e --- /dev/null +++ b/sound/soc/fsl/fsl_ssi.h @@ -0,0 +1,224 @@ +/* + * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 SoC + * + * Author: Timur Tabi + * + * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. + */ + +#ifndef _MPC8610_I2S_H +#define _MPC8610_I2S_H + +/* SSI Register Map */ +struct ccsr_ssi { + __be32 stx0; /* 0x.0000 - SSI Transmit Data Register 0 */ + __be32 stx1; /* 0x.0004 - SSI Transmit Data Register 1 */ + __be32 srx0; /* 0x.0008 - SSI Receive Data Register 0 */ + __be32 srx1; /* 0x.000C - SSI Receive Data Register 1 */ + __be32 scr; /* 0x.0010 - SSI Control Register */ + __be32 sisr; /* 0x.0014 - SSI Interrupt Status Register Mixed */ + __be32 sier; /* 0x.0018 - SSI Interrupt Enable Register */ + __be32 stcr; /* 0x.001C - SSI Transmit Configuration Register */ + __be32 srcr; /* 0x.0020 - SSI Receive Configuration Register */ + __be32 stccr; /* 0x.0024 - SSI Transmit Clock Control Register */ + __be32 srccr; /* 0x.0028 - SSI Receive Clock Control Register */ + __be32 sfcsr; /* 0x.002C - SSI FIFO Control/Status Register */ + __be32 str; /* 0x.0030 - SSI Test Register */ + __be32 sor; /* 0x.0034 - SSI Option Register */ + __be32 sacnt; /* 0x.0038 - SSI AC97 Control Register */ + __be32 sacadd; /* 0x.003C - SSI AC97 Command Address Register */ + __be32 sacdat; /* 0x.0040 - SSI AC97 Command Data Register */ + __be32 satag; /* 0x.0044 - SSI AC97 Tag Register */ + __be32 stmsk; /* 0x.0048 - SSI Transmit Time Slot Mask Register */ + __be32 srmsk; /* 0x.004C - SSI Receive Time Slot Mask Register */ + __be32 saccst; /* 0x.0050 - SSI AC97 Channel Status Register */ + __be32 saccen; /* 0x.0054 - SSI AC97 Channel Enable Register */ + __be32 saccdis; /* 0x.0058 - SSI AC97 Channel Disable Register */ +}; + +#define CCSR_SSI_SCR_RFR_CLK_DIS 0x00000800 +#define CCSR_SSI_SCR_TFR_CLK_DIS 0x00000400 +#define CCSR_SSI_SCR_TCH_EN 0x00000100 +#define CCSR_SSI_SCR_SYS_CLK_EN 0x00000080 +#define CCSR_SSI_SCR_I2S_MODE_MASK 0x00000060 +#define CCSR_SSI_SCR_I2S_MODE_NORMAL 0x00000000 +#define CCSR_SSI_SCR_I2S_MODE_MASTER 0x00000020 +#define CCSR_SSI_SCR_I2S_MODE_SLAVE 0x00000040 +#define CCSR_SSI_SCR_SYN 0x00000010 +#define CCSR_SSI_SCR_NET 0x00000008 +#define CCSR_SSI_SCR_RE 0x00000004 +#define CCSR_SSI_SCR_TE 0x00000002 +#define CCSR_SSI_SCR_SSIEN 0x00000001 + +#define CCSR_SSI_SISR_RFRC 0x01000000 +#define CCSR_SSI_SISR_TFRC 0x00800000 +#define CCSR_SSI_SISR_CMDAU 0x00040000 +#define CCSR_SSI_SISR_CMDDU 0x00020000 +#define CCSR_SSI_SISR_RXT 0x00010000 +#define CCSR_SSI_SISR_RDR1 0x00008000 +#define CCSR_SSI_SISR_RDR0 0x00004000 +#define CCSR_SSI_SISR_TDE1 0x00002000 +#define CCSR_SSI_SISR_TDE0 0x00001000 +#define CCSR_SSI_SISR_ROE1 0x00000800 +#define CCSR_SSI_SISR_ROE0 0x00000400 +#define CCSR_SSI_SISR_TUE1 0x00000200 +#define CCSR_SSI_SISR_TUE0 0x00000100 +#define CCSR_SSI_SISR_TFS 0x00000080 +#define CCSR_SSI_SISR_RFS 0x00000040 +#define CCSR_SSI_SISR_TLS 0x00000020 +#define CCSR_SSI_SISR_RLS 0x00000010 +#define CCSR_SSI_SISR_RFF1 0x00000008 +#define CCSR_SSI_SISR_RFF0 0x00000004 +#define CCSR_SSI_SISR_TFE1 0x00000002 +#define CCSR_SSI_SISR_TFE0 0x00000001 + +#define CCSR_SSI_SIER_RFRC_EN 0x01000000 +#define CCSR_SSI_SIER_TFRC_EN 0x00800000 +#define CCSR_SSI_SIER_RDMAE 0x00400000 +#define CCSR_SSI_SIER_RIE 0x00200000 +#define CCSR_SSI_SIER_TDMAE 0x00100000 +#define CCSR_SSI_SIER_TIE 0x00080000 +#define CCSR_SSI_SIER_CMDAU_EN 0x00040000 +#define CCSR_SSI_SIER_CMDDU_EN 0x00020000 +#define CCSR_SSI_SIER_RXT_EN 0x00010000 +#define CCSR_SSI_SIER_RDR1_EN 0x00008000 +#define CCSR_SSI_SIER_RDR0_EN 0x00004000 +#define CCSR_SSI_SIER_TDE1_EN 0x00002000 +#define CCSR_SSI_SIER_TDE0_EN 0x00001000 +#define CCSR_SSI_SIER_ROE1_EN 0x00000800 +#define CCSR_SSI_SIER_ROE0_EN 0x00000400 +#define CCSR_SSI_SIER_TUE1_EN 0x00000200 +#define CCSR_SSI_SIER_TUE0_EN 0x00000100 +#define CCSR_SSI_SIER_TFS_EN 0x00000080 +#define CCSR_SSI_SIER_RFS_EN 0x00000040 +#define CCSR_SSI_SIER_TLS_EN 0x00000020 +#define CCSR_SSI_SIER_RLS_EN 0x00000010 +#define CCSR_SSI_SIER_RFF1_EN 0x00000008 +#define CCSR_SSI_SIER_RFF0_EN 0x00000004 +#define CCSR_SSI_SIER_TFE1_EN 0x00000002 +#define CCSR_SSI_SIER_TFE0_EN 0x00000001 + +#define CCSR_SSI_STCR_TXBIT0 0x00000200 +#define CCSR_SSI_STCR_TFEN1 0x00000100 +#define CCSR_SSI_STCR_TFEN0 0x00000080 +#define CCSR_SSI_STCR_TFDIR 0x00000040 +#define CCSR_SSI_STCR_TXDIR 0x00000020 +#define CCSR_SSI_STCR_TSHFD 0x00000010 +#define CCSR_SSI_STCR_TSCKP 0x00000008 +#define CCSR_SSI_STCR_TFSI 0x00000004 +#define CCSR_SSI_STCR_TFSL 0x00000002 +#define CCSR_SSI_STCR_TEFS 0x00000001 + +#define CCSR_SSI_SRCR_RXEXT 0x00000400 +#define CCSR_SSI_SRCR_RXBIT0 0x00000200 +#define CCSR_SSI_SRCR_RFEN1 0x00000100 +#define CCSR_SSI_SRCR_RFEN0 0x00000080 +#define CCSR_SSI_SRCR_RFDIR 0x00000040 +#define CCSR_SSI_SRCR_RXDIR 0x00000020 +#define CCSR_SSI_SRCR_RSHFD 0x00000010 +#define CCSR_SSI_SRCR_RSCKP 0x00000008 +#define CCSR_SSI_SRCR_RFSI 0x00000004 +#define CCSR_SSI_SRCR_RFSL 0x00000002 +#define CCSR_SSI_SRCR_REFS 0x00000001 + +/* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR 0x00020000 +#define CCSR_SSI_SxCCR_WL_SHIFT 13 +#define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 +#define CCSR_SSI_SxCCR_WL(x) \ + (((((x) / 2) - 1) << CCSR_SSI_SxCCR_WL_SHIFT) & CCSR_SSI_SxCCR_WL_MASK) +#define CCSR_SSI_SxCCR_DC_SHIFT 8 +#define CCSR_SSI_SxCCR_DC_MASK 0x00001F00 +#define CCSR_SSI_SxCCR_DC(x) \ + ((((x) - 1) << CCSR_SSI_SxCCR_DC_SHIFT) & CCSR_SSI_SxCCR_DC_MASK) +#define CCSR_SSI_SxCCR_PM_SHIFT 0 +#define CCSR_SSI_SxCCR_PM_MASK 0x000000FF +#define CCSR_SSI_SxCCR_PM(x) \ + ((((x) - 1) << CCSR_SSI_SxCCR_PM_SHIFT) & CCSR_SSI_SxCCR_PM_MASK) + +/* + * The xFCNT bits are read-only, and the xFWM bits are read/write. Use the + * CCSR_SSI_SFCSR_xFCNTy() macros to read the FIFO counters, and use the + * CCSR_SSI_SFCSR_xFWMy() macros to set the watermarks. + */ +#define CCSR_SSI_SFCSR_RFCNT1_SHIFT 28 +#define CCSR_SSI_SFCSR_RFCNT1_MASK 0xF0000000 +#define CCSR_SSI_SFCSR_RFCNT1(x) \ + (((x) & CCSR_SSI_SFCSR_RFCNT1_MASK) >> CCSR_SSI_SFCSR_RFCNT1_SHIFT) +#define CCSR_SSI_SFCSR_TFCNT1_SHIFT 24 +#define CCSR_SSI_SFCSR_TFCNT1_MASK 0x0F000000 +#define CCSR_SSI_SFCSR_TFCNT1(x) \ + (((x) & CCSR_SSI_SFCSR_TFCNT1_MASK) >> CCSR_SSI_SFCSR_TFCNT1_SHIFT) +#define CCSR_SSI_SFCSR_RFWM1_SHIFT 20 +#define CCSR_SSI_SFCSR_RFWM1_MASK 0x00F00000 +#define CCSR_SSI_SFCSR_RFWM1(x) \ + (((x) << CCSR_SSI_SFCSR_RFWM1_SHIFT) & CCSR_SSI_SFCSR_RFWM1_MASK) +#define CCSR_SSI_SFCSR_TFWM1_SHIFT 16 +#define CCSR_SSI_SFCSR_TFWM1_MASK 0x000F0000 +#define CCSR_SSI_SFCSR_TFWM1(x) \ + (((x) << CCSR_SSI_SFCSR_TFWM1_SHIFT) & CCSR_SSI_SFCSR_TFWM1_MASK) +#define CCSR_SSI_SFCSR_RFCNT0_SHIFT 12 +#define CCSR_SSI_SFCSR_RFCNT0_MASK 0x0000F000 +#define CCSR_SSI_SFCSR_RFCNT0(x) \ + (((x) & CCSR_SSI_SFCSR_RFCNT0_MASK) >> CCSR_SSI_SFCSR_RFCNT0_SHIFT) +#define CCSR_SSI_SFCSR_TFCNT0_SHIFT 8 +#define CCSR_SSI_SFCSR_TFCNT0_MASK 0x00000F00 +#define CCSR_SSI_SFCSR_TFCNT0(x) \ + (((x) & CCSR_SSI_SFCSR_TFCNT0_MASK) >> CCSR_SSI_SFCSR_TFCNT0_SHIFT) +#define CCSR_SSI_SFCSR_RFWM0_SHIFT 4 +#define CCSR_SSI_SFCSR_RFWM0_MASK 0x000000F0 +#define CCSR_SSI_SFCSR_RFWM0(x) \ + (((x) << CCSR_SSI_SFCSR_RFWM0_SHIFT) & CCSR_SSI_SFCSR_RFWM0_MASK) +#define CCSR_SSI_SFCSR_TFWM0_SHIFT 0 +#define CCSR_SSI_SFCSR_TFWM0_MASK 0x0000000F +#define CCSR_SSI_SFCSR_TFWM0(x) \ + (((x) << CCSR_SSI_SFCSR_TFWM0_SHIFT) & CCSR_SSI_SFCSR_TFWM0_MASK) + +#define CCSR_SSI_STR_TEST 0x00008000 +#define CCSR_SSI_STR_RCK2TCK 0x00004000 +#define CCSR_SSI_STR_RFS2TFS 0x00002000 +#define CCSR_SSI_STR_RXSTATE(x) (((x) >> 8) & 0x1F) +#define CCSR_SSI_STR_TXD2RXD 0x00000080 +#define CCSR_SSI_STR_TCK2RCK 0x00000040 +#define CCSR_SSI_STR_TFS2RFS 0x00000020 +#define CCSR_SSI_STR_TXSTATE(x) ((x) & 0x1F) + +#define CCSR_SSI_SOR_CLKOFF 0x00000040 +#define CCSR_SSI_SOR_RX_CLR 0x00000020 +#define CCSR_SSI_SOR_TX_CLR 0x00000010 +#define CCSR_SSI_SOR_INIT 0x00000008 +#define CCSR_SSI_SOR_WAIT_SHIFT 1 +#define CCSR_SSI_SOR_WAIT_MASK 0x00000006 +#define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT) +#define CCSR_SSI_SOR_SYNRST 0x00000001 + +/* Instantiation data for an SSI interface + * + * This structure contains all the information that the the SSI driver needs + * to instantiate an SSI interface with ALSA. The machine driver should + * create this structure, fill it in, call fsl_ssi_create_dai(), and then + * delete the structure. + * + * id: which SSI this is (0, 1, etc. ) + * ssi: pointer to the SSI's registers + * ssi_phys: physical address of the SSI registers + * irq: IRQ of this SSI + * dev: struct device, used to create the sysfs statistics file +*/ +struct fsl_ssi_info { + unsigned int id; + struct ccsr_ssi __iomem *ssi; + dma_addr_t ssi_phys; + unsigned int irq; + struct device *dev; +}; + +struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); +void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); + +#endif + diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c new file mode 100644 index 0000000..f26c4b2 --- /dev/null +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -0,0 +1,631 @@ +/** + * Freescale MPC8610HPCD ALSA SoC Fabric driver + * + * Author: Timur Tabi + * + * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. + */ + +#include +#include +#include +#include +#include +#include + +#include "../codecs/cs4270.h" +#include "fsl_dma.h" +#include "fsl_ssi.h" + +/** + * mpc8610_hpcd_data: fabric-specific ASoC device data + * + * This structure contains data for a single sound platform device on an + * MPC8610 HPCD. Some of the data is taken from the device tree. + */ +struct mpc8610_hpcd_data { + struct snd_soc_device sound_devdata; + struct snd_soc_dai_link dai; + struct snd_soc_machine machine; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + struct ccsr_guts __iomem *guts; + struct ccsr_ssi __iomem *ssi; + unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ + unsigned int ssi_irq; + unsigned int dma_id; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_irq[2]; + struct ccsr_dma_channel __iomem *dma[2]; + unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ +}; + +/** + * mpc8610_hpcd_machine_probe: initalize the board + * + * This function is called when platform_device_add() is called. It is used + * to initialize the board-specific hardware. + * + * Here we program the DMACR and PMUXCR registers. + */ +static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) +{ + struct mpc8610_hpcd_data *machine_data = + sound_device->dev.platform_data; + + /* Program the signal routing between the SSI and the DMA */ + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); + + guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, + machine_data->dma_channel_id[0], 0); + guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, + machine_data->dma_channel_id[1], 0); + + guts_set_pmuxcr_dma(machine_data->guts, 1, 0, 0); + guts_set_pmuxcr_dma(machine_data->guts, 1, 3, 0); + guts_set_pmuxcr_dma(machine_data->guts, 0, 3, 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); + break; + case 1: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); + break; + } + + return 0; +} + +/** + * mpc8610_hpcd_startup: program the board with various hardware parameters + * + * This function takes board-specific information, like clock frequencies + * and serial data formats, and passes that information to the codec and + * transport drivers. + */ +static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct mpc8610_hpcd_data *machine_data = + rtd->socdev->dev->platform_data; + int ret = 0; + + /* Tell the CPU driver what the serial protocol is. */ + if (cpu_dai->dai_ops.set_fmt) { + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; + } + } + + /* Tell the codec driver what the serial protocol is. */ + if (codec_dai->dai_ops.set_fmt) { + ret = codec_dai->dai_ops.set_fmt(codec_dai, + machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; + } + } + + /* + * Tell the CPU driver what the clock frequency is, and whether it's a + * slave or master. + */ + if (cpu_dai->dai_ops.set_sysclk) { + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0, + machine_data->clk_frequency, + machine_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; + } + } + + /* + * Tell the codec driver what the MCLK frequency is, and whether it's + * a slave or master. + */ + if (codec_dai->dai_ops.set_sysclk) { + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; + } + } + + return 0; +} + +/** + * mpc8610_hpcd_machine_remove: Remove the sound device + * + * This function is called to remove the sound device for one SSI. We + * de-program the DMACR and PMUXCR register. + */ +int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) +{ + struct mpc8610_hpcd_data *machine_data = + sound_device->dev.platform_data; + + /* Restore the signal routing */ + + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[0], 0); + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[1], 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + case 1: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + } + + return 0; +} + +/** + * mpc8610_hpcd_ops: ASoC fabric driver operations + */ +static struct snd_soc_ops mpc8610_hpcd_ops = { + .startup = mpc8610_hpcd_startup, +}; + +/** + * mpc8610_hpcd_machine: ASoC machine data + */ +static struct snd_soc_machine mpc8610_hpcd_machine = { + .probe = mpc8610_hpcd_machine_probe, + .remove = mpc8610_hpcd_machine_remove, + .name = "MPC8610 HPCD", + .num_links = 1, +}; + +/** + * mpc8610_hpcd_probe: OF probe function for the fabric driver + * + * This function gets called when an SSI node is found in the device tree. + * + * Although this is a fabric driver, the SSI node is the "master" node with + * respect to audio hardware connections. Therefore, we create a new ASoC + * device for each new SSI node that has a codec attached. + * + * FIXME: Currently, we only support one DMA controller, so if there are + * multiple SSI nodes with codecs, only the first will be supported. + * + * FIXME: Even if we did support multiple DMA controllers, we have no + * mechanism for assigning DMA controllers and channels to the individual + * SSI devices. We also probably aren't compatible with the generic Elo DMA + * device driver. + */ +static int mpc8610_hpcd_probe(struct of_device *ofdev, + const struct of_device_id *match) +{ + struct device_node *np = ofdev->node; + struct device_node *codec_np = NULL; + struct device_node *guts_np = NULL; + struct device_node *dma_np = NULL; + struct device_node *dma_channel_np = NULL; + const phandle *codec_ph; + const char *sprop; + const u32 *iprop; + struct resource res; + struct platform_device *sound_device = NULL; + struct mpc8610_hpcd_data *machine_data; + struct fsl_ssi_info ssi_info; + struct fsl_dma_info dma_info; + int ret = -ENODEV; + + machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); + if (!machine_data) + return -ENOMEM; + + memset(&ssi_info, 0, sizeof(ssi_info)); + memset(&dma_info, 0, sizeof(dma_info)); + + ssi_info.dev = &ofdev->dev; + + /* + * We are only interested in SSIs with a codec phandle in them, so let's + * make sure this SSI has one. + */ + codec_ph = of_get_property(np, "codec-handle", NULL); + if (!codec_ph) + goto error; + + codec_np = of_find_node_by_phandle(*codec_ph); + if (!codec_np) + goto error; + + /* The MPC8610 HPCD only knows about the CS4270 codec, so reject + anything else. */ + if (!of_device_is_compatible(codec_np, "cirrus,cs4270")) + goto error; + + /* Get the device ID */ + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) { + dev_err(&ofdev->dev, "cell-index property not found\n"); + ret = -EINVAL; + goto error; + } + machine_data->ssi_id = *iprop; + ssi_info.id = *iprop; + + /* Get the serial format and clock direction. */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop) { + dev_err(&ofdev->dev, "fsl,mode property not found\n"); + ret = -EINVAL; + goto error; + } + + if (strcasecmp(sprop, "i2s-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + + /* + * In i2s-slave mode, the codec has its own clock source, so we + * need to get the frequency from the device tree and pass it to + * the codec driver. + */ + iprop = of_get_property(codec_np, "clock-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&ofdev->dev, "codec bus-frequency property " + "is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + machine_data->clk_frequency = *iprop; + } else if (strcasecmp(sprop, "i2s-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "lj-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "lj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "rj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "rj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "ac97-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "ac97-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else { + dev_err(&ofdev->dev, + "unrecognized fsl,mode property \"%s\"\n", sprop); + ret = -EINVAL; + goto error; + } + + if (!machine_data->clk_frequency) { + dev_err(&ofdev->dev, "unknown clock frequency\n"); + ret = -EINVAL; + goto error; + } + + /* Read the SSI information from the device tree */ + ret = of_address_to_resource(np, 0, &res); + if (ret) { + dev_err(&ofdev->dev, "could not obtain SSI address\n"); + goto error; + } + if (!res.start) { + dev_err(&ofdev->dev, "invalid SSI address\n"); + goto error; + } + ssi_info.ssi_phys = res.start; + + machine_data->ssi = ioremap(ssi_info.ssi_phys, sizeof(struct ccsr_ssi)); + if (!machine_data->ssi) { + dev_err(&ofdev->dev, "could not map SSI address %x\n", + ssi_info.ssi_phys); + ret = -EINVAL; + goto error; + } + ssi_info.ssi = machine_data->ssi; + + + /* Get the IRQ of the SSI */ + machine_data->ssi_irq = irq_of_parse_and_map(np, 0); + if (!machine_data->ssi_irq) { + dev_err(&ofdev->dev, "could not get SSI IRQ\n"); + ret = -EINVAL; + goto error; + } + ssi_info.irq = machine_data->ssi_irq; + + + /* Map the global utilities registers. */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); + if (!guts_np) { + dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); + ret = -EINVAL; + goto error; + } + machine_data->guts = of_iomap(guts_np, 0); + of_node_put(guts_np); + if (!machine_data->guts) { + dev_err(&ofdev->dev, "could not map GUTS\n"); + ret = -EINVAL; + goto error; + } + + /* Find the DMA channels to use. For now, we always use the first DMA + controller. */ + for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") { + iprop = of_get_property(dma_np, "cell-index", NULL); + if (iprop && (*iprop == 0)) { + of_node_put(dma_np); + break; + } + } + if (!dma_np) { + dev_err(&ofdev->dev, "could not find DMA node\n"); + ret = -EINVAL; + goto error; + } + machine_data->dma_id = *iprop; + + /* + * Find the DMA channels to use. For now, we always use DMA channel 0 + * for playback, and DMA channel 1 for capture. + */ + while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) { + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + /* Is it DMA channel 0? */ + if (iprop && (*iprop == 0)) { + /* dma_channel[0] and dma_irq[0] are for playback */ + dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0); + dma_info.dma_irq[0] = + irq_of_parse_and_map(dma_channel_np, 0); + machine_data->dma_channel_id[0] = *iprop; + continue; + } + if (iprop && (*iprop == 1)) { + /* dma_channel[1] and dma_irq[1] are for capture */ + dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0); + dma_info.dma_irq[1] = + irq_of_parse_and_map(dma_channel_np, 0); + machine_data->dma_channel_id[1] = *iprop; + continue; + } + } + if (!dma_info.dma_channel[0] || !dma_info.dma_channel[1] || + !dma_info.dma_irq[0] || !dma_info.dma_irq[1]) { + dev_err(&ofdev->dev, "could not find DMA channels\n"); + ret = -EINVAL; + goto error; + } + + dma_info.ssi_stx_phys = ssi_info.ssi_phys + + offsetof(struct ccsr_ssi, stx0); + dma_info.ssi_srx_phys = ssi_info.ssi_phys + + offsetof(struct ccsr_ssi, srx0); + + /* We have the DMA information, so tell the DMA driver what it is */ + if (!fsl_dma_configure(&dma_info)) { + dev_err(&ofdev->dev, "could not instantiate DMA device\n"); + ret = -EBUSY; + goto error; + } + + /* + * Initialize our DAI data structure. We should probably get this + * information from the device tree. + */ + machine_data->dai.name = "CS4270"; + machine_data->dai.stream_name = "CS4270"; + + machine_data->dai.cpu_dai = fsl_ssi_create_dai(&ssi_info); + machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */ + machine_data->dai.ops = &mpc8610_hpcd_ops; + + mpc8610_hpcd_machine.dai_link = &machine_data->dai; + + /* Allocate a new audio platform device structure */ + sound_device = platform_device_alloc("soc-audio", -1); + if (!sound_device) { + dev_err(&ofdev->dev, "platform device allocation failed\n"); + ret = -ENOMEM; + goto error; + } + + machine_data->sound_devdata.machine = &mpc8610_hpcd_machine; + machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; + machine_data->sound_devdata.platform = &fsl_soc_platform; + + sound_device->dev.platform_data = machine_data; + + + /* Set the platform device and ASoC device to point to each other */ + platform_set_drvdata(sound_device, &machine_data->sound_devdata); + + machine_data->sound_devdata.dev = &sound_device->dev; + + + /* Tell ASoC to probe us. This will call mpc8610_hpcd_machine.probe(), + if it exists. */ + ret = platform_device_add(sound_device); + + if (ret) { + dev_err(&ofdev->dev, "platform device add failed\n"); + goto error; + } + + dev_set_drvdata(&ofdev->dev, sound_device); + + return 0; + +error: + of_node_put(codec_np); + of_node_put(guts_np); + of_node_put(dma_np); + of_node_put(dma_channel_np); + + if (sound_device) + platform_device_unregister(sound_device); + + if (machine_data->dai.cpu_dai) + fsl_ssi_destroy_dai(machine_data->dai.cpu_dai); + + if (ssi_info.ssi) + iounmap(ssi_info.ssi); + + if (ssi_info.irq) + irq_dispose_mapping(ssi_info.irq); + + if (dma_info.dma_channel[0]) + iounmap(dma_info.dma_channel[0]); + + if (dma_info.dma_channel[1]) + iounmap(dma_info.dma_channel[1]); + + if (dma_info.dma_irq[0]) + irq_dispose_mapping(dma_info.dma_irq[0]); + + if (dma_info.dma_irq[1]) + irq_dispose_mapping(dma_info.dma_irq[1]); + + if (machine_data->guts) + iounmap(machine_data->guts); + + kfree(machine_data); + + return ret; +} + +/** + * mpc8610_hpcd_remove: remove the OF device + * + * This function is called when the OF device is removed. + */ +static int mpc8610_hpcd_remove(struct of_device *ofdev) +{ + struct platform_device *sound_device = dev_get_drvdata(&ofdev->dev); + struct mpc8610_hpcd_data *machine_data = + sound_device->dev.platform_data; + + platform_device_unregister(sound_device); + + if (machine_data->dai.cpu_dai) + fsl_ssi_destroy_dai(machine_data->dai.cpu_dai); + + if (machine_data->ssi) + iounmap(machine_data->ssi); + + if (machine_data->dma[0]) + iounmap(machine_data->dma[0]); + + if (machine_data->dma[1]) + iounmap(machine_data->dma[1]); + + if (machine_data->dma_irq[0]) + irq_dispose_mapping(machine_data->dma_irq[0]); + + if (machine_data->dma_irq[1]) + irq_dispose_mapping(machine_data->dma_irq[1]); + + if (machine_data->guts) + iounmap(machine_data->guts); + + kfree(machine_data); + sound_device->dev.platform_data = NULL; + + dev_set_drvdata(&ofdev->dev, NULL); + + return 0; +} + +static struct of_device_id mpc8610_hpcd_match[] = { + { + .compatible = "fsl,mpc8610-ssi", + }, + {} +}; +MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); + +static struct of_platform_driver mpc8610_hpcd_of_driver = { + .owner = THIS_MODULE, + .name = "mpc8610_hpcd", + .match_table = mpc8610_hpcd_match, + .probe = mpc8610_hpcd_probe, + .remove = mpc8610_hpcd_remove, +}; + +/** + * mpc8610_hpcd_init: fabric driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init mpc8610_hpcd_init(void) +{ + int ret; + + printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); + + ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); + + if (ret) + printk(KERN_ERR + "mpc8610-hpcd: failed to register platform driver\n"); + + return ret; +} + +/** + * mpc8610_hpcd_exit: fabric driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit mpc8610_hpcd_exit(void) +{ + of_unregister_platform_driver(&mpc8610_hpcd_of_driver); +} + +module_init(mpc8610_hpcd_init); +module_exit(mpc8610_hpcd_exit); + +MODULE_AUTHOR("Timur Tabi "); +MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From e25bcdba1953268c10c308cd2e4526ea89bdbe0f Mon Sep 17 00:00:00 2001 From: Andrew Paprocki Date: Sun, 13 Jan 2008 11:57:17 +0100 Subject: [ALSA] hda_intel: Fix multiple device support by incrementing device count Fixes multiple device support by incrementing the static device counter at the end of the azx_probe() call. Without this, subsequent probes would always use the index specified for the first card. Signed-off-by: Andrew Paprocki Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3ae4b4c..25f35fa 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1933,6 +1933,7 @@ static int __devinit azx_probe(struct pci_dev *pci, chip->running = 1; power_down_all_codecs(chip); + dev++; return err; } -- cgit v0.10.2 From 5795b9e64588192b2b1123fed5f9ecaf0ecef5ba Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Sun, 13 Jan 2008 11:58:27 +0100 Subject: [ALSA] hda: Fix 5.1 sound in Dell 6stack ALC888 HDA This patch fixes 5.1 surround output and headphone detection in the Dell Inspiron 530 and possibly other Dell systems using the ALC888 codec (mode 6stack-dell). Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 959c80d..9f40935 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -872,6 +872,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. haier-w66 Haier W66 6stack-hp HP machines with 6stack (Nettle boards) 3stack-hp HP machines with 3stack (Lucknow, Samba boards) + 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D auto auto-config reading BIOS (default) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 11bd68b..9aa2b58 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -196,6 +196,7 @@ enum { ALC883_HAIER_W66, ALC888_6ST_HP, ALC888_3ST_HP, + ALC888_6ST_DELL, ALC883_MITAC, ALC883_AUTO, ALC883_MODEL_LAST, @@ -6812,6 +6813,46 @@ static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7058,6 +7099,15 @@ static struct hda_verb alc888_3st_hp_verbs[] = { { } }; +static struct hda_verb alc888_6st_dell_verbs[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -7253,6 +7303,33 @@ static struct hda_verb alc883_acer_eapd_verbs[] = { { } }; +static void alc888_6st_dell_front_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +static void alc888_6st_dell_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + printk("hp_event\n"); + alc888_6st_dell_front_automute(codec); + break; + } +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -7369,6 +7446,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_HAIER_W66] = "haier-w66", [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", + [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", [ALC883_AUTO] = "auto", }; @@ -7379,6 +7457,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ + SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), @@ -7653,6 +7732,21 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, + [ALC888_6ST_DELL] = { + .mixers = { alc888_6st_dell_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc888_6st_dell_unsol_event, + .init_hook = alc888_6st_dell_front_automute, + }, [ALC883_MITAC] = { .mixers = { alc883_mitac_mixer }, .init_verbs = { alc883_init_verbs, alc883_mitac_verbs }, -- cgit v0.10.2 From 52a1d4f9d6e40c2473a85105cb5e055914a4779d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 13 Jan 2008 12:03:05 +0100 Subject: [ALSA] hda-codec - Remove obsolete FIXME's Removed 'FIXME' comments that have been already fixed. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b075540..1295931 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1962,7 +1962,6 @@ static struct snd_kcontrol_new ad1988_capture_mixers[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -2978,7 +2977,6 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -3153,7 +3151,6 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -3352,7 +3349,6 @@ static struct snd_kcontrol_new ad1882_base_mixers[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 02e31e4..3d6097b 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -185,7 +185,6 @@ static struct snd_kcontrol_new cmi9880_basic_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9aa2b58..948b576 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1080,7 +1080,6 @@ static struct snd_kcontrol_new alc880_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -1102,7 +1101,6 @@ static struct snd_kcontrol_new alc880_capture_alt_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -1297,7 +1295,6 @@ static struct snd_kcontrol_new alc880_z71v_mixer[] = { }; -/* FIXME! */ /* * ALC880 F1734 model * @@ -1323,7 +1320,6 @@ static struct snd_kcontrol_new alc880_f1734_mixer[] = { }; -/* FIXME! */ /* * ALC880 ASUS model * @@ -1360,7 +1356,6 @@ static struct snd_kcontrol_new alc880_asus_mixer[] = { { } /* end */ }; -/* FIXME! */ /* * ALC880 ASUS W1V model * @@ -1398,7 +1393,6 @@ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -1912,7 +1906,6 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, alc880_uniwill_p53_dcvol_automute(codec); } -/* FIXME! */ /* * F1734 pin configuration: * HP = 0x14, speaker-out = 0x15, mic = 0x18 @@ -1941,7 +1934,6 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = { { } }; -/* FIXME! */ /* * ASUS pin configuration: * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a @@ -2088,7 +2080,6 @@ static struct hda_channel_mode alc880_lg_ch_modes[3] = { }; static struct snd_kcontrol_new alc880_lg_mixer[] = { - /* FIXME: it's not really "master" but front channels */ HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3926,7 +3917,6 @@ static struct snd_kcontrol_new alc260_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -3945,7 +3935,6 @@ static struct snd_kcontrol_new alc260_capture_alt_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -5801,7 +5790,6 @@ static struct snd_kcontrol_new alc882_capture_alt_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -5824,7 +5812,6 @@ static struct snd_kcontrol_new alc882_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -7403,7 +7390,6 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -9548,7 +9534,6 @@ static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -9569,7 +9554,6 @@ static struct snd_kcontrol_new alc268_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -10055,7 +10039,6 @@ static struct snd_kcontrol_new alc269_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -11188,7 +11171,6 @@ static struct snd_kcontrol_new alc861_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - *FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -11675,7 +11657,6 @@ static struct snd_kcontrol_new alc861vd_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - *FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -12916,7 +12897,6 @@ static struct snd_kcontrol_new alc662_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 423b499..d4dc559 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -306,7 +306,6 @@ static struct snd_kcontrol_new vt1708_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -928,7 +927,6 @@ static struct snd_kcontrol_new vt1709_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", @@ -1477,7 +1475,6 @@ static struct snd_kcontrol_new vt1708B_capture_mixer[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! */ /* .name = "Capture Source", */ .name = "Input Source", -- cgit v0.10.2 From bc9abce0de0b180817bc7e9f73145ef0b6a464ef Mon Sep 17 00:00:00 2001 From: Miguel Boton Date: Sun, 13 Jan 2008 12:03:53 +0100 Subject: [ALSA] fix compilation warning in GCC 'snd_shutdown_f_ops' is not a pointer so its address will never be NULL. GCC will complain because 'fops_get' will do an unnecessary check because '&snd_shutdown_f_ops' is always true. Signed-off-by: Miguel Boton Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/init.c b/sound/core/init.c index dc06e79..e3338d6 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -339,8 +339,8 @@ int snd_card_disconnect(struct snd_card *card) list_add(&mfile->shutdown_list, &shutdown_files); spin_unlock(&shutdown_lock); - fops_get(&snd_shutdown_f_ops); mfile->file->f_op = &snd_shutdown_f_ops; + fops_get(mfile->file->f_op); mfile = mfile->next; } -- cgit v0.10.2 From 7113e95812f508bff10f95f2e52ce6ee8cda1875 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 14 Jan 2008 08:55:03 +0100 Subject: [ALSA] oxygen: fix channel routing Do not exchange the surround and back jacks except when in 7.1 mode where the surround jack is not rear but side. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ba7a2a8..ecc0e6f 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -19,10 +19,10 @@ /* * SPI 0 -> 1st AK4396 (front) - * SPI 1 -> 2nd AK4396 (side) + * SPI 1 -> 2nd AK4396 (surround) * SPI 2 -> 3rd AK4396 (center/LFE) * SPI 3 -> WM8785 - * SPI 4 -> 4th AK4396 (rear) + * SPI 4 -> 4th AK4396 (back) * * GPIO 0 -> DFS0 of AK5385 * GPIO 1 -> DFS1 of AK5385 @@ -99,7 +99,7 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, { /* maps ALSA channel pair number to SPI output */ static const u8 codec_spi_map[4] = { - 0, 4, 2, 1 + 0, 1, 2, 4 }; oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER_WRITE | OXYGEN_SPI_DATA_LENGTH_2 | diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 2418ceb..9cd4be2 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -193,7 +193,7 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, 0x010a); oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_MAGIC2); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); - oxygen_write16(chip, OXYGEN_PLAY_ROUTING, 0x6c00); + oxygen_write16(chip, OXYGEN_PLAY_ROUTING, 0xe100); oxygen_write8(chip, OXYGEN_REC_ROUTING, 0x10); oxygen_write8(chip, OXYGEN_ADC_MONITOR, 0x00); oxygen_write8(chip, OXYGEN_A_MONITOR_ROUTING, 0xe4); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 7208b0f..ca72799 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -99,7 +99,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { static const char *const names[3] = { - "Front", "Front+Rear", "Front+Rear+Side" + "Front", "Front+Surround", "Front+Surround+Back" }; info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; info->count = 1; @@ -122,20 +122,22 @@ static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) void oxygen_update_dac_routing(struct oxygen *chip) { - /* - * hardware channel order: front, side, center/lfe, rear - * ALSA channel order: front, rear, center/lfe, side - */ static const unsigned int reg_values[3] = { - 0x6c00, 0x2c00, 0x2000 + 0xe100, /* front <- 0, surround <- 1, center <- 2, back <- 3 */ + 0xe000, /* front <- 0, surround <- 0, center <- 2, back <- 3 */ + 0x2000 /* front <- 0, surround <- 0, center <- 2, back <- 0 */ }; + u8 channels; unsigned int reg_value; - if ((oxygen_read8(chip, OXYGEN_PLAY_CHANNELS) & - OXYGEN_PLAY_CHANNELS_MASK) == OXYGEN_PLAY_CHANNELS_2) + channels = oxygen_read8(chip, OXYGEN_PLAY_CHANNELS) & + OXYGEN_PLAY_CHANNELS_MASK; + if (channels == OXYGEN_PLAY_CHANNELS_2) reg_value = reg_values[chip->dac_routing]; + else if (channels == OXYGEN_PLAY_CHANNELS_8) + reg_value = 0x6c00; /* surround <- 3, back <- 1 */ else - reg_value = 0x6c00; + reg_value = 0xe100; oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value, 0xff00); } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 6603a68..bfd1f3c 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -19,9 +19,9 @@ /* * SPI 0 -> 1st PCM1796 (front) - * SPI 1 -> 2nd PCM1796 (side) + * SPI 1 -> 2nd PCM1796 (surround) * SPI 2 -> 3rd PCM1796 (center/LFE) - * SPI 4 -> 4th PCM1796 (rear) + * SPI 4 -> 4th PCM1796 (back) * * GPIO 2 -> M0 of CS5381 * GPIO 3 -> M1 of CS5381 @@ -76,7 +76,7 @@ static void pcm1796_write(struct oxygen *chip, unsigned int codec, { /* maps ALSA channel pair number to SPI output */ static const u8 codec_map[4] = { - 0, 4, 2, 1 + 0, 1, 2, 4 }; oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER_WRITE | OXYGEN_SPI_DATA_LENGTH_2 | -- cgit v0.10.2 From 01a3affb2eebfd6c996c36d82bbbc6040eb3a7f1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 14 Jan 2008 08:56:01 +0100 Subject: [ALSA] oxygen: use an array of snd_kcontrol pointers Use an array for the pointers to known controls so that it is easier to add more. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 248f7ed..a10c377 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -15,6 +15,12 @@ #define PCM_AC97 5 #define PCM_COUNT 6 +enum { + CONTROL_SPDIF_PCM, + CONTROL_SPDIF_INPUT_BITS, + CONTROL_COUNT +}; + #define OXYGEN_PCI_SUBID(sv, sd) \ .vendor = PCI_VENDOR_ID_CMEDIA, \ .device = 0x8788, \ @@ -50,8 +56,7 @@ struct oxygen { u32 spdif_bits; u32 spdif_pcm_bits; struct snd_pcm_substream *streams[PCM_COUNT]; - struct snd_kcontrol *spdif_pcm_ctl; - struct snd_kcontrol *spdif_input_bits_ctl; + struct snd_kcontrol *controls[CONTROL_COUNT]; struct work_struct spdif_input_bits_work; }; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9cd4be2..5b77c94 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -114,7 +114,7 @@ static void oxygen_spdif_input_bits_changed(struct work_struct *work) } } - if (chip->spdif_input_bits_ctl) { + if (chip->controls[CONTROL_SPDIF_INPUT_BITS]) { spin_lock_irq(&chip->reg_lock); chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_CHANGE; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, @@ -122,7 +122,7 @@ static void oxygen_spdif_input_bits_changed(struct work_struct *work) spin_unlock_irq(&chip->reg_lock); snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->spdif_input_bits_ctl->id); + &chip->controls[CONTROL_SPDIF_INPUT_BITS]->id); } } diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index ca72799..f7350fa 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -586,15 +586,22 @@ static const struct snd_kcontrol_new controls[] = { static void oxygen_any_ctl_free(struct snd_kcontrol *ctl) { struct oxygen *chip = ctl->private_data; + unsigned int i; /* I'm too lazy to write a function for each control :-) */ - chip->spdif_pcm_ctl = NULL; - chip->spdif_input_bits_ctl = NULL; + for (i = 0; i < ARRAY_SIZE(chip->controls); ++i) + chip->controls[i] = NULL; } int oxygen_mixer_init(struct oxygen *chip) { - unsigned int i; + static const char *const known_ctl_names[CONTROL_COUNT] = { + [CONTROL_SPDIF_PCM] = + SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM), + [CONTROL_SPDIF_INPUT_BITS] = + SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), + }; + unsigned int i, j; struct snd_kcontrol *ctl; int err; @@ -610,15 +617,11 @@ int oxygen_mixer_init(struct oxygen *chip) err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; - if (!strcmp(ctl->id.name, - SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM))) { - chip->spdif_pcm_ctl = ctl; - ctl->private_free = oxygen_any_ctl_free; - } else if (!strcmp(ctl->id.name, - SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT))) { - chip->spdif_input_bits_ctl = ctl; - ctl->private_free = oxygen_any_ctl_free; - } + for (j = 0; j < CONTROL_COUNT; ++j) + if (!strcmp(ctl->id.name, known_ctl_names[j])) { + chip->controls[j] = ctl; + ctl->private_free = oxygen_any_ctl_free; + } } return chip->model->mixer_init ? chip->model->mixer_init(chip) : 0; } diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index d4a1d73..5f67a79 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -209,11 +209,11 @@ static int oxygen_open(struct snd_pcm_substream *substream, chip->pcm_active |= 1 << channel; if (channel == PCM_SPDIF) { chip->spdif_pcm_bits = chip->spdif_bits; - chip->spdif_pcm_ctl->vd[0].access &= + chip->controls[CONTROL_SPDIF_PCM]->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, - &chip->spdif_pcm_ctl->id); + &chip->controls[CONTROL_SPDIF_PCM]->id); } mutex_unlock(&chip->mutex); @@ -258,11 +258,11 @@ static int oxygen_close(struct snd_pcm_substream *substream) mutex_lock(&chip->mutex); chip->pcm_active &= ~(1 << channel); if (channel == PCM_SPDIF) { - chip->spdif_pcm_ctl->vd[0].access |= + chip->controls[CONTROL_SPDIF_PCM]->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, - &chip->spdif_pcm_ctl->id); + &chip->controls[CONTROL_SPDIF_PCM]->id); } if (channel == PCM_SPDIF || channel == PCM_MULTICH) oxygen_update_spdif_source(chip); -- cgit v0.10.2 From 893e44ba5a7cc05d66b69806defc17dd762c3ba8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 14 Jan 2008 08:57:05 +0100 Subject: [ALSA] oxygen: make line-in switch exclusive The line input cannot be mixed with the other inputs, so we have to mute the other input switches when it is selected. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index a10c377..4a0c663 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -18,6 +18,10 @@ enum { CONTROL_SPDIF_PCM, CONTROL_SPDIF_INPUT_BITS, + CONTROL_MIC_CAPTURE_SWITCH, + CONTROL_LINE_CAPTURE_SWITCH, + CONTROL_CD_CAPTURE_SWITCH, + CONTROL_AUX_CAPTURE_SWITCH, CONTROL_COUNT }; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index f7350fa..fcb5813 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -402,6 +402,19 @@ static int ac97_switch_get(struct snd_kcontrol *ctl, return 0; } +static void ac97_mute_ctl(struct oxygen *chip, unsigned int control) +{ + unsigned int index = chip->controls[control]->private_value & 0xff; + u16 value; + + value = oxygen_read_ac97(chip, 0, index); + if (!(value & 0x8000)) { + oxygen_write_ac97(chip, 0, index, value | 0x8000); + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->controls[control]->id); + } +} + static int ac97_switch_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -422,9 +435,20 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, change = newreg != oldreg; if (change) { oxygen_write_ac97(chip, 0, index, newreg); - if (index == AC97_LINE) + if (index == AC97_LINE) { oxygen_write_ac97_masked(chip, 0, 0x72, !!(newreg & 0x8000), 0x0001); + if (!(newreg & 0x8000)) { + ac97_mute_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH); + ac97_mute_ctl(chip, CONTROL_CD_CAPTURE_SWITCH); + ac97_mute_ctl(chip, CONTROL_AUX_CAPTURE_SWITCH); + } + } else if ((index == AC97_MIC || index == AC97_CD || + index == AC97_VIDEO || index == AC97_AUX) && + bitnr == 15 && !(newreg & 0x8000)) { + ac97_mute_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH); + oxygen_write_ac97_masked(chip, 0, 0x72, 0x0001, 0x0001); + } } mutex_unlock(&chip->mutex); return change; @@ -600,6 +624,10 @@ int oxygen_mixer_init(struct oxygen *chip) SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM), [CONTROL_SPDIF_INPUT_BITS] = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), + [CONTROL_MIC_CAPTURE_SWITCH] = "Mic Capture Switch", + [CONTROL_LINE_CAPTURE_SWITCH] = "Line Capture Switch", + [CONTROL_CD_CAPTURE_SWITCH] = "CD Capture Switch", + [CONTROL_AUX_CAPTURE_SWITCH] = "Aux Capture Switch", }; unsigned int i, j; struct snd_kcontrol *ctl; -- cgit v0.10.2 From 39516103e683856125b8873fa35f1a7b9172350d Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 14 Jan 2008 12:07:53 +0100 Subject: [ALSA] cs4231: remove one busy wait This busy_wait is not needed after latest changes to the cs4231-lib Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 1cd3fe3..0aa8649 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -332,7 +332,6 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { return; } - snd_cs4231_busy_wait(chip); /* * Wait for (possible -- during init auto-calibration may not be set) -- cgit v0.10.2 From fb920b7d8b65f253671073d40d490d0968151680 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 15 Jan 2008 08:39:06 +0100 Subject: [ALSA] oxygen: rename PCM to Master Rename the 'PCM Playback Volume'/'Switch' mixer controls to 'Master'. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index fcb5813..aa3a5c5 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -524,7 +524,7 @@ static DECLARE_TLV_DB_SCALE(ac97_db_scale, -3450, 150, 0); static const struct snd_kcontrol_new controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Volume", + .name = "Master Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = dac_volume_info, @@ -536,7 +536,7 @@ static const struct snd_kcontrol_new controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Switch", + .name = "Master Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, @@ -637,7 +637,7 @@ int oxygen_mixer_init(struct oxygen *chip) ctl = snd_ctl_new1(&controls[i], chip); if (!ctl) return -ENOMEM; - if (!strcmp(ctl->id.name, "PCM Playback Volume")) + if (!strcmp(ctl->id.name, "Master Playback Volume")) ctl->tlv.p = chip->model->dac_tlv; else if (chip->model->cd_in_from_video_in && !strncmp(ctl->id.name, "CD Capture ", 11)) -- cgit v0.10.2 From bcd7200394bde40e3735054fc660b6f5012638b3 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Tue, 15 Jan 2008 11:23:55 +0100 Subject: [ALSA] HDA: Enable chipset gcap usage This patch removes hardcoded values for the number of streams supported by the southbridge in most chipsets, and reads these values from the chipset directly. Most systems are hardwired for 4 streams in each direction, but newer chipsets change that capability. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 25f35fa..5f2c3ca 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1708,12 +1708,13 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, { struct azx *chip; int err; + unsigned short gcap; static struct snd_device_ops ops = { .dev_free = azx_dev_free, }; *rchip = NULL; - + err = pci_enable_device(pci); if (err < 0) return err; @@ -1775,25 +1776,40 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, pci_set_master(pci); synchronize_irq(chip->irq); - switch (chip->driver_type) { - case AZX_DRIVER_ULI: - chip->playback_streams = ULI_NUM_PLAYBACK; - chip->capture_streams = ULI_NUM_CAPTURE; - chip->playback_index_offset = ULI_PLAYBACK_INDEX; - chip->capture_index_offset = ULI_CAPTURE_INDEX; - break; - case AZX_DRIVER_ATIHDMI: - chip->playback_streams = ATIHDMI_NUM_PLAYBACK; - chip->capture_streams = ATIHDMI_NUM_CAPTURE; - chip->playback_index_offset = ATIHDMI_PLAYBACK_INDEX; - chip->capture_index_offset = ATIHDMI_CAPTURE_INDEX; - break; - default: - chip->playback_streams = ICH6_NUM_PLAYBACK; - chip->capture_streams = ICH6_NUM_CAPTURE; - chip->playback_index_offset = ICH6_PLAYBACK_INDEX; - chip->capture_index_offset = ICH6_CAPTURE_INDEX; - break; + gcap = azx_readw(chip, GCAP); + snd_printdd("chipset global capabilities = 0x%x\n", gcap); + + if (gcap) { + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->playback_streams = (gcap & (0xF << 12)) >> 12; + chip->capture_streams = (gcap & (0xF << 8)) >> 8; + chip->playback_index_offset = (gcap & (0xF << 12)) >> 12; + chip->capture_index_offset = 0; + } else { + /* gcap didn't give any info, switching to old method */ + + switch (chip->driver_type) { + case AZX_DRIVER_ULI: + chip->playback_streams = ULI_NUM_PLAYBACK; + chip->capture_streams = ULI_NUM_CAPTURE; + chip->playback_index_offset = ULI_PLAYBACK_INDEX; + chip->capture_index_offset = ULI_CAPTURE_INDEX; + break; + case AZX_DRIVER_ATIHDMI: + chip->playback_streams = ATIHDMI_NUM_PLAYBACK; + chip->capture_streams = ATIHDMI_NUM_CAPTURE; + chip->playback_index_offset = ATIHDMI_PLAYBACK_INDEX; + chip->capture_index_offset = ATIHDMI_CAPTURE_INDEX; + break; + default: + chip->playback_streams = ICH6_NUM_PLAYBACK; + chip->capture_streams = ICH6_NUM_CAPTURE; + chip->playback_index_offset = ICH6_PLAYBACK_INDEX; + chip->capture_index_offset = ICH6_CAPTURE_INDEX; + break; + } } chip->num_streams = chip->playback_streams + chip->capture_streams; chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), -- cgit v0.10.2 From 76e1ddfbdae590dd4580141b49c4b01f6fb12dab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Jan 2008 11:39:08 +0100 Subject: [ALSA] hda-code - Clean up STAC GPIO enablement code There are two similar GPIO-enablement codes in patch_sigmatel.c. Let's clean up. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c3496de..1d643b9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -121,7 +121,6 @@ struct sigmatel_spec { unsigned int mic_switch: 1; unsigned int alt_switch: 1; unsigned int hp_detect: 1; - unsigned int gpio_mute: 1; unsigned int gpio_mask, gpio_data; unsigned char aloopback_mask; @@ -1681,22 +1680,6 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) spec->pin_configs[i]); } -static void stac92xx_enable_gpio_mask(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - /* Configure GPIOx as output */ - snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, spec->gpio_mask); - /* Configure GPIOx as CMOS */ - snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIOx */ - snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, spec->gpio_data); - /* Enable GPIOx */ - snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, spec->gpio_mask); -} - /* * Analog playback callbacks */ @@ -2678,38 +2661,35 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) * funky external mute control using GPIO pins. */ -static void stac922x_gpio_mute(struct hda_codec *codec, int pin, int muted) +static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, + unsigned int data) { unsigned int gpiostate, gpiomask, gpiodir; gpiostate = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DATA, 0); - - if (!muted) - gpiostate |= (1 << pin); - else - gpiostate &= ~(1 << pin); + gpiostate = (gpiostate & ~mask) | (data & mask); gpiomask = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_MASK, 0); - gpiomask |= (1 << pin); + gpiomask |= mask; gpiodir = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DIRECTION, 0); - gpiodir |= (1 << pin); + gpiodir |= mask; - /* AppleHDA seems to do this -- WTF is this verb?? */ + /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, gpiodir); + snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, gpiodir); /* sync */ msleep(1); - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, gpiostate); + snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, @@ -2791,10 +2771,7 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); - if (spec->gpio_mute) { - stac922x_gpio_mute(codec, 0, 0); - stac922x_gpio_mute(codec, 1, 0); - } + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_data); return 0; } @@ -2950,10 +2927,7 @@ static int stac92xx_resume(struct hda_codec *codec) stac92xx_set_config_regs(codec); snd_hda_sequence_write(codec, spec->init); - if (spec->gpio_mute) { - stac922x_gpio_mute(codec, 0, 0); - stac922x_gpio_mute(codec, 1, 0); - } + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_data); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); /* invoke unsolicited event to reset the HP state */ @@ -3188,7 +3162,6 @@ again: spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ spec->gpio_mask = spec->gpio_data = 0x000001; - stac92xx_enable_gpio_mask(codec); spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; @@ -3263,7 +3236,6 @@ again: spec->aloopback_shift = 0; spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ - stac92xx_enable_gpio_mask(codec); spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -3319,7 +3291,7 @@ static int patch_stac922x(struct hda_codec *codec) stac922x_models, stac922x_cfg_tbl); if (spec->board_config == STAC_INTEL_MAC_V3) { - spec->gpio_mute = 1; + spec->gpio_mask = spec->gpio_data = 0x03; /* Intel Macs have all same PCI SSID, so we need to check * codec SSID to distinguish the exact models */ @@ -3483,7 +3455,6 @@ static int patch_stac927x(struct hda_codec *codec) spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; - stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) { @@ -3568,7 +3539,6 @@ static int patch_stac9205(struct hda_codec *codec) break; } - stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) { if (spec->board_config < 0) { -- cgit v0.10.2 From f0747ee6c12d218c1ccfd90b5413034f043894bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Jan 2008 11:41:41 +0100 Subject: [ALSA] hda-codec - Disable PCBEEP mixer element in test model It turned out that the PCBEEP element (0x1d) is disabled on some hardwares although it's defined in the datasheet. Because of the error at info of this element, the mixer gets totally unusable. Since the PCBEEP isn't that important feature, it's safer to disable this. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 948b576..3b34bb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9596,7 +9596,8 @@ static struct snd_kcontrol_new alc268_test_mixer[] = { HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT), HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT), + /* The below appears problematic on some hardwares */ + /*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/ HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT), -- cgit v0.10.2 From 3866f0b0c2df3d179b2901d084670d3cf711b1da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Jan 2008 12:37:42 +0100 Subject: [ALSA] hda-codec - Add the support of Dell OEM laptops with ALC268 Added the support of Dell OEM laptops (Vostro 1200) with ALC268 codec. The new model=dell is provided. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 9f40935..99b39c8 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -825,6 +825,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops + dell Dell OEM laptops (Vostro 1200) test for testing/debugging purpose, almost all controls can adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3b34bb6..6f652af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -106,6 +106,7 @@ enum { ALC268_3ST, ALC268_TOSHIBA, ALC268_ACER, + ALC268_DELL, #ifdef CONFIG_SND_DEBUG ALC268_TEST, #endif @@ -9424,6 +9425,49 @@ static void alc268_acer_init_hook(struct hda_codec *codec) alc268_acer_automute(codec, 1); } +static struct snd_kcontrol_new alc268_dell_mixer[] = { + /* output mixer control */ + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), + { } +}; + +static struct hda_verb alc268_dell_verbs[] = { + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc268_dell_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned int mute; + + present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0); + if (present & 0x80000000) + mute = HDA_AMP_MUTE; + else + mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); +} + +static void alc268_dell_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc268_dell_automute(codec); +} + +#define alc268_dell_init_hook alc268_dell_automute + /* * generic initialization of ADC, input mixers and output mixers */ @@ -9842,6 +9886,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", + [ALC268_DELL] = "dell", #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = "test", #endif @@ -9851,6 +9896,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), + SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), @@ -9903,6 +9949,19 @@ static struct alc_config_preset alc268_presets[] = { .unsol_event = alc268_acer_unsol_event, .init_hook = alc268_acer_init_hook, }, + [ALC268_DELL] = { + .mixers = { alc268_dell_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc268_dell_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .hp_nid = 0x02, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .unsol_event = alc268_dell_unsol_event, + .init_hook = alc268_dell_init_hook, + .input_mux = &alc268_capture_source, + }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { .mixers = { alc268_test_mixer, alc268_capture_mixer }, @@ -9967,28 +10026,24 @@ static int patch_alc268(struct hda_codec *codec) spec->stream_name_digital = "ALC268 Digital"; spec->stream_digital_playback = &alc268_pcm_digital_playback; - if (board_config == ALC268_AUTO) { - if (!spec->adc_nids && spec->input_mux) { - /* check whether NID 0x07 is valid */ - unsigned int wcap = get_wcaps(codec, 0x07); - - /* get type */ - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wcap != AC_WID_AUD_IN) { - spec->adc_nids = alc268_adc_nids_alt; - spec->num_adc_nids = - ARRAY_SIZE(alc268_adc_nids_alt); - spec->mixers[spec->num_mixers] = + if (!spec->adc_nids && spec->input_mux) { + /* check whether NID 0x07 is valid */ + unsigned int wcap = get_wcaps(codec, 0x07); + + /* get type */ + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc268_adc_nids_alt; + spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt); + spec->mixers[spec->num_mixers] = alc268_capture_alt_mixer; - spec->num_mixers++; - } else { - spec->adc_nids = alc268_adc_nids; - spec->num_adc_nids = - ARRAY_SIZE(alc268_adc_nids); - spec->mixers[spec->num_mixers] = - alc268_capture_mixer; - spec->num_mixers++; - } + spec->num_mixers++; + } else { + spec->adc_nids = alc268_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids); + spec->mixers[spec->num_mixers] = + alc268_capture_mixer; + spec->num_mixers++; } } -- cgit v0.10.2 From 12b74c80cc20dec27b9f9eeb24ee86170c34e5a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Jan 2008 12:39:38 +0100 Subject: [ALSA] hda-codec - Enable VIA SPDIF input pin Enable the SPDIF input-pin on VIA codecs when SPDIF-input is enabled by BIOS. Also, including a bit code clean up. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d4dc559..4e5dd4c 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -566,25 +566,27 @@ static int via_init(struct hda_codec *codec) if (IS_VT1708_VENDORID(codec->vendor_id)) { snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - 0x40); + PIN_OUT); snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) || IS_VT1709_6CH_VENDORID(codec->vendor_id)) { snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - 0x40); + PIN_OUT); snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) || IS_VT1708B_4CH_VENDORID(codec->vendor_id)) { snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - 0x40); + PIN_OUT); snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } - } + } else /* enable SPDIF-input pin */ + snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); return 0; } -- cgit v0.10.2 From 31c77643a06313b3a26f4c38c75ceec2a89ad31a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Jan 2008 08:28:17 +0100 Subject: [ALSA] oxygen: make AC97 codec optional Only initialize and create mixer controls for the first AC97 codec when one has actually been detected. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 4a0c663..66dee95 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -56,7 +56,8 @@ struct oxygen { u8 spdif_playback_enable; u8 ak4396_reg1; u8 revision; - u8 has_2nd_ac97_codec; + u8 has_ac97_0; + u8 has_ac97_1; u32 spdif_bits; u32 spdif_pcm_bits; struct snd_pcm_substream *streams[PCM_COUNT]; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 5b77c94..ba2bb49 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -142,13 +142,25 @@ static void oxygen_proc_read(struct snd_info_entry *entry, } if (mutex_lock_interruptible(&chip->mutex) < 0) return; - snd_iprintf(buffer, "\nAC97\n"); - for (i = 0; i < 0x80; i += 0x10) { - snd_iprintf(buffer, "%02x:", i); - for (j = 0; j < 0x10; j += 2) - snd_iprintf(buffer, " %04x", - oxygen_read_ac97(chip, 0, i + j)); - snd_iprintf(buffer, "\n"); + if (chip->has_ac97_0) { + snd_iprintf(buffer, "\nAC97\n"); + for (i = 0; i < 0x80; i += 0x10) { + snd_iprintf(buffer, "%02x:", i); + for (j = 0; j < 0x10; j += 2) + snd_iprintf(buffer, " %04x", + oxygen_read_ac97(chip, 0, i + j)); + snd_iprintf(buffer, "\n"); + } + } + if (chip->has_ac97_1) { + snd_iprintf(buffer, "\nAC97 2\n"); + for (i = 0; i < 0x80; i += 0x10) { + snd_iprintf(buffer, "%02x:", i); + for (j = 0; j < 0x10; j += 2) + snd_iprintf(buffer, " %04x", + oxygen_read_ac97(chip, 1, i + j)); + snd_iprintf(buffer, "\n"); + } } mutex_unlock(&chip->mutex); } @@ -184,6 +196,10 @@ static void __devinit oxygen_init(struct oxygen *chip) if (chip->revision == 1) oxygen_set_bits8(chip, OXYGEN_MISC, OXYGEN_MISC_MAGIC); + i = oxygen_read16(chip, OXYGEN_AC97_CONTROL); + chip->has_ac97_0 = (i & OXYGEN_AC97_CODEC_0) != 0; + chip->has_ac97_1 = (i & OXYGEN_AC97_CODEC_1) != 0; + oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC | OXYGEN_FUNCTION_ENABLE_SPI_4_5); @@ -202,31 +218,33 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0x00); - oxygen_clear_bits16(chip, OXYGEN_AC97_OUT_CONFIG, - OXYGEN_AC97_OUT_MAGIC3); - oxygen_set_bits16(chip, OXYGEN_AC97_IN_CONFIG, - OXYGEN_AC97_IN_MAGIC3); - oxygen_write_ac97(chip, 0, AC97_RESET, 0); - msleep(1); - oxygen_ac97_set_bits(chip, 0, 0x70, 0x0300); - oxygen_ac97_set_bits(chip, 0, 0x64, 0x8043); - oxygen_ac97_set_bits(chip, 0, 0x62, 0x180f); - oxygen_write_ac97(chip, 0, AC97_MASTER, 0x0000); - oxygen_write_ac97(chip, 0, AC97_PC_BEEP, 0x8000); - oxygen_write_ac97(chip, 0, AC97_MIC, 0x8808); - oxygen_write_ac97(chip, 0, AC97_LINE, 0x0808); - oxygen_write_ac97(chip, 0, AC97_CD, 0x8808); - oxygen_write_ac97(chip, 0, AC97_VIDEO, 0x8808); - oxygen_write_ac97(chip, 0, AC97_AUX, 0x8808); - oxygen_write_ac97(chip, 0, AC97_REC_GAIN, 0x8000); - oxygen_write_ac97(chip, 0, AC97_CENTER_LFE_MASTER, 0x8080); - oxygen_write_ac97(chip, 0, AC97_SURROUND_MASTER, 0x8080); - oxygen_ac97_clear_bits(chip, 0, 0x72, 0x0001); - /* power down unused ADCs and DACs */ - oxygen_ac97_set_bits(chip, 0, AC97_POWERDOWN, - AC97_PD_PR0 | AC97_PD_PR1); - oxygen_ac97_set_bits(chip, 0, AC97_EXTENDED_STATUS, - AC97_EA_PRI | AC97_EA_PRJ | AC97_EA_PRK); + if (chip->has_ac97_0) { + oxygen_clear_bits16(chip, OXYGEN_AC97_OUT_CONFIG, + OXYGEN_AC97_OUT_MAGIC3); + oxygen_set_bits16(chip, OXYGEN_AC97_IN_CONFIG, + OXYGEN_AC97_IN_MAGIC3); + oxygen_write_ac97(chip, 0, AC97_RESET, 0); + msleep(1); + oxygen_ac97_set_bits(chip, 0, 0x70, 0x0300); + oxygen_ac97_set_bits(chip, 0, 0x64, 0x8043); + oxygen_ac97_set_bits(chip, 0, 0x62, 0x180f); + oxygen_write_ac97(chip, 0, AC97_MASTER, 0x0000); + oxygen_write_ac97(chip, 0, AC97_PC_BEEP, 0x8000); + oxygen_write_ac97(chip, 0, AC97_MIC, 0x8808); + oxygen_write_ac97(chip, 0, AC97_LINE, 0x0808); + oxygen_write_ac97(chip, 0, AC97_CD, 0x8808); + oxygen_write_ac97(chip, 0, AC97_VIDEO, 0x8808); + oxygen_write_ac97(chip, 0, AC97_AUX, 0x8808); + oxygen_write_ac97(chip, 0, AC97_REC_GAIN, 0x8000); + oxygen_write_ac97(chip, 0, AC97_CENTER_LFE_MASTER, 0x8080); + oxygen_write_ac97(chip, 0, AC97_SURROUND_MASTER, 0x8080); + oxygen_ac97_clear_bits(chip, 0, 0x72, 0x0001); + /* power down unused ADCs and DACs */ + oxygen_ac97_set_bits(chip, 0, AC97_POWERDOWN, + AC97_PD_PR0 | AC97_PD_PR1); + oxygen_ac97_set_bits(chip, 0, AC97_EXTENDED_STATUS, + AC97_EA_PRI | AC97_EA_PRJ | AC97_EA_PRK); + } } static void oxygen_card_free(struct snd_card *card) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index aa3a5c5..8b08e6d 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -597,6 +597,9 @@ static const struct snd_kcontrol_new controls[] = { .info = spdif_info, .get = spdif_input_default_get, }, +}; + +static const struct snd_kcontrol_new ac97_controls[] = { AC97_VOLUME("Mic Capture Volume", AC97_MIC), AC97_SWITCH("Mic Capture Switch", AC97_MIC, 15, 1), AC97_SWITCH("Mic Boost (+20dB)", AC97_MIC, 6, 0), @@ -617,7 +620,9 @@ static void oxygen_any_ctl_free(struct snd_kcontrol *ctl) chip->controls[i] = NULL; } -int oxygen_mixer_init(struct oxygen *chip) +static int add_controls(struct oxygen *chip, + const struct snd_kcontrol_new controls[], + unsigned int count) { static const char *const known_ctl_names[CONTROL_COUNT] = { [CONTROL_SPDIF_PCM] = @@ -633,7 +638,7 @@ int oxygen_mixer_init(struct oxygen *chip) struct snd_kcontrol *ctl; int err; - for (i = 0; i < ARRAY_SIZE(controls); ++i) { + for (i = 0; i < count; ++i) { ctl = snd_ctl_new1(&controls[i], chip); if (!ctl) return -ENOMEM; @@ -651,5 +656,21 @@ int oxygen_mixer_init(struct oxygen *chip) ctl->private_free = oxygen_any_ctl_free; } } + return 0; +} + +int oxygen_mixer_init(struct oxygen *chip) +{ + int err; + + err = add_controls(chip, controls, ARRAY_SIZE(controls)); + if (err < 0) + return err; + if (chip->has_ac97_0) { + err = add_controls(chip, ac97_controls, + ARRAY_SIZE(ac97_controls)); + if (err < 0) + return err; + } return chip->model->mixer_init ? chip->model->mixer_init(chip) : 0; } diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 5f67a79..0f67def 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -714,7 +714,7 @@ int __devinit oxygen_pcm_init(struct oxygen *chip) snd_dma_pci_data(chip->pci), 128 * 1024, 256 * 1024); - if (chip->has_2nd_ac97_codec) { + if (chip->has_ac97_1) { err = snd_pcm_new(chip->card, "AC97", 2, 1, 0, &pcm); if (err < 0) return err; -- cgit v0.10.2 From 84aa6b7ba746e6f637444d0e14a9b75c0b49a612 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Jan 2008 08:28:54 +0100 Subject: [ALSA] oxygen: make SPI configuration configurable Add a field to the model structure so that it is possible to have a card where the SPI outputs 4 and 5 are used for an EEPROM. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ecc0e6f..83c6fab 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -256,6 +256,7 @@ static const struct oxygen_model model_generic = { .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, .dac_tlv = ak4396_db_scale, + .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, }; static const struct oxygen_model model_meridian = { .shortname = "C-Media CMI8788", @@ -270,6 +271,7 @@ static const struct oxygen_model model_meridian = { .update_dac_mute = update_ak4396_mute, .dac_tlv = ak4396_db_scale, .record_from_dma_b = 1, + .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, }; static int __devinit generic_oxygen_probe(struct pci_dev *pci, diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 66dee95..fb9cb36b 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -83,6 +83,7 @@ struct oxygen_model { u8 record_from_dma_b; u8 cd_in_from_video_in; u8 dac_minimum_volume; + u8 function_flags; }; /* oxygen_lib.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index ba2bb49..4edf40b 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -202,7 +202,7 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC | - OXYGEN_FUNCTION_ENABLE_SPI_4_5); + chip->model->function_flags); oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT, 0x010a); oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, 0x010a); oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, 0x010a); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bfd1f3c..247d5e1 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -227,6 +227,7 @@ static const struct oxygen_model model_xonar = { .record_from_dma_b = 1, .cd_in_from_video_in = 1, .dac_minimum_volume = 15, + .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, }; static int __devinit xonar_probe(struct pci_dev *pci, -- cgit v0.10.2 From e85e09250ab552fab6925bcde7c77746101b2d40 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Jan 2008 08:30:38 +0100 Subject: [ALSA] oxygen: make all DMA channels configurable Allow the card models to specify whether each of the hardware DMA channels is used. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 83c6fab..e2dda16 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -256,6 +256,11 @@ static const struct oxygen_model model_generic = { .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, .dac_tlv = ak4396_db_scale, + .used_channels = OXYGEN_CHANNEL_A | + OXYGEN_CHANNEL_C | + OXYGEN_CHANNEL_SPDIF | + OXYGEN_CHANNEL_MULTICH | + OXYGEN_CHANNEL_AC97, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, }; static const struct oxygen_model model_meridian = { @@ -270,7 +275,11 @@ static const struct oxygen_model model_meridian = { .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, .dac_tlv = ak4396_db_scale, - .record_from_dma_b = 1, + .used_channels = OXYGEN_CHANNEL_B | + OXYGEN_CHANNEL_C | + OXYGEN_CHANNEL_SPDIF | + OXYGEN_CHANNEL_MULTICH | + OXYGEN_CHANNEL_AC97, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, }; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index fb9cb36b..9b0234d 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -80,7 +80,7 @@ struct oxygen_model { void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); const unsigned int *dac_tlv; - u8 record_from_dma_b; + u8 used_channels; u8 cd_in_from_video_in; u8 dac_minimum_volume; u8 function_flags; diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 0f67def..5f15d35 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -681,15 +681,22 @@ static void oxygen_pcm_free(struct snd_pcm *pcm) int __devinit oxygen_pcm_init(struct oxygen *chip) { struct snd_pcm *pcm; + int outs, ins; int err; - err = snd_pcm_new(chip->card, "Analog", 0, 1, 1, &pcm); + outs = 1; /* OXYGEN_CHANNEL_MULTICH is always used */ + ins = !!(chip->model->used_channels & (OXYGEN_CHANNEL_A | + OXYGEN_CHANNEL_B)); + err = snd_pcm_new(chip->card, "Analog", 0, outs, ins, &pcm); if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &oxygen_multich_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - chip->model->record_from_dma_b ? - &oxygen_rec_b_ops : &oxygen_rec_a_ops); + if (chip->model->used_channels & OXYGEN_CHANNEL_A) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &oxygen_rec_a_ops); + else if (chip->model->used_channels & OXYGEN_CHANNEL_B) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &oxygen_rec_b_ops); pcm->private_data = chip; pcm->private_free = oxygen_pcm_free; strcpy(pcm->name, "Analog"); @@ -697,32 +704,51 @@ int __devinit oxygen_pcm_init(struct oxygen *chip) SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 512 * 1024, 2048 * 1024); - snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, - SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); - - err = snd_pcm_new(chip->card, "Digital", 1, 1, 1, &pcm); - if (err < 0) - return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &oxygen_spdif_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &oxygen_rec_c_ops); - pcm->private_data = chip; - pcm->private_free = oxygen_pcm_free; - strcpy(pcm->name, "Digital"); - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + if (ins) + snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, + SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 128 * 1024, 256 * 1024); - if (chip->has_ac97_1) { - err = snd_pcm_new(chip->card, "AC97", 2, 1, 0, &pcm); + outs = !!(chip->model->used_channels & OXYGEN_CHANNEL_SPDIF); + ins = !!(chip->model->used_channels & OXYGEN_CHANNEL_C); + if (outs | ins) { + err = snd_pcm_new(chip->card, "Digital", 1, outs, ins, &pcm); + if (err < 0) + return err; + if (outs) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &oxygen_spdif_ops); + if (ins) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &oxygen_rec_c_ops); + pcm->private_data = chip; + pcm->private_free = oxygen_pcm_free; + strcpy(pcm->name, "Digital"); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 128 * 1024, 256 * 1024); + } + + outs = chip->has_ac97_1 && + (chip->model->used_channels & OXYGEN_CHANNEL_AC97); + ins = (chip->model->used_channels & (OXYGEN_CHANNEL_A | + OXYGEN_CHANNEL_B)) + == (OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_B); + if (outs | ins) { + err = snd_pcm_new(chip->card, ins ? "Analog2" : "AC97", + 2, outs, ins, &pcm); if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &oxygen_ac97_ops); + if (outs) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &oxygen_ac97_ops); + if (ins) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &oxygen_rec_b_ops); pcm->private_data = chip; pcm->private_free = oxygen_pcm_free; - strcpy(pcm->name, "Front Panel"); + strcpy(pcm->name, ins ? "Analog 2" : "Front Panel"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 128 * 1024, 256 * 1024); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 247d5e1..0574fa1 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -224,7 +224,10 @@ static const struct oxygen_model model_xonar = { .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, .dac_tlv = pcm1796_db_scale, - .record_from_dma_b = 1, + .used_channels = OXYGEN_CHANNEL_B | + OXYGEN_CHANNEL_C | + OXYGEN_CHANNEL_SPDIF | + OXYGEN_CHANNEL_MULTICH, .cd_in_from_video_in = 1, .dac_minimum_volume = 15, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, -- cgit v0.10.2 From ccc80fb467a88ceb7ce1b68546632b91e5ba6c18 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Jan 2008 08:32:08 +0100 Subject: [ALSA] oxygen: add control filter to model struct Allow the models to modify mixer controls before they are added to the card. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index e2dda16..adf91cc 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -29,6 +29,7 @@ */ #include +#include #include #include #include @@ -244,18 +245,27 @@ static void set_ak5385_params(struct oxygen *chip, static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); +static int ak4396_control_filter(struct snd_kcontrol_new *template) +{ + if (!strcmp(template->name, "Master Playback Volume")) { + template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + template->tlv.p = ak4396_db_scale; + } + return 0; +} + static const struct oxygen_model model_generic = { .shortname = "C-Media CMI8788", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", .owner = THIS_MODULE, .init = generic_init, + .control_filter = ak4396_control_filter, .cleanup = generic_cleanup, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, - .dac_tlv = ak4396_db_scale, .used_channels = OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_C | OXYGEN_CHANNEL_SPDIF | @@ -269,12 +279,12 @@ static const struct oxygen_model model_meridian = { .chip = "CMI8788", .owner = THIS_MODULE, .init = meridian_init, + .control_filter = ak4396_control_filter, .cleanup = generic_cleanup, .set_dac_params = set_ak4396_params, .set_adc_params = set_ak5385_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, - .dac_tlv = ak4396_db_scale, .used_channels = OXYGEN_CHANNEL_B | OXYGEN_CHANNEL_C | OXYGEN_CHANNEL_SPDIF | diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 9b0234d..7278c15 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -35,6 +35,7 @@ struct pci_dev; struct snd_card; struct snd_pcm_substream; struct snd_pcm_hw_params; +struct snd_kcontrol_new; struct snd_rawmidi; struct oxygen_model; @@ -71,6 +72,7 @@ struct oxygen_model { const char *chip; struct module *owner; void (*init)(struct oxygen *chip); + int (*control_filter)(struct snd_kcontrol_new *template); int (*mixer_init)(struct oxygen *chip); void (*cleanup)(struct oxygen *chip); void (*set_dac_params)(struct oxygen *chip, @@ -79,10 +81,7 @@ struct oxygen_model { struct snd_pcm_hw_params *params); void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); - const unsigned int *dac_tlv; u8 used_channels; - u8 cd_in_from_video_in; - u8 dac_minimum_volume; u8 function_flags; }; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 8b08e6d..fae7c0f 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -27,11 +27,9 @@ static int dac_volume_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { - struct oxygen *chip = ctl->private_data; - info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; info->count = 8; - info->value.integer.min = chip->model->dac_minimum_volume; + info->value.integer.min = 0; info->value.integer.max = 0xff; return 0; } @@ -525,14 +523,10 @@ static const struct snd_kcontrol_new controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, - .tlv = { - .p = NULL, /* set later */ - }, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -635,18 +629,18 @@ static int add_controls(struct oxygen *chip, [CONTROL_AUX_CAPTURE_SWITCH] = "Aux Capture Switch", }; unsigned int i, j; + struct snd_kcontrol_new template; struct snd_kcontrol *ctl; int err; for (i = 0; i < count; ++i) { + template = controls[i]; + err = chip->model->control_filter(&template); + if (err < 0) + return err; ctl = snd_ctl_new1(&controls[i], chip); if (!ctl) return -ENOMEM; - if (!strcmp(ctl->id.name, "Master Playback Volume")) - ctl->tlv.p = chip->model->dac_tlv; - else if (chip->model->cd_in_from_video_in && - !strncmp(ctl->id.name, "CD Capture ", 11)) - ctl->private_value ^= AC97_CD ^ AC97_VIDEO; err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 0574fa1..bea34f1 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -32,6 +32,8 @@ #include #include +#include +#include #include #include #include @@ -167,6 +169,16 @@ static void set_cs5381_params(struct oxygen *chip, oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, value, 0x000c); } +static int pcm1796_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 8; + info->value.integer.min = 0x0f; + info->value.integer.max = 0xff; + return 0; +} + static int alt_switch_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -207,6 +219,18 @@ static const struct snd_kcontrol_new alt_switch = { static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0); +static int xonar_control_filter(struct snd_kcontrol_new *template) +{ + if (!strcmp(template->name, "Master Playback Volume")) { + template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + template->info = pcm1796_volume_info, + template->tlv.p = pcm1796_db_scale; + } else if (!strncmp(template->name, "CD Capture ", 11)) { + template->private_value ^= AC97_CD ^ AC97_VIDEO; + } + return 0; +} + static int xonar_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); @@ -217,19 +241,17 @@ static const struct oxygen_model model_xonar = { .longname = "Asus Virtuoso 200", .chip = "AV200", .init = xonar_init, + .control_filter = xonar_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, .set_dac_params = set_pcm1796_params, .set_adc_params = set_cs5381_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, - .dac_tlv = pcm1796_db_scale, .used_channels = OXYGEN_CHANNEL_B | OXYGEN_CHANNEL_C | OXYGEN_CHANNEL_SPDIF | OXYGEN_CHANNEL_MULTICH, - .cd_in_from_video_in = 1, - .dac_minimum_volume = 15, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, }; -- cgit v0.10.2 From 747c6016ced692db18d7e849e8cbdc523fba0874 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 16 Jan 2008 08:32:53 +0100 Subject: [ALSA] oxygen: make PCM limits configurable Add a callback to the model structure to allow modification of the hardware PCM limits. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index adf91cc..f8e3fd3 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -163,6 +163,18 @@ static void generic_cleanup(struct oxygen *chip) { } +static void generic_pcm_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware) +{ + if (channel == PCM_A) { + hardware->rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000; + hardware->rate_min = 44100; + } +} + static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -262,6 +274,7 @@ static const struct oxygen_model model_generic = { .init = generic_init, .control_filter = ak4396_control_filter, .cleanup = generic_cleanup, + .pcm_hardware_filter = generic_pcm_hardware_filter, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 7278c15..602105c 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -34,6 +34,7 @@ enum { struct pci_dev; struct snd_card; struct snd_pcm_substream; +struct snd_pcm_hardware; struct snd_pcm_hw_params; struct snd_kcontrol_new; struct snd_rawmidi; @@ -75,6 +76,8 @@ struct oxygen_model { int (*control_filter)(struct snd_kcontrol_new *template); int (*mixer_init)(struct oxygen *chip); void (*cleanup)(struct oxygen *chip); + void (*pcm_hardware_filter)(unsigned int channel, + struct snd_pcm_hardware *hardware); void (*set_dac_params)(struct oxygen *chip, struct snd_pcm_hw_params *params); void (*set_adc_params)(struct oxygen *chip, diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 5f15d35..5515c75 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -33,11 +33,15 @@ static struct snd_pcm_hardware oxygen_hardware[PCM_COUNT] = { SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_44100 | + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, - .rate_min = 44100, + .rate_min = 32000, .rate_max = 192000, .channels_min = 2, .channels_max = 2, @@ -182,6 +186,8 @@ static int oxygen_open(struct snd_pcm_substream *substream, runtime->private_data = (void *)(uintptr_t)channel; runtime->hw = oxygen_hardware[channel]; + if (chip->model->pcm_hardware_filter) + chip->model->pcm_hardware_filter(channel, &runtime->hw); err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (err < 0) -- cgit v0.10.2 From e7d7c2e28890e1b3944246af1a668b6da91ab411 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 16 Jan 2008 14:54:21 +0100 Subject: [ALSA] sound: fix mts64 section mismatches Fix section mismatches in mts64 by making a static variable __devinitdata. WARNING: vmlinux.o(.data+0x2e33f0): Section mismatch: reference to .init.data:mts64_ctl_smpte_switch (between 'control.19929' and 'snd_mts64_rawmidi_output_ops') WARNING: vmlinux.o(.data+0x2e33f8): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_hours (between 'control.19929' and 'snd_mts64_rawmidi_output_ops') WARNING: vmlinux.o(.data+0x2e3400): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_minutes (between 'control.19929' and 'snd_mts64_rawmidi_output_ops') WARNING: vmlinux.o(.data+0x2e3408): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_seconds (between 'control.19929' and 'snd_mts64_rawmidi_output_ops') WARNING: vmlinux.o(.data+0x2e3410): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_frames (between 'control.19929' and 'snd_mts64_rawmidi_output_ops') WARNING: vmlinux.o(.data+0x2e3418): Section mismatch: reference to .init.data:mts64_ctl_smpte_fps (between 'control.19929' and 'snd_mts64_rawmidi_output_ops') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index f057d92..87ba1dd 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -665,7 +665,7 @@ static int __devinit snd_mts64_ctl_create(struct snd_card *card, struct mts64 *mts) { int err, i; - static struct snd_kcontrol_new *control[] = { + static struct snd_kcontrol_new *control[] __devinitdata = { &mts64_ctl_smpte_switch, &mts64_ctl_smpte_time_hours, &mts64_ctl_smpte_time_minutes, -- cgit v0.10.2 From e4d76815c2d6477f4b77f1c7dbfbde113af89e67 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 16 Jan 2008 14:54:46 +0100 Subject: [ALSA] sound: fix ad1889 section mismatch Fix section mismatch in ad1889 by renaming the pci_driver variable to a whitelisted variable name. WARNING: vmlinux.o(.data+0x2e5ff0): Section mismatch: reference to .init.text:snd_ad1889_probe (between 'ad1889_pci' and 'index') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index be9f1a2..a66d515 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1054,7 +1054,7 @@ static struct pci_device_id snd_ad1889_ids[] = { }; MODULE_DEVICE_TABLE(pci, snd_ad1889_ids); -static struct pci_driver ad1889_pci = { +static struct pci_driver ad1889_pci_driver = { .name = "AD1889 Audio", .id_table = snd_ad1889_ids, .probe = snd_ad1889_probe, @@ -1064,13 +1064,13 @@ static struct pci_driver ad1889_pci = { static int __init alsa_ad1889_init(void) { - return pci_register_driver(&ad1889_pci); + return pci_register_driver(&ad1889_pci_driver); } static void __exit alsa_ad1889_fini(void) { - pci_unregister_driver(&ad1889_pci); + pci_unregister_driver(&ad1889_pci_driver); } module_init(alsa_ad1889_init); -- cgit v0.10.2 From 32a41b026e65e9430a8c24f5d400cc21d44a897d Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 16 Jan 2008 14:55:07 +0100 Subject: [ALSA] sound: fix atiixp section mismatch Fix section mismatch in atiixp by making some functions __devinit. WARNING: vmlinux.o(.text+0xfd9304): Section mismatch: reference to .init.data:atiixp_quirks (between 'ac97_probing_bugs' and 'snd_atiixp_codec_detect') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index e9c87f5..4594186 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -559,7 +559,7 @@ static int snd_atiixp_aclink_down(struct atiixp *chip) ATI_REG_ISR_CODEC2_NOT_READY) #define CODEC_CHECK_BITS (ALL_CODEC_NOT_READY|ATI_REG_ISR_NEW_FRAME) -static int ac97_probing_bugs(struct pci_dev *pci) +static int __devinit ac97_probing_bugs(struct pci_dev *pci) { const struct snd_pci_quirk *q; @@ -573,7 +573,7 @@ static int ac97_probing_bugs(struct pci_dev *pci) return -1; } -static int snd_atiixp_codec_detect(struct atiixp *chip) +static int __devinit snd_atiixp_codec_detect(struct atiixp *chip) { int timeout; -- cgit v0.10.2 From 1374f8ceeefcb24194c29b9a81ecc40118d2c4d1 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 16 Jan 2008 14:55:42 +0100 Subject: [ALSA] sound: fix rme9652 section mismatch Fix section mismatch in hdsp: snd_hdsp_proc_init() can be called from an ioctl at any time. WARNING: vmlinux.o(.text+0x1089bc2): Section mismatch: reference to .init.text: (between 'snd_hdsp_create_alsa_devices' and 'snd_hdsp_free') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4ba9e01..763e4c9 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3555,7 +3555,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } -static void __devinit snd_hdsp_proc_init(struct hdsp *hdsp) +static void snd_hdsp_proc_init(struct hdsp *hdsp) { struct snd_info_entry *entry; -- cgit v0.10.2 From 599c3e76fe89b314667e699a20ad08f8d16d0454 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 16 Jan 2008 14:56:04 +0100 Subject: [ALSA] sound: fix caiaq section mismatches Fix section mismatch in caiaq: these __devinit functions can be called at any time so they should not be __devinit. WARNING: vmlinux.o(.text+0x10a8dae): Section mismatch: reference to .init.text:snd_usb_caiaq_audio_init (between 'setup_card' and 'create_card') WARNING: vmlinux.o(.text+0x10a8dd6): Section mismatch: reference to .init.text:snd_usb_caiaq_midi_init (between 'setup_card' and 'create_card') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index f7e5284..2f9e9b2 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -603,7 +603,7 @@ static void free_urbs(struct urb **urbs) kfree(urbs); } -int __devinit snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) +int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; diff --git a/sound/usb/caiaq/caiaq-midi.c b/sound/usb/caiaq/caiaq-midi.c index 5427032..30b57f9 100644 --- a/sound/usb/caiaq/caiaq-midi.c +++ b/sound/usb/caiaq/caiaq-midi.c @@ -123,7 +123,7 @@ void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, snd_rawmidi_receive(dev->midi_receive_substream, buf, len); } -int __devinit snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) +int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) { int ret; struct snd_rawmidi *rmidi; -- cgit v0.10.2 From e70515dd518bbd5b9e2e5c90a56347df0e871389 Mon Sep 17 00:00:00 2001 From: "T. H. Huth" Date: Wed, 16 Jan 2008 15:57:08 +0100 Subject: [ALSA] snd-powermac: handle dead DMA transfers This patch provides the snd-powermac sound driver with the ability to handle dead DMA transfers. If a dead DMA transfer is detected, the driver now sets up a new DMA transfer to continue with the sound output at the point where the old transfer died. This dead DMA transfer handling has become necessary with recent kernels on certain G4 PowerMacs. Please refer to the following URLs for more information: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3126 https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/87652 http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=436723 The patch is based on the dead DMA transfer handling code from the old dmasound driver which can be found in the file sound/oss/dmasound/dmasound_awacs.c in the Linux source code. Signed-off-by: T. H. Huth Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index aada1d7..613a565 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -44,6 +44,18 @@ static int tumbler_freqs[1] = { 44100 }; + +/* + * we will allocate a single 'emergency' dbdma cmd block to use if the + * tx status comes up "DEAD". This happens on some PowerComputing Pmac + * clones, either owing to a bug in dbdma or some interaction between + * IDE and sound. However, this measure would deal with DEAD status if + * it appeared elsewhere. + */ +static struct pmac_dbdma emergency_dbdma; +static int emergency_in_use; + + /* * allocate DBDMA command arrays */ @@ -375,6 +387,75 @@ static snd_pcm_uframes_t snd_pmac_capture_pointer(struct snd_pcm_substream *subs /* + * Handle DEAD DMA transfers: + * if the TX status comes up "DEAD" - reported on some Power Computing machines + * we need to re-start the dbdma - but from a different physical start address + * and with a different transfer length. It would get very messy to do this + * with the normal dbdma_cmd blocks - we would have to re-write the buffer start + * addresses each time. So, we will keep a single dbdma_cmd block which can be + * fiddled with. + * When DEAD status is first reported the content of the faulted dbdma block is + * copied into the emergency buffer and we note that the buffer is in use. + * we then bump the start physical address by the amount that was successfully + * output before it died. + * On any subsequent DEAD result we just do the bump-ups (we know that we are + * already using the emergency dbdma_cmd). + * CHECK: this just tries to "do it". It is possible that we should abandon + * xfers when the number of residual bytes gets below a certain value - I can + * see that this might cause a loop-forever if a too small transfer causes + * DEAD status. However this is a TODO for now - we'll see what gets reported. + * When we get a successful transfer result with the emergency buffer we just + * pretend that it completed using the original dmdma_cmd and carry on. The + * 'next_cmd' field will already point back to the original loop of blocks. + */ +static inline void snd_pmac_pcm_dead_xfer(struct pmac_stream *rec, + volatile struct dbdma_cmd __iomem *cp) +{ + unsigned short req, res ; + unsigned int phy ; + + /* printk(KERN_WARNING "snd-powermac: DMA died - patching it up!\n"); */ + + /* to clear DEAD status we must first clear RUN + set it to quiescent to be on the safe side */ + (void)in_le32(&rec->dma->status); + out_le32(&rec->dma->control, (RUN|PAUSE|FLUSH|WAKE) << 16); + + if (!emergency_in_use) { /* new problem */ + memcpy((void *)emergency_dbdma.cmds, (void *)cp, + sizeof(struct dbdma_cmd)); + emergency_in_use = 1; + st_le16(&cp->xfer_status, 0); + st_le16(&cp->req_count, rec->period_size); + cp = emergency_dbdma.cmds; + } + + /* now bump the values to reflect the amount + we haven't yet shifted */ + req = ld_le16(&cp->req_count); + res = ld_le16(&cp->res_count); + phy = ld_le32(&cp->phy_addr); + phy += (req - res); + st_le16(&cp->req_count, res); + st_le16(&cp->res_count, 0); + st_le16(&cp->xfer_status, 0); + st_le32(&cp->phy_addr, phy); + + st_le32(&cp->cmd_dep, rec->cmd.addr + + sizeof(struct dbdma_cmd)*((rec->cur_period+1)%rec->nperiods)); + + st_le16(&cp->command, OUTPUT_MORE | BR_ALWAYS | INTR_ALWAYS); + + /* point at our patched up command block */ + out_le32(&rec->dma->cmdptr, emergency_dbdma.addr); + + /* we must re-start the controller */ + (void)in_le32(&rec->dma->status); + /* should complete clearing the DEAD status */ + out_le32(&rec->dma->control, ((RUN|WAKE) << 16) + (RUN|WAKE)); +} + +/* * update playback/capture pointer from interrupts */ static void snd_pmac_pcm_update(struct snd_pmac *chip, struct pmac_stream *rec) @@ -385,11 +466,26 @@ static void snd_pmac_pcm_update(struct snd_pmac *chip, struct pmac_stream *rec) spin_lock(&chip->reg_lock); if (rec->running) { - cp = &rec->cmd.cmds[rec->cur_period]; for (c = 0; c < rec->nperiods; c++) { /* at most all fragments */ + + if (emergency_in_use) /* already using DEAD xfer? */ + cp = emergency_dbdma.cmds; + else + cp = &rec->cmd.cmds[rec->cur_period]; + stat = ld_le16(&cp->xfer_status); + + if (stat & DEAD) { + snd_pmac_pcm_dead_xfer(rec, cp); + break; /* this block is still going */ + } + + if (emergency_in_use) + emergency_in_use = 0 ; /* done that */ + if (! (stat & ACTIVE)) break; + /*printk("update frag %d\n", rec->cur_period);*/ st_le16(&cp->xfer_status, 0); st_le16(&cp->req_count, rec->period_size); @@ -397,9 +493,8 @@ static void snd_pmac_pcm_update(struct snd_pmac *chip, struct pmac_stream *rec) rec->cur_period++; if (rec->cur_period >= rec->nperiods) { rec->cur_period = 0; - cp = rec->cmd.cmds; - } else - cp++; + } + spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(rec->substream); spin_lock(&chip->reg_lock); @@ -769,6 +864,7 @@ static int snd_pmac_free(struct snd_pmac *chip) snd_pmac_dbdma_free(chip, &chip->playback.cmd); snd_pmac_dbdma_free(chip, &chip->capture.cmd); snd_pmac_dbdma_free(chip, &chip->extra_dma); + snd_pmac_dbdma_free(chip, &emergency_dbdma); if (chip->macio_base) iounmap(chip->macio_base); if (chip->latch_base) @@ -1107,7 +1203,8 @@ int __init snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) if (snd_pmac_dbdma_alloc(chip, &chip->playback.cmd, PMAC_MAX_FRAGS + 1) < 0 || snd_pmac_dbdma_alloc(chip, &chip->capture.cmd, PMAC_MAX_FRAGS + 1) < 0 || - snd_pmac_dbdma_alloc(chip, &chip->extra_dma, 2) < 0) { + snd_pmac_dbdma_alloc(chip, &chip->extra_dma, 2) < 0 || + snd_pmac_dbdma_alloc(chip, &emergency_dbdma, 2) < 0) { err = -ENOMEM; goto __error; } -- cgit v0.10.2 From 192b8e3922c916cbacac7e5a190d9412ae39a7ee Mon Sep 17 00:00:00 2001 From: Andreas Degert Date: Wed, 16 Jan 2008 15:59:48 +0100 Subject: [ALSA] hdsp: make Multiface II work again This device has io_type == 1 (Multiface) and firmware_rev > 0xa (fixes regression from changeset 5326) Signed-off-by: Andreas Degert Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 763e4c9..c2bd438 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -607,7 +607,10 @@ static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) case Multiface: case Digiface: default: - return (64 * out) + (32 + (in)); + if (hdsp->firmware_rev == 0xa) + return (64 * out) + (32 + (in)); + else + return (52 * out) + (26 + (in)); case H9632: return (32 * out) + (16 + (in)); case H9652: @@ -621,7 +624,10 @@ static int hdsp_input_to_output_key (struct hdsp *hdsp, int in, int out) case Multiface: case Digiface: default: - return (64 * out) + in; + if (hdsp->firmware_rev == 0xa) + return (64 * out) + in; + else + return (52 * out) + in; case H9632: return (32 * out) + in; case H9652: -- cgit v0.10.2 From 52987656fb3d43192639a7d585feb564c075c864 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 16 Jan 2008 16:09:47 +0100 Subject: [ALSA] hda-intel - Add workarounds for STAC codecs Some machines with STAC codecs seem to have problems (e.g. no audible playback) when the delay in codec-read routine is too short. I still don't figure out which command sequence causes this problem (due to lack of test hardware), but it's known that increasing the delay fixes. So, added a stupid workaround here temporarily... Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 20be776..719e46f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -464,6 +464,9 @@ struct hda_bus { struct hda_bus_unsolicited *unsol; struct snd_info_entry *proc; + + /* misc op flags */ + unsigned int needs_damn_long_delay :1; }; /* diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5f2c3ca..fe07bdf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -559,8 +559,12 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) } if (!chip->rirb.cmds) return chip->rirb.res; /* the last value */ - udelay(10); - cond_resched(); + if (codec->bus->needs_damn_long_delay) + msleep(2); /* temporary workaround */ + else { + udelay(10); + cond_resched(); + } } while (time_after_eq(timeout, jiffies)); if (chip->msi) { diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d643b9..24137bc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3472,6 +3472,18 @@ static int patch_stac927x(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; + /* + * !!FIXME!! + * The STAC927x seem to require fairly long delays for certain + * command sequences. With too short delays (even if the answer + * is set to RIRB properly), it results in the silence output + * on some hardwares like Dell. + * + * The below flag enables the longer delay (see get_response + * in hda_intel.c). + */ + codec->bus->needs_damn_long_delay = 1; + return 0; } -- cgit v0.10.2 From da65fd3bf10c3500502e448c76917f5e86e28b28 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Wed, 16 Jan 2008 17:05:13 +0100 Subject: [ALSA] hda-codec - remove 11c1:1040 from patch_si3054.c id list Codec with id 11c1:1040 sitting on hda bus isn't si3054-compatible. It should be removed from patch_si3054.c id list. The detailed information http://archives.linmodems.org/26457 From: Vasily Khoruzhick Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index f5e43e8..d22f5a6 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -286,7 +286,6 @@ static int patch_si3054(struct hda_codec *codec) struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 }, - { .id = 0x11c11040, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x11c13055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x11c13155, .name = "Si3054", .patch = patch_si3054 }, -- cgit v0.10.2 From 6b8d6e5518e2812b150c2d7c1e975a1bd33f0ccc Mon Sep 17 00:00:00 2001 From: Julian Scheel Date: Wed, 16 Jan 2008 19:50:00 +0100 Subject: [ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV See ALSA bug#2384 for more details. Signed-off-by: Julian Scheel Signed-off-by: Konstantin Kletschke Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile index ee86a1d..f99fe08 100644 --- a/sound/pci/ice1712/Makefile +++ b/sound/pci/ice1712/Makefile @@ -5,7 +5,7 @@ snd-ice17xx-ak4xxx-objs := ak4xxx.o snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o -snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o juli.o phase.o wtm.o se.o +snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o # Toplevel Module Dependency obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 4dc576a..86e418e 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -407,7 +407,13 @@ struct snd_ice1712 { unsigned char ch1, ch2; } vol[8]; } se; - + struct prodigy_hifi_spec { + unsigned short master[2]; + unsigned short vol[8]; + } prodigy_hifi; + struct prodigy_hd2_spec { + unsigned short vol[2]; + } prodigy_hd2; } spec; }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 3147cbc..c89a4fe 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -47,6 +47,7 @@ #include "vt1720_mobo.h" #include "pontis.h" #include "prodigy192.h" +#include "prodigy_hifi.h" #include "juli.h" #include "phase.h" #include "wtm.h" @@ -62,6 +63,7 @@ MODULE_SUPPORTED_DEVICE("{" VT1720_MOBO_DEVICE_DESC PONTIS_DEVICE_DESC PRODIGY192_DEVICE_DESC + PRODIGY_HIFI_DEVICE_DESC JULI_DEVICE_DESC PHASE_DEVICE_DESC WTM_DEVICE_DESC @@ -1930,6 +1932,7 @@ static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_aureon_cards, snd_vt1720_mobo_cards, snd_vt1720_pontis_cards, + snd_vt1724_prodigy_hifi_cards, snd_vt1724_prodigy192_cards, snd_vt1724_juli_cards, snd_vt1724_phase_cards, diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c new file mode 100644 index 0000000..6ec1aa4 --- /dev/null +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -0,0 +1,1195 @@ +/* + * ALSA driver for ICEnsemble VT1724 (Envy24HT) + * + * Lowlevel functions for Audiotrak Prodigy 7.1 Hifi + * based on pontis.c + * + * Copyright (c) 2007 Julian Scheel + * Copyright (c) 2007 allank + * Copyright (c) 2004 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + + +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "ice1712.h" +#include "envy24ht.h" +#include "prodigy_hifi.h" + +/* I2C addresses */ +#define WM_DEV 0x34 + +/* WM8776 registers */ +#define WM_HP_ATTEN_L 0x00 /* headphone left attenuation */ +#define WM_HP_ATTEN_R 0x01 /* headphone left attenuation */ +#define WM_HP_MASTER 0x02 /* headphone master (both channels), + override LLR */ +#define WM_DAC_ATTEN_L 0x03 /* digital left attenuation */ +#define WM_DAC_ATTEN_R 0x04 +#define WM_DAC_MASTER 0x05 +#define WM_PHASE_SWAP 0x06 /* DAC phase swap */ +#define WM_DAC_CTRL1 0x07 +#define WM_DAC_MUTE 0x08 +#define WM_DAC_CTRL2 0x09 +#define WM_DAC_INT 0x0a +#define WM_ADC_INT 0x0b +#define WM_MASTER_CTRL 0x0c +#define WM_POWERDOWN 0x0d +#define WM_ADC_ATTEN_L 0x0e +#define WM_ADC_ATTEN_R 0x0f +#define WM_ALC_CTRL1 0x10 +#define WM_ALC_CTRL2 0x11 +#define WM_ALC_CTRL3 0x12 +#define WM_NOISE_GATE 0x13 +#define WM_LIMITER 0x14 +#define WM_ADC_MUX 0x15 +#define WM_OUT_MUX 0x16 +#define WM_RESET 0x17 + +/* Analog Recording Source :- Mic, LineIn, CD/Video, */ + +/* implement capture source select control for WM8776 */ + +#define WM_AIN1 "AIN1" +#define WM_AIN2 "AIN2" +#define WM_AIN3 "AIN3" +#define WM_AIN4 "AIN4" +#define WM_AIN5 "AIN5" + +/* GPIO pins of envy24ht connected to wm8766 */ +#define WM8766_SPI_CLK (1<<17) /* CLK, Pin97 on ICE1724 */ +#define WM8766_SPI_MD (1<<16) /* DATA VT1724 -> WM8766, Pin96 */ +#define WM8766_SPI_ML (1<<18) /* Latch, Pin98 */ + +/* WM8766 registers */ +#define WM8766_DAC_CTRL 0x02 /* DAC Control */ +#define WM8766_INT_CTRL 0x03 /* Interface Control */ +#define WM8766_DAC_CTRL2 0x09 +#define WM8766_DAC_CTRL3 0x0a +#define WM8766_RESET 0x1f +#define WM8766_LDA1 0x00 +#define WM8766_LDA2 0x04 +#define WM8766_LDA3 0x06 +#define WM8766_RDA1 0x01 +#define WM8766_RDA2 0x05 +#define WM8766_RDA3 0x07 +#define WM8766_MUTE1 0x0C +#define WM8766_MUTE2 0x0F + + +/* + * Prodigy HD2 + */ +#define AK4396_ADDR 0x00 +#define AK4396_CSN (1 << 8) /* CSN->GPIO8, pin 75 */ +#define AK4396_CCLK (1 << 9) /* CCLK->GPIO9, pin 76 */ +#define AK4396_CDTI (1 << 10) /* CDTI->GPIO10, pin 77 */ + +/* ak4396 registers */ +#define AK4396_CTRL1 0x00 +#define AK4396_CTRL2 0x01 +#define AK4396_CTRL3 0x02 +#define AK4396_LCH_ATT 0x03 +#define AK4396_RCH_ATT 0x04 + + +/* + * get the current register value of WM codec + */ +static unsigned short wm_get(struct snd_ice1712 *ice, int reg) +{ + reg <<= 1; + return ((unsigned short)ice->akm[0].images[reg] << 8) | + ice->akm[0].images[reg + 1]; +} + +/* + * set the register value of WM codec and remember it + */ +static void wm_put_nocache(struct snd_ice1712 *ice, int reg, unsigned short val) +{ + unsigned short cval; + cval = (reg << 9) | val; + snd_vt1724_write_i2c(ice, WM_DEV, cval >> 8, cval & 0xff); +} + +static void wm_put(struct snd_ice1712 *ice, int reg, unsigned short val) +{ + wm_put_nocache(ice, reg, val); + reg <<= 1; + ice->akm[0].images[reg] = val >> 8; + ice->akm[0].images[reg + 1] = val; +} + +/* + * write data in the SPI mode + */ + +static void set_gpio_bit(struct snd_ice1712 *ice, unsigned int bit, int val) +{ + unsigned int tmp = snd_ice1712_gpio_read(ice); + if (val) + tmp |= bit; + else + tmp &= ~bit; + snd_ice1712_gpio_write(ice, tmp); +} + +/* + * SPI implementation for WM8766 codec - only writing supported, no readback + */ + +static void wm8766_spi_send_word(struct snd_ice1712 *ice, unsigned int data) +{ + int i; + for (i = 0; i < 16; i++) { + set_gpio_bit(ice, WM8766_SPI_CLK, 0); + udelay(1); + set_gpio_bit(ice, WM8766_SPI_MD, data & 0x8000); + udelay(1); + set_gpio_bit(ice, WM8766_SPI_CLK, 1); + udelay(1); + data <<= 1; + } +} + +static void wm8766_spi_write(struct snd_ice1712 *ice, unsigned int reg, unsigned int data) +{ + unsigned int block; + + snd_ice1712_gpio_set_dir(ice, WM8766_SPI_MD| + WM8766_SPI_CLK|WM8766_SPI_ML); + snd_ice1712_gpio_set_mask(ice, ~(WM8766_SPI_MD| + WM8766_SPI_CLK|WM8766_SPI_ML)); + /* latch must be low when writing */ + set_gpio_bit(ice, WM8766_SPI_ML, 0); + block = (reg << 9) | (data & 0x1ff); + wm8766_spi_send_word(ice, block); /* REGISTER ADDRESS */ + /* release latch */ + set_gpio_bit(ice, WM8766_SPI_ML, 1); + udelay(1); + /* restore */ + snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask); + snd_ice1712_gpio_set_dir(ice, ice->gpio.direction); +} + + +/* + * serial interface for ak4396 - only writing supported, no readback + */ + +static void ak4396_send_word(struct snd_ice1712 *ice, unsigned int data) +{ + int i; + for (i = 0; i < 16; i++) { + set_gpio_bit(ice, AK4396_CCLK, 0); + udelay(1); + set_gpio_bit(ice, AK4396_CDTI, data & 0x8000); + udelay(1); + set_gpio_bit(ice, AK4396_CCLK, 1); + udelay(1); + data <<= 1; + } +} + +static void ak4396_write(struct snd_ice1712 *ice, unsigned int reg, unsigned int data) +{ + unsigned int block; + + snd_ice1712_gpio_set_dir(ice, AK4396_CSN|AK4396_CCLK|AK4396_CDTI); + snd_ice1712_gpio_set_mask(ice, ~(AK4396_CSN|AK4396_CCLK|AK4396_CDTI)); + /* latch must be low when writing */ + set_gpio_bit(ice, AK4396_CSN, 0); + block = ((AK4396_ADDR & 0x03) << 14) | (1 << 13) | + ((reg & 0x1f) << 8) | (data & 0xff); + ak4396_send_word(ice, block); /* REGISTER ADDRESS */ + /* release latch */ + set_gpio_bit(ice, AK4396_CSN, 1); + udelay(1); + /* restore */ + snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask); + snd_ice1712_gpio_set_dir(ice, ice->gpio.direction); +} + + +/* + * ak4396 mixers + */ + + + +/* + * DAC volume attenuation mixer control (-64dB to 0dB) + */ + +static int ak4396_dac_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; /* mute */ + uinfo->value.integer.max = 0xFF; /* linear */ + return 0; +} + +static int ak4396_dac_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int i; + + for (i = 0; i < 2; i++) { + ucontrol->value.integer.value[i] =ice->spec.prodigy_hd2.vol[i]; + } + return 0; +} + +static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int i; + int change = 0; + + mutex_lock(&ice->gpio_mutex); + for (i = 0; i < 2; i++) { + if (ucontrol->value.integer.value[i] != + ice->spec.prodigy_hd2.vol[i]) { + ice->spec.prodigy_hd2.vol[i] = + ucontrol->value.integer.value[i]; + ak4396_write(ice, AK4396_LCH_ATT+i, + (ice->spec.prodigy_hd2.vol[i] & 0xff)); + change = 1; + } + } + mutex_unlock(&ice->gpio_mutex); + return change; +} + +static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); + +static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Front Playback Volume", + .info = ak4396_dac_vol_info, + .get = ak4396_dac_vol_get, + .put = ak4396_dac_vol_put, + .tlv = { .p = db_scale_wm_dac }, + }, +}; + + +/* --------------- */ + +/* + * Logarithmic volume values for WM87*6 + * Computed as 20 * Log10(255 / x) + */ +static const unsigned char wm_vol[256] = { + 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23, + 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17, + 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13, + 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11, + 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8, + 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6, + 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, + 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, + 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0 +}; + +#define WM_VOL_MAX (sizeof(wm_vol) - 1) +#define WM_VOL_MUTE 0x8000 + + +#define DAC_0dB 0xff +#define DAC_RES 128 +#define DAC_MIN (DAC_0dB - DAC_RES) + + +static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) +{ + unsigned char nvol; + + if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) + nvol = 0; + else { + nvol = (((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 128) + & WM_VOL_MAX; + nvol = (nvol ? (nvol + DAC_MIN) : 0) & 0xff; + } + + wm_put(ice, index, nvol); + wm_put_nocache(ice, index, 0x100 | nvol); +} + +static void wm8766_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) +{ + unsigned char nvol; + + if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) + nvol = 0; + else { + nvol = (((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 128) + & WM_VOL_MAX; + nvol = (nvol ? (nvol + DAC_MIN) : 0) & 0xff; + } + + + wm8766_spi_write(ice, index, (0x0100 | nvol)); +} + + +/* + * DAC volume attenuation mixer control (-64dB to 0dB) + */ + +static int wm_dac_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; /* mute */ + uinfo->value.integer.max = DAC_RES; /* 0dB, 0.5dB step */ + return 0; +} + +static int wm_dac_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int i; + + for (i = 0; i < 2; i++) { + ucontrol->value.integer.value[i] = + ice->spec.prodigy_hifi.vol[2+i] & ~WM_VOL_MUTE; + } + return 0; +} + +static int wm_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int i, idx, change = 0; + + mutex_lock(&ice->gpio_mutex); + for (i = 0; i < 2; i++) { + if (ucontrol->value.integer.value[i] != + ice->spec.prodigy_hifi.vol[2+i]) { + idx = WM_DAC_ATTEN_L + i; + ice->spec.prodigy_hifi.vol[2+i] &= WM_VOL_MUTE; + ice->spec.prodigy_hifi.vol[2+i] |= + ucontrol->value.integer.value[i]; + wm_set_vol(ice, idx, ice->spec.prodigy_hifi.vol[2+i], + ice->spec.prodigy_hifi.master[i]); + change = 1; + } + } + mutex_unlock(&ice->gpio_mutex); + return change; +} + + +/* + * WM8766 DAC volume attenuation mixer control + */ +static int wm8766_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + int voices = kcontrol->private_value >> 8; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = voices; + uinfo->value.integer.min = 0; /* mute */ + uinfo->value.integer.max = DAC_RES; /* 0dB */ + return 0; +} + +static int wm8766_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int i, ofs, voices; + + voices = kcontrol->private_value >> 8; + ofs = kcontrol->private_value & 0xff; + for (i = 0; i < voices; i++) + ucontrol->value.integer.value[i] = + ice->spec.prodigy_hifi.vol[ofs+i]; + return 0; +} + +static int wm8766_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int i, idx, ofs, voices; + int change = 0; + + voices = kcontrol->private_value >> 8; + ofs = kcontrol->private_value & 0xff; + mutex_lock(&ice->gpio_mutex); + for (i = 0; i < voices; i++) { + if (ucontrol->value.integer.value[i] != + ice->spec.prodigy_hifi.vol[ofs+i]) { + idx = WM8766_LDA1 + ofs + i; + ice->spec.prodigy_hifi.vol[ofs+i] &= WM_VOL_MUTE; + ice->spec.prodigy_hifi.vol[ofs+i] |= + ucontrol->value.integer.value[i]; + wm8766_set_vol(ice, idx, + ice->spec.prodigy_hifi.vol[ofs+i], + ice->spec.prodigy_hifi.master[i]); + change = 1; + } + } + mutex_unlock(&ice->gpio_mutex); + return change; +} + +/* + * Master volume attenuation mixer control / applied to WM8776+WM8766 + */ +static int wm_master_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = DAC_RES; + return 0; +} + +static int wm_master_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int i; + for (i=0; i<2; i++) + ucontrol->value.integer.value[i] = + ice->spec.prodigy_hifi.master[i]; + return 0; +} + +static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int ch, change = 0; + + mutex_lock(&ice->gpio_mutex); + for (ch = 0; ch < 2; ch++) { + if (ucontrol->value.integer.value[ch] != + ice->spec.prodigy_hifi.master[ch]) { + ice->spec.prodigy_hifi.master[ch] &= 0x00; + ice->spec.prodigy_hifi.master[ch] |= + ucontrol->value.integer.value[ch]; + + /* Apply to front DAC */ + wm_set_vol(ice, WM_DAC_ATTEN_L + ch, + ice->spec.prodigy_hifi.vol[2 + ch], + ice->spec.prodigy_hifi.master[ch]); + + wm8766_set_vol(ice, WM8766_LDA1 + ch, + ice->spec.prodigy_hifi.vol[0 + ch], + ice->spec.prodigy_hifi.master[ch]); + + wm8766_set_vol(ice, WM8766_LDA2 + ch, + ice->spec.prodigy_hifi.vol[4 + ch], + ice->spec.prodigy_hifi.master[ch]); + + wm8766_set_vol(ice, WM8766_LDA3 + ch, + ice->spec.prodigy_hifi.vol[6 + ch], + ice->spec.prodigy_hifi.master[ch]); + change = 1; + } + } + mutex_unlock(&ice->gpio_mutex); + return change; +} + + + +/* KONSTI */ + +static int wm_adc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char* texts[32] = {"NULL", WM_AIN1, WM_AIN2, WM_AIN1 "+" WM_AIN2, + WM_AIN3, WM_AIN1 "+" WM_AIN3, WM_AIN2 "+" WM_AIN3, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN3, + WM_AIN4, WM_AIN1 "+" WM_AIN4, WM_AIN2 "+" WM_AIN4, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN4, + WM_AIN3 "+" WM_AIN4, WM_AIN1 "+" WM_AIN3 "+" WM_AIN4, + WM_AIN2 "+" WM_AIN3 "+" WM_AIN4, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4, + WM_AIN5, WM_AIN1 "+" WM_AIN5, WM_AIN2 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN5, + WM_AIN3 "+" WM_AIN5, WM_AIN1 "+" WM_AIN3 "+" WM_AIN5, + WM_AIN2 "+" WM_AIN3 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN5, + WM_AIN4 "+" WM_AIN5, WM_AIN1 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN2 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5}; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 32; + if (uinfo->value.enumerated.item > 31) + uinfo->value.enumerated.item = 31; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + + mutex_lock(&ice->gpio_mutex); + ucontrol->value.integer.value[0]=wm_get(ice, WM_ADC_MUX) & 0x1f; + mutex_unlock(&ice->gpio_mutex); + return 0; +} + +static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned short oval, nval; + + mutex_lock(&ice->gpio_mutex); + oval = wm_get(ice, WM_ADC_MUX); + nval = ( oval & 0xe0 ) | ucontrol->value.integer.value[0] ; + if ( nval != oval ) { + wm_put(ice, WM_ADC_MUX, nval); + } + mutex_unlock(&ice->gpio_mutex); + return 0; +} + +/* KONSTI */ + +/* + * ADC gain mixer control (-64dB to 0dB) + */ + +#define ADC_0dB 0xcf +#define ADC_RES 128 +#define ADC_MIN (ADC_0dB - ADC_RES) + +static int wm_adc_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; /* mute (-64dB) */ + uinfo->value.integer.max = ADC_RES; /* 0dB, 0.5dB step */ + return 0; +} + +static int wm_adc_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned short val; + int i; + + mutex_lock(&ice->gpio_mutex); + for (i = 0; i < 2; i++) { + val = wm_get(ice, WM_ADC_ATTEN_L + i) & 0xff; + val = val > ADC_MIN ? (val - ADC_MIN) : 0; + ucontrol->value.integer.value[i] = val; + } + mutex_unlock(&ice->gpio_mutex); + return 0; +} + +static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned short ovol, nvol; + int i, idx, change = 0; + + mutex_lock(&ice->gpio_mutex); + for (i = 0; i < 2; i++) { + nvol = ucontrol->value.integer.value[i]; + nvol = nvol ? (nvol + ADC_MIN) : 0; + idx = WM_ADC_ATTEN_L + i; + ovol = wm_get(ice, idx) & 0xff; + if (ovol != nvol) { + wm_put(ice, idx, nvol); + change = 1; + } + } + mutex_unlock(&ice->gpio_mutex); + return change; +} + +/* + * ADC input mux mixer control + */ +static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int bit = kcontrol->private_value; + + mutex_lock(&ice->gpio_mutex); + ucontrol->value.integer.value[0] = (wm_get(ice, WM_ADC_MUX) & (1 << bit)) ? 1 : 0; + mutex_unlock(&ice->gpio_mutex); + return 0; +} + +static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + int bit = kcontrol->private_value; + unsigned short oval, nval; + int change; + + mutex_lock(&ice->gpio_mutex); + nval = oval = wm_get(ice, WM_ADC_MUX); + if (ucontrol->value.integer.value[0]) + nval |= (1 << bit); + else + nval &= ~(1 << bit); + change = nval != oval; + if (change) { + wm_put(ice, WM_ADC_MUX, nval); + } + mutex_unlock(&ice->gpio_mutex); + return 0; +} + +/* + * Analog bypass (In -> Out) + */ +static int wm_bypass_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int wm_bypass_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + + mutex_lock(&ice->gpio_mutex); + ucontrol->value.integer.value[0] = (wm_get(ice, WM_OUT_MUX) & 0x04) ? 1 : 0; + mutex_unlock(&ice->gpio_mutex); + return 0; +} + +static int wm_bypass_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned short val, oval; + int change = 0; + + mutex_lock(&ice->gpio_mutex); + val = oval = wm_get(ice, WM_OUT_MUX); + if (ucontrol->value.integer.value[0]) + val |= 0x04; + else + val &= ~0x04; + if (val != oval) { + wm_put(ice, WM_OUT_MUX, val); + change = 1; + } + mutex_unlock(&ice->gpio_mutex); + return change; +} + +/* + * Left/Right swap + */ +static int wm_chswap_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int wm_chswap_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + + mutex_lock(&ice->gpio_mutex); + ucontrol->value.integer.value[0] = + (wm_get(ice, WM_DAC_CTRL1) & 0xf0) != 0x90; + mutex_unlock(&ice->gpio_mutex); + return 0; +} + +static int wm_chswap_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned short val, oval; + int change = 0; + + mutex_lock(&ice->gpio_mutex); + oval = wm_get(ice, WM_DAC_CTRL1); + val = oval & 0x0f; + if (ucontrol->value.integer.value[0]) + val |= 0x60; + else + val |= 0x90; + if (val != oval) { + wm_put(ice, WM_DAC_CTRL1, val); + wm_put_nocache(ice, WM_DAC_CTRL1, val); + change = 1; + } + mutex_unlock(&ice->gpio_mutex); + return change; +} + + +/* + * mixers + */ + +static struct snd_kcontrol_new prodigy_hifi_controls[] __devinitdata = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Master Playback Volume", + .info = wm_master_vol_info, + .get = wm_master_vol_get, + .put = wm_master_vol_put, + .tlv = { .p = db_scale_wm_dac } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Front Playback Volume", + .info = wm_dac_vol_info, + .get = wm_dac_vol_get, + .put = wm_dac_vol_put, + .tlv = { .p = db_scale_wm_dac }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Rear Playback Volume", + .info = wm8766_vol_info, + .get = wm8766_vol_get, + .put = wm8766_vol_put, + .private_value = (2 << 8) | 0, + .tlv = { .p = db_scale_wm_dac }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Center Playback Volume", + .info = wm8766_vol_info, + .get = wm8766_vol_get, + .put = wm8766_vol_put, + .private_value = (1 << 8) | 4, + .tlv = { .p = db_scale_wm_dac } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "LFE Playback Volume", + .info = wm8766_vol_info, + .get = wm8766_vol_get, + .put = wm8766_vol_put, + .private_value = (1 << 8) | 5, + .tlv = { .p = db_scale_wm_dac } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Side Playback Volume", + .info = wm8766_vol_info, + .get = wm8766_vol_get, + .put = wm8766_vol_put, + .private_value = (2 << 8) | 6, + .tlv = { .p = db_scale_wm_dac }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Capture Volume", + .info = wm_adc_vol_info, + .get = wm_adc_vol_get, + .put = wm_adc_vol_put, + .tlv = { .p = db_scale_wm_dac }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "CD Capture Switch", + .info = wm_adc_mux_info, + .get = wm_adc_mux_get, + .put = wm_adc_mux_put, + .private_value = 0, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Switch", + .info = wm_adc_mux_info, + .get = wm_adc_mux_get, + .put = wm_adc_mux_put, + .private_value = 1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Bypass Switch", + .info = wm_bypass_info, + .get = wm_bypass_get, + .put = wm_bypass_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Swap Output Channels", + .info = wm_chswap_info, + .get = wm_chswap_get, + .put = wm_chswap_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Capture Source", + .info = wm_adc_mux_enum_info, + .get = wm_adc_mux_enum_get, + .put = wm_adc_mux_enum_put, + }, +}; + +/* + * WM codec registers + */ +static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + char line[64]; + unsigned int reg, val; + mutex_lock(&ice->gpio_mutex); + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "%x %x", ®, &val) != 2) + continue; + if (reg <= 0x17 && val <= 0xffff) + wm_put(ice, reg, val); + } + mutex_unlock(&ice->gpio_mutex); +} + +static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + int reg, val; + + mutex_lock(&ice->gpio_mutex); + for (reg = 0; reg <= 0x17; reg++) { + val = wm_get(ice, reg); + snd_iprintf(buffer, "%02x = %04x\n", reg, val); + } + mutex_unlock(&ice->gpio_mutex); +} + +static void wm_proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { + snd_info_set_text_ops(entry, ice, wm_proc_regs_read); + entry->mode |= S_IWUSR; + entry->c.text.write = wm_proc_regs_write; + } +} + +static int __devinit prodigy_hifi_add_controls(struct snd_ice1712 *ice) +{ + unsigned int i; + int err; + + for (i = 0; i < ARRAY_SIZE(prodigy_hifi_controls); i++) { + err = snd_ctl_add(ice->card, snd_ctl_new1(&prodigy_hifi_controls[i], ice)); + if (err < 0) + return err; + } + + wm_proc_init(ice); + + return 0; +} + +static int __devinit prodigy_hd2_add_controls(struct snd_ice1712 *ice) +{ + unsigned int i; + int err; + + for (i = 0; i < ARRAY_SIZE(prodigy_hd2_controls); i++) { + err = snd_ctl_add(ice->card, snd_ctl_new1(&prodigy_hd2_controls[i], ice)); + if (err < 0) + return err; + } + + wm_proc_init(ice); + + return 0; +} + + +/* + * initialize the chip + */ +static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) +{ + static unsigned short wm_inits[] = { + /* These come first to reduce init pop noise */ + WM_ADC_MUX, 0x0003, /* ADC mute */ + /* 0x00c0 replaced by 0x0003 */ + + WM_DAC_MUTE, 0x0001, /* DAC softmute */ + WM_DAC_CTRL1, 0x0000, /* DAC mute */ + + WM_POWERDOWN, 0x0008, /* All power-up except HP */ + WM_RESET, 0x0000, /* reset */ + }; + static unsigned short wm_inits2[] = { + WM_MASTER_CTRL, 0x0022, /* 256fs, slave mode */ + WM_DAC_INT, 0x0022, /* I2S, normal polarity, 24bit */ + WM_ADC_INT, 0x0022, /* I2S, normal polarity, 24bit */ + WM_DAC_CTRL1, 0x0090, /* DAC L/R */ + WM_OUT_MUX, 0x0001, /* OUT DAC */ + WM_HP_ATTEN_L, 0x0179, /* HP 0dB */ + WM_HP_ATTEN_R, 0x0179, /* HP 0dB */ + WM_DAC_ATTEN_L, 0x0000, /* DAC 0dB */ + WM_DAC_ATTEN_L, 0x0100, /* DAC 0dB */ + WM_DAC_ATTEN_R, 0x0000, /* DAC 0dB */ + WM_DAC_ATTEN_R, 0x0100, /* DAC 0dB */ + WM_PHASE_SWAP, 0x0000, /* phase normal */ +#if 0 + WM_DAC_MASTER, 0x0100, /* DAC master muted */ +#endif + WM_DAC_CTRL2, 0x0000, /* no deemphasis, no ZFLG */ + WM_ADC_ATTEN_L, 0x0000, /* ADC muted */ + WM_ADC_ATTEN_R, 0x0000, /* ADC muted */ +#if 1 + WM_ALC_CTRL1, 0x007b, /* */ + WM_ALC_CTRL2, 0x0000, /* */ + WM_ALC_CTRL3, 0x0000, /* */ + WM_NOISE_GATE, 0x0000, /* */ +#endif + WM_DAC_MUTE, 0x0000, /* DAC unmute */ + WM_ADC_MUX, 0x0003, /* ADC unmute, both CD/Line On */ + }; + static unsigned short wm8766_inits[] = { + WM8766_RESET, 0x0000, + WM8766_DAC_CTRL, 0x0120, + WM8766_INT_CTRL, 0x0022, /* I2S Normal Mode, 24 bit */ + WM8766_DAC_CTRL2, 0x0001, + WM8766_DAC_CTRL3, 0x0080, + WM8766_LDA1, 0x0100, + WM8766_LDA2, 0x0100, + WM8766_LDA3, 0x0100, + WM8766_RDA1, 0x0100, + WM8766_RDA2, 0x0100, + WM8766_RDA3, 0x0100, + WM8766_MUTE1, 0x0000, + WM8766_MUTE2, 0x0000, + }; + + + unsigned int i; + + ice->vt1720 = 0; + ice->vt1724 = 1; + + ice->num_total_dacs = 8; + ice->num_total_adcs = 1; + ice->akm_codecs = 2; + + /* HACK - use this as the SPDIF source. + * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten + */ + ice->gpio.saved[0] = 0; + /* to remeber the register values */ + + ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); + if (! ice->akm) + return -ENOMEM; + ice->akm_codecs = 1; + + /* initialize WM8776 codec */ + for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2) + wm_put(ice, wm_inits[i], wm_inits[i+1]); + schedule_timeout_uninterruptible(1); + for (i = 0; i < ARRAY_SIZE(wm_inits2); i += 2) + wm_put(ice, wm_inits2[i], wm_inits2[i+1]); + + /* initialize WM8766 codec */ + + for (i = 0; i < ARRAY_SIZE(wm8766_inits); i += 2) + wm8766_spi_write(ice, wm8766_inits[i], wm8766_inits[i+1]); + + + return 0; +} + + +/* + * initialize the chip + */ +static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) +{ + static unsigned short ak4396_inits[] = { + AK4396_CTRL1, 0x87, /* I2S Normal Mode, 24 bit */ + AK4396_CTRL2, 0x02, + AK4396_CTRL3, 0x00, + AK4396_LCH_ATT, 0x00, + AK4396_RCH_ATT, 0x00, + }; + + + unsigned int i; + + ice->vt1720 = 0; + ice->vt1724 = 1; + + ice->num_total_dacs = 1; + ice->num_total_adcs = 1; + ice->akm_codecs = 1; + + /* HACK - use this as the SPDIF source. + * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten + */ + ice->gpio.saved[0] = 0; + /* to remeber the register values */ + + ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); + if (! ice->akm) + return -ENOMEM; + ice->akm_codecs = 1; + + /* initialize ak4396 codec */ + /* reset codec */ + ak4396_write(ice, AK4396_CTRL1, 0x86); + msleep(100); + ak4396_write(ice, AK4396_CTRL1, 0x87); + + for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2) + ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]); + + return 0; +} + + +static unsigned char prodigy71hifi_eeprom[] __devinitdata = { + 0x4b, /* SYSCONF: clock 512, spdif-in/ADC, 4DACs */ + 0x80, /* ACLINK: I2S */ + 0xfc, /* I2S: vol, 96k, 24bit, 192k */ + 0xc3, /* SPDIF: out-en, out-int, spdif-in */ + 0xff, /* GPIO_DIR */ + 0xff, /* GPIO_DIR1 */ + 0x5f, /* GPIO_DIR2 */ + 0x00, /* GPIO_MASK */ + 0x00, /* GPIO_MASK1 */ + 0x00, /* GPIO_MASK2 */ + 0x00, /* GPIO_STATE */ + 0x00, /* GPIO_STATE1 */ + 0x00, /* GPIO_STATE2 */ +}; + +static unsigned char prodigyhd2_eeprom[] __devinitdata = { + 0x4b, /* SYSCONF: clock 512, spdif-in/ADC, 4DACs */ + 0x80, /* ACLINK: I2S */ + 0xfc, /* I2S: vol, 96k, 24bit, 192k */ + 0xc3, /* SPDIF: out-en, out-int, spdif-in */ + 0xff, /* GPIO_DIR */ + 0xff, /* GPIO_DIR1 */ + 0x5f, /* GPIO_DIR2 */ + 0x00, /* GPIO_MASK */ + 0x00, /* GPIO_MASK1 */ + 0x00, /* GPIO_MASK2 */ + 0x00, /* GPIO_STATE */ + 0x00, /* GPIO_STATE1 */ + 0x00, /* GPIO_STATE2 */ +}; + +static unsigned char fortissimo4_eeprom[] __devinitdata = { + 0x43, /* SYSCONF: clock 512, ADC, 4DACs */ + 0x80, /* ACLINK: I2S */ + 0xfc, /* I2S: vol, 96k, 24bit, 192k */ + 0xc1, /* SPDIF: out-en, out-int */ + 0xff, /* GPIO_DIR */ + 0xff, /* GPIO_DIR1 */ + 0x5f, /* GPIO_DIR2 */ + 0x00, /* GPIO_MASK */ + 0x00, /* GPIO_MASK1 */ + 0x00, /* GPIO_MASK2 */ + 0x00, /* GPIO_STATE */ + 0x00, /* GPIO_STATE1 */ + 0x00, /* GPIO_STATE2 */ +}; + +/* entry point */ +struct snd_ice1712_card_info snd_vt1724_prodigy_hifi_cards[] __devinitdata = { + { + .subvendor = VT1724_SUBDEVICE_PRODIGY_HIFI, + .name = "Audiotrak Prodigy 7.1 HiFi", + .model = "prodigy71hifi", + .chip_init = prodigy_hifi_init, + .build_controls = prodigy_hifi_add_controls, + .eeprom_size = sizeof(prodigy71hifi_eeprom), + .eeprom_data = prodigy71hifi_eeprom, + .driver = "Prodigy71HIFI", + }, + { + .subvendor = VT1724_SUBDEVICE_PRODIGY_HD2, + .name = "Audiotrak Prodigy HD2", + .model = "prodigyhd2", + .chip_init = prodigy_hd2_init, + .build_controls = prodigy_hd2_add_controls, + .eeprom_size = sizeof(prodigyhd2_eeprom), + .eeprom_data = prodigyhd2_eeprom, + .driver = "Prodigy71HD2", + }, + { + .subvendor = VT1724_SUBDEVICE_FORTISSIMO4, + .name = "Hercules Fortissimo IV", + .model = "fortissimo4", + .chip_init = prodigy_hifi_init, + .build_controls = prodigy_hifi_add_controls, + .eeprom_size = sizeof(fortissimo4_eeprom), + .eeprom_data = fortissimo4_eeprom, + .driver = "Fortissimo4", + }, + { } /* terminator */ +}; + -- cgit v0.10.2 From b8c5b53e1f104a9f877f947cdb86d5c7208d6c9a Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Jan 2008 20:12:34 +0100 Subject: [ALSA] ice1712 - fixed midi input for Hoontech C-Ports See ALSA bug#1846 for more details. Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index b2cfba16..b042e5c 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -205,13 +205,13 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) snd_ice1712_stdsp24_darear(ice, (ice->spec.hoontech.config & ICE1712_STDSP24_DAREAR) ? 1 : 0); snd_ice1712_stdsp24_mute(ice, (ice->spec.hoontech.config & ICE1712_STDSP24_MUTE) ? 1 : 0); snd_ice1712_stdsp24_insel(ice, (ice->spec.hoontech.config & ICE1712_STDSP24_INSEL) ? 1 : 0); - for (box = 0; box < 4; box++) { + for (box = 0; box < 1; box++) { + if (ice->spec.hoontech.boxconfig[box] & ICE1712_STDSP24_BOX_MIDI2) + snd_ice1712_stdsp24_midi2(ice, 1); for (chn = 0; chn < 4; chn++) snd_ice1712_stdsp24_box_channel(ice, box, chn, (ice->spec.hoontech.boxconfig[box] & (1 << chn)) ? 1 : 0); snd_ice1712_stdsp24_box_midi(ice, box, (ice->spec.hoontech.boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1) ? 1 : 0); - if (ice->spec.hoontech.boxconfig[box] & ICE1712_STDSP24_BOX_MIDI2) - snd_ice1712_stdsp24_midi2(ice, 1); } return 0; -- cgit v0.10.2 From 05855ba3f405d02c4530072527d2b1c72e3b38a9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 17 Jan 2008 09:05:09 +0100 Subject: [ALSA] oxygen: make the I2S format configurable Add proper register bit symbols for the I2S format field, and allow card models to configure the I2S format to be used for the DACs and ADCs. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index f8e3fd3..e618cde 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -285,6 +285,8 @@ static const struct oxygen_model model_generic = { OXYGEN_CHANNEL_MULTICH | OXYGEN_CHANNEL_AC97, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; static const struct oxygen_model model_meridian = { .shortname = "C-Media CMI8788", @@ -304,6 +306,8 @@ static const struct oxygen_model model_meridian = { OXYGEN_CHANNEL_MULTICH | OXYGEN_CHANNEL_AC97, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; static int __devinit generic_oxygen_probe(struct pci_dev *pci, diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 602105c..98cccc6 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -86,6 +86,8 @@ struct oxygen_model { void (*update_dac_mute)(struct oxygen *chip); u8 used_channels; u8 function_flags; + u16 dac_i2s_format; + u16 adc_i2s_format; }; /* oxygen_lib.c */ diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 5515c75..f24659a 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -313,12 +313,12 @@ static unsigned int oxygen_i2s_magic2(struct snd_pcm_hw_params *hw_params) return params_rate(hw_params) <= 96000 ? 0x10 : 0x00; } -static unsigned int oxygen_i2s_format(struct snd_pcm_hw_params *hw_params) +static unsigned int oxygen_i2s_bits(struct snd_pcm_hw_params *hw_params) { if (params_format(hw_params) == SNDRV_PCM_FORMAT_S32_LE) - return OXYGEN_I2S_FORMAT_24; + return OXYGEN_I2S_BITS_24; else - return OXYGEN_I2S_FORMAT_16; + return OXYGEN_I2S_BITS_16; } static unsigned int oxygen_play_channels(struct snd_pcm_hw_params *hw_params) @@ -386,13 +386,15 @@ static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, oxygen_format(hw_params) << OXYGEN_REC_FORMAT_A_SHIFT, OXYGEN_REC_FORMAT_A_MASK); - oxygen_write8_masked(chip, OXYGEN_I2S_A_FORMAT, - oxygen_rate(hw_params) | - oxygen_i2s_magic2(hw_params) | - oxygen_i2s_format(hw_params), - OXYGEN_I2S_RATE_MASK | - OXYGEN_I2S_MAGIC2_MASK | - OXYGEN_I2S_FORMAT_MASK); + oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, + oxygen_rate(hw_params) | + oxygen_i2s_magic2(hw_params) | + chip->model->adc_i2s_format | + oxygen_i2s_bits(hw_params), + OXYGEN_I2S_RATE_MASK | + OXYGEN_I2S_FORMAT_MASK | + OXYGEN_I2S_MAGIC2_MASK | + OXYGEN_I2S_BITS_MASK); oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x08); spin_unlock_irq(&chip->reg_lock); @@ -416,13 +418,15 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, oxygen_format(hw_params) << OXYGEN_REC_FORMAT_B_SHIFT, OXYGEN_REC_FORMAT_B_MASK); - oxygen_write8_masked(chip, OXYGEN_I2S_B_FORMAT, - oxygen_rate(hw_params) | - oxygen_i2s_magic2(hw_params) | - oxygen_i2s_format(hw_params), - OXYGEN_I2S_RATE_MASK | - OXYGEN_I2S_MAGIC2_MASK | - OXYGEN_I2S_FORMAT_MASK); + oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT, + oxygen_rate(hw_params) | + oxygen_i2s_magic2(hw_params) | + chip->model->adc_i2s_format | + oxygen_i2s_bits(hw_params), + OXYGEN_I2S_RATE_MASK | + OXYGEN_I2S_FORMAT_MASK | + OXYGEN_I2S_MAGIC2_MASK | + OXYGEN_I2S_BITS_MASK); oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x10); spin_unlock_irq(&chip->reg_lock); @@ -493,8 +497,12 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, oxygen_format(hw_params) << OXYGEN_MULTICH_FORMAT_SHIFT, OXYGEN_MULTICH_FORMAT_MASK); oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, - oxygen_rate(hw_params) | oxygen_i2s_format(hw_params), - OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK); + oxygen_rate(hw_params) | + chip->model->dac_i2s_format | + oxygen_i2s_bits(hw_params), + OXYGEN_I2S_RATE_MASK | + OXYGEN_I2S_FORMAT_MASK | + OXYGEN_I2S_BITS_MASK); oxygen_clear_bits16(chip, OXYGEN_PLAY_ROUTING, 0x001f); oxygen_update_dac_routing(chip); oxygen_update_spdif_source(chip); diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index 7a4726d..b3491f7 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -105,18 +105,20 @@ #define OXYGEN_RATE_96000 0x0005 #define OXYGEN_RATE_176400 0x0006 #define OXYGEN_RATE_192000 0x0007 -#define OXYGEN_I2S_MAGIC1_MASK 0x0008 +#define OXYGEN_I2S_FORMAT_MASK 0x0008 +#define OXYGEN_I2S_FORMAT_I2S 0x0000 +#define OXYGEN_I2S_FORMAT_LJUST 0x0008 #define OXYGEN_I2S_MAGIC2_MASK 0x0030 -#define OXYGEN_I2S_FORMAT_MASK 0x00c0 -#define OXYGEN_I2S_FORMAT_16 0x0000 -#define OXYGEN_I2S_FORMAT_20 0x0040 -#define OXYGEN_I2S_FORMAT_24 0x0080 -#define OXYGEN_I2S_FORMAT_32 0x00c0 +#define OXYGEN_I2S_BITS_MASK 0x00c0 +#define OXYGEN_I2S_BITS_16 0x0000 +#define OXYGEN_I2S_BITS_20 0x0040 +#define OXYGEN_I2S_BITS_24 0x0080 +#define OXYGEN_I2S_BITS_32 0x00c0 #define OXYGEN_I2S_A_FORMAT 0x62 #define OXYGEN_I2S_B_FORMAT 0x64 #define OXYGEN_I2S_C_FORMAT 0x66 -/* OXYGEN_I2S_RATE_* and OXYGEN_I2S_FORMAT_* */ +/* like OXYGEN_I2S_MULTICH_FORMAT */ #define OXYGEN_SPDIF_CONTROL 0x70 #define OXYGEN_SPDIF_OUT_ENABLE 0x00000002 diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bea34f1..83c2c43 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -102,10 +102,6 @@ static void xonar_init(struct oxygen *chip) oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, 0x8c); oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, 0x00, 0x8c); -#if 0 - oxygen_clear_bits16(chip, OXYGEN_I2S_MULTICH_FORMAT, - OXYGEN_I2S_MAGIC1_MASK); -#endif oxygen_ac97_set_bits(chip, 0, 0x62, 0x0080); msleep(300); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, 0x100); @@ -253,6 +249,8 @@ static const struct oxygen_model model_xonar = { OXYGEN_CHANNEL_SPDIF | OXYGEN_CHANNEL_MULTICH, .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; static int __devinit xonar_probe(struct pci_dev *pci, -- cgit v0.10.2 From 88d1612cc22dcd4e336bfa5cc459c61551274550 Mon Sep 17 00:00:00 2001 From: Julian Scheel Date: Thu, 17 Jan 2008 11:12:13 +0100 Subject: [ALSA] ice1724 - Add missing prodigy_hifi.h Signed-off-by: Julian Scheel Signed-off-by: Konstantin Kletschke Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ice1712/prodigy_hifi.h b/sound/pci/ice1712/prodigy_hifi.h new file mode 100644 index 0000000..a4415d4 --- /dev/null +++ b/sound/pci/ice1712/prodigy_hifi.h @@ -0,0 +1,38 @@ +#ifndef __SOUND_PRODIGY_HIFI_H +#define __SOUND_PRODIGY_HIFI_H + +/* + * ALSA driver for VIA VT1724 (Envy24HT) + * + * Lowlevel functions for Audiotrak Prodigy Hifi + * + * Copyright (c) 2004 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#define PRODIGY_HIFI_DEVICE_DESC "{Audiotrak,Prodigy 7.1 HIFI},"\ + "{Audiotrak Prodigy HD2},"\ + "{Hercules Fortissimo IV}," + +#define VT1724_SUBDEVICE_PRODIGY_HIFI 0x38315441 /* PRODIGY 7.1 HIFI */ +#define VT1724_SUBDEVICE_PRODIGY_HD2 0x37315441 /* PRODIGY HD2 */ +#define VT1724_SUBDEVICE_FORTISSIMO4 0x81160100 /* Fortissimo IV */ + + +extern struct snd_ice1712_card_info snd_vt1724_prodigy_hifi_cards[]; + +#endif /* __SOUND_PRODIGY_HIFI_H */ -- cgit v0.10.2 From 178532348c215d23d8d64548c6ff0e38b50fca6c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Jan 2008 11:17:39 +0100 Subject: [ALSA] Add missing model for HD-audio Cx5045 codec Added the description of the model fujitsu for Cx5045 codec chip. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 99b39c8..bbe4b7c 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -946,6 +946,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Conexant 5045 laptop Laptop config + fujitsu Fujitsu Si1520 laptop test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y -- cgit v0.10.2 From 5218c892652c1e8f89964a7fd4cf8b71cc863094 Mon Sep 17 00:00:00 2001 From: Jiang Zhe Date: Thu, 17 Jan 2008 11:18:41 +0100 Subject: [ALSA] hda-codec - New model for conexant 5045 codec to support benq r55e The benq r55e laptop have 3 jacks on the front panel. One for HP, one for Line In and one for Mic In. This patch implemented a new model to support it. Signed-off-by: Jiang Zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index bbe4b7c..66390d7 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -947,6 +947,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Conexant 5045 laptop Laptop config fujitsu Fujitsu Si1520 laptop + benq Benq R55E test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8b8de96..23a3be5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -457,6 +457,15 @@ static struct hda_input_mux cxt5045_capture_source = { } }; +static struct hda_input_mux cxt5045_capture_source_benq = { + .num_items = 3, + .items = { + { "IntMic", 0x1 }, + { "ExtMic", 0x2 }, + { "LineIn", 0x3 }, + } +}; + /* turn on/off EAPD (+ mute HP) as a master switch */ static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -576,6 +585,15 @@ static struct snd_kcontrol_new cxt5045_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5045_benq_mixers[] = { + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + + {} +}; + static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -601,6 +619,30 @@ static struct hda_verb cxt5045_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5045_benq_init_verbs[] = { + /* Int Mic, Mic */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, + /* Line In,HP, Amp */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x11, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Record selector: Int mic */ + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, + /* SPDIF route: PCM */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + { } /* end */ +}; static struct hda_verb cxt5045_hp_sense_init_verbs[] = { /* pin sensing on HP jack */ @@ -741,6 +783,7 @@ static int cxt5045_init(struct hda_codec *codec) enum { CXT5045_LAPTOP, /* Laptops w/ EAPD support */ CXT5045_FUJITSU, /* Laptops w/ EAPD support */ + CXT5045_BENQ, #ifdef CONFIG_SND_DEBUG CXT5045_TEST, #endif @@ -750,6 +793,7 @@ enum { static const char *cxt5045_models[CXT5045_MODELS] = { [CXT5045_LAPTOP] = "laptop", [CXT5045_FUJITSU] = "fujitsu", + [CXT5045_BENQ] = "benq", #ifdef CONFIG_SND_DEBUG [CXT5045_TEST] = "test", #endif @@ -763,6 +807,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP), @@ -819,6 +864,16 @@ static int patch_cxt5045(struct hda_codec *codec) spec->mixers[0] = cxt5045_mixers; codec->patch_ops.init = cxt5045_init; break; + case CXT5045_BENQ: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; + spec->input_mux = &cxt5045_capture_source_benq; + spec->num_init_verbs = 1; + spec->init_verbs[0] = cxt5045_benq_init_verbs; + spec->mixers[0] = cxt5045_mixers; + spec->mixers[1] = cxt5045_benq_mixers; + spec->num_mixers = 2; + codec->patch_ops.init = cxt5045_init; + break; #ifdef CONFIG_SND_DEBUG case CXT5045_TEST: spec->input_mux = &cxt5045_test_capture_source; -- cgit v0.10.2 From f4beee9420773faefd3eac23bb5ba65b0395c1a1 Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Thu, 17 Jan 2008 11:19:26 +0100 Subject: [ALSA] hda-codec - Fix capture source for Cx5045 codec For codec conexant 5045, I found that the name of 'Capture Source Items' is different from the name of mixer. The mixer is: HDA_CODEC_VOLUME('Ext Mic Playback Volume', 0x17, 0x2, HDA_INPUT), HDA_CODEC_MUTE('Ext Mic Playback Switch', 0x17, 0x2, HDA_INPUT), But the capture source item is : static struct hda_input_mux cxt5045_capture_source = { .num_items = 2, .items = { { 'IntMic', 0x1 }, { 'LineIn', 0x2 }, } }; I think that it's better to change the name of capture_source to avoid misunderstanding. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 23a3be5..dab2ce1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -453,7 +453,7 @@ static struct hda_input_mux cxt5045_capture_source = { .num_items = 2, .items = { { "IntMic", 0x1 }, - { "LineIn", 0x2 }, + { "ExtMic", 0x2 }, } }; -- cgit v0.10.2 From 4052ce4cbf48531bdd8ff43b673ccb5c005dec79 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 17 Jan 2008 17:44:49 +0100 Subject: [ALSA] mpc8610: Add mmap support Enable mmap support in the MPC8610 ASoC driver. The driver can use ALSA's default mmap functionality, it was just not enabled previously. Signed-off-by: Timur Tabi Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 2173203..652514f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -136,7 +136,9 @@ struct fsl_dma_private { */ static const struct snd_pcm_hardware fsl_dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED, + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, -- cgit v0.10.2 From c2353a0826d2b8fe9f5c6a6aca99149e4ee7b196 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Jan 2008 09:17:53 +0100 Subject: [ALSA] oxygen: add register definitions Add more symbols for registers and register fields. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index e618cde..e0e54ab 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -102,18 +102,21 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, static const u8 codec_spi_map[4] = { 0, 1, 2, 4 }; - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER_WRITE | + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | (codec_spi_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | - OXYGEN_SPI_MAGIC, + OXYGEN_SPI_CEN_LATCH_CLOCK_HI, AK4396_WRITE | (reg << 8) | value); } static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) { - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER_WRITE | + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | - (3 << OXYGEN_SPI_CODEC_SHIFT), + OXYGEN_SPI_CLOCK_160 | + (3 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); } diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 616087c..ebafc65 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -119,7 +119,7 @@ void oxygen_write_ac97(struct oxygen *chip, unsigned int codec, udelay(5); oxygen_write32(chip, OXYGEN_AC97_REGS, reg); /* require two "completed" writes, just to be sure */ - if (oxygen_ac97_wait(chip, OXYGEN_AC97_WRITE_COMPLETE) >= 0 && + if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_WRITE_DONE) >= 0 && ++succeeded >= 2) return; } @@ -141,7 +141,7 @@ u16 oxygen_read_ac97(struct oxygen *chip, unsigned int codec, udelay(5); oxygen_write32(chip, OXYGEN_AC97_REGS, reg); udelay(10); - if (oxygen_ac97_wait(chip, OXYGEN_AC97_READ_COMPLETE) >= 0) { + if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_READ_DONE) >= 0) { u16 value = oxygen_read16(chip, OXYGEN_AC97_REGS); /* we require two consecutive reads of the same value */ if (value == last_read) diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 4edf40b..79e7c16 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -51,11 +51,11 @@ static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) OXYGEN_CHANNEL_SPDIF | OXYGEN_CHANNEL_MULTICH | OXYGEN_CHANNEL_AC97 | - OXYGEN_INT_SPDIF_IN_CHANGE | + OXYGEN_INT_SPDIF_IN_DETECT | OXYGEN_INT_GPIO); if (clear) { - if (clear & OXYGEN_INT_SPDIF_IN_CHANGE) - chip->interrupt_mask &= ~OXYGEN_INT_SPDIF_IN_CHANGE; + if (clear & OXYGEN_INT_SPDIF_IN_DETECT) + chip->interrupt_mask &= ~OXYGEN_INT_SPDIF_IN_DETECT; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask & ~clear); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, @@ -70,10 +70,10 @@ static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) if ((elapsed_streams & (1 << i)) && chip->streams[i]) snd_pcm_period_elapsed(chip->streams[i]); - if (status & OXYGEN_INT_SPDIF_IN_CHANGE) { + if (status & OXYGEN_INT_SPDIF_IN_DETECT) { spin_lock(&chip->reg_lock); i = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); - if (i & OXYGEN_SPDIF_IN_CHANGE) { + if (i & OXYGEN_SPDIF_RATE_INT) { oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, i); schedule_work(&chip->spdif_input_bits_work); } @@ -95,28 +95,32 @@ static void oxygen_spdif_input_bits_changed(struct work_struct *work) spdif_input_bits_work); spin_lock_irq(&chip->reg_lock); - oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_IN_INVERT); + oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_IN_CLOCK_96, + OXYGEN_SPDIF_IN_CLOCK_MASK); spin_unlock_irq(&chip->reg_lock); msleep(1); if (!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) - & OXYGEN_SPDIF_IN_VALID)) { + & OXYGEN_SPDIF_LOCK_STATUS)) { spin_lock_irq(&chip->reg_lock); - oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, - OXYGEN_SPDIF_IN_INVERT); + oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_IN_CLOCK_192, + OXYGEN_SPDIF_IN_CLOCK_MASK); spin_unlock_irq(&chip->reg_lock); msleep(1); if (!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) - & OXYGEN_SPDIF_IN_VALID)) { + & OXYGEN_SPDIF_LOCK_STATUS)) { spin_lock_irq(&chip->reg_lock); - oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, - OXYGEN_SPDIF_IN_INVERT); + oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_IN_CLOCK_96, + OXYGEN_SPDIF_IN_CLOCK_MASK); spin_unlock_irq(&chip->reg_lock); } } if (chip->controls[CONTROL_SPDIF_INPUT_BITS]) { spin_lock_irq(&chip->reg_lock); - chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_CHANGE; + chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); spin_unlock_irq(&chip->reg_lock); @@ -194,7 +198,8 @@ static void __devinit oxygen_init(struct oxygen *chip) chip->revision = 1; if (chip->revision == 1) - oxygen_set_bits8(chip, OXYGEN_MISC, OXYGEN_MISC_MAGIC); + oxygen_set_bits8(chip, OXYGEN_MISC, + OXYGEN_MISC_PCI_MEM_W_1_CLOCK); i = oxygen_read16(chip, OXYGEN_AC97_CONTROL); chip->has_ac97_0 = (i & OXYGEN_AC97_CODEC_0) != 0; @@ -207,7 +212,7 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, 0x010a); oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, 0x010a); oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, 0x010a); - oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_MAGIC2); + oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_RATE_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); oxygen_write16(chip, OXYGEN_PLAY_ROUTING, 0xe100); oxygen_write8(chip, OXYGEN_REC_ROUTING, 0x10); @@ -220,9 +225,17 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0x00); if (chip->has_ac97_0) { oxygen_clear_bits16(chip, OXYGEN_AC97_OUT_CONFIG, - OXYGEN_AC97_OUT_MAGIC3); + OXYGEN_AC97_CODEC0_FRONTL | + OXYGEN_AC97_CODEC0_FRONTR | + OXYGEN_AC97_CODEC0_SIDEL | + OXYGEN_AC97_CODEC0_SIDER | + OXYGEN_AC97_CODEC0_CENTER | + OXYGEN_AC97_CODEC0_BASE | + OXYGEN_AC97_CODEC0_REARL | + OXYGEN_AC97_CODEC0_REARR); oxygen_set_bits16(chip, OXYGEN_AC97_IN_CONFIG, - OXYGEN_AC97_IN_MAGIC3); + OXYGEN_AC97_CODEC0_LINEL | + OXYGEN_AC97_CODEC0_LINER); oxygen_write_ac97(chip, 0, AC97_RESET, 0); msleep(1); oxygen_ac97_set_bits(chip, 0, 0x70, 0x0300); @@ -349,7 +362,7 @@ int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, oxygen_proc_init(chip); spin_lock_irq(&chip->reg_lock); - chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_CHANGE; + chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); spin_unlock_irq(&chip->reg_lock); diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index f24659a..72481fd 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -308,9 +308,10 @@ static unsigned int oxygen_rate(struct snd_pcm_hw_params *hw_params) } } -static unsigned int oxygen_i2s_magic2(struct snd_pcm_hw_params *hw_params) +static unsigned int oxygen_i2s_mclk(struct snd_pcm_hw_params *hw_params) { - return params_rate(hw_params) <= 96000 ? 0x10 : 0x00; + return params_rate(hw_params) <= 96000 + ? OXYGEN_I2S_MCLK_256 : OXYGEN_I2S_MCLK_128; } static unsigned int oxygen_i2s_bits(struct snd_pcm_hw_params *hw_params) @@ -388,12 +389,12 @@ static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, OXYGEN_REC_FORMAT_A_MASK); oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, oxygen_rate(hw_params) | - oxygen_i2s_magic2(hw_params) | + oxygen_i2s_mclk(hw_params) | chip->model->adc_i2s_format | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | - OXYGEN_I2S_MAGIC2_MASK | + OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x08); spin_unlock_irq(&chip->reg_lock); @@ -420,12 +421,12 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, OXYGEN_REC_FORMAT_B_MASK); oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT, oxygen_rate(hw_params) | - oxygen_i2s_magic2(hw_params) | + oxygen_i2s_mclk(hw_params) | chip->model->adc_i2s_format | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | - OXYGEN_I2S_MAGIC2_MASK | + OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x10); spin_unlock_irq(&chip->reg_lock); @@ -514,24 +515,6 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, return 0; } -static int oxygen_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct oxygen *chip = snd_pcm_substream_chip(substream); - int err; - - err = oxygen_hw_params(substream, hw_params); - if (err < 0) - return err; - - spin_lock_irq(&chip->reg_lock); - oxygen_write8_masked(chip, OXYGEN_PLAY_FORMAT, - oxygen_format(hw_params) << OXYGEN_AC97_FORMAT_SHIFT, - OXYGEN_AC97_FORMAT_MASK); - spin_unlock_irq(&chip->reg_lock); - return 0; -} - static int oxygen_hw_free(struct snd_pcm_substream *substream) { struct oxygen *chip = snd_pcm_substream_chip(substream); @@ -680,7 +663,7 @@ static struct snd_pcm_ops oxygen_ac97_ops = { .open = oxygen_ac97_open, .close = oxygen_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = oxygen_ac97_hw_params, + .hw_params = oxygen_hw_params, .hw_free = oxygen_hw_free, .prepare = oxygen_prepare, .trigger = oxygen_trigger, diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index b3491f7..530f148 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -23,8 +23,8 @@ /* multichannel playback channel */ #define OXYGEN_DMA_MULTICH_ADDRESS 0x20 -#define OXYGEN_DMA_MULTICH_COUNT 0x24 /* 32 bits */ -#define OXYGEN_DMA_MULTICH_TCOUNT 0x28 /* 32 bits */ +#define OXYGEN_DMA_MULTICH_COUNT 0x24 /* 24 bits */ +#define OXYGEN_DMA_MULTICH_TCOUNT 0x28 /* 24 bits */ /* AC'97 (front panel) playback channel */ #define OXYGEN_DMA_AC97_ADDRESS 0x30 @@ -41,6 +41,9 @@ #define OXYGEN_CHANNEL_MULTICH 0x10 #define OXYGEN_CHANNEL_AC97 0x20 +#define OXYGEN_DMA_PAUSE 0x41 /* 1 = pause */ +/* OXYGEN_CHANNEL_* */ + #define OXYGEN_DMA_RESET 0x42 /* OXYGEN_CHANNEL_* */ @@ -50,19 +53,37 @@ #define OXYGEN_PLAY_CHANNELS_4 0x01 #define OXYGEN_PLAY_CHANNELS_6 0x02 #define OXYGEN_PLAY_CHANNELS_8 0x03 +#define OXYGEN_DMA_A_BURST_MASK 0x04 +#define OXYGEN_DMA_A_BURST_8 0x00 /* dwords */ +#define OXYGEN_DMA_A_BURST_16 0x04 +#define OXYGEN_DMA_MULTICH_BURST_MASK 0x08 +#define OXYGEN_DMA_MULTICH_BURST_8 0x00 +#define OXYGEN_DMA_MULTICH_BURST_16 0x08 #define OXYGEN_INTERRUPT_MASK 0x44 /* OXYGEN_CHANNEL_* */ -#define OXYGEN_INT_SPDIF_IN_CHANGE 0x0100 +#define OXYGEN_INT_SPDIF_IN_DETECT 0x0100 +#define OXYGEN_INT_MCU 0x0200 +#define OXYGEN_INT_2WIRE 0x0400 #define OXYGEN_INT_GPIO 0x0800 +#define OXYGEN_INT_MCB 0x2000 +#define OXYGEN_INT_AC97 0x4000 #define OXYGEN_INTERRUPT_STATUS 0x46 /* OXYGEN_CHANNEL_* amd OXYGEN_INT_* */ #define OXYGEN_INT_MIDI 0x1000 #define OXYGEN_MISC 0x48 -#define OXYGEN_MISC_MAGIC 0x20 +#define OXYGEN_MISC_WRITE_PCI_SUBID 0x01 +#define OXYGEN_MISC_LATENCY_3F 0x02 +#define OXYGEN_MISC_REC_C_FROM_SPDIF 0x04 +#define OXYGEN_MISC_REC_B_FROM_AC97 0x08 +#define OXYGEN_MISC_REC_A_FROM_MULTICH 0x10 +#define OXYGEN_MISC_PCI_MEM_W_1_CLOCK 0x20 #define OXYGEN_MISC_MIDI 0x40 +#define OXYGEN_MISC_CRYSTAL_MASK 0x80 +#define OXYGEN_MISC_CRYSTAL_24576 0x00 +#define OXYGEN_MISC_CRYSTAL_27 0x80 /* MHz */ #define OXYGEN_REC_FORMAT 0x4a #define OXYGEN_REC_FORMAT_A_MASK 0x03 @@ -80,23 +101,32 @@ #define OXYGEN_SPDIF_FORMAT_SHIFT 0 #define OXYGEN_MULTICH_FORMAT_MASK 0x0c #define OXYGEN_MULTICH_FORMAT_SHIFT 2 -#define OXYGEN_AC97_FORMAT_MASK 0x30 -#define OXYGEN_AC97_FORMAT_SHIFT 4 /* OXYGEN_FORMAT_* */ #define OXYGEN_REC_CHANNELS 0x4c -#define OXYGEN_REC_A_CHANNELS_MASK 0x07 -#define OXYGEN_REC_CHANNELS_2 0x00 -#define OXYGEN_REC_CHANNELS_4 0x01 -#define OXYGEN_REC_CHANNELS_6 0x03 /* or 0x02 */ -#define OXYGEN_REC_CHANNELS_8 0x04 +#define OXYGEN_REC_CHANNELS_MASK 0x07 +#define OXYGEN_REC_CHANNELS_2_2_2 0x00 /* DMA A, B, C */ +#define OXYGEN_REC_CHANNELS_4_2_2 0x01 +#define OXYGEN_REC_CHANNELS_6_0_2 0x02 +#define OXYGEN_REC_CHANNELS_6_2_0 0x03 +#define OXYGEN_REC_CHANNELS_8_0_0 0x04 #define OXYGEN_FUNCTION 0x50 +#define OXYGEN_FUNCTION_CLOCK_MASK 0x01 +#define OXYGEN_FUNCTION_CLOCK_PLL 0x00 +#define OXYGEN_FUNCTION_CLOCK_CRYSTAL 0x01 #define OXYGEN_FUNCTION_RESET_CODEC 0x02 -#define OXYGEN_FUNCTION_ENABLE_SPI_4_5 0x80 +#define OXYGEN_FUNCTION_RESET_POL 0x04 +#define OXYGEN_FUNCTION_PWDN 0x08 +#define OXYGEN_FUNCTION_PWDN_EN 0x10 +#define OXYGEN_FUNCTION_PWDN_POL 0x20 +#define OXYGEN_FUNCTION_2WIRE_SPI_MASK 0x40 +#define OXYGEN_FUNCTION_SPI 0x00 +#define OXYGEN_FUNCTION_2WIRE 0x40 +#define OXYGEN_FUNCTION_ENABLE_SPI_4_5 0x80 /* 0 = EEPROM */ #define OXYGEN_I2S_MULTICH_FORMAT 0x60 -#define OXYGEN_I2S_RATE_MASK 0x0007 +#define OXYGEN_I2S_RATE_MASK 0x0007 /* LRCK */ #define OXYGEN_RATE_32000 0x0000 #define OXYGEN_RATE_44100 0x0001 #define OXYGEN_RATE_48000 0x0002 @@ -108,12 +138,21 @@ #define OXYGEN_I2S_FORMAT_MASK 0x0008 #define OXYGEN_I2S_FORMAT_I2S 0x0000 #define OXYGEN_I2S_FORMAT_LJUST 0x0008 -#define OXYGEN_I2S_MAGIC2_MASK 0x0030 +#define OXYGEN_I2S_MCLK_MASK 0x0030 /* MCLK/LRCK */ +#define OXYGEN_I2S_MCLK_128 0x0000 +#define OXYGEN_I2S_MCLK_256 0x0010 +#define OXYGEN_I2S_MCLK_512 0x0020 #define OXYGEN_I2S_BITS_MASK 0x00c0 #define OXYGEN_I2S_BITS_16 0x0000 #define OXYGEN_I2S_BITS_20 0x0040 #define OXYGEN_I2S_BITS_24 0x0080 #define OXYGEN_I2S_BITS_32 0x00c0 +#define OXYGEN_I2S_MASTER 0x0100 +#define OXYGEN_I2S_BCLK_MASK 0x0600 /* BCLK/LRCK */ +#define OXYGEN_I2S_BCLK_64 0x0000 +#define OXYGEN_I2S_BCLK_128 0x0200 +#define OXYGEN_I2S_BCLK_256 0x0400 +#define OXYGEN_I2S_MUTE_MCLK 0x0800 #define OXYGEN_I2S_A_FORMAT 0x62 #define OXYGEN_I2S_B_FORMAT 0x64 @@ -122,12 +161,21 @@ #define OXYGEN_SPDIF_CONTROL 0x70 #define OXYGEN_SPDIF_OUT_ENABLE 0x00000002 -#define OXYGEN_SPDIF_LOOPBACK 0x00000004 -#define OXYGEN_SPDIF_MAGIC2 0x00000020 -#define OXYGEN_SPDIF_MAGIC3 0x00000040 -#define OXYGEN_SPDIF_IN_VALID 0x00001000 -#define OXYGEN_SPDIF_IN_CHANGE 0x00008000 /* r/wc */ -#define OXYGEN_SPDIF_IN_INVERT 0x00010000 /* ? */ +#define OXYGEN_SPDIF_LOOPBACK 0x00000004 /* in to out */ +#define OXYGEN_SPDIF_SENSE_MASK 0x00000008 +#define OXYGEN_SPDIF_LOCK_MASK 0x00000010 +#define OXYGEN_SPDIF_RATE_MASK 0x00000020 +#define OXYGEN_SPDIF_SPDVALID 0x00000040 +#define OXYGEN_SPDIF_SENSE_PAR 0x00000200 +#define OXYGEN_SPDIF_LOCK_PAR 0x00000400 +#define OXYGEN_SPDIF_SENSE_STATUS 0x00000800 +#define OXYGEN_SPDIF_LOCK_STATUS 0x00001000 +#define OXYGEN_SPDIF_SENSE_INT 0x00002000 /* r/wc */ +#define OXYGEN_SPDIF_LOCK_INT 0x00004000 /* r/wc */ +#define OXYGEN_SPDIF_RATE_INT 0x00008000 /* r/wc */ +#define OXYGEN_SPDIF_IN_CLOCK_MASK 0x00010000 +#define OXYGEN_SPDIF_IN_CLOCK_96 0x00000000 /* <= 96 kHz */ +#define OXYGEN_SPDIF_IN_CLOCK_192 0x00010000 /* > 96 kHz */ #define OXYGEN_SPDIF_OUT_RATE_MASK 0x07000000 #define OXYGEN_SPDIF_OUT_RATE_SHIFT 24 /* OXYGEN_RATE_* << OXYGEN_SPDIF_OUT_RATE_SHIFT */ @@ -146,10 +194,22 @@ #define OXYGEN_SPDIF_INPUT_BITS 0x78 /* 32 bits, IEC958_AES_* */ +#define OXYGEN_EEPROM_CONTROL 0x80 +#define OXYGEN_EEPROM_ADDRESS_MASK 0x7f +#define OXYGEN_EEPROM_DIR_MASK 0x80 +#define OXYGEN_EEPROM_DIR_READ 0x00 +#define OXYGEN_EEPROM_DIR_WRITE 0x80 + +#define OXYGEN_EEPROM_STATUS 0x81 +#define OXYGEN_EEPROM_VALID 0x40 +#define OXYGEN_EEPROM_BUSY 0x80 + +#define OXYGEN_EEPROM_DATA 0x82 /* 16 bits */ + #define OXYGEN_2WIRE_CONTROL 0x90 #define OXYGEN_2WIRE_DIR_MASK 0x01 -#define OXYGEN_2WIRE_DIR_WRITE 0x00 /* ? */ -#define OXYGEN_2WIRE_DIR_READ 0x01 /* ? */ +#define OXYGEN_2WIRE_DIR_WRITE 0x00 +#define OXYGEN_2WIRE_DIR_READ 0x01 #define OXYGEN_2WIRE_ADDRESS_MASK 0xfe /* slave device address */ #define OXYGEN_2WIRE_ADDRESS_SHIFT 1 @@ -157,17 +217,37 @@ #define OXYGEN_2WIRE_DATA 0x92 /* data, 16 bits */ #define OXYGEN_2WIRE_BUS_STATUS 0x94 -#define OXYGEN_2WIRE_BUSY 0x01 +#define OXYGEN_2WIRE_BUSY 0x0001 +#define OXYGEN_2WIRE_LENGTH_MASK 0x0002 +#define OXYGEN_2WIRE_LENGTH_8 0x0000 +#define OXYGEN_2WIRE_LENGTH_16 0x0002 +#define OXYGEN_2WIRE_MANUAL_READ 0x0004 /* 0 = auto read */ +#define OXYGEN_2WIRE_WRITE_MAP_ONLY 0x0008 +#define OXYGEN_2WIRE_SLAVE_AD_MASK 0x0030 /* AD0, AD1 */ +#define OXYGEN_2WIRE_INTERRUPT_MASK 0x0040 /* 0 = int. if not responding */ +#define OXYGEN_2WIRE_SLAVE_NO_RESPONSE 0x0080 +#define OXYGEN_2WIRE_SPEED_MASK 0x0100 +#define OXYGEN_2WIRE_SPEED_STANDARD 0x0000 +#define OXYGEN_2WIRE_SPEED_FAST 0x0100 +#define OXYGEN_2WIRE_CLOCK_SYNC 0x0200 +#define OXYGEN_2WIRE_BUS_RESET 0x0400 #define OXYGEN_SPI_CONTROL 0x98 #define OXYGEN_SPI_BUSY 0x01 /* read */ -#define OXYGEN_SPI_TRIGGER_WRITE 0x01 /* write */ +#define OXYGEN_SPI_TRIGGER 0x01 /* write */ #define OXYGEN_SPI_DATA_LENGTH_MASK 0x02 #define OXYGEN_SPI_DATA_LENGTH_2 0x00 #define OXYGEN_SPI_DATA_LENGTH_3 0x02 +#define OXYGEN_SPI_CLOCK_MASK 0xc0 +#define OXYGEN_SPI_CLOCK_160 0x00 /* ns */ +#define OXYGEN_SPI_CLOCK_320 0x40 +#define OXYGEN_SPI_CLOCK_640 0x80 +#define OXYGEN_SPI_CLOCK_1280 0xc0 #define OXYGEN_SPI_CODEC_MASK 0x70 /* 0..5 */ #define OXYGEN_SPI_CODEC_SHIFT 4 -#define OXYGEN_SPI_MAGIC 0x80 +#define OXYGEN_SPI_CEN_MASK 0x80 +#define OXYGEN_SPI_CEN_LATCH_CLOCK_LO 0x00 +#define OXYGEN_SPI_CEN_LATCH_CLOCK_HI 0x80 #define OXYGEN_SPI_DATA1 0x99 #define OXYGEN_SPI_DATA2 0x9a @@ -175,56 +255,161 @@ #define OXYGEN_MPU401 0xa0 +#define OXYGEN_MPU401_CONTROL 0xa2 +#define OXYGEN_MPU401_LOOPBACK 0x01 /* TXD to RXD */ + #define OXYGEN_GPI_DATA 0xa4 +/* bits 0..5 = pin XGPI0..XGPI5 */ #define OXYGEN_GPI_INTERRUPT_MASK 0xa5 +/* bits 0..5, 1 = enable */ #define OXYGEN_GPIO_DATA 0xa6 +/* bits 0..9 */ #define OXYGEN_GPIO_CONTROL 0xa8 -/* 0: input, 1: output */ +/* bits 0..9, 0 = input, 1 = output */ +#define OXYGEN_GPIO1_XSLAVE_RDY 0x8000 #define OXYGEN_GPIO_INTERRUPT_MASK 0xaa - -#define OXYGEN_DEVICE_SENSE 0xac /* ? */ +/* bits 0..9, 1 = enable */ + +#define OXYGEN_DEVICE_SENSE 0xac +#define OXYGEN_HEAD_PHONE_DETECT 0x01 +#define OXYGEN_HEAD_PHONE_MASK 0x06 +#define OXYGEN_HEAD_PHONE_PASSIVE_SPK 0x00 +#define OXYGEN_HEAD_PHONE_HP 0x02 +#define OXYGEN_HEAD_PHONE_ACTIVE_SPK 0x04 + +#define OXYGEN_MCU_2WIRE_DATA 0xb0 + +#define OXYGEN_MCU_2WIRE_MAP 0xb2 + +#define OXYGEN_MCU_2WIRE_STATUS 0xb3 +#define OXYGEN_MCU_2WIRE_BUSY 0x01 +#define OXYGEN_MCU_2WIRE_LENGTH_MASK 0x06 +#define OXYGEN_MCU_2WIRE_LENGTH_1 0x00 +#define OXYGEN_MCU_2WIRE_LENGTH_2 0x02 +#define OXYGEN_MCU_2WIRE_LENGTH_3 0x04 +#define OXYGEN_MCU_2WIRE_WRITE 0x08 /* r/wc */ +#define OXYGEN_MCU_2WIRE_READ 0x10 /* r/wc */ +#define OXYGEN_MCU_2WIRE_DRV_XACT_FAIL 0x20 /* r/wc */ +#define OXYGEN_MCU_2WIRE_RESET 0x40 + +#define OXYGEN_MCU_2WIRE_CONTROL 0xb4 +#define OXYGEN_MCU_2WIRE_DRV_ACK 0x01 +#define OXYGEN_MCU_2WIRE_DRV_XACT 0x02 +#define OXYGEN_MCU_2WIRE_INT_MASK 0x04 +#define OXYGEN_MCU_2WIRE_SYNC_MASK 0x08 +#define OXYGEN_MCU_2WIRE_SYNC_RDY_PIN 0x00 +#define OXYGEN_MCU_2WIRE_SYNC_DATA 0x08 +#define OXYGEN_MCU_2WIRE_ADDRESS_MASK 0x30 +#define OXYGEN_MCU_2WIRE_ADDRESS_10 0x00 +#define OXYGEN_MCU_2WIRE_ADDRESS_12 0x10 +#define OXYGEN_MCU_2WIRE_ADDRESS_14 0x20 +#define OXYGEN_MCU_2WIRE_ADDRESS_16 0x30 +#define OXYGEN_MCU_2WIRE_INT_POL 0x40 +#define OXYGEN_MCU_2WIRE_SYNC_ENABLE 0x80 #define OXYGEN_PLAY_ROUTING 0xc0 +#define OXYGEN_PLAY_MUTE01 0x0001 +#define OXYGEN_PLAY_MUTE23 0x0002 +#define OXYGEN_PLAY_MUTE45 0x0004 +#define OXYGEN_PLAY_MUTE67 0x0008 +#define OXYGEN_PLAY_MULTICH_MASK 0x0010 +#define OXYGEN_PLAY_MULTICH_I2S_DAC 0x0000 +#define OXYGEN_PLAY_MULTICH_AC97 0x0010 +#define OXYGEN_PLAY_SPDIF_MASK 0x00e0 +#define OXYGEN_PLAY_SPDIF_SPDIF 0x0000 +#define OXYGEN_PLAY_SPDIF_MULTICH_01 0x0020 +#define OXYGEN_PLAY_SPDIF_MULTICH_23 0x0040 +#define OXYGEN_PLAY_SPDIF_MULTICH_45 0x0060 +#define OXYGEN_PLAY_SPDIF_MULTICH_67 0x0080 +#define OXYGEN_PLAY_SPDIF_REC_A 0x00a0 +#define OXYGEN_PLAY_SPDIF_REC_B 0x00c0 +#define OXYGEN_PLAY_SPDIF_I2S_ADC_3 0x00e0 #define OXYGEN_PLAY_DAC0_SOURCE_MASK 0x0300 +#define OXYGEN_PLAY_DAC0_SOURCE_SHIFT 8 #define OXYGEN_PLAY_DAC1_SOURCE_MASK 0x0700 +#define OXYGEN_PLAY_DAC1_SOURCE_SHIFT 10 #define OXYGEN_PLAY_DAC2_SOURCE_MASK 0x3000 +#define OXYGEN_PLAY_DAC2_SOURCE_SHIFT 12 #define OXYGEN_PLAY_DAC3_SOURCE_MASK 0x7000 +#define OXYGEN_PLAY_DAC3_SOURCE_SHIFT 14 #define OXYGEN_REC_ROUTING 0xc2 +#define OXYGEN_MUTE_I2S_ADC_1 0x01 +#define OXYGEN_MUTE_I2S_ADC_2 0x02 +#define OXYGEN_MUTE_I2S_ADC_3 0x04 +#define OXYGEN_REC_A_ROUTE_MASK 0x08 +#define OXYGEN_REC_A_ROUTE_I2S_ADC_1 0x00 +#define OXYGEN_REC_A_ROUTE_AC97_0 0x08 +#define OXYGEN_REC_B_ROUTE_MASK 0x10 +#define OXYGEN_REC_B_ROUTE_I2S_ADC_2 0x00 +#define OXYGEN_REC_B_ROUTE_AC97_1 0x10 +#define OXYGEN_REC_C_ROUTE_MASK 0x20 +#define OXYGEN_REC_C_ROUTE_SPDIF 0x00 +#define OXYGEN_REC_C_ROUTE_I2S_ADC_3 0x20 #define OXYGEN_ADC_MONITOR 0xc3 -#define OXYGEN_ADC_MONITOR_MULTICH 0x01 -#define OXYGEN_ADC_MONITOR_AC97 0x04 -#define OXYGEN_ADC_MONITOR_SPDIF 0x10 +#define OXYGEN_ADC_MONITOR_A 0x01 +#define OXYGEN_ADC_MONITOR_A_HALF_VOL 0x02 +#define OXYGEN_ADC_MONITOR_B 0x04 +#define OXYGEN_ADC_MONITOR_B_HALF_VOL 0x08 +#define OXYGEN_ADC_MONITOR_C 0x10 +#define OXYGEN_ADC_MONITOR_C_HALF_VOL 0x20 #define OXYGEN_A_MONITOR_ROUTING 0xc4 +#define OXYGEN_A_MONITOR_ROUTE_01_MASK 0x03 +#define OXYGEN_A_MONITOR_ROUTE_23_MASK 0x0c +#define OXYGEN_A_MONITOR_ROUTE_45_MASK 0x30 +#define OXYGEN_A_MONITOR_ROUTE_67_MASK 0xc0 #define OXYGEN_AC97_CONTROL 0xd0 -#define OXYGEN_AC97_RESET1 0x0001 -#define OXYGEN_AC97_RESET1_BUSY 0x0002 -#define OXYGEN_AC97_RESET2 0x0008 +#define OXYGEN_AC97_COLD_RESET 0x0001 +#define OXYGEN_AC97_SUSPENDED 0x0002 /* read */ +#define OXYGEN_AC97_RESUME 0x0002 /* write */ +#define OXYGEN_AC97_CLOCK_DISABLE 0x0004 +#define OXYGEN_AC97_NO_CODEC_0 0x0008 #define OXYGEN_AC97_CODEC_0 0x0010 #define OXYGEN_AC97_CODEC_1 0x0020 #define OXYGEN_AC97_INTERRUPT_MASK 0xd2 +#define OXYGEN_AC97_INT_READ_DONE 0x01 +#define OXYGEN_AC97_INT_WRITE_DONE 0x02 +#define OXYGEN_AC97_INT_CODEC_0 0x10 +#define OXYGEN_AC97_INT_CODEC_1 0x20 #define OXYGEN_AC97_INTERRUPT_STATUS 0xd3 -#define OXYGEN_AC97_READ_COMPLETE 0x01 -#define OXYGEN_AC97_WRITE_COMPLETE 0x02 +/* OXYGEN_AC97_INT_* */ #define OXYGEN_AC97_OUT_CONFIG 0xd4 -#define OXYGEN_AC97_OUT_MAGIC1 0x00000011 -#define OXYGEN_AC97_OUT_MAGIC2 0x00000033 -#define OXYGEN_AC97_OUT_MAGIC3 0x0000ff00 +#define OXYGEN_AC97_CODEC1_SLOT3 0x00000001 +#define OXYGEN_AC97_CODEC1_SLOT3_VSR 0x00000002 +#define OXYGEN_AC97_CODEC1_SLOT4 0x00000010 +#define OXYGEN_AC97_CODEC1_SLOT4_VSR 0x00000020 +#define OXYGEN_AC97_CODEC0_FRONTL 0x00000100 +#define OXYGEN_AC97_CODEC0_FRONTR 0x00000200 +#define OXYGEN_AC97_CODEC0_SIDEL 0x00000400 +#define OXYGEN_AC97_CODEC0_SIDER 0x00000800 +#define OXYGEN_AC97_CODEC0_CENTER 0x00001000 +#define OXYGEN_AC97_CODEC0_BASE 0x00002000 +#define OXYGEN_AC97_CODEC0_REARL 0x00004000 +#define OXYGEN_AC97_CODEC0_REARR 0x00008000 #define OXYGEN_AC97_IN_CONFIG 0xd8 -#define OXYGEN_AC97_IN_MAGIC1 0x00000011 -#define OXYGEN_AC97_IN_MAGIC2 0x00000033 -#define OXYGEN_AC97_IN_MAGIC3 0x00000300 +#define OXYGEN_AC97_CODEC1_LINEL 0x00000001 +#define OXYGEN_AC97_CODEC1_LINEL_VSR 0x00000002 +#define OXYGEN_AC97_CODEC1_LINEL_16 0x00000000 +#define OXYGEN_AC97_CODEC1_LINEL_18 0x00000004 +#define OXYGEN_AC97_CODEC1_LINEL_20 0x00000008 +#define OXYGEN_AC97_CODEC1_LINER 0x00000010 +#define OXYGEN_AC97_CODEC1_LINER_VSR 0x00000020 +#define OXYGEN_AC97_CODEC1_LINER_16 0x00000000 +#define OXYGEN_AC97_CODEC1_LINER_18 0x00000040 +#define OXYGEN_AC97_CODEC1_LINER_20 0x00000080 +#define OXYGEN_AC97_CODEC0_LINEL 0x00000100 +#define OXYGEN_AC97_CODEC0_LINER 0x00000200 #define OXYGEN_AC97_REGS 0xdc #define OXYGEN_AC97_REG_DATA_MASK 0x0000ffff @@ -236,13 +421,29 @@ #define OXYGEN_AC97_REG_CODEC_MASK 0x01000000 #define OXYGEN_AC97_REG_CODEC_SHIFT 24 +#define OXYGEN_TEST 0xe0 +#define OXYGEN_TEST_RAM_SUCCEEDED 0x01 +#define OXYGEN_TEST_PLAYBACK_RAM 0x02 +#define OXYGEN_TEST_RECORD_RAM 0x04 +#define OXYGEN_TEST_PLL 0x08 +#define OXYGEN_TEST_2WIRE_LOOPBACK 0x10 + #define OXYGEN_DMA_FLUSH 0xe1 /* OXYGEN_CHANNEL_* */ #define OXYGEN_CODEC_VERSION 0xe4 +#define OXYGEN_XCID_MASK 0x07 #define OXYGEN_REVISION 0xe6 -#define OXYGEN_REVISION_2 0x08 /* bit flag */ -#define OXYGEN_REVISION_8787 0x14 /* all 8 bits */ +#define OXYGEN_REVISION_XPKGID_MASK 0x0007 +#define OXYGEN_REVISION_MASK 0xfff8 +#define OXYGEN_REVISION_2 0x0008 /* bit flag */ +#define OXYGEN_REVISION_8787 0x0014 /* 8 bits */ + +#define OXYGEN_OFFSIN_48K 0xe8 +#define OXYGEN_OFFSBASE_48K 0xe9 +#define OXYGEN_OFFSBASE_MASK 0x0fff +#define OXYGEN_OFFSIN_44K 0xec +#define OXYGEN_OFFSBASE_44K 0xed #endif diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 83c2c43..1a0367b 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -80,10 +80,11 @@ static void pcm1796_write(struct oxygen *chip, unsigned int codec, static const u8 codec_map[4] = { 0, 1, 2, 4 }; - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER_WRITE | + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | - OXYGEN_SPI_MAGIC, + OXYGEN_SPI_CEN_LATCH_CLOCK_HI, (reg << 8) | value); } -- cgit v0.10.2 From e0059549345903195d6eb796c22048204c40a785 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Jan 2008 09:18:32 +0100 Subject: [ALSA] oxygen: fix playback routing The default playback routing must be 0xe4, not 0xe1; the front and surround DACs were exchanged. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 79e7c16..3e61f94 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -214,7 +214,7 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, 0x010a); oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_RATE_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); - oxygen_write16(chip, OXYGEN_PLAY_ROUTING, 0xe100); + oxygen_write16(chip, OXYGEN_PLAY_ROUTING, 0xe400); oxygen_write8(chip, OXYGEN_REC_ROUTING, 0x10); oxygen_write8(chip, OXYGEN_ADC_MONITOR, 0x00); oxygen_write8(chip, OXYGEN_A_MONITOR_ROUTING, 0xe4); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index fae7c0f..6fd2a59 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -121,7 +121,7 @@ static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) void oxygen_update_dac_routing(struct oxygen *chip) { static const unsigned int reg_values[3] = { - 0xe100, /* front <- 0, surround <- 1, center <- 2, back <- 3 */ + 0xe400, /* front <- 0, surround <- 1, center <- 2, back <- 3 */ 0xe000, /* front <- 0, surround <- 0, center <- 2, back <- 3 */ 0x2000 /* front <- 0, surround <- 0, center <- 2, back <- 0 */ }; @@ -135,7 +135,7 @@ void oxygen_update_dac_routing(struct oxygen *chip) else if (channels == OXYGEN_PLAY_CHANNELS_8) reg_value = 0x6c00; /* surround <- 3, back <- 1 */ else - reg_value = 0xe100; + reg_value = 0xe400; oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value, 0xff00); } -- cgit v0.10.2 From 797760ab14db4e82a50c06a9916dd5c6147b415b Mon Sep 17 00:00:00 2001 From: Andrew Paprocki Date: Fri, 18 Jan 2008 12:51:11 +0100 Subject: [ALSA] hda_proc - Add a number of new settings to proc codec output This patch adds additional output to the /proc codec#X info. The following pieces of information are added to the output: - Balanced, L/R swap, trigger, impedance sense pin capabilities - Vref pin capabilities - Current Vref pin widget control setting - Default configuration association, sequence, and misc bit test - EAPD/BTL bits conveying balanced mode, EAPD, and L/R swap - Power state modified to show state name as well as setting vs actual value - GPIO parameter output on Audio Function Group, including enumeration of IO pins which are indicated present (Any I and O pins are not output at this time) - Stripe and L/R swap widget capabilities - All digital converter bits: enable, validity, validity config, preemphasis, copyright, non-audio, professional, generation level, and content category - Converter stream and channel values for in/out widgets - SDI select value for in widgets - Unsolicited response widget capability tag and enabled bit - Delay widget capability value - Processing widget capability benign bit and number of coefficients - Realtek Define Registers: processing coefficient, coefficient index [Also, fixed space/tab issues and make codes a bit more readable -- Takashi] Signed-off-by: Andrew Paprocki Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 719e46f..eb4a2ae 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -84,7 +84,9 @@ enum { #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18 #define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 +#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -112,7 +114,9 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_WAKE_MASK 0x718 #define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 +#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -185,6 +189,27 @@ enum { #define AC_SUPFMT_FLOAT32 (1<<1) #define AC_SUPFMT_AC3 (1<<2) +/* GP I/O count */ +#define AC_GPIO_IO_COUNT (0xff<<0) +#define AC_GPIO_O_COUNT (0xff<<8) +#define AC_GPIO_O_COUNT_SHIFT 8 +#define AC_GPIO_I_COUNT (0xff<<16) +#define AC_GPIO_I_COUNT_SHIFT 16 +#define AC_GPIO_UNSOLICITED (1<<30) +#define AC_GPIO_WAKE (1<<31) + +/* Converter stream, channel */ +#define AC_CONV_CHANNEL (0xf<<0) +#define AC_CONV_STREAM (0xf<<4) +#define AC_CONV_STREAM_SHIFT 4 + +/* Input converter SDI select */ +#define AC_SDI_SELECT (0xf<<0) + +/* Unsolicited response */ +#define AC_UNSOL_TAG (0x3f<<0) +#define AC_UNSOL_ENABLED (1<<7) + /* Pin widget capabilies */ #define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ #define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */ @@ -230,6 +255,9 @@ enum { #define AC_PWRST_D3SUP (1<<3) /* Power state values */ +#define AC_PWRST_SETTING (0xf<<0) +#define AC_PWRST_ACTUAL (0xf<<4) +#define AC_PWRST_ACTUAL_SHIFT 4 #define AC_PWRST_D0 0x00 #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 @@ -238,6 +266,7 @@ enum { /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) #define AC_PCAP_NUM_COEF (0xff<<8) +#define AC_PCAP_NUM_COEF_SHIFT 8 /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) @@ -274,6 +303,9 @@ enum { #define AC_DIG1_PROFESSIONAL (1<<6) #define AC_DIG1_LEVEL (1<<7) +/* DIGITAL2 bits */ +#define AC_DIG2_CC (0x7f<<0) + /* Pin widget control - 8bit */ #define AC_PINCTL_VREFEN (0x7<<0) #define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */ @@ -288,12 +320,22 @@ enum { /* Unsolicited response - 8bit */ #define AC_USRSP_EN (1<<7) +/* Pin sense - 32bit */ +#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff) +#define AC_PINSENSE_PRESENCE (1<<31) + +/* EAPD/BTL enable - 32bit */ +#define AC_EAPDBTL_BALANCED (1<<0) +#define AC_EAPDBTL_EAPD (1<<1) +#define AC_EAPDBTL_LR_SWAP (1<<2) + /* configuration default - 32bit */ #define AC_DEFCFG_SEQUENCE (0xf<<0) #define AC_DEFCFG_DEF_ASSOC (0xf<<4) #define AC_DEFCFG_ASSOC_SHIFT 4 #define AC_DEFCFG_MISC (0xf<<8) #define AC_DEFCFG_MISC_SHIFT 8 +#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0) #define AC_DEFCFG_COLOR (0xf<<12) #define AC_DEFCFG_COLOR_SHIFT 12 #define AC_DEFCFG_CONN_TYPE (0xf<<16) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 4b8d644..35a630d 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -202,7 +202,8 @@ static const char *get_jack_color(u32 cfg) } static void print_pin_caps(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) + struct hda_codec *codec, hda_nid_t nid, + int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; static char *jack_types[16] = { @@ -226,7 +227,45 @@ static void print_pin_caps(struct snd_info_buffer *buffer, snd_iprintf(buffer, " EAPD"); if (caps & AC_PINCAP_PRES_DETECT) snd_iprintf(buffer, " Detect"); + if (caps & AC_PINCAP_BALANCE) + snd_iprintf(buffer, " Balanced"); + if (caps & AC_PINCAP_LR_SWAP) + snd_iprintf(buffer, " R/L"); + if (caps & AC_PINCAP_TRIG_REQ) + snd_iprintf(buffer, " Trigger"); + if (caps & AC_PINCAP_IMP_SENSE) + snd_iprintf(buffer, " ImpSense"); snd_iprintf(buffer, "\n"); + if (caps & AC_PINCAP_VREF) { + unsigned int vref = + (caps & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + snd_iprintf(buffer, " Vref caps:"); + if (vref & AC_PINCAP_VREF_HIZ) + snd_iprintf(buffer, " HIZ"); + if (vref & AC_PINCAP_VREF_50) + snd_iprintf(buffer, " 50"); + if (vref & AC_PINCAP_VREF_GRD) + snd_iprintf(buffer, " GRD"); + if (vref & AC_PINCAP_VREF_80) + snd_iprintf(buffer, " 80"); + if (vref & AC_PINCAP_VREF_100) + snd_iprintf(buffer, " 100"); + snd_iprintf(buffer, "\n"); + *supports_vref = 1; + } else + *supports_vref = 0; + if (caps & AC_PINCAP_EAPD) { + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_EAPD_BTLENABLE, 0); + snd_iprintf(buffer, " EAPD 0x%x:", val); + if (val & AC_EAPDBTL_BALANCED) + snd_iprintf(buffer, " BALANCED"); + if (val & AC_EAPDBTL_EAPD) + snd_iprintf(buffer, " EAPD"); + if (val & AC_EAPDBTL_LR_SWAP) + snd_iprintf(buffer, " R/L"); + snd_iprintf(buffer, "\n"); + } caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], @@ -236,13 +275,233 @@ static void print_pin_caps(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps), get_jack_color(caps)); - if (caps & AC_PINCAP_EAPD) { - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_EAPD_BTLENABLE, 0); - snd_iprintf(buffer, " EAPD: 0x%x\n", val); + /* Default association and sequence values refer to default grouping + * of pin complexes and their sequence within the group. This is used + * for priority and resource allocation. + */ + snd_iprintf(buffer, " DefAssociation = 0x%x, Sequence = 0x%x\n", + (caps & AC_DEFCFG_DEF_ASSOC) >> AC_DEFCFG_ASSOC_SHIFT, + caps & AC_DEFCFG_SEQUENCE); + if (((caps & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) & + AC_DEFCFG_MISC_NO_PRESENCE) { + /* Miscellaneous bit indicates external hardware does not + * support presence detection even if the pin complex + * indicates it is supported. + */ + snd_iprintf(buffer, " Misc = NO_PRESENCE\n"); } } +static void print_pin_ctls(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid, + int supports_vref) +{ + unsigned int pinctls; + + pinctls = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_iprintf(buffer, " Pin-ctls: 0x%02x:", pinctls); + if (pinctls & AC_PINCTL_IN_EN) + snd_iprintf(buffer, " IN"); + if (pinctls & AC_PINCTL_OUT_EN) + snd_iprintf(buffer, " OUT"); + if (pinctls & AC_PINCTL_HP_EN) + snd_iprintf(buffer, " HP"); + if (supports_vref) { + int vref = pinctls & AC_PINCTL_VREFEN; + switch (vref) { + case AC_PINCTL_VREF_HIZ: + snd_iprintf(buffer, " VREF_HIZ"); + break; + case AC_PINCTL_VREF_50: + snd_iprintf(buffer, " VREF_50"); + break; + case AC_PINCTL_VREF_GRD: + snd_iprintf(buffer, " VREF_GRD"); + break; + case AC_PINCTL_VREF_80: + snd_iprintf(buffer, " VREF_80"); + break; + case AC_PINCTL_VREF_100: + snd_iprintf(buffer, " VREF_100"); + break; + } + } + snd_iprintf(buffer, "\n"); +} + +static void print_vol_knob(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int cap = snd_hda_param_read(codec, nid, + AC_PAR_VOL_KNB_CAP); + snd_iprintf(buffer, " Volume-Knob: delta=%d, steps=%d, ", + (cap >> 7) & 1, cap & 0x7f); + cap = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); + snd_iprintf(buffer, "direct=%d, val=%d\n", + (cap >> 7) & 1, cap & 0x7f); +} + +static void print_audio_io(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid, + unsigned int wid_type) +{ + int conv = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + snd_iprintf(buffer, + " Converter: stream=%d, channel=%d\n", + (conv & AC_CONV_STREAM) >> AC_CONV_STREAM_SHIFT, + conv & AC_CONV_CHANNEL); + + if (wid_type == AC_WID_AUD_IN && (conv & AC_CONV_CHANNEL) == 0) { + int sdi = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_SDI_SELECT, 0); + snd_iprintf(buffer, " SDI-Select: %d\n", + sdi & AC_SDI_SELECT); + } +} + +static void print_digital_conv(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int digi1 = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + unsigned int digi2 = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_2, 0); + snd_iprintf(buffer, " Digital:"); + if (digi1 & AC_DIG1_ENABLE) + snd_iprintf(buffer, " Enabled"); + if (digi1 & AC_DIG1_V) + snd_iprintf(buffer, " Validity"); + if (digi1 & AC_DIG1_VCFG) + snd_iprintf(buffer, " ValidityCfg"); + if (digi1 & AC_DIG1_EMPHASIS) + snd_iprintf(buffer, " Preemphasis"); + if (digi1 & AC_DIG1_COPYRIGHT) + snd_iprintf(buffer, " Copyright"); + if (digi1 & AC_DIG1_NONAUDIO) + snd_iprintf(buffer, " Non-Audio"); + if (digi1 & AC_DIG1_PROFESSIONAL) + snd_iprintf(buffer, " Pro"); + if (digi1 & AC_DIG1_LEVEL) + snd_iprintf(buffer, " GenLevel"); + snd_iprintf(buffer, "\n"); + snd_iprintf(buffer, " Digital category: 0x%x\n", digi2 & AC_DIG2_CC); +} + +static const char *get_pwr_state(u32 state) +{ + static const char *buf[4] = { + "D0", "D1", "D2", "D3" + }; + if (state < 4) + return buf[state]; + return "UNKNOWN"; +} + +static void print_power_state(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int pwr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + snd_iprintf(buffer, " Power: setting=%s, actual=%s\n", + get_pwr_state(pwr & AC_PWRST_SETTING), + get_pwr_state((pwr & AC_PWRST_ACTUAL) >> + AC_PWRST_ACTUAL_SHIFT)); +} + +static void print_unsol_cap(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int unsol = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_UNSOLICITED_RESPONSE, 0); + snd_iprintf(buffer, + " Unsolicited: tag=%02x, enabled=%d\n", + unsol & AC_UNSOL_TAG, + (unsol & AC_UNSOL_ENABLED) ? 1 : 0); +} + +static void print_proc_caps(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int proc_caps = snd_hda_param_read(codec, nid, + AC_PAR_PROC_CAP); + snd_iprintf(buffer, " Processing caps: benign=%d, ncoeff=%d\n", + proc_caps & AC_PCAP_BENIGN, + (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT); +} + +static void print_conn_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid, + unsigned int wid_type, hda_nid_t *conn, + int conn_len) +{ + int c, curr = -1; + + if (conn_len > 1 && wid_type != AC_WID_AUD_MIX) + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + snd_iprintf(buffer, " Connection: %d\n", conn_len); + if (conn_len > 0) { + snd_iprintf(buffer, " "); + for (c = 0; c < conn_len; c++) { + snd_iprintf(buffer, " 0x%02x", conn[c]); + if (c == curr) + snd_iprintf(buffer, "*"); + } + snd_iprintf(buffer, "\n"); + } +} + +static void print_realtek_coef(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int coeff = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff); + coeff = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_COEF_INDEX, 0); + snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff); +} + +static void print_gpio(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int gpio = + snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + unsigned int enable, direction, wake, unsol, sticky, data; + int i, max; + snd_iprintf(buffer, "GPIO: io=%d, o=%d, i=%d, " + "unsolicited=%d, wake=%d\n", + gpio & AC_GPIO_IO_COUNT, + (gpio & AC_GPIO_O_COUNT) >> AC_GPIO_O_COUNT_SHIFT, + (gpio & AC_GPIO_I_COUNT) >> AC_GPIO_I_COUNT_SHIFT, + (gpio & AC_GPIO_UNSOLICITED) ? 1 : 0, + (gpio & AC_GPIO_WAKE) ? 1 : 0); + max = gpio & AC_GPIO_IO_COUNT; + enable = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_MASK, 0); + direction = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + wake = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_WAKE_MASK, 0); + unsol = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, 0); + sticky = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_STICKY_MASK, 0); + data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DATA, 0); + for (i = 0; i < max; ++i) + snd_iprintf(buffer, + " IO[%d]: enable=%d, dir=%d, wake=%d, " + "sticky=%d, data=%d\n", i, + (enable & (1<afg); + for (i = 0; i < nodes; i++, nid++) { unsigned int wid_caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - int conn_len = 0; hda_nid_t conn[HDA_MAX_CONNECTIONS]; - unsigned int pinctls; + int conn_len = 0; snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, get_wid_type_name(wid_type), wid_caps); @@ -302,6 +563,10 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-In"); if (wid_caps & AC_WCAP_OUT_AMP) snd_iprintf(buffer, " Amp-Out"); + if (wid_caps & AC_WCAP_STRIPE) + snd_iprintf(buffer, " Stripe"); + if (wid_caps & AC_WCAP_LR_SWAP) + snd_iprintf(buffer, " R/L"); snd_iprintf(buffer, "\n"); /* volume knob is a special widget that always have connection @@ -330,33 +595,20 @@ static void print_codec_info(struct snd_info_entry *entry, } switch (wid_type) { - case AC_WID_PIN: - print_pin_caps(buffer, codec, nid); - pinctls = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, - 0); - snd_iprintf(buffer, " Pin-ctls: 0x%02x:", pinctls); - if (pinctls & AC_PINCTL_IN_EN) - snd_iprintf(buffer, " IN"); - if (pinctls & AC_PINCTL_OUT_EN) - snd_iprintf(buffer, " OUT"); - if (pinctls & AC_PINCTL_HP_EN) - snd_iprintf(buffer, " HP"); - snd_iprintf(buffer, "\n"); + case AC_WID_PIN: { + int supports_vref; + print_pin_caps(buffer, codec, nid, &supports_vref); + print_pin_ctls(buffer, codec, nid, supports_vref); break; + } case AC_WID_VOL_KNB: - pinctls = snd_hda_param_read(codec, nid, - AC_PAR_VOL_KNB_CAP); - snd_iprintf(buffer, " Volume-Knob: delta=%d, " - "steps=%d, ", - (pinctls >> 7) & 1, pinctls & 0x7f); - pinctls = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); - snd_iprintf(buffer, "direct=%d, val=%d\n", - (pinctls >> 7) & 1, pinctls & 0x7f); + print_vol_knob(buffer, codec, nid); break; case AC_WID_AUD_OUT: case AC_WID_AUD_IN: + print_audio_io(buffer, codec, nid, wid_type); + if (wid_caps & AC_WCAP_DIGITAL) + print_digital_conv(buffer, codec, nid); if (wid_caps & AC_WCAP_FORMAT_OVRD) { snd_iprintf(buffer, " PCM:\n"); print_pcm_caps(buffer, codec, nid); @@ -364,28 +616,27 @@ static void print_codec_info(struct snd_info_entry *entry, break; } + if (wid_caps & AC_WCAP_UNSOL_CAP) + print_unsol_cap(buffer, codec, nid); + if (wid_caps & AC_WCAP_POWER) - snd_iprintf(buffer, " Power: 0x%x\n", - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_POWER_STATE, - 0)); - - if (wid_caps & AC_WCAP_CONN_LIST) { - int c, curr = -1; - if (conn_len > 1 && wid_type != AC_WID_AUD_MIX) - curr = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_SEL, 0); - snd_iprintf(buffer, " Connection: %d\n", conn_len); - if (conn_len > 0) { - snd_iprintf(buffer, " "); - for (c = 0; c < conn_len; c++) { - snd_iprintf(buffer, " 0x%02x", conn[c]); - if (c == curr) - snd_iprintf(buffer, "*"); - } - snd_iprintf(buffer, "\n"); - } - } + print_power_state(buffer, codec, nid); + + if (wid_caps & AC_WCAP_DELAY) + snd_iprintf(buffer, " Delay: %d samples\n", + (wid_caps & AC_WCAP_DELAY) >> + AC_WCAP_DELAY_SHIFT); + + if (wid_caps & AC_WCAP_CONN_LIST) + print_conn_list(buffer, codec, nid, wid_type, + conn, conn_len); + + if (wid_caps & AC_WCAP_PROC_WID) + print_proc_caps(buffer, codec, nid); + + /* NID 0x20 == Realtek Define Registers */ + if (codec->vendor_id == 0x10ec && nid == 0x20) + print_realtek_coef(buffer, codec, nid); } snd_hda_power_down(codec); } -- cgit v0.10.2 From 7cda8ba9f4e471dfba914ecf67fd14ebffb17c16 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Jan 2008 13:36:07 +0100 Subject: [ALSA] ice1712, ice1724 - Code clean up Clean up ice1712/ice1724 codes. The board-specific data is allocated locally in each code instead of having an ungly union in struct ice1712. Also, fix coding issues in prodigy_hifi.c. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 3374891..868ae29 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -61,6 +61,15 @@ #include "aureon.h" #include +/* AC97 register cache for Aureon */ +struct aureon_spec { + unsigned short stac9744[64]; + unsigned int cs8415_mux; + unsigned short master[2]; + unsigned short vol[8]; + unsigned char pca9554_out; +}; + /* WM8770 registers */ #define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */ #define WM_DAC_MASTER_ATTEN 0x08 /* DAC master analog attenuation */ @@ -204,7 +213,8 @@ static int aureon_universe_inmux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = ice->spec.aureon.pca9554_out; + struct aureon_spec *spec = ice->spec; + ucontrol->value.enumerated.item[0] = spec->pca9554_out; return 0; } @@ -212,6 +222,7 @@ static int aureon_universe_inmux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; unsigned char oval, nval; int change; @@ -219,10 +230,10 @@ static int aureon_universe_inmux_put(struct snd_kcontrol *kcontrol, if (nval >= 3) return -EINVAL; snd_ice1712_save_gpio_status(ice); - oval = ice->spec.aureon.pca9554_out; + oval = spec->pca9554_out; if ((change = (oval != nval))) { aureon_pca9554_write(ice, PCA9554_OUT, nval); - ice->spec.aureon.pca9554_out = nval; + spec->pca9554_out = nval; } snd_ice1712_restore_gpio_status(ice); @@ -233,6 +244,7 @@ static int aureon_universe_inmux_put(struct snd_kcontrol *kcontrol, static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, unsigned short val) { + struct aureon_spec *spec = ice->spec; unsigned int tmp; /* Send address to XILINX chip */ @@ -280,12 +292,13 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, udelay(10); /* Store the data in out private buffer */ - ice->spec.aureon.stac9744[(reg & 0x7F) >> 1] = val; + spec->stac9744[(reg & 0x7F) >> 1] = val; } static unsigned short aureon_ac97_read(struct snd_ice1712 *ice, unsigned short reg) { - return ice->spec.aureon.stac9744[(reg & 0x7F) >> 1]; + struct aureon_spec *spec = ice->spec; + return spec->stac9744[(reg & 0x7F) >> 1]; } /* @@ -293,6 +306,7 @@ static unsigned short aureon_ac97_read(struct snd_ice1712 *ice, unsigned short r */ static int aureon_ac97_init (struct snd_ice1712 *ice) { + struct aureon_spec *spec = ice->spec; int i; static const unsigned short ac97_defaults[] = { 0x00, 0x9640, @@ -330,9 +344,9 @@ static int aureon_ac97_init (struct snd_ice1712 *ice) snd_ice1712_gpio_write(ice, tmp); udelay(3); - memset(&ice->spec.aureon.stac9744, 0, sizeof(ice->spec.aureon.stac9744)); + memset(&spec->stac9744, 0, sizeof(spec->stac9744)); for (i=0; ac97_defaults[i] != (unsigned short)-1; i+=2) - ice->spec.aureon.stac9744[(ac97_defaults[i]) >> 1] = ac97_defaults[i+1]; + spec->stac9744[(ac97_defaults[i]) >> 1] = ac97_defaults[i+1]; aureon_ac97_write(ice, AC97_MASTER, 0x0000); // Unmute AC'97 master volume permanently - muting is done by WM8770 @@ -744,15 +758,18 @@ static int wm_master_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int wm_master_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; int i; for (i=0; i<2; i++) - ucontrol->value.integer.value[i] = ice->spec.aureon.master[i] & ~WM_VOL_MUTE; + ucontrol->value.integer.value[i] = + spec->master[i] & ~WM_VOL_MUTE; return 0; } static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; int ch, change = 0; snd_ice1712_save_gpio_status(ice); @@ -760,14 +777,14 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ unsigned int vol = ucontrol->value.integer.value[ch]; if (vol > WM_VOL_MAX) continue; - vol |= ice->spec.aureon.master[ch] & WM_VOL_MUTE; - if (vol != ice->spec.aureon.master[ch]) { + vol |= spec->master[ch] & WM_VOL_MUTE; + if (vol != spec->master[ch]) { int dac; - ice->spec.aureon.master[ch] = vol; + spec->master[ch] = vol; for (dac = 0; dac < ice->num_total_dacs; dac += 2) wm_set_vol(ice, WM_DAC_ATTEN + dac + ch, - ice->spec.aureon.vol[dac + ch], - ice->spec.aureon.master[ch]); + spec->vol[dac + ch], + spec->master[ch]); change = 1; } } @@ -791,18 +808,21 @@ static int wm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info * static int wm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; int i, ofs, voices; voices = kcontrol->private_value >> 8; ofs = kcontrol->private_value & 0xff; for (i = 0; i < voices; i++) - ucontrol->value.integer.value[i] = ice->spec.aureon.vol[ofs+i] & ~WM_VOL_MUTE; + ucontrol->value.integer.value[i] = + spec->vol[ofs+i] & ~WM_VOL_MUTE; return 0; } static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; int i, idx, ofs, voices; int change = 0; @@ -813,12 +833,12 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * unsigned int vol = ucontrol->value.integer.value[i]; if (vol > 0x7f) continue; - vol |= ice->spec.aureon.vol[ofs+i]; - if (vol != ice->spec.aureon.vol[ofs+i]) { - ice->spec.aureon.vol[ofs+i] = vol; + vol |= spec->vol[ofs+i]; + if (vol != spec->vol[ofs+i]) { + spec->vol[ofs+i] = vol; idx = WM_DAC_ATTEN + ofs + i; - wm_set_vol(ice, idx, ice->spec.aureon.vol[ofs+i], - ice->spec.aureon.master[i]); + wm_set_vol(ice, idx, spec->vol[ofs + i], + spec->master[i]); change = 1; } } @@ -840,19 +860,22 @@ static int wm_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info static int wm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; int voices, ofs, i; voices = kcontrol->private_value >> 8; ofs = kcontrol->private_value & 0xFF; for (i = 0; i < voices; i++) - ucontrol->value.integer.value[i] = (ice->spec.aureon.vol[ofs+i] & WM_VOL_MUTE) ? 0 : 1; + ucontrol->value.integer.value[i] = + (spec->vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1; return 0; } static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; int change = 0, voices, ofs, i; voices = kcontrol->private_value >> 8; @@ -860,13 +883,13 @@ static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value snd_ice1712_save_gpio_status(ice); for (i = 0; i < voices; i++) { - int val = (ice->spec.aureon.vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1; + int val = (spec->vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1; if (ucontrol->value.integer.value[i] != val) { - ice->spec.aureon.vol[ofs + i] &= ~WM_VOL_MUTE; - ice->spec.aureon.vol[ofs + i] |= + spec->vol[ofs + i] &= ~WM_VOL_MUTE; + spec->vol[ofs + i] |= ucontrol->value.integer.value[i] ? 0 : WM_VOL_MUTE; - wm_set_vol(ice, ofs + i, ice->spec.aureon.vol[ofs + i], - ice->spec.aureon.master[i]); + wm_set_vol(ice, ofs + i, spec->vol[ofs + i], + spec->master[i]); change = 1; } } @@ -883,29 +906,33 @@ static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value static int wm_master_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; - ucontrol->value.integer.value[0] = (ice->spec.aureon.master[0] & WM_VOL_MUTE) ? 0 : 1; - ucontrol->value.integer.value[1] = (ice->spec.aureon.master[1] & WM_VOL_MUTE) ? 0 : 1; + ucontrol->value.integer.value[0] = + (spec->master[0] & WM_VOL_MUTE) ? 0 : 1; + ucontrol->value.integer.value[1] = + (spec->master[1] & WM_VOL_MUTE) ? 0 : 1; return 0; } static int wm_master_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; int change = 0, i; snd_ice1712_save_gpio_status(ice); for (i = 0; i < 2; i++) { - int val = (ice->spec.aureon.master[i] & WM_VOL_MUTE) ? 0 : 1; + int val = (spec->master[i] & WM_VOL_MUTE) ? 0 : 1; if (ucontrol->value.integer.value[i] != val) { int dac; - ice->spec.aureon.master[i] &= ~WM_VOL_MUTE; - ice->spec.aureon.master[i] |= + spec->master[i] &= ~WM_VOL_MUTE; + spec->master[i] |= ucontrol->value.integer.value[i] ? 0 : WM_VOL_MUTE; for (dac = 0; dac < ice->num_total_dacs; dac += 2) wm_set_vol(ice, WM_DAC_ATTEN + dac + i, - ice->spec.aureon.vol[dac + i], - ice->spec.aureon.master[i]); + spec->vol[dac + i], + spec->master[i]); change = 1; } } @@ -1151,10 +1178,11 @@ static int aureon_cs8415_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int aureon_cs8415_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; //snd_ice1712_save_gpio_status(ice); //val = aureon_cs8415_get(ice, CS8415_CTRL2); - ucontrol->value.enumerated.item[0] = ice->spec.aureon.cs8415_mux; + ucontrol->value.enumerated.item[0] = spec->cs8415_mux; //snd_ice1712_restore_gpio_status(ice); return 0; } @@ -1162,6 +1190,7 @@ static int aureon_cs8415_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int aureon_cs8415_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct aureon_spec *spec = ice->spec; unsigned short oval, nval; int change; @@ -1173,7 +1202,7 @@ static int aureon_cs8415_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (change) aureon_cs8415_put(ice, CS8415_CTRL2, nval); snd_ice1712_restore_gpio_status(ice); - ice->spec.aureon.cs8415_mux = ucontrol->value.enumerated.item[0]; + spec->cs8415_mux = ucontrol->value.enumerated.item[0]; return change; } @@ -2009,10 +2038,16 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) 0x0605, /* slave, 24bit, MSB on second OSCLK, SDOUT for right channel when OLRCK is high */ (unsigned short)-1 }; + struct aureon_spec *spec; unsigned int tmp; const unsigned short *p; int err, i; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY) { ice->num_total_dacs = 6; ice->num_total_adcs = 2; @@ -2063,7 +2098,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT) { for (p = cs_inits; *p != (unsigned short)-1; p++) aureon_spi_write(ice, AUREON_CS8415_CS, *p | 0x200000, 24); - ice->spec.aureon.cs8415_mux = 1; + spec->cs8415_mux = 1; aureon_set_headphone_amp(ice, 1); } @@ -2074,11 +2109,11 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) aureon_pca9554_write(ice, PCA9554_DIR, 0x00); aureon_pca9554_write(ice, PCA9554_OUT, 0x00); /* internal AUX */ - ice->spec.aureon.master[0] = WM_VOL_MUTE; - ice->spec.aureon.master[1] = WM_VOL_MUTE; + spec->master[0] = WM_VOL_MUTE; + spec->master[1] = WM_VOL_MUTE; for (i = 0; i < ice->num_total_dacs; i++) { - ice->spec.aureon.vol[i] = WM_VOL_MUTE; - wm_set_vol(ice, i, ice->spec.aureon.vol[i], ice->spec.aureon.master[i % 2]); + spec->vol[i] = WM_VOL_MUTE; + wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); } return 0; diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 6f65da4..064760d 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -44,6 +44,11 @@ enum { }; +/* additional i2c devices for EWS boards */ +struct ews_spec { + struct snd_i2c_device *i2cdevs[3]; +}; + /* * access via i2c mode (for EWX 24/96, EWS 88MT&D) */ @@ -141,15 +146,17 @@ static struct snd_i2c_bit_ops snd_ice1712_ewx_cs8427_bit_ops = { /* AK4524 chip select; address 0x48 bit 0-3 */ static int snd_ice1712_ews88mt_chip_select(struct snd_ice1712 *ice, int chip_mask) { + struct ews_spec *spec = ice->spec; unsigned char data, ndata; snd_assert(chip_mask >= 0 && chip_mask <= 0x0f, return -EINVAL); snd_i2c_lock(ice->i2c); - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_PCF2], &data, 1) != 1) + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF2], &data, 1) != 1) goto __error; ndata = (data & 0xf0) | chip_mask; if (ndata != data) - if (snd_i2c_sendbytes(ice->spec.i2cdevs[EWS_I2C_PCF2], &ndata, 1) != 1) + if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_PCF2], &ndata, 1) + != 1) goto __error; snd_i2c_unlock(ice->i2c); return 0; @@ -223,6 +230,7 @@ static void dmx6fire_ak4524_lock(struct snd_akm4xxx *ak, int chip) static void snd_ice1712_ews_cs8404_spdif_write(struct snd_ice1712 *ice, unsigned char bits) { + struct ews_spec *spec = ice->spec; unsigned char bytes[2]; snd_i2c_lock(ice->i2c); @@ -230,15 +238,18 @@ static void snd_ice1712_ews_cs8404_spdif_write(struct snd_ice1712 *ice, unsigned case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: - if (snd_i2c_sendbytes(ice->spec.i2cdevs[EWS_I2C_CS8404], &bits, 1) != 1) + if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_CS8404], &bits, 1) + != 1) goto _error; break; case ICE1712_SUBDEVICE_EWS88D: - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_88D], bytes, 2) != 2) + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_88D], bytes, 2) + != 2) goto _error; if (bits != bytes[1]) { bytes[1] = bits; - if (snd_i2c_sendbytes(ice->spec.i2cdevs[EWS_I2C_88D], bytes, 2) != 2) + if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_88D], + bytes, 2) != 2) goto _error; } break; @@ -411,6 +422,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) { int err; struct snd_akm4xxx *ak; + struct ews_spec *spec; /* set the analog DACs */ switch (ice->eeprom.subvendor) { @@ -435,6 +447,11 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) break; } + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + /* create i2c */ if ((err = snd_i2c_bus_create(ice->card, "ICE1712 GPIO 1", NULL, &ice->i2c)) < 0) { snd_printk(KERN_ERR "unable to create I2C bus\n"); @@ -446,7 +463,10 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) /* create i2c devices */ switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_DMX6FIRE: - if ((err = snd_i2c_device_create(ice->i2c, "PCF9554", ICE1712_6FIRE_PCF9554_ADDR, &ice->spec.i2cdevs[EWS_I2C_6FIRE])) < 0) { + err = snd_i2c_device_create(ice->i2c, "PCF9554", + ICE1712_6FIRE_PCF9554_ADDR, + &spec->i2cdevs[EWS_I2C_6FIRE]); + if (err < 0) { snd_printk(KERN_ERR "PCF9554 initialization failed\n"); return err; } @@ -455,18 +475,30 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: - if ((err = snd_i2c_device_create(ice->i2c, "CS8404", ICE1712_EWS88MT_CS8404_ADDR, &ice->spec.i2cdevs[EWS_I2C_CS8404])) < 0) + err = snd_i2c_device_create(ice->i2c, "CS8404", + ICE1712_EWS88MT_CS8404_ADDR, + &spec->i2cdevs[EWS_I2C_CS8404]); + if (err < 0) return err; - if ((err = snd_i2c_device_create(ice->i2c, "PCF8574 (1st)", ICE1712_EWS88MT_INPUT_ADDR, &ice->spec.i2cdevs[EWS_I2C_PCF1])) < 0) + err = snd_i2c_device_create(ice->i2c, "PCF8574 (1st)", + ICE1712_EWS88MT_INPUT_ADDR, + &spec->i2cdevs[EWS_I2C_PCF1]); + if (err < 0) return err; - if ((err = snd_i2c_device_create(ice->i2c, "PCF8574 (2nd)", ICE1712_EWS88MT_OUTPUT_ADDR, &ice->spec.i2cdevs[EWS_I2C_PCF2])) < 0) + err = snd_i2c_device_create(ice->i2c, "PCF8574 (2nd)", + ICE1712_EWS88MT_OUTPUT_ADDR, + &spec->i2cdevs[EWS_I2C_PCF2]); + if (err < 0) return err; /* Check if the front module is connected */ if ((err = snd_ice1712_ews88mt_chip_select(ice, 0x0f)) < 0) return err; break; case ICE1712_SUBDEVICE_EWS88D: - if ((err = snd_i2c_device_create(ice->i2c, "PCF8575", ICE1712_EWS88D_PCF_ADDR, &ice->spec.i2cdevs[EWS_I2C_88D])) < 0) + err = snd_i2c_device_create(ice->i2c, "PCF8575", + ICE1712_EWS88D_PCF_ADDR, + &spec->i2cdevs[EWS_I2C_88D]); + if (err < 0) return err; break; } @@ -506,7 +538,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) } /* analog section */ - ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); + ak = ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ak) return -ENOMEM; ice->akm_codecs = 1; @@ -604,10 +636,11 @@ static struct snd_kcontrol_new snd_ice1712_ewx2496_controls[] __devinitdata = { static int snd_ice1712_ews88mt_output_sense_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct ews_spec *spec = ice->spec; unsigned char data; snd_i2c_lock(ice->i2c); - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_PCF2], &data, 1) != 1) { + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF2], &data, 1) != 1) { snd_i2c_unlock(ice->i2c); return -EIO; } @@ -620,15 +653,17 @@ static int snd_ice1712_ews88mt_output_sense_get(struct snd_kcontrol *kcontrol, s static int snd_ice1712_ews88mt_output_sense_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct ews_spec *spec = ice->spec; unsigned char data, ndata; snd_i2c_lock(ice->i2c); - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_PCF2], &data, 1) != 1) { + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF2], &data, 1) != 1) { snd_i2c_unlock(ice->i2c); return -EIO; } ndata = (data & ~ICE1712_EWS88MT_OUTPUT_SENSE) | (ucontrol->value.enumerated.item[0] ? ICE1712_EWS88MT_OUTPUT_SENSE : 0); - if (ndata != data && snd_i2c_sendbytes(ice->spec.i2cdevs[EWS_I2C_PCF2], &ndata, 1) != 1) { + if (ndata != data && snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_PCF2], + &ndata, 1) != 1) { snd_i2c_unlock(ice->i2c); return -EIO; } @@ -640,12 +675,13 @@ static int snd_ice1712_ews88mt_output_sense_put(struct snd_kcontrol *kcontrol, s static int snd_ice1712_ews88mt_input_sense_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct ews_spec *spec = ice->spec; int channel = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned char data; snd_assert(channel >= 0 && channel <= 7, return 0); snd_i2c_lock(ice->i2c); - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_PCF1], &data, 1) != 1) { + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF1], &data, 1) != 1) { snd_i2c_unlock(ice->i2c); return -EIO; } @@ -659,17 +695,19 @@ static int snd_ice1712_ews88mt_input_sense_get(struct snd_kcontrol *kcontrol, st static int snd_ice1712_ews88mt_input_sense_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct ews_spec *spec = ice->spec; int channel = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned char data, ndata; snd_assert(channel >= 0 && channel <= 7, return 0); snd_i2c_lock(ice->i2c); - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_PCF1], &data, 1) != 1) { + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF1], &data, 1) != 1) { snd_i2c_unlock(ice->i2c); return -EIO; } ndata = (data & ~(1 << channel)) | (ucontrol->value.enumerated.item[0] ? 0 : (1 << channel)); - if (ndata != data && snd_i2c_sendbytes(ice->spec.i2cdevs[EWS_I2C_PCF1], &ndata, 1) != 1) { + if (ndata != data && snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_PCF1], + &ndata, 1) != 1) { snd_i2c_unlock(ice->i2c); return -EIO; } @@ -704,12 +742,13 @@ static struct snd_kcontrol_new snd_ice1712_ews88mt_output_sense __devinitdata = static int snd_ice1712_ews88d_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct ews_spec *spec = ice->spec; int shift = kcontrol->private_value & 0xff; int invert = (kcontrol->private_value >> 8) & 1; unsigned char data[2]; snd_i2c_lock(ice->i2c); - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_88D], data, 2) != 2) { + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_88D], data, 2) != 2) { snd_i2c_unlock(ice->i2c); return -EIO; } @@ -724,13 +763,14 @@ static int snd_ice1712_ews88d_control_get(struct snd_kcontrol *kcontrol, struct static int snd_ice1712_ews88d_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct ews_spec *spec = ice->spec; int shift = kcontrol->private_value & 0xff; int invert = (kcontrol->private_value >> 8) & 1; unsigned char data[2], ndata[2]; int change; snd_i2c_lock(ice->i2c); - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_88D], data, 2) != 2) { + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_88D], data, 2) != 2) { snd_i2c_unlock(ice->i2c); return -EIO; } @@ -743,7 +783,8 @@ static int snd_ice1712_ews88d_control_put(struct snd_kcontrol *kcontrol, struct ndata[shift >> 3] |= (1 << (shift & 7)); } change = (data[shift >> 3] != ndata[shift >> 3]); - if (change && snd_i2c_sendbytes(ice->spec.i2cdevs[EWS_I2C_88D], data, 2) != 2) { + if (change && + snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_88D], data, 2) != 2) { snd_i2c_unlock(ice->i2c); return -EIO; } @@ -777,11 +818,13 @@ static struct snd_kcontrol_new snd_ice1712_ews88d_controls[] __devinitdata = { static int snd_ice1712_6fire_read_pca(struct snd_ice1712 *ice, unsigned char reg) { unsigned char byte; + struct ews_spec *spec = ice->spec; + snd_i2c_lock(ice->i2c); byte = reg; - snd_i2c_sendbytes(ice->spec.i2cdevs[EWS_I2C_6FIRE], &byte, 1); + snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1); byte = 0; - if (snd_i2c_readbytes(ice->spec.i2cdevs[EWS_I2C_6FIRE], &byte, 1) != 1) { + if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1) != 1) { snd_i2c_unlock(ice->i2c); printk(KERN_ERR "cannot read pca\n"); return -EIO; @@ -793,10 +836,12 @@ static int snd_ice1712_6fire_read_pca(struct snd_ice1712 *ice, unsigned char reg static int snd_ice1712_6fire_write_pca(struct snd_ice1712 *ice, unsigned char reg, unsigned char data) { unsigned char bytes[2]; + struct ews_spec *spec = ice->spec; + snd_i2c_lock(ice->i2c); bytes[0] = reg; bytes[1] = data; - if (snd_i2c_sendbytes(ice->spec.i2cdevs[EWS_I2C_6FIRE], bytes, 2) != 2) { + if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], bytes, 2) != 2) { snd_i2c_unlock(ice->i2c); return -EIO; } diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index b042e5c..cf5c7c0 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -33,6 +33,12 @@ #include "ice1712.h" #include "hoontech.h" +/* Hoontech-specific setting */ +struct hoontech_spec { + unsigned char boxbits[4]; + unsigned int config; + unsigned short boxconfig[4]; +}; static void __devinit snd_ice1712_stdsp24_gpio_write(struct snd_ice1712 *ice, unsigned char byte) { @@ -49,169 +55,182 @@ static void __devinit snd_ice1712_stdsp24_gpio_write(struct snd_ice1712 *ice, un static void __devinit snd_ice1712_stdsp24_darear(struct snd_ice1712 *ice, int activate) { + struct hoontech_spec *spec = ice->spec; mutex_lock(&ice->gpio_mutex); - ICE1712_STDSP24_0_DAREAR(ice->spec.hoontech.boxbits, activate); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[0]); + ICE1712_STDSP24_0_DAREAR(spec->boxbits, activate); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[0]); mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_mute(struct snd_ice1712 *ice, int activate) { + struct hoontech_spec *spec = ice->spec; mutex_lock(&ice->gpio_mutex); - ICE1712_STDSP24_3_MUTE(ice->spec.hoontech.boxbits, activate); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[3]); + ICE1712_STDSP24_3_MUTE(spec->boxbits, activate); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]); mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_insel(struct snd_ice1712 *ice, int activate) { + struct hoontech_spec *spec = ice->spec; mutex_lock(&ice->gpio_mutex); - ICE1712_STDSP24_3_INSEL(ice->spec.hoontech.boxbits, activate); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[3]); + ICE1712_STDSP24_3_INSEL(spec->boxbits, activate); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]); mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_box_channel(struct snd_ice1712 *ice, int box, int chn, int activate) { + struct hoontech_spec *spec = ice->spec; + mutex_lock(&ice->gpio_mutex); /* select box */ - ICE1712_STDSP24_0_BOX(ice->spec.hoontech.boxbits, box); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[0]); + ICE1712_STDSP24_0_BOX(spec->boxbits, box); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[0]); /* prepare for write */ if (chn == 3) - ICE1712_STDSP24_2_CHN4(ice->spec.hoontech.boxbits, 0); - ICE1712_STDSP24_2_MIDI1(ice->spec.hoontech.boxbits, activate); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[3]); - - ICE1712_STDSP24_1_CHN1(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_1_CHN2(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_1_CHN3(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_2_CHN4(ice->spec.hoontech.boxbits, 1); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[1]); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); + ICE1712_STDSP24_2_CHN4(spec->boxbits, 0); + ICE1712_STDSP24_2_MIDI1(spec->boxbits, activate); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]); + + ICE1712_STDSP24_1_CHN1(spec->boxbits, 1); + ICE1712_STDSP24_1_CHN2(spec->boxbits, 1); + ICE1712_STDSP24_1_CHN3(spec->boxbits, 1); + ICE1712_STDSP24_2_CHN4(spec->boxbits, 1); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[1]); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); udelay(100); if (chn == 3) { - ICE1712_STDSP24_2_CHN4(ice->spec.hoontech.boxbits, 0); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); + ICE1712_STDSP24_2_CHN4(spec->boxbits, 0); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); } else { switch (chn) { - case 0: ICE1712_STDSP24_1_CHN1(ice->spec.hoontech.boxbits, 0); break; - case 1: ICE1712_STDSP24_1_CHN2(ice->spec.hoontech.boxbits, 0); break; - case 2: ICE1712_STDSP24_1_CHN3(ice->spec.hoontech.boxbits, 0); break; + case 0: ICE1712_STDSP24_1_CHN1(spec->boxbits, 0); break; + case 1: ICE1712_STDSP24_1_CHN2(spec->boxbits, 0); break; + case 2: ICE1712_STDSP24_1_CHN3(spec->boxbits, 0); break; } - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[1]); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[1]); } udelay(100); - ICE1712_STDSP24_1_CHN1(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_1_CHN2(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_1_CHN3(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_2_CHN4(ice->spec.hoontech.boxbits, 1); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[1]); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); + ICE1712_STDSP24_1_CHN1(spec->boxbits, 1); + ICE1712_STDSP24_1_CHN2(spec->boxbits, 1); + ICE1712_STDSP24_1_CHN3(spec->boxbits, 1); + ICE1712_STDSP24_2_CHN4(spec->boxbits, 1); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[1]); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); udelay(100); - ICE1712_STDSP24_2_MIDI1(ice->spec.hoontech.boxbits, 0); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); + ICE1712_STDSP24_2_MIDI1(spec->boxbits, 0); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_box_midi(struct snd_ice1712 *ice, int box, int master) { + struct hoontech_spec *spec = ice->spec; + mutex_lock(&ice->gpio_mutex); /* select box */ - ICE1712_STDSP24_0_BOX(ice->spec.hoontech.boxbits, box); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[0]); + ICE1712_STDSP24_0_BOX(spec->boxbits, box); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[0]); - ICE1712_STDSP24_2_MIDIIN(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_2_MIDI1(ice->spec.hoontech.boxbits, master); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[3]); + ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 1); + ICE1712_STDSP24_2_MIDI1(spec->boxbits, master); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]); udelay(100); - ICE1712_STDSP24_2_MIDIIN(ice->spec.hoontech.boxbits, 0); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); + ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 0); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); mdelay(10); - ICE1712_STDSP24_2_MIDIIN(ice->spec.hoontech.boxbits, 1); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); + ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 1); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_midi2(struct snd_ice1712 *ice, int activate) { + struct hoontech_spec *spec = ice->spec; mutex_lock(&ice->gpio_mutex); - ICE1712_STDSP24_3_MIDI2(ice->spec.hoontech.boxbits, activate); - snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[3]); + ICE1712_STDSP24_3_MIDI2(spec->boxbits, activate); + snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]); mutex_unlock(&ice->gpio_mutex); } static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) { + struct hoontech_spec *spec; int box, chn; ice->num_total_dacs = 8; ice->num_total_adcs = 8; - ice->spec.hoontech.boxbits[0] = - ice->spec.hoontech.boxbits[1] = - ice->spec.hoontech.boxbits[2] = - ice->spec.hoontech.boxbits[3] = 0; /* should be already */ - - ICE1712_STDSP24_SET_ADDR(ice->spec.hoontech.boxbits, 0); - ICE1712_STDSP24_CLOCK(ice->spec.hoontech.boxbits, 0, 1); - ICE1712_STDSP24_0_BOX(ice->spec.hoontech.boxbits, 0); - ICE1712_STDSP24_0_DAREAR(ice->spec.hoontech.boxbits, 0); - - ICE1712_STDSP24_SET_ADDR(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_CLOCK(ice->spec.hoontech.boxbits, 1, 1); - ICE1712_STDSP24_1_CHN1(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_1_CHN2(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_1_CHN3(ice->spec.hoontech.boxbits, 1); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + + ICE1712_STDSP24_SET_ADDR(spec->boxbits, 0); + ICE1712_STDSP24_CLOCK(spec->boxbits, 0, 1); + ICE1712_STDSP24_0_BOX(spec->boxbits, 0); + ICE1712_STDSP24_0_DAREAR(spec->boxbits, 0); + + ICE1712_STDSP24_SET_ADDR(spec->boxbits, 1); + ICE1712_STDSP24_CLOCK(spec->boxbits, 1, 1); + ICE1712_STDSP24_1_CHN1(spec->boxbits, 1); + ICE1712_STDSP24_1_CHN2(spec->boxbits, 1); + ICE1712_STDSP24_1_CHN3(spec->boxbits, 1); - ICE1712_STDSP24_SET_ADDR(ice->spec.hoontech.boxbits, 2); - ICE1712_STDSP24_CLOCK(ice->spec.hoontech.boxbits, 2, 1); - ICE1712_STDSP24_2_CHN4(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_2_MIDIIN(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_2_MIDI1(ice->spec.hoontech.boxbits, 0); - - ICE1712_STDSP24_SET_ADDR(ice->spec.hoontech.boxbits, 3); - ICE1712_STDSP24_CLOCK(ice->spec.hoontech.boxbits, 3, 1); - ICE1712_STDSP24_3_MIDI2(ice->spec.hoontech.boxbits, 0); - ICE1712_STDSP24_3_MUTE(ice->spec.hoontech.boxbits, 1); - ICE1712_STDSP24_3_INSEL(ice->spec.hoontech.boxbits, 0); + ICE1712_STDSP24_SET_ADDR(spec->boxbits, 2); + ICE1712_STDSP24_CLOCK(spec->boxbits, 2, 1); + ICE1712_STDSP24_2_CHN4(spec->boxbits, 1); + ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 1); + ICE1712_STDSP24_2_MIDI1(spec->boxbits, 0); + + ICE1712_STDSP24_SET_ADDR(spec->boxbits, 3); + ICE1712_STDSP24_CLOCK(spec->boxbits, 3, 1); + ICE1712_STDSP24_3_MIDI2(spec->boxbits, 0); + ICE1712_STDSP24_3_MUTE(spec->boxbits, 1); + ICE1712_STDSP24_3_INSEL(spec->boxbits, 0); /* let's go - activate only functions in first box */ - ice->spec.hoontech.config = 0; + spec->config = 0; /* ICE1712_STDSP24_MUTE | ICE1712_STDSP24_INSEL | ICE1712_STDSP24_DAREAR; */ - ice->spec.hoontech.boxconfig[0] = ICE1712_STDSP24_BOX_CHN1 | + spec->boxconfig[0] = ICE1712_STDSP24_BOX_CHN1 | ICE1712_STDSP24_BOX_CHN2 | ICE1712_STDSP24_BOX_CHN3 | ICE1712_STDSP24_BOX_CHN4 | ICE1712_STDSP24_BOX_MIDI1 | ICE1712_STDSP24_BOX_MIDI2; - ice->spec.hoontech.boxconfig[1] = - ice->spec.hoontech.boxconfig[2] = - ice->spec.hoontech.boxconfig[3] = 0; - snd_ice1712_stdsp24_darear(ice, (ice->spec.hoontech.config & ICE1712_STDSP24_DAREAR) ? 1 : 0); - snd_ice1712_stdsp24_mute(ice, (ice->spec.hoontech.config & ICE1712_STDSP24_MUTE) ? 1 : 0); - snd_ice1712_stdsp24_insel(ice, (ice->spec.hoontech.config & ICE1712_STDSP24_INSEL) ? 1 : 0); + spec->boxconfig[1] = + spec->boxconfig[2] = + spec->boxconfig[3] = 0; + snd_ice1712_stdsp24_darear(ice, + (spec->config & ICE1712_STDSP24_DAREAR) ? 1 : 0); + snd_ice1712_stdsp24_mute(ice, + (spec->config & ICE1712_STDSP24_MUTE) ? 1 : 0); + snd_ice1712_stdsp24_insel(ice, + (spec->config & ICE1712_STDSP24_INSEL) ? 1 : 0); for (box = 0; box < 1; box++) { - if (ice->spec.hoontech.boxconfig[box] & ICE1712_STDSP24_BOX_MIDI2) + if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI2) snd_ice1712_stdsp24_midi2(ice, 1); for (chn = 0; chn < 4; chn++) - snd_ice1712_stdsp24_box_channel(ice, box, chn, (ice->spec.hoontech.boxconfig[box] & (1 << chn)) ? 1 : 0); + snd_ice1712_stdsp24_box_channel(ice, box, chn, + (spec->boxconfig[box] & (1 << chn)) ? 1 : 0); snd_ice1712_stdsp24_box_midi(ice, box, - (ice->spec.hoontech.boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1) ? 1 : 0); + (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1) ? 1 : 0); } return 0; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 47d7737..df292af 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2490,6 +2490,7 @@ static int snd_ice1712_free(struct snd_ice1712 *ice) pci_release_regions(ice->pci); snd_ice1712_akm4xxx_free(ice); pci_disable_device(ice->pci); + kfree(ice->spec); kfree(ice); return 0; } diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 86e418e..303cffe 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -366,56 +366,7 @@ struct snd_ice1712 { struct mutex gpio_mutex; /* other board-specific data */ - union { - /* additional i2c devices for EWS boards */ - struct snd_i2c_device *i2cdevs[3]; - /* AC97 register cache for Aureon */ - struct aureon_spec { - unsigned short stac9744[64]; - unsigned int cs8415_mux; - unsigned short master[2]; - unsigned short vol[8]; - unsigned char pca9554_out; - } aureon; - /* AC97 register cache for Phase28 */ - struct phase28_spec { - unsigned short master[2]; - unsigned short vol[8]; - } phase28; - /* a non-standard I2C device for revo51 */ - struct revo51_spec { - struct snd_i2c_device *dev; - struct snd_pt2258 *pt2258; - } revo51; - /* Hoontech-specific setting */ - struct hoontech_spec { - unsigned char boxbits[4]; - unsigned int config; - unsigned short boxconfig[4]; - } hoontech; - struct { - struct ak4114 *ak4114; - unsigned int analog: 1; - } juli; - struct { - struct ak4114 *ak4114; - /* rate change needs atomic mute/unmute of all dacs*/ - struct mutex mute_mutex; - } prodigy192; - struct { - struct { - unsigned char ch1, ch2; - } vol[8]; - } se; - struct prodigy_hifi_spec { - unsigned short master[2]; - unsigned short vol[8]; - } prodigy_hifi; - struct prodigy_hd2_spec { - unsigned short vol[2]; - } prodigy_hd2; - } spec; - + void *spec; }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index c89a4fe..f533850 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2176,6 +2176,7 @@ static int snd_vt1724_free(struct snd_ice1712 *ice) pci_release_regions(ice->pci); snd_ice1712_akm4xxx_free(ice); pci_disable_device(ice->pci); + kfree(ice->spec); kfree(ice); return 0; } diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 1a435df..e8038c0 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -32,6 +32,11 @@ #include "envy24ht.h" #include "juli.h" +struct juli_spec { + struct ak4114 *ak4114; + unsigned int analog: 1; +}; + /* * chip addresses on I2C bus */ @@ -137,12 +142,13 @@ static struct snd_akm4xxx akm_juli_dac __devinitdata = { static int __devinit juli_add_controls(struct snd_ice1712 *ice) { + struct juli_spec *spec = ice->spec; int err; err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; /* only capture SPDIF over AK4114 */ - err = snd_ak4114_build(ice->spec.juli.ak4114, NULL, + err = snd_ak4114_build(spec->ak4114, NULL, ice->pcm_pro->streams[SNDRV_PCM_STREAM_CAPTURE].substream); if (err < 0) return err; @@ -166,13 +172,19 @@ static int __devinit juli_init(struct snd_ice1712 *ice) 0x41, 0x02, 0x2c, 0x00, 0x00 }; int err; + struct juli_spec *spec; struct snd_akm4xxx *ak; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + err = snd_ak4114_create(ice->card, juli_ak4114_read, juli_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, - ice, &ice->spec.juli.ak4114); + ice, &spec->ak4114); if (err < 0) return err; @@ -180,12 +192,12 @@ static int __devinit juli_init(struct snd_ice1712 *ice) /* it seems that the analog doughter board detection does not work reliably, so force the analog flag; it should be very rare to use Juli@ without the analog doughter board */ - ice->spec.juli.analog = (ice->gpio.get_data(ice) & GPIO_ANALOG_PRESENT) ? 0 : 1; + spec->analog = (ice->gpio.get_data(ice) & GPIO_ANALOG_PRESENT) ? 0 : 1; #else - ice->spec.juli.analog = 1; + spec->analog = 1; #endif - if (ice->spec.juli.analog) { + if (spec->analog) { printk(KERN_INFO "juli@: analog I/O detected\n"); ice->num_total_dacs = 2; ice->num_total_adcs = 2; diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index 718e935..9ab4a9f 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -47,6 +47,12 @@ #include "phase.h" #include +/* AC97 register cache for Phase28 */ +struct phase28_spec { + unsigned short master[2]; + unsigned short vol[8]; +} phase28; + /* WM8770 registers */ #define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */ #define WM_DAC_MASTER_ATTEN 0x08 /* DAC master analog attenuation */ @@ -312,15 +318,17 @@ static int wm_master_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int wm_master_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct phase28_spec *spec = ice->spec; int i; for (i=0; i<2; i++) - ucontrol->value.integer.value[i] = ice->spec.phase28.master[i] & ~WM_VOL_MUTE; + ucontrol->value.integer.value[i] = spec->master[i] & ~WM_VOL_MUTE; return 0; } static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct phase28_spec *spec = ice->spec; int ch, change = 0; snd_ice1712_save_gpio_status(ice); @@ -328,14 +336,14 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ unsigned int vol = ucontrol->value.integer.value[ch]; if (vol > WM_VOL_MAX) continue; - vol |= ice->spec.phase28.master[ch] & WM_VOL_MUTE; - if (vol != ice->spec.phase28.master[ch]) { + vol |= spec->master[ch] & WM_VOL_MUTE; + if (vol != spec->master[ch]) { int dac; - ice->spec.phase28.master[ch] = vol; + spec->master[ch] = vol; for (dac = 0; dac < ice->num_total_dacs; dac += 2) wm_set_vol(ice, WM_DAC_ATTEN + dac + ch, - ice->spec.phase28.vol[dac + ch], - ice->spec.phase28.master[ch]); + spec->vol[dac + ch], + spec->master[ch]); change = 1; } } @@ -384,12 +392,18 @@ static int __devinit phase28_init(struct snd_ice1712 *ice) unsigned int tmp; struct snd_akm4xxx *ak; + struct phase28_spec *spec; const unsigned short *p; int i; ice->num_total_dacs = 8; ice->num_total_adcs = 2; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + // Initialize analog chips ak = ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (!ak) @@ -419,11 +433,11 @@ static int __devinit phase28_init(struct snd_ice1712 *ice) snd_ice1712_restore_gpio_status(ice); - ice->spec.phase28.master[0] = WM_VOL_MUTE; - ice->spec.phase28.master[1] = WM_VOL_MUTE; + spec->master[0] = WM_VOL_MUTE; + spec->master[1] = WM_VOL_MUTE; for (i = 0; i < ice->num_total_dacs; i++) { - ice->spec.phase28.vol[i] = WM_VOL_MUTE; - wm_set_vol(ice, i, ice->spec.phase28.vol[i], ice->spec.phase28.master[i % 2]); + spec->vol[i] = WM_VOL_MUTE; + wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); } return 0; @@ -445,18 +459,21 @@ static int wm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info * static int wm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct phase28_spec *spec = ice->spec; int i, ofs, voices; voices = kcontrol->private_value >> 8; ofs = kcontrol->private_value & 0xff; for (i = 0; i < voices; i++) - ucontrol->value.integer.value[i] = ice->spec.phase28.vol[ofs+i] & ~WM_VOL_MUTE; + ucontrol->value.integer.value[i] = + spec->vol[ofs+i] & ~WM_VOL_MUTE; return 0; } static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct phase28_spec *spec = ice->spec; int i, idx, ofs, voices; int change = 0; @@ -468,12 +485,12 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * vol = ucontrol->value.integer.value[i]; if (vol > 0x7f) continue; - vol |= ice->spec.phase28.vol[ofs+i] & WM_VOL_MUTE; - if (vol != ice->spec.phase28.vol[ofs+i]) { - ice->spec.phase28.vol[ofs+i] = vol; + vol |= spec->vol[ofs+i] & WM_VOL_MUTE; + if (vol != spec->vol[ofs+i]) { + spec->vol[ofs+i] = vol; idx = WM_DAC_ATTEN + ofs + i; - wm_set_vol(ice, idx, ice->spec.phase28.vol[ofs+i], - ice->spec.phase28.master[i]); + wm_set_vol(ice, idx, spec->vol[ofs+i], + spec->master[i]); change = 1; } } @@ -495,19 +512,22 @@ static int wm_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info static int wm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct phase28_spec *spec = ice->spec; int voices, ofs, i; voices = kcontrol->private_value >> 8; ofs = kcontrol->private_value & 0xFF; for (i = 0; i < voices; i++) - ucontrol->value.integer.value[i] = (ice->spec.phase28.vol[ofs+i] & WM_VOL_MUTE) ? 0 : 1; + ucontrol->value.integer.value[i] = + (spec->vol[ofs+i] & WM_VOL_MUTE) ? 0 : 1; return 0; } static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct phase28_spec *spec = ice->spec; int change = 0, voices, ofs, i; voices = kcontrol->private_value >> 8; @@ -515,13 +535,13 @@ static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value snd_ice1712_save_gpio_status(ice); for (i = 0; i < voices; i++) { - int val = (ice->spec.phase28.vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1; + int val = (spec->vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1; if (ucontrol->value.integer.value[i] != val) { - ice->spec.phase28.vol[ofs + i] &= ~WM_VOL_MUTE; - ice->spec.phase28.vol[ofs + i] |= + spec->vol[ofs + i] &= ~WM_VOL_MUTE; + spec->vol[ofs + i] |= ucontrol->value.integer.value[i] ? 0 : WM_VOL_MUTE; - wm_set_vol(ice, ofs + i, ice->spec.phase28.vol[ofs + i], - ice->spec.phase28.master[i]); + wm_set_vol(ice, ofs + i, spec->vol[ofs + i], + spec->master[i]); change = 1; } } @@ -538,29 +558,33 @@ static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value static int wm_master_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct phase28_spec *spec = ice->spec; - ucontrol->value.integer.value[0] = (ice->spec.phase28.master[0] & WM_VOL_MUTE) ? 0 : 1; - ucontrol->value.integer.value[1] = (ice->spec.phase28.master[1] & WM_VOL_MUTE) ? 0 : 1; + ucontrol->value.integer.value[0] = + (spec->master[0] & WM_VOL_MUTE) ? 0 : 1; + ucontrol->value.integer.value[1] = + (spec->master[1] & WM_VOL_MUTE) ? 0 : 1; return 0; } static int wm_master_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct phase28_spec *spec = ice->spec; int change = 0, i; snd_ice1712_save_gpio_status(ice); for (i = 0; i < 2; i++) { - int val = (ice->spec.phase28.master[i] & WM_VOL_MUTE) ? 0 : 1; + int val = (spec->master[i] & WM_VOL_MUTE) ? 0 : 1; if (ucontrol->value.integer.value[i] != val) { int dac; - ice->spec.phase28.master[i] &= ~WM_VOL_MUTE; - ice->spec.phase28.master[i] |= + spec->master[i] &= ~WM_VOL_MUTE; + spec->master[i] |= ucontrol->value.integer.value[i] ? 0 : WM_VOL_MUTE; for (dac = 0; dac < ice->num_total_dacs; dac += 2) wm_set_vol(ice, WM_DAC_ATTEN + dac + i, - ice->spec.phase28.vol[dac + i], - ice->spec.phase28.master[i]); + spec->vol[dac + i], + spec->master[i]); change = 1; } } diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 7339378..48cf40a 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -67,6 +67,12 @@ #include "stac946x.h" #include +struct prodigy192_spec { + struct ak4114 *ak4114; + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + static inline void stac9460_put(struct snd_ice1712 *ice, int reg, unsigned char val) { snd_vt1724_write_i2c(ice, PRODIGY192_STAC9460_ADDR, reg, val); @@ -118,6 +124,7 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy192_spec *spec = ice->spec; int idx, change; if (kcontrol->private_value) @@ -125,11 +132,11 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e else idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME; /* due to possible conflicts with stac9460_set_rate_val, mutexing */ - mutex_lock(&ice->spec.prodigy192.mute_mutex); + mutex_lock(&spec->mute_mutex); /*printk("Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx, ucontrol->value.integer.value[0]);*/ change = stac9460_dac_mute(ice, idx, ucontrol->value.integer.value[0]); - mutex_unlock(&ice->spec.prodigy192.mute_mutex); + mutex_unlock(&spec->mute_mutex); return change; } @@ -318,6 +325,7 @@ static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) int idx; unsigned char changed[7]; struct snd_ice1712 *ice = ak->private_data[0]; + struct prodigy192_spec *spec = ice->spec; if (rate == 0) /* no hint - S/PDIF input is master, simply return */ return; @@ -332,7 +340,7 @@ static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) return; /* change detected, setting master clock, muting first */ /* due to possible conflicts with mute controls - mutexing */ - mutex_lock(&ice->spec.prodigy192.mute_mutex); + mutex_lock(&spec->mute_mutex); /* we have to remember current mute status for each DAC */ for (idx = 0; idx < 7 ; ++idx) changed[idx] = stac9460_dac_mute(ice, @@ -346,7 +354,7 @@ static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) if (changed[idx]) stac9460_dac_mute(ice, STAC946X_MASTER_VOLUME + idx, 1); } - mutex_unlock(&ice->spec.prodigy192.mute_mutex); + mutex_unlock(&spec->mute_mutex); } /* using akm infrastructure for setting rate of the codec */ @@ -633,12 +641,13 @@ static int prodigy192_ak4114_init(struct snd_ice1712 *ice) static const unsigned char ak4114_init_txcsb[] = { 0x41, 0x02, 0x2c, 0x00, 0x00 }; + struct prodigy192_spec *spec = ice->spec; return snd_ak4114_create(ice->card, prodigy192_ak4114_read, prodigy192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, - ice, &ice->spec.prodigy192.ak4114); + ice, &spec->ak4114); } static void stac9460_proc_regs_read(struct snd_info_entry *entry, @@ -664,6 +673,7 @@ static void stac9460_proc_init(struct snd_ice1712 *ice) static int __devinit prodigy192_add_controls(struct snd_ice1712 *ice) { + struct prodigy192_spec *spec = ice->spec; unsigned int i; int err; @@ -673,7 +683,7 @@ static int __devinit prodigy192_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; } - if (ice->spec.prodigy192.ak4114) { + if (spec->ak4114) { /* ak4114 is connected */ for (i = 0; i < ARRAY_SIZE(ak4114_controls); i++) { err = snd_ctl_add(ice->card, @@ -682,7 +692,7 @@ static int __devinit prodigy192_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; } - err = snd_ak4114_build(ice->spec.prodigy192.ak4114, + err = snd_ak4114_build(spec->ak4114, NULL, /* ak4114 in MIO/DI/O handles no IEC958 output */ ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); if (err < 0) @@ -734,12 +744,19 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) const unsigned short *p; int err = 0; struct snd_akm4xxx *ak; + struct prodigy192_spec *spec; /* prodigy 192 */ ice->num_total_dacs = 6; ice->num_total_adcs = 2; ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + /* initialize codec */ p = stac_inits_prodigy; for (; *p != (unsigned short)-1; p += 2) @@ -758,7 +775,7 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) if (prodigy192_miodio_exists(ice)) { err = prodigy192_ak4114_init(ice); /* from this moment if err = 0 then - * ice->spec.prodigy192.ak4114 should not be null + * spec->ak4114 should not be null */ snd_printdd("AK4114 initialized with status %d\n", err); } else @@ -766,8 +783,6 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) if (err < 0) return err; - mutex_init(&ice->spec.prodigy192.mute_mutex); - return 0; } diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 6ec1aa4..043a938 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -40,6 +40,11 @@ #include "envy24ht.h" #include "prodigy_hifi.h" +struct prodigy_hifi_spec { + unsigned short master[2]; + unsigned short vol[8]; +}; + /* I2C addresses */ #define WM_DEV 0x34 @@ -177,7 +182,8 @@ static void wm8766_spi_send_word(struct snd_ice1712 *ice, unsigned int data) } } -static void wm8766_spi_write(struct snd_ice1712 *ice, unsigned int reg, unsigned int data) +static void wm8766_spi_write(struct snd_ice1712 *ice, unsigned int reg, + unsigned int data) { unsigned int block; @@ -216,7 +222,8 @@ static void ak4396_send_word(struct snd_ice1712 *ice, unsigned int data) } } -static void ak4396_write(struct snd_ice1712 *ice, unsigned int reg, unsigned int data) +static void ak4396_write(struct snd_ice1712 *ice, unsigned int reg, + unsigned int data) { unsigned int block; @@ -246,7 +253,8 @@ static void ak4396_write(struct snd_ice1712 *ice, unsigned int reg, unsigned int * DAC volume attenuation mixer control (-64dB to 0dB) */ -static int ak4396_dac_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int ak4396_dac_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -255,31 +263,32 @@ static int ak4396_dac_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ele return 0; } -static int ak4396_dac_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int ak4396_dac_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy_hifi_spec *spec = ice->spec; int i; - for (i = 0; i < 2; i++) { - ucontrol->value.integer.value[i] =ice->spec.prodigy_hd2.vol[i]; - } + for (i = 0; i < 2; i++) + ucontrol->value.integer.value[i] = spec->vol[i]; + return 0; } static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy_hifi_spec *spec = ice->spec; int i; int change = 0; mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { - if (ucontrol->value.integer.value[i] != - ice->spec.prodigy_hd2.vol[i]) { - ice->spec.prodigy_hd2.vol[i] = - ucontrol->value.integer.value[i]; - ak4396_write(ice, AK4396_LCH_ATT+i, - (ice->spec.prodigy_hd2.vol[i] & 0xff)); + if (ucontrol->value.integer.value[i] != spec->vol[i]) { + spec->vol[i] = ucontrol->value.integer.value[i]; + ak4396_write(ice, AK4396_LCH_ATT + i, + spec->vol[i] & 0xff); change = 1; } } @@ -333,7 +342,8 @@ static const unsigned char wm_vol[256] = { #define DAC_MIN (DAC_0dB - DAC_RES) -static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) +static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, + unsigned short vol, unsigned short master) { unsigned char nvol; @@ -349,7 +359,8 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho wm_put_nocache(ice, index, 0x100 | nvol); } -static void wm8766_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) +static void wm8766_set_vol(struct snd_ice1712 *ice, unsigned int index, + unsigned short vol, unsigned short master) { unsigned char nvol; @@ -360,7 +371,6 @@ static void wm8766_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned & WM_VOL_MAX; nvol = (nvol ? (nvol + DAC_MIN) : 0) & 0xff; } - wm8766_spi_write(ice, index, (0x0100 | nvol)); } @@ -370,7 +380,8 @@ static void wm8766_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned * DAC volume attenuation mixer control (-64dB to 0dB) */ -static int wm_dac_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int wm_dac_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -379,33 +390,32 @@ static int wm_dac_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_in return 0; } -static int wm_dac_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_dac_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy_hifi_spec *spec = ice->spec; int i; - for (i = 0; i < 2; i++) { + for (i = 0; i < 2; i++) ucontrol->value.integer.value[i] = - ice->spec.prodigy_hifi.vol[2+i] & ~WM_VOL_MUTE; - } + spec->vol[2 + i] & ~WM_VOL_MUTE; return 0; } static int wm_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy_hifi_spec *spec = ice->spec; int i, idx, change = 0; mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { - if (ucontrol->value.integer.value[i] != - ice->spec.prodigy_hifi.vol[2+i]) { + if (ucontrol->value.integer.value[i] != spec->vol[2 + i]) { idx = WM_DAC_ATTEN_L + i; - ice->spec.prodigy_hifi.vol[2+i] &= WM_VOL_MUTE; - ice->spec.prodigy_hifi.vol[2+i] |= - ucontrol->value.integer.value[i]; - wm_set_vol(ice, idx, ice->spec.prodigy_hifi.vol[2+i], - ice->spec.prodigy_hifi.master[i]); + spec->vol[2 + i] &= WM_VOL_MUTE; + spec->vol[2 + i] |= ucontrol->value.integer.value[i]; + wm_set_vol(ice, idx, spec->vol[2 + i], spec->master[i]); change = 1; } } @@ -417,7 +427,8 @@ static int wm_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val /* * WM8766 DAC volume attenuation mixer control */ -static int wm8766_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int wm8766_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { int voices = kcontrol->private_value >> 8; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -427,22 +438,24 @@ static int wm8766_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_in return 0; } -static int wm8766_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm8766_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy_hifi_spec *spec = ice->spec; int i, ofs, voices; voices = kcontrol->private_value >> 8; ofs = kcontrol->private_value & 0xff; for (i = 0; i < voices; i++) - ucontrol->value.integer.value[i] = - ice->spec.prodigy_hifi.vol[ofs+i]; + ucontrol->value.integer.value[i] = spec->vol[ofs + i]; return 0; } static int wm8766_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy_hifi_spec *spec = ice->spec; int i, idx, ofs, voices; int change = 0; @@ -450,15 +463,12 @@ static int wm8766_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val ofs = kcontrol->private_value & 0xff; mutex_lock(&ice->gpio_mutex); for (i = 0; i < voices; i++) { - if (ucontrol->value.integer.value[i] != - ice->spec.prodigy_hifi.vol[ofs+i]) { + if (ucontrol->value.integer.value[i] != spec->vol[ofs + i]) { idx = WM8766_LDA1 + ofs + i; - ice->spec.prodigy_hifi.vol[ofs+i] &= WM_VOL_MUTE; - ice->spec.prodigy_hifi.vol[ofs+i] |= - ucontrol->value.integer.value[i]; + spec->vol[ofs + i] &= WM_VOL_MUTE; + spec->vol[ofs + i] |= ucontrol->value.integer.value[i]; wm8766_set_vol(ice, idx, - ice->spec.prodigy_hifi.vol[ofs+i], - ice->spec.prodigy_hifi.master[i]); + spec->vol[ofs + i], spec->master[i]); change = 1; } } @@ -469,7 +479,8 @@ static int wm8766_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val /* * Master volume attenuation mixer control / applied to WM8776+WM8766 */ -static int wm_master_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int wm_master_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -478,45 +489,41 @@ static int wm_master_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem return 0; } -static int wm_master_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_master_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy_hifi_spec *spec = ice->spec; int i; - for (i=0; i<2; i++) - ucontrol->value.integer.value[i] = - ice->spec.prodigy_hifi.master[i]; + for (i = 0; i < 2; i++) + ucontrol->value.integer.value[i] = spec->master[i]; return 0; } -static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct prodigy_hifi_spec *spec = ice->spec; int ch, change = 0; mutex_lock(&ice->gpio_mutex); for (ch = 0; ch < 2; ch++) { - if (ucontrol->value.integer.value[ch] != - ice->spec.prodigy_hifi.master[ch]) { - ice->spec.prodigy_hifi.master[ch] &= 0x00; - ice->spec.prodigy_hifi.master[ch] |= - ucontrol->value.integer.value[ch]; + if (ucontrol->value.integer.value[ch] != spec->master[ch]) { + spec->master[ch] = ucontrol->value.integer.value[ch]; /* Apply to front DAC */ wm_set_vol(ice, WM_DAC_ATTEN_L + ch, - ice->spec.prodigy_hifi.vol[2 + ch], - ice->spec.prodigy_hifi.master[ch]); + spec->vol[2 + ch], spec->master[ch]); wm8766_set_vol(ice, WM8766_LDA1 + ch, - ice->spec.prodigy_hifi.vol[0 + ch], - ice->spec.prodigy_hifi.master[ch]); + spec->vol[0 + ch], spec->master[ch]); wm8766_set_vol(ice, WM8766_LDA2 + ch, - ice->spec.prodigy_hifi.vol[4 + ch], - ice->spec.prodigy_hifi.master[ch]); + spec->vol[4 + ch], spec->master[ch]); wm8766_set_vol(ice, WM8766_LDA3 + ch, - ice->spec.prodigy_hifi.vol[6 + ch], - ice->spec.prodigy_hifi.master[ch]); + spec->vol[6 + ch], spec->master[ch]); change = 1; } } @@ -525,64 +532,71 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ } - /* KONSTI */ -static int wm_adc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int wm_adc_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { - static char* texts[32] = {"NULL", WM_AIN1, WM_AIN2, WM_AIN1 "+" WM_AIN2, - WM_AIN3, WM_AIN1 "+" WM_AIN3, WM_AIN2 "+" WM_AIN3, - WM_AIN1 "+" WM_AIN2 "+" WM_AIN3, - WM_AIN4, WM_AIN1 "+" WM_AIN4, WM_AIN2 "+" WM_AIN4, - WM_AIN1 "+" WM_AIN2 "+" WM_AIN4, - WM_AIN3 "+" WM_AIN4, WM_AIN1 "+" WM_AIN3 "+" WM_AIN4, - WM_AIN2 "+" WM_AIN3 "+" WM_AIN4, - WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4, - WM_AIN5, WM_AIN1 "+" WM_AIN5, WM_AIN2 "+" WM_AIN5, - WM_AIN1 "+" WM_AIN2 "+" WM_AIN5, - WM_AIN3 "+" WM_AIN5, WM_AIN1 "+" WM_AIN3 "+" WM_AIN5, - WM_AIN2 "+" WM_AIN3 "+" WM_AIN5, - WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN5, - WM_AIN4 "+" WM_AIN5, WM_AIN1 "+" WM_AIN4 "+" WM_AIN5, - WM_AIN2 "+" WM_AIN4 "+" WM_AIN5, - WM_AIN1 "+" WM_AIN2 "+" WM_AIN4 "+" WM_AIN5, - WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, - WM_AIN1 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, - WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, - WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5}; + static char* texts[32] = { + "NULL", WM_AIN1, WM_AIN2, WM_AIN1 "+" WM_AIN2, + WM_AIN3, WM_AIN1 "+" WM_AIN3, WM_AIN2 "+" WM_AIN3, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN3, + WM_AIN4, WM_AIN1 "+" WM_AIN4, WM_AIN2 "+" WM_AIN4, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN4, + WM_AIN3 "+" WM_AIN4, WM_AIN1 "+" WM_AIN3 "+" WM_AIN4, + WM_AIN2 "+" WM_AIN3 "+" WM_AIN4, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4, + WM_AIN5, WM_AIN1 "+" WM_AIN5, WM_AIN2 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN5, + WM_AIN3 "+" WM_AIN5, WM_AIN1 "+" WM_AIN3 "+" WM_AIN5, + WM_AIN2 "+" WM_AIN3 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN5, + WM_AIN4 "+" WM_AIN5, WM_AIN1 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN2 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5, + WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5 + }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 32; if (uinfo->value.enumerated.item > 31) uinfo->value.enumerated.item = 31; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); return 0; } -static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0]=wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } -static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short oval, nval; + int change = 0; mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = ( oval & 0xe0 ) | ucontrol->value.integer.value[0] ; - if ( nval != oval ) { + nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); + change = 1; } mutex_unlock(&ice->gpio_mutex); - return 0; + return change; } /* KONSTI */ @@ -595,7 +609,8 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele #define ADC_RES 128 #define ADC_MIN (ADC_0dB - ADC_RES) -static int wm_adc_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int wm_adc_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -604,7 +619,8 @@ static int wm_adc_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_in return 0; } -static int wm_adc_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_adc_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -620,7 +636,8 @@ static int wm_adc_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val return 0; } -static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short ovol, nvol; @@ -644,27 +661,23 @@ static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val /* * ADC input mux mixer control */ -static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define wm_adc_mux_info snd_ctl_boolean_mono_info -static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int bit = kcontrol->private_value; mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = (wm_get(ice, WM_ADC_MUX) & (1 << bit)) ? 1 : 0; + ucontrol->value.integer.value[0] = + (wm_get(ice, WM_ADC_MUX) & (1 << bit)) ? 1 : 0; mutex_unlock(&ice->gpio_mutex); return 0; } -static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int bit = kcontrol->private_value; @@ -688,26 +701,22 @@ static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val /* * Analog bypass (In -> Out) */ -static int wm_bypass_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define wm_bypass_info snd_ctl_boolean_mono_info -static int wm_bypass_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = (wm_get(ice, WM_OUT_MUX) & 0x04) ? 1 : 0; + ucontrol->value.integer.value[0] = + (wm_get(ice, WM_OUT_MUX) & 0x04) ? 1 : 0; mutex_unlock(&ice->gpio_mutex); return 0; } -static int wm_bypass_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short val, oval; @@ -730,16 +739,10 @@ static int wm_bypass_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu /* * Left/Right swap */ -static int wm_chswap_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define wm_chswap_info snd_ctl_boolean_mono_info -static int wm_chswap_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_chswap_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); @@ -750,7 +753,8 @@ static int wm_chswap_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu return 0; } -static int wm_chswap_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int wm_chswap_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short val, oval; @@ -894,9 +898,10 @@ static struct snd_kcontrol_new prodigy_hifi_controls[] __devinitdata = { /* * WM codec registers */ -static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +static void wm_proc_regs_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; char line[64]; unsigned int reg, val; mutex_lock(&ice->gpio_mutex); @@ -909,9 +914,10 @@ static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buf mutex_unlock(&ice->gpio_mutex); } -static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +static void wm_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; int reg, val; mutex_lock(&ice->gpio_mutex); @@ -925,7 +931,7 @@ static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buff static void wm_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; - if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { + if (!snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); entry->mode |= S_IWUSR; entry->c.text.write = wm_proc_regs_write; @@ -938,7 +944,8 @@ static int __devinit prodigy_hifi_add_controls(struct snd_ice1712 *ice) int err; for (i = 0; i < ARRAY_SIZE(prodigy_hifi_controls); i++) { - err = snd_ctl_add(ice->card, snd_ctl_new1(&prodigy_hifi_controls[i], ice)); + err = snd_ctl_add(ice->card, + snd_ctl_new1(&prodigy_hifi_controls[i], ice)); if (err < 0) return err; } @@ -950,18 +957,19 @@ static int __devinit prodigy_hifi_add_controls(struct snd_ice1712 *ice) static int __devinit prodigy_hd2_add_controls(struct snd_ice1712 *ice) { - unsigned int i; - int err; + unsigned int i; + int err; - for (i = 0; i < ARRAY_SIZE(prodigy_hd2_controls); i++) { - err = snd_ctl_add(ice->card, snd_ctl_new1(&prodigy_hd2_controls[i], ice)); - if (err < 0) - return err; - } + for (i = 0; i < ARRAY_SIZE(prodigy_hd2_controls); i++) { + err = snd_ctl_add(ice->card, + snd_ctl_new1(&prodigy_hd2_controls[i], ice)); + if (err < 0) + return err; + } - wm_proc_init(ice); + wm_proc_init(ice); - return 0; + return 0; } @@ -1025,7 +1033,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) WM8766_MUTE2, 0x0000, }; - + struct prodigy_hifi_spec *spec; unsigned int i; ice->vt1720 = 0; @@ -1033,7 +1041,6 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) ice->num_total_dacs = 8; ice->num_total_adcs = 1; - ice->akm_codecs = 2; /* HACK - use this as the SPDIF source. * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten @@ -1044,7 +1051,12 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ice->akm) return -ENOMEM; - ice->akm_codecs = 1; + ice->akm_codecs = 1; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; /* initialize WM8776 codec */ for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2) @@ -1054,7 +1066,6 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) wm_put(ice, wm_inits2[i], wm_inits2[i+1]); /* initialize WM8766 codec */ - for (i = 0; i < ARRAY_SIZE(wm8766_inits); i += 2) wm8766_spi_write(ice, wm8766_inits[i], wm8766_inits[i+1]); @@ -1076,7 +1087,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) AK4396_RCH_ATT, 0x00, }; - + struct prodigy_hifi_spec *spec; unsigned int i; ice->vt1720 = 0; @@ -1084,7 +1095,6 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) ice->num_total_dacs = 1; ice->num_total_adcs = 1; - ice->akm_codecs = 1; /* HACK - use this as the SPDIF source. * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten @@ -1097,6 +1107,11 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) return -ENOMEM; ice->akm_codecs = 1; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + /* initialize ak4396 codec */ /* reset codec */ ak4396_write(ice, AK4396_CTRL1, 0x86); diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 05a751c..ddd5fc8 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -32,6 +32,12 @@ #include "envy24ht.h" #include "revo.h" +/* a non-standard I2C device for revo51 */ +struct revo51_spec { + struct snd_i2c_device *dev; + struct snd_pt2258 *pt2258; +} revo51; + static void revo_i2s_mclk_changed(struct snd_ice1712 *ice) { /* assert PRST# to converters; MT05 bit 7 */ @@ -152,8 +158,14 @@ static struct snd_i2c_bit_ops revo51_bit_ops = { static int revo51_i2c_init(struct snd_ice1712 *ice, struct snd_pt2258 *pt) { + struct revo51_spec *spec; int err; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + /* create the I2C bus */ err = snd_i2c_bus_create(ice->card, "ICE1724 GPIO6", NULL, &ice->i2c); if (err < 0) @@ -163,15 +175,14 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, ice->i2c->hw_ops.bit = &revo51_bit_ops; /* create the I2C device */ - err = snd_i2c_device_create(ice->i2c, "PT2258", 0x40, - &ice->spec.revo51.dev); + err = snd_i2c_device_create(ice->i2c, "PT2258", 0x40, &spec->dev); if (err < 0) return err; pt->card = ice->card; pt->i2c_bus = ice->i2c; - pt->i2c_dev = ice->spec.revo51.dev; - ice->spec.revo51.pt2258 = pt; + pt->i2c_dev = spec->dev; + spec->pt2258 = pt; snd_pt2258_reset(pt); @@ -555,6 +566,7 @@ static int __devinit revo_init(struct snd_ice1712 *ice) static int __devinit revo_add_controls(struct snd_ice1712 *ice) { + struct revo51_spec *spec; int err; switch (ice->eeprom.subvendor) { @@ -567,7 +579,8 @@ static int __devinit revo_add_controls(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; - err = snd_pt2258_build_controls(ice->spec.revo51.pt2258); + spec = ice->spec; + err = snd_pt2258_build_controls(spec->pt2258); if (err < 0) return err; break; diff --git a/sound/pci/ice1712/se.c b/sound/pci/ice1712/se.c index 6661f65..69673b9 100644 --- a/sound/pci/ice1712/se.c +++ b/sound/pci/ice1712/se.c @@ -34,6 +34,11 @@ #include "envy24ht.h" #include "se.h" +struct se_spec { + struct { + unsigned char ch1, ch2; + } vol[8]; +}; /****************************************************************************/ /* ONKYO WAVIO SE-200PCI */ @@ -462,9 +467,10 @@ static int se200pci_cont_volume_get(struct snd_kcontrol *kc, struct snd_ctl_elem_value *uc) { struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct se_spec *spec = ice->spec; int n = kc->private_value; - uc->value.integer.value[0] = ice->spec.se.vol[n].ch1; - uc->value.integer.value[1] = ice->spec.se.vol[n].ch2; + uc->value.integer.value[0] = spec->vol[n].ch1; + uc->value.integer.value[1] = spec->vol[n].ch2; return 0; } @@ -472,8 +478,9 @@ static int se200pci_cont_boolean_get(struct snd_kcontrol *kc, struct snd_ctl_elem_value *uc) { struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct se_spec *spec = ice->spec; int n = kc->private_value; - uc->value.integer.value[0] = ice->spec.se.vol[n].ch1; + uc->value.integer.value[0] = spec->vol[n].ch1; return 0; } @@ -481,44 +488,46 @@ static int se200pci_cont_enum_get(struct snd_kcontrol *kc, struct snd_ctl_elem_value *uc) { struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct se_spec *spec = ice->spec; int n = kc->private_value; - uc->value.enumerated.item[0] = ice->spec.se.vol[n].ch1; + uc->value.enumerated.item[0] = spec->vol[n].ch1; return 0; } static void se200pci_cont_update(struct snd_ice1712 *ice, int n) { + struct se_spec *spec = ice->spec; switch (se200pci_cont[n].target) { case WM8766: se200pci_WM8766_set_volume(ice, se200pci_cont[n].ch, - ice->spec.se.vol[n].ch1, - ice->spec.se.vol[n].ch2); + spec->vol[n].ch1, + spec->vol[n].ch2); break; case WM8776in: se200pci_WM8776_set_input_volume(ice, - ice->spec.se.vol[n].ch1, - ice->spec.se.vol[n].ch2); + spec->vol[n].ch1, + spec->vol[n].ch2); break; case WM8776out: se200pci_WM8776_set_output_volume(ice, - ice->spec.se.vol[n].ch1, - ice->spec.se.vol[n].ch2); + spec->vol[n].ch1, + spec->vol[n].ch2); break; case WM8776sel: se200pci_WM8776_set_input_selector(ice, - ice->spec.se.vol[n].ch1); + spec->vol[n].ch1); break; case WM8776agc: - se200pci_WM8776_set_agc(ice, ice->spec.se.vol[n].ch1); + se200pci_WM8776_set_agc(ice, spec->vol[n].ch1); break; case WM8776afl: - se200pci_WM8776_set_afl(ice, ice->spec.se.vol[n].ch1); + se200pci_WM8776_set_afl(ice, spec->vol[n].ch1); break; default: @@ -530,6 +539,7 @@ static int se200pci_cont_volume_put(struct snd_kcontrol *kc, struct snd_ctl_elem_value *uc) { struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct se_spec *spec = ice->spec; int n = kc->private_value; unsigned int vol1, vol2; int changed; @@ -537,12 +547,12 @@ static int se200pci_cont_volume_put(struct snd_kcontrol *kc, changed = 0; vol1 = uc->value.integer.value[0] & 0xff; vol2 = uc->value.integer.value[1] & 0xff; - if (ice->spec.se.vol[n].ch1 != vol1) { - ice->spec.se.vol[n].ch1 = vol1; + if (spec->vol[n].ch1 != vol1) { + spec->vol[n].ch1 = vol1; changed = 1; } - if (ice->spec.se.vol[n].ch2 != vol2) { - ice->spec.se.vol[n].ch2 = vol2; + if (spec->vol[n].ch2 != vol2) { + spec->vol[n].ch2 = vol2; changed = 1; } if (changed) @@ -555,12 +565,13 @@ static int se200pci_cont_boolean_put(struct snd_kcontrol *kc, struct snd_ctl_elem_value *uc) { struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct se_spec *spec = ice->spec; int n = kc->private_value; unsigned int vol1; vol1 = !!uc->value.integer.value[0]; - if (ice->spec.se.vol[n].ch1 != vol1) { - ice->spec.se.vol[n].ch1 = vol1; + if (spec->vol[n].ch1 != vol1) { + spec->vol[n].ch1 = vol1; se200pci_cont_update(ice, n); return 1; } @@ -571,14 +582,15 @@ static int se200pci_cont_enum_put(struct snd_kcontrol *kc, struct snd_ctl_elem_value *uc) { struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct se_spec *spec = ice->spec; int n = kc->private_value; unsigned int vol1; vol1 = uc->value.enumerated.item[0]; if (vol1 >= se200pci_get_enum_count(n)) return -EINVAL; - if (ice->spec.se.vol[n].ch1 != vol1) { - ice->spec.se.vol[n].ch1 = vol1; + if (spec->vol[n].ch1 != vol1) { + spec->vol[n].ch1 = vol1; se200pci_cont_update(ice, n); return 1; } @@ -668,6 +680,13 @@ static int __devinit se200pci_add_controls(struct snd_ice1712 *ice) static int __devinit se_init(struct snd_ice1712 *ice) { + struct se_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_SE90PCI) { ice->num_total_dacs = 2; ice->num_total_adcs = 0; -- cgit v0.10.2 From 28a0d9df0706c523d3bb8de18892c76b78c2e428 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Jan 2008 15:32:32 +0100 Subject: [ALSA] hda-intel - Make azx_get_response() a bit more robust In azx_[rirb_]get_response(), the timeout is checked at the end of the loop. It's better to be checked just after the check of the RIRB index to avoid a bogus error with a too long msleep(). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fe07bdf..be5cbbc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -551,7 +551,7 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) again: timeout = jiffies + msecs_to_jiffies(1000); - do { + for (;;) { if (chip->polling_mode) { spin_lock_irq(&chip->reg_lock); azx_update_rirb(chip); @@ -559,13 +559,15 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) } if (!chip->rirb.cmds) return chip->rirb.res; /* the last value */ + if (time_after(jiffies, timeout)) + break; if (codec->bus->needs_damn_long_delay) msleep(2); /* temporary workaround */ else { udelay(10); cond_resched(); } - } while (time_after_eq(timeout, jiffies)); + } if (chip->msi) { snd_printk(KERN_WARNING "hda_intel: No response from codec, " -- cgit v0.10.2 From 6735e5723b5b4bc9bc0a24a841fbd0f2e7944b3f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 19 Jan 2008 10:33:07 +0100 Subject: [ALSA] emu10k1 - Fix over-sized kmalloc for TLV Reported by Al Viro: In copy_tlv(), the size of kmalloc is wrongly calculated. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index c35d9e1..354a892 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -665,7 +665,7 @@ static unsigned int *copy_tlv(const unsigned int __user *_tlv) return NULL; if (data[1] >= MAX_TLV_SIZE) return NULL; - tlv = kmalloc(data[1] * 4 + sizeof(data), GFP_KERNEL); + tlv = kmalloc(data[1] + sizeof(data), GFP_KERNEL); if (!tlv) return NULL; memcpy(tlv, data, sizeof(data)); -- cgit v0.10.2 From 48a8a26dd8ea1f7aeacef6b48f2d4d45508f46ba Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:43:06 +0100 Subject: [ALSA] oxygen: update ALSA-Configuration.txt Add documentation entries for snd-oxygen and snd-virtuoso. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 66390d7..a607111 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1559,6 +1559,23 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports only one card, autoprobe and PnP. + Module snd-oxygen + ----------------- + + Module for sound cards based on the C-Media CMI8788 chip: + * Asound A-8788 + * AuzenTech X-Meridian + * Bgears b-Enspirer + * Club3D Theatron DTS + * HT-Omega Claro + * Razer Barracuda AC-1 + * Sondigo Inferno + * TempoTec HIFIER + + This module supports autoprobe and multiple cards. + + Power management is _not_ supported. + Module snd-pcxhr ---------------- @@ -1997,6 +2014,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards. + Module snd-virtuoso + ------------------- + + Module for sound cards based on the Asus AV200 chip, i.e., + Xonar D2 and Xonar D2X. + + This module supports autoprobe and multiple cards. + + Power management is _not_ supported. + Module snd-vx222 ---------------- -- cgit v0.10.2 From c9946b2c807aa2e6829765accc267415a893f74a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:44:24 +0100 Subject: [ALSA] oxygen: remove magic numbers Replace some magic numbers with register symbols. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 3e61f94..bd0050b 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -208,21 +208,45 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC | chip->model->function_flags); - oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT, 0x010a); - oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, 0x010a); - oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, 0x010a); - oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, 0x010a); + oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT, + OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | + OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, + OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | + OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, + OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | + OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, + OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | + OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_RATE_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); - oxygen_write16(chip, OXYGEN_PLAY_ROUTING, 0xe400); - oxygen_write8(chip, OXYGEN_REC_ROUTING, 0x10); - oxygen_write8(chip, OXYGEN_ADC_MONITOR, 0x00); - oxygen_write8(chip, OXYGEN_A_MONITOR_ROUTING, 0xe4); + oxygen_write16(chip, OXYGEN_PLAY_ROUTING, + OXYGEN_PLAY_MULTICH_I2S_DAC | OXYGEN_PLAY_SPDIF_SPDIF | + (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT)); + oxygen_write8(chip, OXYGEN_REC_ROUTING, + OXYGEN_REC_A_ROUTE_I2S_ADC_1 | + OXYGEN_REC_B_ROUTE_AC97_1 | + OXYGEN_REC_C_ROUTE_SPDIF); + oxygen_write8(chip, OXYGEN_ADC_MONITOR, 0); + oxygen_write8(chip, OXYGEN_A_MONITOR_ROUTING, + (0 << OXYGEN_A_MONITOR_ROUTE_0_SHIFT) | + (1 << OXYGEN_A_MONITOR_ROUTE_1_SHIFT) | + (2 << OXYGEN_A_MONITOR_ROUTE_2_SHIFT) | + (3 << OXYGEN_A_MONITOR_ROUTE_3_SHIFT)); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); - oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0x00); + oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0); if (chip->has_ac97_0) { oxygen_clear_bits16(chip, OXYGEN_AC97_OUT_CONFIG, OXYGEN_AC97_CODEC0_FRONTL | diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 6fd2a59..d23d18a 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -120,10 +120,23 @@ static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) void oxygen_update_dac_routing(struct oxygen *chip) { + /* DAC 0: front, DAC 1: surround, DAC 2: center/LFE, DAC 3: back */ static const unsigned int reg_values[3] = { - 0xe400, /* front <- 0, surround <- 1, center <- 2, back <- 3 */ - 0xe000, /* front <- 0, surround <- 0, center <- 2, back <- 3 */ - 0x2000 /* front <- 0, surround <- 0, center <- 2, back <- 0 */ + /* stereo -> front */ + (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), + /* stereo -> front+surround */ + (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), + /* stereo -> front+surround+back */ + (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), }; u8 channels; unsigned int reg_value; @@ -133,10 +146,21 @@ void oxygen_update_dac_routing(struct oxygen *chip) if (channels == OXYGEN_PLAY_CHANNELS_2) reg_value = reg_values[chip->dac_routing]; else if (channels == OXYGEN_PLAY_CHANNELS_8) - reg_value = 0x6c00; /* surround <- 3, back <- 1 */ + /* in 7.1 mode, "rear" channels go to the "back" jack */ + reg_value = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (3 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (1 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); else - reg_value = 0xe400; - oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value, 0xff00); + reg_value = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); + oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value, + OXYGEN_PLAY_DAC0_SOURCE_MASK | + OXYGEN_PLAY_DAC1_SOURCE_MASK | + OXYGEN_PLAY_DAC2_SOURCE_MASK | + OXYGEN_PLAY_DAC3_SOURCE_MASK); } static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) @@ -199,13 +223,15 @@ void oxygen_update_spdif_source(struct oxygen *chip) old_routing = oxygen_read16(chip, OXYGEN_PLAY_ROUTING); if (chip->pcm_active & (1 << PCM_SPDIF)) { new_control = old_control | OXYGEN_SPDIF_OUT_ENABLE; - new_routing = (old_routing & ~0x00e0) | 0x0000; + new_routing = (old_routing & ~OXYGEN_PLAY_SPDIF_MASK) + | OXYGEN_PLAY_SPDIF_SPDIF; oxygen_rate = (old_control >> OXYGEN_SPDIF_OUT_RATE_SHIFT) & OXYGEN_I2S_RATE_MASK; /* S/PDIF rate was already set by the caller */ } else if ((chip->pcm_active & (1 << PCM_MULTICH)) && chip->spdif_playback_enable) { - new_routing = (old_routing & ~0x00e0) | 0x0020; + new_routing = (old_routing & ~OXYGEN_PLAY_SPDIF_MASK) + | OXYGEN_PLAY_SPDIF_MULTICH_01; oxygen_rate = oxygen_read16(chip, OXYGEN_I2S_MULTICH_FORMAT) & OXYGEN_I2S_RATE_MASK; new_control = (old_control & ~OXYGEN_SPDIF_OUT_RATE_MASK) | diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 72481fd..f147f97 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -396,7 +396,9 @@ static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, OXYGEN_I2S_FORMAT_MASK | OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); - oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x08); + oxygen_write8_masked(chip, OXYGEN_REC_ROUTING, + OXYGEN_REC_A_ROUTE_I2S_ADC_1, + OXYGEN_REC_A_ROUTE_MASK); spin_unlock_irq(&chip->reg_lock); mutex_lock(&chip->mutex); @@ -428,7 +430,9 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, OXYGEN_I2S_FORMAT_MASK | OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); - oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x10); + oxygen_write8_masked(chip, OXYGEN_REC_ROUTING, + OXYGEN_REC_B_ROUTE_I2S_ADC_2, + OXYGEN_REC_B_ROUTE_MASK); spin_unlock_irq(&chip->reg_lock); mutex_lock(&chip->mutex); @@ -451,7 +455,9 @@ static int oxygen_rec_c_hw_params(struct snd_pcm_substream *substream, oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, oxygen_format(hw_params) << OXYGEN_REC_FORMAT_C_SHIFT, OXYGEN_REC_FORMAT_C_MASK); - oxygen_clear_bits8(chip, OXYGEN_REC_ROUTING, 0x20); + oxygen_write8_masked(chip, OXYGEN_REC_ROUTING, + OXYGEN_REC_C_ROUTE_SPDIF, + OXYGEN_REC_C_ROUTE_MASK); spin_unlock_irq(&chip->reg_lock); return 0; } @@ -504,7 +510,11 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | OXYGEN_I2S_BITS_MASK); - oxygen_clear_bits16(chip, OXYGEN_PLAY_ROUTING, 0x001f); + oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, + OXYGEN_PLAY_MULTICH_I2S_DAC, + OXYGEN_PLAY_MUTE01 | OXYGEN_PLAY_MUTE23 | + OXYGEN_PLAY_MUTE45 | OXYGEN_PLAY_MUTE67 | + OXYGEN_PLAY_MULTICH_MASK); oxygen_update_dac_routing(chip); oxygen_update_spdif_source(chip); spin_unlock_irq(&chip->reg_lock); diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index 530f148..3e0cdae 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -360,10 +360,14 @@ #define OXYGEN_ADC_MONITOR_C_HALF_VOL 0x20 #define OXYGEN_A_MONITOR_ROUTING 0xc4 -#define OXYGEN_A_MONITOR_ROUTE_01_MASK 0x03 -#define OXYGEN_A_MONITOR_ROUTE_23_MASK 0x0c -#define OXYGEN_A_MONITOR_ROUTE_45_MASK 0x30 -#define OXYGEN_A_MONITOR_ROUTE_67_MASK 0xc0 +#define OXYGEN_A_MONITOR_ROUTE_0_MASK 0x03 +#define OXYGEN_A_MONITOR_ROUTE_0_SHIFT 0 +#define OXYGEN_A_MONITOR_ROUTE_1_MASK 0x0c +#define OXYGEN_A_MONITOR_ROUTE_1_SHIFT 2 +#define OXYGEN_A_MONITOR_ROUTE_2_MASK 0x30 +#define OXYGEN_A_MONITOR_ROUTE_2_SHIFT 4 +#define OXYGEN_A_MONITOR_ROUTE_3_MASK 0xc0 +#define OXYGEN_A_MONITOR_ROUTE_3_SHIFT 6 #define OXYGEN_AC97_CONTROL 0xd0 #define OXYGEN_AC97_COLD_RESET 0x0001 -- cgit v0.10.2 From db2396d4959340dbe2b617bde3beb2268f1e3658 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:44:52 +0100 Subject: [ALSA] oxygen: fix pause handling Use the DMA_PAUSE register for pausing instead of stopping DMA. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index f147f97..31b0ccd 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -570,16 +570,16 @@ static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd) struct oxygen *chip = snd_pcm_substream_chip(substream); struct snd_pcm_substream *s; unsigned int mask = 0; - int running; + int pausing; switch (cmd) { case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - running = 0; - break; case SNDRV_PCM_TRIGGER_START: + pausing = 0; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - running = 1; + pausing = 1; break; default: return -EINVAL; @@ -593,11 +593,18 @@ static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd) } spin_lock(&chip->reg_lock); - if (running) - chip->pcm_running |= mask; - else - chip->pcm_running &= ~mask; - oxygen_write8(chip, OXYGEN_DMA_STATUS, chip->pcm_running); + if (!pausing) { + if (cmd == SNDRV_PCM_TRIGGER_START) + chip->pcm_running |= mask; + else + chip->pcm_running &= ~mask; + oxygen_write8(chip, OXYGEN_DMA_STATUS, chip->pcm_running); + } else { + if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH) + oxygen_set_bits8(chip, OXYGEN_DMA_PAUSE, mask); + else + oxygen_clear_bits8(chip, OXYGEN_DMA_PAUSE, mask); + } spin_unlock(&chip->reg_lock); return 0; } -- cgit v0.10.2 From 44fb7aae82b37f5bb66cb1423e2babb11d90969e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:45:37 +0100 Subject: [ALSA] oxygen: remove MIDI autodetection The MIDI bit in the MISC register is set by default and cannot be used to detect the presence of a MIDI port. Instead, add a parameter to the oxygen_pci_probe() function so that model drivers can specify this. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index e0e54ab..b11341f 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -327,7 +327,7 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci, return -ENOENT; } model = pci_id->driver_data ? &model_meridian : &model_generic; - err = oxygen_pci_probe(pci, index[dev], id[dev], model); + err = oxygen_pci_probe(pci, index[dev], id[dev], 1, model); if (err >= 0) ++dev; return err; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 98cccc6..1c402cb 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -92,7 +92,7 @@ struct oxygen_model { /* oxygen_lib.c */ -int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, +int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, int midi, const struct oxygen_model *model); void oxygen_pci_remove(struct pci_dev *pci); diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index bd0050b..b7079ad 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -306,7 +306,7 @@ static void oxygen_card_free(struct snd_card *card) } int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, - const struct oxygen_model *model) + int midi, const struct oxygen_model *model) { struct snd_card *card; struct oxygen *chip; @@ -374,7 +374,9 @@ int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, if (err < 0) goto err_card; - if (oxygen_read8(chip, OXYGEN_MISC) & OXYGEN_MISC_MIDI) { + oxygen_write8_masked(chip, OXYGEN_MISC, + midi ? OXYGEN_MISC_MIDI : 0, OXYGEN_MISC_MIDI); + if (midi) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, chip->addr + OXYGEN_MPU401, MPU401_INFO_INTEGRATED, 0, 0, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 1a0367b..7397571 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -266,7 +266,7 @@ static int __devinit xonar_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], &model_xonar); + err = oxygen_pci_probe(pci, index[dev], id[dev], 1, &model_xonar); if (err >= 0) ++dev; return err; -- cgit v0.10.2 From 878ac3ee76a5abb4952396570207f6ebe0597e52 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:50:19 +0100 Subject: [ALSA] oxygen: add more symbols Add symbol definitions for the various codecs and GPIO pins. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/cm9780.h b/sound/pci/oxygen/cm9780.h new file mode 100644 index 0000000..1445967 --- /dev/null +++ b/sound/pci/oxygen/cm9780.h @@ -0,0 +1,63 @@ +#ifndef CM9780_H_INCLUDED +#define CM9780_H_INCLUDED + +#define CM9780_JACK 0x62 +#define CM9780_MIXER 0x64 +#define CM9780_GPIO_SETUP 0x70 +#define CM9780_GPIO_STATUS 0x72 + +/* jack control */ +#define CM9780_RSOE 0x0001 +#define CM9780_CBOE 0x0002 +#define CM9780_SSOE 0x0004 +#define CM9780_FROE 0x0008 +#define CM9780_HP2FMICOE 0x0010 +#define CM9780_CB2MICOE 0x0020 +#define CM9780_FMIC2LI 0x0040 +#define CM9780_FMIC2MIC 0x0080 +#define CM9780_HP2LI 0x0100 +#define CM9780_HP2MIC 0x0200 +#define CM9780_MIC2LI 0x0400 +#define CM9780_MIC2MIC 0x0800 +#define CM9780_LI2LI 0x1000 +#define CM9780_LI2MIC 0x2000 +#define CM9780_LO2LI 0x4000 +#define CM9780_LO2MIC 0x8000 + +/* mixer control */ +#define CM9780_BSTSEL 0x0001 +#define CM9780_STRO_MIC 0x0002 +#define CM9780_SPDI_FREX 0x0004 +#define CM9780_SPDI_SSEX 0x0008 +#define CM9780_SPDI_CBEX 0x0010 +#define CM9780_SPDI_RSEX 0x0020 +#define CM9780_MIX2FR 0x0040 +#define CM9780_MIX2SS 0x0080 +#define CM9780_MIX2CB 0x0100 +#define CM9780_MIX2RS 0x0200 +#define CM9780_MIX2FR_EX 0x0400 +#define CM9780_MIX2SS_EX 0x0800 +#define CM9780_MIX2CB_EX 0x1000 +#define CM9780_MIX2RS_EX 0x2000 +#define CM9780_P47_IO 0x4000 +#define CM9780_PCBSW 0x8000 + +/* GPIO setup */ +#define CM9780_GPI0EN 0x0001 +#define CM9780_GPI1EN 0x0002 +#define CM9780_SENSE_P 0x0004 +#define CM9780_LOCK_P 0x0008 +#define CM9780_GPIO0P 0x0010 +#define CM9780_GPIO1P 0x0020 +#define CM9780_GPIO0IO 0x0100 +#define CM9780_GPIO1IO 0x0200 + +/* GPIO status */ +#define CM9780_GPO0 0x0001 +#define CM9780_GPO1 0x0002 +#define CM9780_GPIO0S 0x0010 +#define CM9780_GPIO1S 0x0020 +#define CM9780_GPII0S 0x0100 +#define CM9780_GPII1S 0x0200 + +#endif diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index b11341f..78aa969 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -70,30 +70,92 @@ static struct pci_device_id oxygen_ids[] __devinitdata = { }; MODULE_DEVICE_TABLE(pci, oxygen_ids); + +#define GPIO_AK5385_DFS_MASK 0x0003 +#define GPIO_AK5385_DFS_NORMAL 0x0000 +#define GPIO_AK5385_DFS_DOUBLE 0x0001 +#define GPIO_AK5385_DFS_QUAD 0x0002 + #define AK4396_WRITE 0x2000 -/* register 0 */ +#define AK4396_CONTROL_1 0 +#define AK4396_CONTROL_2 1 +#define AK4396_CONTROL_3 2 +#define AK4396_LCH_ATT 3 +#define AK4396_RCH_ATT 4 + +/* control 1 */ #define AK4396_RSTN 0x01 +#define AK4396_DIF_MASK 0x0e +#define AK4396_DIF_16_LSB 0x00 +#define AK4396_DIF_20_LSB 0x02 #define AK4396_DIF_24_MSB 0x04 -/* register 1 */ +#define AK4396_DIF_24_I2S 0x06 +#define AK4396_DIF_24_LSB 0x08 +#define AK4396_ACKS 0x80 +/* control 2 */ #define AK4396_SMUTE 0x01 +#define AK4396_DEM_MASK 0x06 +#define AK4396_DEM_441 0x00 #define AK4396_DEM_OFF 0x02 +#define AK4396_DEM_48 0x04 +#define AK4396_DEM_32 0x06 #define AK4396_DFS_MASK 0x18 #define AK4396_DFS_NORMAL 0x00 #define AK4396_DFS_DOUBLE 0x08 #define AK4396_DFS_QUAD 0x10 - -/* register 0 */ +#define AK4396_SLOW 0x20 +#define AK4396_DZFM 0x40 +#define AK4396_DZFE 0x80 +/* control 3 */ +#define AK4396_DZFB 0x04 +#define AK4396_DCKB 0x10 +#define AK4396_DCKS 0x20 +#define AK4396_DSDM 0x40 +#define AK4396_D_P_MASK 0x80 +#define AK4396_PCM 0x00 +#define AK4396_DSD 0x80 + +#define WM8785_R0 0 +#define WM8785_R1 1 +#define WM8785_R2 2 +#define WM8785_R7 7 + +/* R0 */ +#define WM8785_MCR_MASK 0x007 +#define WM8785_MCR_SLAVE 0x000 +#define WM8785_MCR_MASTER_128 0x001 +#define WM8785_MCR_MASTER_192 0x002 +#define WM8785_MCR_MASTER_256 0x003 +#define WM8785_MCR_MASTER_384 0x004 +#define WM8785_MCR_MASTER_512 0x005 +#define WM8785_MCR_MASTER_768 0x006 +#define WM8785_OSR_MASK 0x018 #define WM8785_OSR_SINGLE 0x000 #define WM8785_OSR_DOUBLE 0x008 #define WM8785_OSR_QUAD 0x010 +#define WM8785_FORMAT_MASK 0x060 +#define WM8785_FORMAT_RJUST 0x000 #define WM8785_FORMAT_LJUST 0x020 #define WM8785_FORMAT_I2S 0x040 -/* register 1 */ +#define WM8785_FORMAT_DSP 0x060 +/* R1 */ +#define WM8785_WL_MASK 0x003 #define WM8785_WL_16 0x000 #define WM8785_WL_20 0x001 #define WM8785_WL_24 0x002 #define WM8785_WL_32 0x003 +#define WM8785_LRP 0x004 +#define WM8785_BCLKINV 0x008 +#define WM8785_LRSWAP 0x010 +#define WM8785_DEVNO_MASK 0x0e0 +/* R2 */ +#define WM8785_HPFR 0x001 +#define WM8785_HPFL 0x002 +#define WM8785_SDODIS 0x004 +#define WM8785_PWRDNR 0x008 +#define WM8785_PWRDNL 0x010 +#define WM8785_TDM_MASK 0x1c0 static void ak4396_write(struct oxygen *chip, unsigned int codec, u8 reg, u8 value) @@ -124,29 +186,33 @@ static void ak4396_init(struct oxygen *chip) { unsigned int i; - chip->ak4396_reg1 = AK4396_DEM_OFF | AK4396_DFS_NORMAL; + chip->ak4396_ctl2 = AK4396_DEM_OFF | AK4396_DFS_NORMAL; for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, 0, AK4396_DIF_24_MSB | AK4396_RSTN); - ak4396_write(chip, i, 1, chip->ak4396_reg1); - ak4396_write(chip, i, 2, 0); - ak4396_write(chip, i, 3, 0xff); - ak4396_write(chip, i, 4, 0xff); + ak4396_write(chip, i, + AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write(chip, i, + AK4396_CONTROL_2, chip->ak4396_ctl2); + ak4396_write(chip, i, + AK4396_CONTROL_3, AK4396_PCM); + ak4396_write(chip, i, AK4396_LCH_ATT, 0xff); + ak4396_write(chip, i, AK4396_RCH_ATT, 0xff); } snd_component_add(chip->card, "AK4396"); } static void ak5385_init(struct oxygen *chip) { - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, 0x0003); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, 0x0003); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_AK5385_DFS_MASK); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_AK5385_DFS_MASK); snd_component_add(chip->card, "AK5385"); } static void wm8785_init(struct oxygen *chip) { - wm8785_write(chip, 7, 0); - wm8785_write(chip, 0, WM8785_FORMAT_LJUST | WM8785_OSR_SINGLE); - wm8785_write(chip, 1, WM8785_WL_24); + wm8785_write(chip, WM8785_R7, 0); + wm8785_write(chip, WM8785_R0, WM8785_MCR_SLAVE | + WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST); + wm8785_write(chip, WM8785_R1, WM8785_WL_24); snd_component_add(chip->card, "WM8785"); } @@ -184,18 +250,21 @@ static void set_ak4396_params(struct oxygen *chip, unsigned int i; u8 value; - value = chip->ak4396_reg1 & ~AK4396_DFS_MASK; + value = chip->ak4396_ctl2 & ~AK4396_DFS_MASK; if (params_rate(params) <= 54000) value |= AK4396_DFS_NORMAL; else if (params_rate(params) < 120000) value |= AK4396_DFS_DOUBLE; else value |= AK4396_DFS_QUAD; - chip->ak4396_reg1 = value; + chip->ak4396_ctl2 = value; for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, 0, AK4396_DIF_24_MSB); - ak4396_write(chip, i, 1, value); - ak4396_write(chip, i, 0, AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write(chip, i, + AK4396_CONTROL_1, AK4396_DIF_24_MSB); + ak4396_write(chip, i, + AK4396_CONTROL_2, value); + ak4396_write(chip, i, + AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); } } @@ -204,8 +273,10 @@ static void update_ak4396_volume(struct oxygen *chip) unsigned int i; for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, 3, chip->dac_volume[i * 2]); - ak4396_write(chip, i, 4, chip->dac_volume[i * 2 + 1]); + ak4396_write(chip, i, + AK4396_LCH_ATT, chip->dac_volume[i * 2]); + ak4396_write(chip, i, + AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); } } @@ -214,11 +285,11 @@ static void update_ak4396_mute(struct oxygen *chip) unsigned int i; u8 value; - value = chip->ak4396_reg1 & ~AK4396_SMUTE; + value = chip->ak4396_ctl2 & ~AK4396_SMUTE; if (chip->dac_mute) value |= AK4396_SMUTE; for (i = 0; i < 4; ++i) - ak4396_write(chip, i, 1, value); + ak4396_write(chip, i, AK4396_CONTROL_2, value); } static void set_wm8785_params(struct oxygen *chip, @@ -226,22 +297,22 @@ static void set_wm8785_params(struct oxygen *chip, { unsigned int value; - wm8785_write(chip, 7, 0); + wm8785_write(chip, WM8785_R7, 0); - value = WM8785_FORMAT_LJUST; + value = WM8785_MCR_SLAVE | WM8785_FORMAT_LJUST; if (params_rate(params) == 96000) value |= WM8785_OSR_DOUBLE; else if (params_rate(params) == 192000) value |= WM8785_OSR_QUAD; else value |= WM8785_OSR_SINGLE; - wm8785_write(chip, 0, value); + wm8785_write(chip, WM8785_R0, value); if (snd_pcm_format_width(params_format(params)) <= 16) value = WM8785_WL_16; else value = WM8785_WL_24; - wm8785_write(chip, 1, value); + wm8785_write(chip, WM8785_R1, value); } static void set_ak5385_params(struct oxygen *chip, @@ -250,12 +321,13 @@ static void set_ak5385_params(struct oxygen *chip, unsigned int value; if (params_rate(params) <= 54000) - value = 0; + value = GPIO_AK5385_DFS_NORMAL; else if (params_rate(params) <= 108000) - value = 1; + value = GPIO_AK5385_DFS_DOUBLE; else - value = 2; - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, value, 0x0003); + value = GPIO_AK5385_DFS_QUAD; + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + value, GPIO_AK5385_DFS_MASK); } static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 1c402cb..4f4a56a 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -56,7 +56,7 @@ struct oxygen { u8 pcm_running; u8 dac_routing; u8 spdif_playback_enable; - u8 ak4396_reg1; + u8 ak4396_ctl2; u8 revision; u8 has_ac97_0; u8 has_ac97_1; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index b7079ad..aceb1f9 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -28,6 +28,7 @@ #include #include #include "oxygen.h" +#include "cm9780.h" MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 helper library"); @@ -262,9 +263,15 @@ static void __devinit oxygen_init(struct oxygen *chip) OXYGEN_AC97_CODEC0_LINER); oxygen_write_ac97(chip, 0, AC97_RESET, 0); msleep(1); - oxygen_ac97_set_bits(chip, 0, 0x70, 0x0300); - oxygen_ac97_set_bits(chip, 0, 0x64, 0x8043); - oxygen_ac97_set_bits(chip, 0, 0x62, 0x180f); + oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_SETUP, + CM9780_GPIO0IO | CM9780_GPIO1IO); + oxygen_ac97_set_bits(chip, 0, CM9780_MIXER, + CM9780_BSTSEL | CM9780_STRO_MIC | + CM9780_MIX2FR | CM9780_PCBSW); + oxygen_ac97_set_bits(chip, 0, CM9780_JACK, + CM9780_RSOE | CM9780_CBOE | + CM9780_SSOE | CM9780_FROE | + CM9780_MIC2MIC | CM9780_LI2LI); oxygen_write_ac97(chip, 0, AC97_MASTER, 0x0000); oxygen_write_ac97(chip, 0, AC97_PC_BEEP, 0x8000); oxygen_write_ac97(chip, 0, AC97_MIC, 0x8808); @@ -275,7 +282,8 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_write_ac97(chip, 0, AC97_REC_GAIN, 0x8000); oxygen_write_ac97(chip, 0, AC97_CENTER_LFE_MASTER, 0x8080); oxygen_write_ac97(chip, 0, AC97_SURROUND_MASTER, 0x8080); - oxygen_ac97_clear_bits(chip, 0, 0x72, 0x0001); + oxygen_ac97_clear_bits(chip, 0, + CM9780_GPIO_STATUS, CM9780_GPO0); /* power down unused ADCs and DACs */ oxygen_ac97_set_bits(chip, 0, AC97_POWERDOWN, AC97_PD_PR0 | AC97_PD_PR1); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index d23d18a..11114ce 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -23,6 +23,7 @@ #include #include #include "oxygen.h" +#include "cm9780.h" static int dac_volume_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) @@ -460,8 +461,9 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, if (change) { oxygen_write_ac97(chip, 0, index, newreg); if (index == AC97_LINE) { - oxygen_write_ac97_masked(chip, 0, 0x72, - !!(newreg & 0x8000), 0x0001); + oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, + newreg & 0x8000 ? + CM9780_GPO0 : 0, CM9780_GPO0); if (!(newreg & 0x8000)) { ac97_mute_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH); ac97_mute_ctl(chip, CONTROL_CD_CAPTURE_SWITCH); @@ -471,7 +473,8 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, index == AC97_VIDEO || index == AC97_AUX) && bitnr == 15 && !(newreg & 0x8000)) { ac97_mute_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH); - oxygen_write_ac97_masked(chip, 0, 0x72, 0x0001, 0x0001); + oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, + CM9780_GPO0, CM9780_GPO0); } } mutex_unlock(&chip->mutex); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 7397571..7b240c7 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -18,6 +18,8 @@ */ /* + * CMI8788: + * * SPI 0 -> 1st PCM1796 (front) * SPI 1 -> 2nd PCM1796 (surround) * SPI 2 -> 3rd PCM1796 (center/LFE) @@ -25,9 +27,13 @@ * * GPIO 2 -> M0 of CS5381 * GPIO 3 -> M1 of CS5381 - * GPIO 5 <- ? (D2X only) + * GPIO 5 <- external power present (D2X only) * GPIO 7 -> ALT - * GPIO 8 -> ? (amps enable?) + * GPIO 8 -> enable output to speakers + * + * CM9780: + * + * GPIO 0 -> enable AC'97 bypass (line in -> ADC) */ #include @@ -40,6 +46,7 @@ #include #include #include "oxygen.h" +#include "cm9780.h" MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("Asus AV200 driver"); @@ -64,14 +71,68 @@ static struct pci_device_id xonar_ids[] __devinitdata = { }; MODULE_DEVICE_TABLE(pci, xonar_ids); -/* register 0x12 */ + +#define GPIO_CS5381_M_MASK 0x000c +#define GPIO_CS5381_M_SINGLE 0x0000 +#define GPIO_CS5381_M_DOUBLE 0x0004 +#define GPIO_CS5381_M_QUAD 0x0008 +#define GPIO_EXT_POWER 0x0020 +#define GPIO_ALT 0x0080 +#define GPIO_OUTPUT_ENABLE 0x0100 + +/* register 16 */ +#define PCM1796_ATL_MASK 0xff +/* register 17 */ +#define PCM1796_ATR_MASK 0xff +/* register 18 */ #define PCM1796_MUTE 0x01 -#define PCM1796_FMT_24_MSB 0x30 +#define PCM1796_DME 0x02 +#define PCM1796_DMF_MASK 0x0c +#define PCM1796_DMF_DISABLED 0x00 +#define PCM1796_DMF_48 0x04 +#define PCM1796_DMF_441 0x08 +#define PCM1796_DMF_32 0x0c +#define PCM1796_FMT_MASK 0x70 +#define PCM1796_FMT_16_RJUST 0x00 +#define PCM1796_FMT_20_RJUST 0x10 +#define PCM1796_FMT_24_RJUST 0x20 +#define PCM1796_FMT_24_LJUST 0x30 +#define PCM1796_FMT_16_I2S 0x40 +#define PCM1796_FMT_24_I2S 0x50 #define PCM1796_ATLD 0x80 -/* register 0x14 */ +/* register 19 */ +#define PCM1796_INZD 0x01 +#define PCM1796_FLT_MASK 0x02 +#define PCM1796_FLT_SHARP 0x00 +#define PCM1796_FLT_SLOW 0x02 +#define PCM1796_DFMS 0x04 +#define PCM1796_OPE 0x10 +#define PCM1796_ATS_MASK 0x60 +#define PCM1796_ATS_1 0x00 +#define PCM1796_ATS_2 0x20 +#define PCM1796_ATS_4 0x40 +#define PCM1796_ATS_8 0x60 +#define PCM1796_REV 0x80 +/* register 20 */ +#define PCM1796_OS_MASK 0x03 #define PCM1796_OS_64 0x00 #define PCM1796_OS_32 0x01 #define PCM1796_OS_128 0x02 +#define PCM1796_CHSL_MASK 0x04 +#define PCM1796_CHSL_LEFT 0x00 +#define PCM1796_CHSL_RIGHT 0x04 +#define PCM1796_MONO 0x08 +#define PCM1796_DFTH 0x10 +#define PCM1796_DSD 0x20 +#define PCM1796_SRST 0x40 +/* register 21 */ +#define PCM1796_PCMZ 0x01 +#define PCM1796_DZ_MASK 0x06 +/* register 22 */ +#define PCM1796_ZFGL 0x01 +#define PCM1796_ZFGR 0x02 +/* register 23 */ +#define PCM1796_ID_MASK 0x1f static void pcm1796_write(struct oxygen *chip, unsigned int codec, u8 reg, u8 value) @@ -93,20 +154,23 @@ static void xonar_init(struct oxygen *chip) unsigned int i; for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 0x12, PCM1796_FMT_24_MSB | PCM1796_ATLD); - pcm1796_write(chip, i, 0x13, 0); - pcm1796_write(chip, i, 0x14, PCM1796_OS_64); - pcm1796_write(chip, i, 0x15, 0); - pcm1796_write(chip, i, 0x10, 0xff); - pcm1796_write(chip, i, 0x11, 0xff); + pcm1796_write(chip, i, 18, PCM1796_FMT_24_LJUST | PCM1796_ATLD); + pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); + pcm1796_write(chip, i, 20, PCM1796_OS_64); + pcm1796_write(chip, i, 21, 0); + pcm1796_write(chip, i, 16, 0xff); /* set ATL/ATR after ATLD */ + pcm1796_write(chip, i, 17, 0xff); } - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, 0x8c); - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, 0x00, 0x8c); - oxygen_ac97_set_bits(chip, 0, 0x62, 0x0080); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_CS5381_M_MASK | GPIO_ALT); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + GPIO_CS5381_M_SINGLE, + GPIO_CS5381_M_MASK | GPIO_ALT); + oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); msleep(300); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, 0x100); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, 0x100); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); snd_component_add(chip->card, "PCM1796"); snd_component_add(chip->card, "CS5381"); @@ -114,7 +178,7 @@ static void xonar_init(struct oxygen *chip) static void xonar_cleanup(struct oxygen *chip) { - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, 0x100); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); } static void set_pcm1796_params(struct oxygen *chip, @@ -126,7 +190,7 @@ static void set_pcm1796_params(struct oxygen *chip, value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; for (i = 0; i < 4; ++i) - pcm1796_write(chip, i, 0x14, value); + pcm1796_write(chip, i, 20, value); #endif } @@ -135,8 +199,8 @@ static void update_pcm1796_volume(struct oxygen *chip) unsigned int i; for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 0x10, chip->dac_volume[i * 2]); - pcm1796_write(chip, i, 0x11, chip->dac_volume[i * 2 + 1]); + pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); + pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); } } @@ -145,11 +209,11 @@ static void update_pcm1796_mute(struct oxygen *chip) unsigned int i; u8 value; - value = PCM1796_FMT_24_MSB | PCM1796_ATLD; + value = PCM1796_FMT_24_LJUST | PCM1796_ATLD; if (chip->dac_mute) value |= PCM1796_MUTE; for (i = 0; i < 4; ++i) - pcm1796_write(chip, i, 0x12, value); + pcm1796_write(chip, i, 18, value); } static void set_cs5381_params(struct oxygen *chip, @@ -158,12 +222,13 @@ static void set_cs5381_params(struct oxygen *chip, unsigned int value; if (params_rate(params) <= 54000) - value = 0; + value = GPIO_CS5381_M_SINGLE; else if (params_rate(params) <= 108000) - value = 4; + value = GPIO_CS5381_M_DOUBLE; else - value = 8; - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, value, 0x000c); + value = GPIO_CS5381_M_QUAD; + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + value, GPIO_CS5381_M_MASK); } static int pcm1796_volume_info(struct snd_kcontrol *ctl, @@ -182,7 +247,7 @@ static int alt_switch_get(struct snd_kcontrol *ctl, struct oxygen *chip = ctl->private_data; value->value.integer.value[0] = - !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & 0x80); + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_ALT); return 0; } @@ -196,9 +261,9 @@ static int alt_switch_put(struct snd_kcontrol *ctl, spin_lock_irq(&chip->reg_lock); old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); if (value->value.integer.value[0]) - new_bits = old_bits | 0x80; + new_bits = old_bits | GPIO_ALT; else - new_bits = old_bits & ~0x80; + new_bits = old_bits & ~GPIO_ALT; changed = new_bits != old_bits; if (changed) oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits); -- cgit v0.10.2 From 71e22a4b77fb36ce3205122454f5500843bed3ea Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:50:51 +0100 Subject: [ALSA] oxygen: allow more sample rates with WM8785 Allow to record with 32 kHz, 64 kHz, 88.2 kHz and 176.4 kHz with cards that have a WM8785 ADC. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 78aa969..0a7c613 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -232,18 +232,6 @@ static void generic_cleanup(struct oxygen *chip) { } -static void generic_pcm_hardware_filter(unsigned int channel, - struct snd_pcm_hardware *hardware) -{ - if (channel == PCM_A) { - hardware->rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000; - hardware->rate_min = 44100; - } -} - static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -300,12 +288,12 @@ static void set_wm8785_params(struct oxygen *chip, wm8785_write(chip, WM8785_R7, 0); value = WM8785_MCR_SLAVE | WM8785_FORMAT_LJUST; - if (params_rate(params) == 96000) + if (params_rate(params) <= 48000) + value |= WM8785_OSR_SINGLE; + else if (params_rate(params) <= 96000) value |= WM8785_OSR_DOUBLE; - else if (params_rate(params) == 192000) - value |= WM8785_OSR_QUAD; else - value |= WM8785_OSR_SINGLE; + value |= WM8785_OSR_QUAD; wm8785_write(chip, WM8785_R0, value); if (snd_pcm_format_width(params_format(params)) <= 16) @@ -349,7 +337,6 @@ static const struct oxygen_model model_generic = { .init = generic_init, .control_filter = ak4396_control_filter, .cleanup = generic_cleanup, - .pcm_hardware_filter = generic_pcm_hardware_filter, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, -- cgit v0.10.2 From 3b94253bc9c950d2038a2db4f9c804b50f82001a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:51:19 +0100 Subject: [ALSA] oxygen: reduce SPI clock frequency for AK4396/WM8785 According to the datasheets, the SPI clock cycle must be at least 200 ns for the AK4396 and the WM8785, so we cannot use the default 160 ns. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 0a7c613..35b2601 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -166,7 +166,7 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, }; oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_160 | + OXYGEN_SPI_CLOCK_320 | (codec_spi_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, AK4396_WRITE | (reg << 8) | value); @@ -176,7 +176,7 @@ static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) { oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_160 | + OXYGEN_SPI_CLOCK_320 | (3 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index ebafc65..d0cdce0 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -174,7 +174,7 @@ void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; - /* should not need more than 3.84 us (24 * 160 ns) */ + /* should not need more than 7.68 us (24 * 320 ns) */ count = 10; while ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & OXYGEN_SPI_BUSY) && count > 0) { -- cgit v0.10.2 From 7ef37cd95494a0a9be425c4d75f21ee8d2807b5a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:51:55 +0100 Subject: [ALSA] oxygen: move model-specific data out of common header Instead of having model-specific fields in the common struct oxygen, put them into a private structure that is allocated together with the card structure. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 35b2601..af6e802 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -157,6 +157,10 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define WM8785_PWRDNL 0x010 #define WM8785_TDM_MASK 0x1c0 +struct generic_data { + u8 ak4396_ctl2; +}; + static void ak4396_write(struct oxygen *chip, unsigned int codec, u8 reg, u8 value) { @@ -184,14 +188,15 @@ static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) static void ak4396_init(struct oxygen *chip) { + struct generic_data *data = chip->model_data; unsigned int i; - chip->ak4396_ctl2 = AK4396_DEM_OFF | AK4396_DFS_NORMAL; + data->ak4396_ctl2 = AK4396_DEM_OFF | AK4396_DFS_NORMAL; for (i = 0; i < 4; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); ak4396_write(chip, i, - AK4396_CONTROL_2, chip->ak4396_ctl2); + AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, i, AK4396_CONTROL_3, AK4396_PCM); ak4396_write(chip, i, AK4396_LCH_ATT, 0xff); @@ -235,17 +240,18 @@ static void generic_cleanup(struct oxygen *chip) static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { + struct generic_data *data = chip->model_data; unsigned int i; u8 value; - value = chip->ak4396_ctl2 & ~AK4396_DFS_MASK; + value = data->ak4396_ctl2 & ~AK4396_DFS_MASK; if (params_rate(params) <= 54000) value |= AK4396_DFS_NORMAL; else if (params_rate(params) < 120000) value |= AK4396_DFS_DOUBLE; else value |= AK4396_DFS_QUAD; - chip->ak4396_ctl2 = value; + data->ak4396_ctl2 = value; for (i = 0; i < 4; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB); @@ -270,12 +276,14 @@ static void update_ak4396_volume(struct oxygen *chip) static void update_ak4396_mute(struct oxygen *chip) { + struct generic_data *data = chip->model_data; unsigned int i; u8 value; - value = chip->ak4396_ctl2 & ~AK4396_SMUTE; + value = data->ak4396_ctl2 & ~AK4396_SMUTE; if (chip->dac_mute) value |= AK4396_SMUTE; + data->ak4396_ctl2 = value; for (i = 0; i < 4; ++i) ak4396_write(chip, i, AK4396_CONTROL_2, value); } @@ -341,6 +349,7 @@ static const struct oxygen_model model_generic = { .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, + .model_data_size = sizeof(struct generic_data), .used_channels = OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_C | OXYGEN_CHANNEL_SPDIF | @@ -362,6 +371,7 @@ static const struct oxygen_model model_meridian = { .set_adc_params = set_ak5385_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, + .model_data_size = sizeof(struct generic_data), .used_channels = OXYGEN_CHANNEL_B | OXYGEN_CHANNEL_C | OXYGEN_CHANNEL_SPDIF | diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 4f4a56a..4894dbd 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -49,6 +49,7 @@ struct oxygen { struct snd_rawmidi *midi; int irq; const struct oxygen_model *model; + void *model_data; unsigned int interrupt_mask; u8 dac_volume[8]; u8 dac_mute; @@ -56,7 +57,6 @@ struct oxygen { u8 pcm_running; u8 dac_routing; u8 spdif_playback_enable; - u8 ak4396_ctl2; u8 revision; u8 has_ac97_0; u8 has_ac97_1; @@ -84,6 +84,7 @@ struct oxygen_model { struct snd_pcm_hw_params *params); void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); + size_t model_data_size; u8 used_channels; u8 function_flags; u16 dac_i2s_format; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index aceb1f9..0927e04 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -320,7 +320,8 @@ int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, struct oxygen *chip; int err; - card = snd_card_new(index, id, model->owner, sizeof *chip); + card = snd_card_new(index, id, model->owner, + sizeof *chip + model->model_data_size); if (!card) return -ENOMEM; @@ -329,6 +330,7 @@ int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, chip->pci = pci; chip->irq = -1; chip->model = model; + chip->model_data = chip + 1; spin_lock_init(&chip->reg_lock); mutex_init(&chip->mutex); INIT_WORK(&chip->spdif_input_bits_work, -- cgit v0.10.2 From e9d88a8bd9fb85d2a4a0adb89c47dcae437fa308 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:52:11 +0100 Subject: [ALSA] oxygen: fix control filter Actually use the template that was maybe changed by the control filter instead of the original one. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 11114ce..0993c29 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -667,7 +667,7 @@ static int add_controls(struct oxygen *chip, err = chip->model->control_filter(&template); if (err < 0) return err; - ctl = snd_ctl_new1(&controls[i], chip); + ctl = snd_ctl_new1(&template, chip); if (!ctl) return -ENOMEM; err = snd_ctl_add(chip->card, ctl); -- cgit v0.10.2 From 5a256f862c2a9155456b718edb303e37cda2d153 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:52:29 +0100 Subject: [ALSA] oxygen: fix DAC source register fields Fix some wrong values for the definitions of the source masks for DACS 1 and 3. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index 3e0cdae..72de159 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -330,11 +330,11 @@ #define OXYGEN_PLAY_SPDIF_I2S_ADC_3 0x00e0 #define OXYGEN_PLAY_DAC0_SOURCE_MASK 0x0300 #define OXYGEN_PLAY_DAC0_SOURCE_SHIFT 8 -#define OXYGEN_PLAY_DAC1_SOURCE_MASK 0x0700 +#define OXYGEN_PLAY_DAC1_SOURCE_MASK 0x0c00 #define OXYGEN_PLAY_DAC1_SOURCE_SHIFT 10 #define OXYGEN_PLAY_DAC2_SOURCE_MASK 0x3000 #define OXYGEN_PLAY_DAC2_SOURCE_SHIFT 12 -#define OXYGEN_PLAY_DAC3_SOURCE_MASK 0x7000 +#define OXYGEN_PLAY_DAC3_SOURCE_MASK 0xc000 #define OXYGEN_PLAY_DAC3_SOURCE_SHIFT 14 #define OXYGEN_REC_ROUTING 0xc2 -- cgit v0.10.2 From 7f0b89465bb94eb3273ea5af5e009332351a54c9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:53:30 +0100 Subject: [ALSA] oxygen: add 192 kHz SPDIF input support Change the oxygen_spdif_input_bits_changed() function so that clock changes on the SPDIF input are correctly detected. This means that sample rates greater than 96 kHz are now supported. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 0927e04..49eabda 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -74,7 +74,9 @@ static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) if (status & OXYGEN_INT_SPDIF_IN_DETECT) { spin_lock(&chip->reg_lock); i = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); - if (i & OXYGEN_SPDIF_RATE_INT) { + if (i & (OXYGEN_SPDIF_SENSE_INT | OXYGEN_SPDIF_LOCK_INT | + OXYGEN_SPDIF_RATE_INT)) { + /* write the interrupt bit(s) to clear */ oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, i); schedule_work(&chip->spdif_input_bits_work); } @@ -94,30 +96,46 @@ static void oxygen_spdif_input_bits_changed(struct work_struct *work) { struct oxygen *chip = container_of(work, struct oxygen, spdif_input_bits_work); + u32 reg; - spin_lock_irq(&chip->reg_lock); - oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, - OXYGEN_SPDIF_IN_CLOCK_96, - OXYGEN_SPDIF_IN_CLOCK_MASK); - spin_unlock_irq(&chip->reg_lock); + /* + * This function gets called when there is new activity on the SPDIF + * input, or when we lose lock on the input signal, or when the rate + * changes. + */ msleep(1); - if (!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) - & OXYGEN_SPDIF_LOCK_STATUS)) { - spin_lock_irq(&chip->reg_lock); - oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, - OXYGEN_SPDIF_IN_CLOCK_192, - OXYGEN_SPDIF_IN_CLOCK_MASK); + spin_lock_irq(&chip->reg_lock); + reg = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); + if ((reg & (OXYGEN_SPDIF_SENSE_STATUS | + OXYGEN_SPDIF_LOCK_STATUS)) + == OXYGEN_SPDIF_SENSE_STATUS) { + /* + * If we detect activity on the SPDIF input but cannot lock to + * a signal, the clock bit is likely to be wrong. + */ + reg ^= OXYGEN_SPDIF_IN_CLOCK_MASK; + oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, reg); spin_unlock_irq(&chip->reg_lock); msleep(1); - if (!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) - & OXYGEN_SPDIF_LOCK_STATUS)) { - spin_lock_irq(&chip->reg_lock); - oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, - OXYGEN_SPDIF_IN_CLOCK_96, - OXYGEN_SPDIF_IN_CLOCK_MASK); - spin_unlock_irq(&chip->reg_lock); + spin_lock_irq(&chip->reg_lock); + reg = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); + if ((reg & (OXYGEN_SPDIF_SENSE_STATUS | + OXYGEN_SPDIF_LOCK_STATUS)) + == OXYGEN_SPDIF_SENSE_STATUS) { + /* nothing detected with either clock; give up */ + if ((reg & OXYGEN_SPDIF_IN_CLOCK_MASK) + == OXYGEN_SPDIF_IN_CLOCK_192) { + /* + * Reset clock to <= 96 kHz because this is + * more likely to be received next time. + */ + reg &= ~OXYGEN_SPDIF_IN_CLOCK_MASK; + reg |= OXYGEN_SPDIF_IN_CLOCK_96; + oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, reg); + } } } + spin_unlock_irq(&chip->reg_lock); if (chip->controls[CONTROL_SPDIF_INPUT_BITS]) { spin_lock_irq(&chip->reg_lock); @@ -126,6 +144,10 @@ static void oxygen_spdif_input_bits_changed(struct work_struct *work) chip->interrupt_mask); spin_unlock_irq(&chip->reg_lock); + /* + * We don't actually know that any channel status bits have + * changed, but let's send a notification just to be sure. + */ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->controls[CONTROL_SPDIF_INPUT_BITS]->id); } @@ -225,7 +247,20 @@ static void __devinit oxygen_init(struct oxygen *chip) OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); - oxygen_set_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_RATE_MASK); + oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_SENSE_MASK | + OXYGEN_SPDIF_LOCK_MASK | + OXYGEN_SPDIF_RATE_MASK | + OXYGEN_SPDIF_LOCK_PAR | + OXYGEN_SPDIF_IN_CLOCK_96, + OXYGEN_SPDIF_OUT_ENABLE | + OXYGEN_SPDIF_LOOPBACK | + OXYGEN_SPDIF_SENSE_MASK | + OXYGEN_SPDIF_LOCK_MASK | + OXYGEN_SPDIF_RATE_MASK | + OXYGEN_SPDIF_SENSE_PAR | + OXYGEN_SPDIF_LOCK_PAR | + OXYGEN_SPDIF_IN_CLOCK_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); oxygen_write16(chip, OXYGEN_PLAY_ROUTING, OXYGEN_PLAY_MULTICH_I2S_DAC | OXYGEN_PLAY_SPDIF_SPDIF | diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 31b0ccd..2002900 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -85,12 +85,16 @@ static struct snd_pcm_hardware oxygen_hardware[PCM_COUNT] = { SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_44100 | + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000, - .rate_min = 44100, - .rate_max = 96000, + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 32000, + .rate_max = 192000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 256 * 1024, -- cgit v0.10.2 From c57cccc0382fedd95373ffdb192f45d278210686 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 08:54:06 +0100 Subject: [ALSA] oxygen: optimize snd_pcm_hardware structures Add one more indirection to the lookup of the snd_pcm_hardware structures so that we can save the space of the duplicate ones. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 2002900..272ef08 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -24,155 +24,84 @@ #include #include "oxygen.h" -static struct snd_pcm_hardware oxygen_hardware[PCM_COUNT] = { - [PCM_A] = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 32000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, - .periods_min = 2, - .periods_max = 2048, - }, - [PCM_B] = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 32000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, - .periods_min = 2, - .periods_max = 2048, - }, - [PCM_C] = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 32000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, - .periods_min = 2, - .periods_max = 2048, - }, - [PCM_SPDIF] = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 32000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, - .periods_min = 2, - .periods_max = 2048, - }, - [PCM_MULTICH] = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 32000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 8, - .buffer_bytes_max = 2048 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 256 * 1024, - .periods_min = 2, - .periods_max = 16384, - }, - [PCM_AC97] = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, - .periods_min = 2, - .periods_max = 2048, - }, +static const struct snd_pcm_hardware oxygen_stereo_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 32000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 256 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 2048, +}; +static const struct snd_pcm_hardware oxygen_multichannel_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 32000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = 2048 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 256 * 1024, + .periods_min = 2, + .periods_max = 16384, +}; +static const struct snd_pcm_hardware oxygen_ac97_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 256 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 2048, +}; + +static const struct snd_pcm_hardware *const oxygen_hardware[PCM_COUNT] = { + [PCM_A] = &oxygen_stereo_hardware, + [PCM_B] = &oxygen_stereo_hardware, + [PCM_C] = &oxygen_stereo_hardware, + [PCM_SPDIF] = &oxygen_stereo_hardware, + [PCM_MULTICH] = &oxygen_multichannel_hardware, + [PCM_AC97] = &oxygen_ac97_hardware, }; static inline unsigned int @@ -189,7 +118,7 @@ static int oxygen_open(struct snd_pcm_substream *substream, int err; runtime->private_data = (void *)(uintptr_t)channel; - runtime->hw = oxygen_hardware[channel]; + runtime->hw = *oxygen_hardware[channel]; if (chip->model->pcm_hardware_filter) chip->model->pcm_hardware_filter(channel, &runtime->hw); err = snd_pcm_hw_constraint_step(runtime, 0, -- cgit v0.10.2 From 0de27c10d3524eba9e81f56ff315e97d95b3faa0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Jan 2008 11:49:03 +0100 Subject: [ALSA] cs5535audio - Fix available sample rates The available sample rates on CS5535 depend on AC97 codec chip. Set the additional hw params limit. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 499e279..cdcda87 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -97,6 +97,8 @@ static int snd_cs5535audio_playback_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; runtime->hw = snd_cs5535audio_playback; + runtime->hw.rates = cs5535au->ac97->rates[AC97_RATES_FRONT_DAC]; + snd_pcm_limit_hw_rates(runtime); cs5535au->playback_substream = substream; runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_PLAYBACK]); if ((err = snd_pcm_hw_constraint_integer(runtime, @@ -342,6 +344,8 @@ static int snd_cs5535audio_capture_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; runtime->hw = snd_cs5535audio_capture; + runtime->hw.rates = cs5535au->ac97->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); cs5535au->capture_substream = substream; runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_CAPTURE]); if ((err = snd_pcm_hw_constraint_integer(runtime, -- cgit v0.10.2 From c518b35ea2ef2b7b3dbe6b1cc5299daf0c9de3f7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 21 Jan 2008 12:59:40 +0100 Subject: [ALSA] oxygen: TempoTec HiFier is probably not supported The TempoTec HiFier has a somwhat different architecture; remove it from the list of cards that are known to be supported. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index a607111..ede4f34 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1570,7 +1570,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * HT-Omega Claro * Razer Barracuda AC-1 * Sondigo Inferno - * TempoTec HIFIER This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 5baee6b..10c3694 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -203,7 +203,6 @@ config SND_OXYGEN * HT-Omega Claro * Razer Barracuda AC-1 * Sondigo Inferno - * TempoTec HIFIER To compile this driver as a module, choose M here: the module will be called snd-oxygen. -- cgit v0.10.2 From 08c8efe6925ba712d65ca07877fa169b45f8d502 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Jan 2008 14:33:37 +0100 Subject: [ALSA] Fix lockdep warning in ASoC machine probe Don't take the codec mutex during machine probe until we have registered with ALSA, fixing a lockdep warning reported by Dmitry Baryshkov. Cc: Dmitry Baryshkov Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5f86e03..9eb5479 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1101,7 +1101,6 @@ int snd_soc_register_card(struct snd_soc_device *socdev) struct snd_soc_machine *machine = socdev->machine; int ret = 0, i, ac97 = 0, err = 0; - mutex_lock(&codec->mutex); for(i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { err = socdev->machine->dai_link[i].init(codec); @@ -1127,12 +1126,14 @@ int snd_soc_register_card(struct snd_soc_device *socdev) goto out; } + mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS if (ac97) { ret = soc_ac97_dev_register(codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); snd_card_free(codec->card); + mutex_unlock(&codec->mutex); goto out; } } @@ -1145,8 +1146,10 @@ int snd_soc_register_card(struct snd_soc_device *socdev) err = device_create_file(socdev->dev, &dev_attr_codec_reg); if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n"); -out: + mutex_unlock(&codec->mutex); + +out: return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); -- cgit v0.10.2 From 02f21c9d6ba863aa86c33e1335cb1307322f7fb8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 22 Jan 2008 08:36:03 +0100 Subject: [ALSA] oxygen: add SPDIF loopback control Add a mixer control for the SPDIF loopback function. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 0993c29..6b7420f 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -408,6 +408,37 @@ static int spdif_input_default_get(struct snd_kcontrol *ctl, return 0; } +static int spdif_loopback_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + value->value.integer.value[0] = + !!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) + & OXYGEN_SPDIF_LOOPBACK); + return 0; +} + +static int spdif_loopback_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u32 oldreg, newreg; + int changed; + + spin_lock_irq(&chip->reg_lock); + oldreg = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); + if (value->value.integer.value[0]) + newreg = oldreg | OXYGEN_SPDIF_LOOPBACK; + else + newreg = oldreg & ~OXYGEN_SPDIF_LOOPBACK; + changed = newreg != oldreg; + if (changed) + oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, newreg); + spin_unlock_irq(&chip->reg_lock); + return changed; +} + static int ac97_switch_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -620,6 +651,13 @@ static const struct snd_kcontrol_new controls[] = { .info = spdif_info, .get = spdif_input_default_get, }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("Loopback ", NONE, SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = spdif_loopback_get, + .put = spdif_loopback_put, + }, }; static const struct snd_kcontrol_new ac97_controls[] = { -- cgit v0.10.2 From b22b48214ff492379dfc89b3ea6dc9fb4d157d2a Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 22 Jan 2008 12:32:30 +0100 Subject: [ALSA] hda: Mono mux mixer support Add support for the mono mux on several 92HD7xxx codecs. Creates a dynamic mixer for the mux selection. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 24137bc..e014ae3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -131,6 +131,8 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; /* playback */ + struct hda_input_mux *mono_mux; + unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -144,6 +146,7 @@ struct sigmatel_spec { hda_nid_t *dmux_nids; unsigned int num_dmuxes; hda_nid_t dig_in_nid; + hda_nid_t mono_nid; /* pin widgets */ hda_nid_t *pin_nids; @@ -174,6 +177,7 @@ struct sigmatel_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_dimux; struct hda_input_mux private_imux; + struct hda_input_mux private_mono_mux; /* virtual master */ unsigned int vmaster_tlv[4]; @@ -401,6 +405,34 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); } +static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->mono_mux, uinfo); +} + +static int stac92xx_mono_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_mmux; + return 0; +} + +static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, spec->mono_mux, ucontrol, + spec->mono_nid, &spec->cur_mmux); +} + #define stac92xx_aloopback_info snd_ctl_boolean_mono_info static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, @@ -603,6 +635,16 @@ static struct hda_verb stac9205_core_init[] = { {} }; +#define STAC_MONO_MUX \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Mono Mux", \ + .count = 1, \ + .info = stac92xx_mono_mux_enum_info, \ + .get = stac92xx_mono_mux_enum_get, \ + .put = stac92xx_mono_mux_enum_put, \ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -714,6 +756,7 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -733,6 +776,7 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), + STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1180,7 +1224,7 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01a19020, 0x90a000f0, + 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, }; @@ -2318,6 +2362,35 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, return 0; } +/* labels for mono mux outputs */ +static const char *stac92xx_mono_labels[3] = { + "DAC0", "DAC1", "Mixer" +}; + +/* create mono mux for mono out on capable codecs */ +static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *mono_mux = &spec->private_mono_mux; + int i, num_cons; + hda_nid_t con_lst[ARRAY_SIZE(stac92xx_mono_labels)]; + + num_cons = snd_hda_get_connections(codec, + spec->mono_nid, + con_lst, + HDA_MAX_NUM_INPUTS); + if (!num_cons || num_cons > ARRAY_SIZE(stac92xx_mono_labels)) + return -EINVAL; + + for (i = 0; i < num_cons; i++) { + mono_mux->items[mono_mux->num_items].label = + stac92xx_mono_labels[i]; + mono_mux->items[mono_mux->num_items].index = i; + mono_mux->num_items++; + } + return 0; +} + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -2532,6 +2605,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; + if (spec->mono_nid > 0) { + err = stac92xx_auto_create_mono_output_ctls(codec); + if (err < 0) + return err; + } + if (spec->num_dmics > 0) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) @@ -2552,6 +2631,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->input_mux = &spec->private_imux; if (!spec->dinput_mux) spec->dinput_mux = &spec->private_dimux; + spec->mono_mux = &spec->private_mono_mux; return 1; } @@ -3237,6 +3317,7 @@ again: spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + spec->mono_nid = 0x15; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; -- cgit v0.10.2 From 15908c36aaefe9f1300fd2f0ee6d5d848131b217 Mon Sep 17 00:00:00 2001 From: Marc Boucher Date: Tue, 22 Jan 2008 15:15:59 +0100 Subject: [ALSA] hda-codec - Fix laptop models for Cxt5045 Change laptop models to three different models, laptop-hpsense, laptop-micsense and laptop-hpmicsense. The first two correspond to the old 'laptop' and 'fujitsu' models. Reassigned the quirk table for the new models. Signed-off-by: Marc Boucher Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index ede4f34..07317da 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -945,8 +945,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. auto auto-config reading BIOS (default) Conexant 5045 - laptop Laptop config - fujitsu Fujitsu Si1520 laptop + laptop-hpsense Laptop with HP sense (old model laptop) + laptop-micsense Laptop with Mic sense (old model fujitsu) + laptop-hpmicsense Laptop with HP and Mic senses benq Benq R55E test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index dab2ce1..ea3559fe 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -781,8 +781,9 @@ static int cxt5045_init(struct hda_codec *codec) enum { - CXT5045_LAPTOP, /* Laptops w/ EAPD support */ - CXT5045_FUJITSU, /* Laptops w/ EAPD support */ + CXT5045_LAPTOP_HPSENSE, + CXT5045_LAPTOP_MICSENSE, + CXT5045_LAPTOP_HPMICSENSE, CXT5045_BENQ, #ifdef CONFIG_SND_DEBUG CXT5045_TEST, @@ -791,27 +792,34 @@ enum { }; static const char *cxt5045_models[CXT5045_MODELS] = { - [CXT5045_LAPTOP] = "laptop", - [CXT5045_FUJITSU] = "fujitsu", - [CXT5045_BENQ] = "benq", + [CXT5045_LAPTOP_HPSENSE] = "laptop-hpsense", + [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", + [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", + [CXT5045_BENQ] = "benq", #ifdef CONFIG_SND_DEBUG [CXT5045_TEST] = "test", #endif }; static struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30a5, "HP", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), - SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), - SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP), - SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), + SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK(0x1631, 0xc106, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK(0x1631, 0xc107, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE), {} }; @@ -849,7 +857,7 @@ static int patch_cxt5045(struct hda_codec *codec) cxt5045_models, cxt5045_cfg_tbl); switch (board_config) { - case CXT5045_LAPTOP: + case CXT5045_LAPTOP_HPSENSE: codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; spec->input_mux = &cxt5045_capture_source; spec->num_init_verbs = 2; @@ -857,13 +865,23 @@ static int patch_cxt5045(struct hda_codec *codec) spec->mixers[0] = cxt5045_mixers; codec->patch_ops.init = cxt5045_init; break; - case CXT5045_FUJITSU: + case CXT5045_LAPTOP_MICSENSE: spec->input_mux = &cxt5045_capture_source; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; spec->mixers[0] = cxt5045_mixers; codec->patch_ops.init = cxt5045_init; break; + default: + case CXT5045_LAPTOP_HPMICSENSE: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; + spec->input_mux = &cxt5045_capture_source; + spec->num_init_verbs = 3; + spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; + spec->init_verbs[2] = cxt5045_mic_sense_init_verbs; + spec->mixers[0] = cxt5045_mixers; + codec->patch_ops.init = cxt5045_init; + break; case CXT5045_BENQ: codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; spec->input_mux = &cxt5045_capture_source_benq; @@ -879,6 +897,8 @@ static int patch_cxt5045(struct hda_codec *codec) spec->input_mux = &cxt5045_test_capture_source; spec->mixers[0] = cxt5045_test_mixer; spec->init_verbs[0] = cxt5045_test_init_verbs; + break; + #endif } -- cgit v0.10.2 From 9f113e0ef47451defa950eed738d87387c0aa8f6 Mon Sep 17 00:00:00 2001 From: Marc Boucher Date: Tue, 22 Jan 2008 15:18:08 +0100 Subject: [ALSA] hda-codec - Add missing input controls for Cxt5047 test model The input volume/switch elements are missing in Cxt5047 test model. Also the patch includes some code clean ups. Signed-off-by: Marc Boucher Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ea3559fe..1ed0f075 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1020,13 +1020,13 @@ static void cxt5047_hp2_automute(struct hda_codec *codec) static void cxt5047_hp_automic(struct hda_codec *codec) { static struct hda_verb mic_jack_on[] = { - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {} }; static struct hda_verb mic_jack_off[] = { - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {} }; unsigned int present; @@ -1043,8 +1043,7 @@ static void cxt5047_hp_automic(struct hda_codec *codec) static void cxt5047_hp_unsol_event(struct hda_codec *codec, unsigned int res) { - res >>= 26; - switch (res) { + switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5047_hp_automute(codec); break; @@ -1253,6 +1252,17 @@ static struct snd_kcontrol_new cxt5047_test_mixer[] = { .get = conexant_mux_enum_get, .put = conexant_mux_enum_put, }, + HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT), + { } /* end */ }; -- cgit v0.10.2 From d2595d86e5095f724c17ce97d679f30aa3eaabf8 Mon Sep 17 00:00:00 2001 From: Marc Boucher Date: Tue, 22 Jan 2008 15:23:30 +0100 Subject: [ALSA] hda-codec - Add a delay after power state change Added a delay after the power state change as a partial workaround for 'hda_intel: azx_get_response timeout' problem on Cxt codecs. Signed-off-by: Marc Boucher Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b5e69b1..6b93f45 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1685,6 +1685,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); + msleep(10); /* partial workaround for "azx_get_response timeout" */ nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { -- cgit v0.10.2 From 1cfd52bc1ad516dbdc23839d40013dea4c19c70a Mon Sep 17 00:00:00 2001 From: Marc Boucher Date: Tue, 22 Jan 2008 15:29:26 +0100 Subject: [ALSA] hda-intel - Add ratelimit to timeout messages Signed-off-by: Marc Boucher Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index be5cbbc..0003021 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -629,8 +629,9 @@ static int azx_single_send_cmd(struct hda_codec *codec, u32 val) } udelay(1); } - snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", - azx_readw(chip, IRS), val); + if (printk_ratelimit()) + snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", + azx_readw(chip, IRS), val); return -EIO; } @@ -646,8 +647,9 @@ static unsigned int azx_single_get_response(struct hda_codec *codec) return azx_readl(chip, IR); udelay(1); } - snd_printd(SFX "get_response timeout: IRS=0x%x\n", - azx_readw(chip, IRS)); + if (printk_ratelimit()) + snd_printd(SFX "get_response timeout: IRS=0x%x\n", + azx_readw(chip, IRS)); return (unsigned int)-1; } -- cgit v0.10.2 From ca7cfae9eb91c012ac448dc6816ded6285f6f39a Mon Sep 17 00:00:00 2001 From: Marc Boucher Date: Tue, 22 Jan 2008 15:32:25 +0100 Subject: [ALSA] hda-codec - Add afg and mfg preset mask Added afg and mfg preset masks for more finer codec-preset selection. Signed-off-by: Marc Boucher Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6b93f45..d4fd948 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -429,6 +429,10 @@ find_codec_preset(struct hda_codec *codec) for (tbl = hda_preset_tables; *tbl; tbl++) { for (preset = *tbl; preset->id; preset++) { u32 mask = preset->mask; + if (preset->afg && preset->afg != codec->afg) + continue; + if (preset->mfg && preset->mfg != codec->mfg) + continue; if (!mask) mask = ~0; if (preset->id == (codec->vendor_id & mask) && diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index eb4a2ae..f148711 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -523,6 +523,7 @@ struct hda_codec_preset { unsigned int subs; unsigned int subs_mask; unsigned int rev; + hda_nid_t afg, mfg; const char *name; int (*patch)(struct hda_codec *codec); }; -- cgit v0.10.2 From 0ef6ce7b6acddc3dc9e898125b3bada98a21130a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Jan 2008 15:35:37 +0100 Subject: [ALSA] hda-codec - Optimize snd_hda_pser_pin_def_config() Don't read the widget list again as we already have it at the beginning. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d4fd948..f6a9a5d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2671,8 +2671,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg, hda_nid_t *ignore_nids) { - hda_nid_t nid, nid_start; - int nodes; + hda_nid_t nid, end_nid; short seq, assoc_line_out, assoc_speaker; short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)]; short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)]; @@ -2685,8 +2684,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, memset(sequences_hp, 0, sizeof(sequences_hp)); assoc_line_out = assoc_speaker = 0; - nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid_start); - for (nid = nid_start; nid < nodes + nid_start; nid++) { + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { unsigned int wid_caps = get_wcaps(codec, nid); unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; -- cgit v0.10.2 From dca008f367586f73bd1c766836e4f7a38ce9814f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Jan 2008 15:37:13 +0100 Subject: [ALSA] hda-codec - Don't query widget parameter for invalid NID Don't query a widget parameter for an invalid NID in get_wcaps() but rather returns zero (i.e. no attribute). The read to an non-existing widget may result in a fatal codec communication error. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ddc61a1..448c4ce 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -370,7 +370,7 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) { if (nid < codec->start_nid || nid >= codec->start_nid + codec->num_nodes) - return snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + return 0; return codec->wcaps[nid - codec->start_nid]; } -- cgit v0.10.2 From 7c4dbbd87c0dc62849f0f72449464dc37da0a82a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Jan 2008 08:41:46 +0100 Subject: [ALSA] ASoC documentation updates Update the ASoC documentation. Along with minor formatting and grammar cleanups it moves the ASoC overview into the present tense to reflect the fact that it has now been merged. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt index 3feeb9e..0ebd7ea 100644 --- a/Documentation/sound/alsa/soc/DAI.txt +++ b/Documentation/sound/alsa/soc/DAI.txt @@ -1,5 +1,5 @@ ASoC currently supports the three main Digital Audio Interfaces (DAI) found on -SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM. +SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM. AC97 @@ -25,7 +25,7 @@ left/right clock (LRC) synchronise the link. I2S is flexible in that either the controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock usually varies depending on the sample rate and the master system clock (SYSCLK). LRCLK is the same as the sample rate. A few devices support separate -ADC and DAC LRCLK's, this allows for simultaneous capture and playback at +ADC and DAC LRCLKs, this allows for simultaneous capture and playback at different sample rates. I2S has several different operating modes:- @@ -35,7 +35,7 @@ I2S has several different operating modes:- o Left Justified - MSB is transmitted on transition of LRC. - o Right Justified - MSB is transmitted sample size BCLK's before LRC + o Right Justified - MSB is transmitted sample size BCLKs before LRC transition. PCM diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index 1493088..b130016 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -13,7 +13,7 @@ or SYSCLK). This audio master clock can be derived from a number of sources (e.g. crystal, PLL, CPU clock) and is responsible for producing the correct audio playback and capture sample rates. -Some master clocks (e.g. PLL's and CPU based clocks) are configurable in that +Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that their speed can be altered by software (depending on the system use and to save power). Other master clocks are fixed at a set frequency (i.e. crystals). @@ -41,11 +41,11 @@ BCLK = LRC * x BCLK = LRC * Channels * Word Size This relationship depends on the codec or SoC CPU in particular. In general -it's best to configure BCLK to the lowest possible speed (depending on your -rate, number of channels and wordsize) to save on power. +it is best to configure BCLK to the lowest possible speed (depending on your +rate, number of channels and word size) to save on power. -It's also desirable to use the codec (if possible) to drive (or master) the -audio clocks as it's usually gives more accurate sample rates than the CPU. +It is also desirable to use the codec (if possible) to drive (or master) the +audio clocks as it usually gives more accurate sample rates than the CPU. diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index 1e766ad..1e95342 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -9,7 +9,7 @@ code should be added to the platform and machine drivers respectively. Each codec driver *must* provide the following features:- 1) Codec DAI and PCM configuration - 2) Codec control IO - using I2C, 3 Wire(SPI) or both API's + 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs 3) Mixers and audio controls 4) Codec audio operations @@ -19,7 +19,7 @@ Optionally, codec drivers can also provide:- 6) DAPM event handler. 7) DAC Digital mute control. -It's probably best to use this guide in conjunction with the existing codec +Its probably best to use this guide in conjunction with the existing codec driver code in sound/soc/codecs/ ASoC Codec driver breakdown @@ -27,8 +27,8 @@ ASoC Codec driver breakdown 1 - Codec DAI and PCM configuration ----------------------------------- -Each codec driver must have a struct snd_soc_codec_dai to define it's DAI and -PCM's capabilities and operations. This struct is exported so that it can be +Each codec driver must have a struct snd_soc_codec_dai to define its DAI and +PCM capabilities and operations. This struct is exported so that it can be registered with the core by your machine driver. e.g. @@ -67,18 +67,18 @@ EXPORT_SYMBOL_GPL(wm8731_dai); 2 - Codec control IO -------------------- -The codec can usually be controlled via an I2C or SPI style interface (AC97 -combines control with data in the DAI). The codec drivers will have to provide -functions to read and write the codec registers along with supplying a register -cache:- +The codec can usually be controlled via an I2C or SPI style interface +(AC97 combines control with data in the DAI). The codec drivers provide +functions to read and write the codec registers along with supplying a +register cache:- /* IO control data and register cache */ - void *control_data; /* codec control (i2c/3wire) data */ - void *reg_cache; + void *control_data; /* codec control (i2c/3wire) data */ + void *reg_cache; -Codec read/write should do any data formatting and call the hardware read write -below to perform the IO. These functions are called by the core and alsa when -performing DAPM or changing the mixer:- +Codec read/write should do any data formatting and call the hardware +read write below to perform the IO. These functions are called by the +core and ALSA when performing DAPM or changing the mixer:- unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); @@ -131,7 +131,7 @@ Defines a stereo enumerated control 4 - Codec Audio Operations -------------------------- -The codec driver also supports the following alsa operations:- +The codec driver also supports the following ALSA operations:- /* SoC audio ops */ struct snd_soc_ops { @@ -142,15 +142,15 @@ struct snd_soc_ops { int (*prepare)(struct snd_pcm_substream *); }; -Please refer to the alsa driver PCM documentation for details. +Please refer to the ALSA driver PCM documentation for details. http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm 5 - DAPM description. --------------------- -The Dynamic Audio Power Management description describes the codec's power -components, their relationships and registers to the ASoC core. Please read -dapm.txt for details of building the description. +The Dynamic Audio Power Management description describes the codec power +components and their relationships and registers to the ASoC core. +Please read dapm.txt for details of building the description. Please also see the examples in other codec drivers. @@ -158,8 +158,8 @@ Please also see the examples in other codec drivers. 6 - DAPM event handler ---------------------- This function is a callback that handles codec domain PM calls and system -domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep -when not in use. +domain PM calls (e.g. suspend and resume). It is used to put the codec +to sleep when not in use. Power states:- @@ -175,13 +175,14 @@ Power states:- SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ -7 - Codec DAC digital mute control. ------------------------------------- -Most codecs have a digital mute before the DAC's that can be used to minimise -any system noise. The mute stops any digital data from entering the DAC. +7 - Codec DAC digital mute control +---------------------------------- +Most codecs have a digital mute before the DACs that can be used to +minimise any system noise. The mute stops any digital data from +entering the DAC. -A callback can be created that is called by the core for each codec DAI when the -mute is applied or freed. +A callback can be created that is called by the core for each codec DAI +when the mute is applied or freed. i.e. diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index ab0766f..c784a18 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -4,20 +4,20 @@ Dynamic Audio Power Management for Portable Devices 1. Description ============== -Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices -to use the minimum amount of power within the audio subsystem at all times. It -is independent of other kernel PM and as such, can easily co-exist with the -other PM systems. +Dynamic Audio Power Management (DAPM) is designed to allow portable +Linux devices to use the minimum amount of power within the audio +subsystem at all times. It is independent of other kernel PM and as +such, can easily co-exist with the other PM systems. -DAPM is also completely transparent to all user space applications as all power -switching is done within the ASoC core. No code changes or recompiling are -required for user space applications. DAPM makes power switching decisions based -upon any audio stream (capture/playback) activity and audio mixer settings -within the device. +DAPM is also completely transparent to all user space applications as +all power switching is done within the ASoC core. No code changes or +recompiling are required for user space applications. DAPM makes power +switching decisions based upon any audio stream (capture/playback) +activity and audio mixer settings within the device. -DAPM spans the whole machine. It covers power control within the entire audio -subsystem, this includes internal codec power blocks and machine level power -systems. +DAPM spans the whole machine. It covers power control within the entire +audio subsystem, this includes internal codec power blocks and machine +level power systems. There are 4 power domains within DAPM @@ -34,7 +34,7 @@ There are 4 power domains within DAPM Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. - 4. Stream domain - DAC's and ADC's. + 4. Stream domain - DACs and ADCs. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. @@ -51,7 +51,7 @@ widgets hereafter. Audio DAPM widgets fall into a number of types:- o Mixer - Mixes several analog signals into a single analog signal. - o Mux - An analog switch that outputs only 1 of it's inputs. + o Mux - An analog switch that outputs only one of many inputs. o PGA - A programmable gain amplifier or attenuation widget. o ADC - Analog to Digital Converter o DAC - Digital to Analog Converter @@ -78,14 +78,14 @@ parameters for stream name and kcontrols. 2.1 Stream Domain Widgets ------------------------- -Stream Widgets relate to the stream power domain and only consist of ADC's -(analog to digital converters) and DAC's (digital to analog converters). +Stream Widgets relate to the stream power domain and only consist of ADCs +(analog to digital converters) and DACs (digital to analog converters). Stream widgets have the following format:- SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), -NOTE: the stream name must match the corresponding stream name in your codecs +NOTE: the stream name must match the corresponding stream name in your codec snd_soc_codec_dai. e.g. stream widgets for HiFi playback and capture @@ -97,7 +97,7 @@ SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), 2.2 Path Domain Widgets ----------------------- -Path domain widgets have a ability to control or effect the audio signal or +Path domain widgets have a ability to control or affect the audio signal or audio paths within the audio subsystem. They have the following form:- SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) @@ -149,7 +149,7 @@ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), 2.4 Codec Domain ---------------- -The Codec power domain has no widgets and is handled by the codecs DAPM event +The codec power domain has no widgets and is handled by the codecs DAPM event handler. This handler is called when the codec powerstate is changed wrt to any stream event or by kernel PM events. @@ -158,8 +158,8 @@ stream event or by kernel PM events. ------------------- Sometimes widgets exist in the codec or machine audio map that don't have any -corresponding register bit for power control. In this case it's necessary to -create a virtual widget - a widget with no control bits e.g. +corresponding soft power control. In this case it is necessary to create +a virtual widget - a widget with no control bits e.g. SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), @@ -172,13 +172,14 @@ subsystem individually with a call to snd_soc_dapm_new_control(). 3. Codec Widget Interconnections ================================ -Widgets are connected to each other within the codec and machine by audio -paths (called interconnections). Each interconnection must be defined in order -to create a map of all audio paths between widgets. +Widgets are connected to each other within the codec and machine by audio paths +(called interconnections). Each interconnection must be defined in order to +create a map of all audio paths between widgets. + This is easiest with a diagram of the codec (and schematic of the machine audio system), as it requires joining widgets together via their audio signal paths. -i.e. from the WM8731 codec's output mixer (wm8731.c) +e.g., from the WM8731 output mixer (wm8731.c) The WM8731 output mixer has 3 inputs (sources) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 72bd222..f370e7d 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -16,7 +16,7 @@ struct snd_soc_machine { int (*remove)(struct platform_device *pdev); /* the pre and post PM functions are used to do any PM work before and - * after the codec and DAI's do any PM work. */ + * after the codec and DAIs do any PM work. */ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); int (*suspend_post)(struct platform_device *pdev, pm_message_t state); int (*resume_pre)(struct platform_device *pdev); @@ -38,7 +38,7 @@ probe/remove are optional. Do any machine specific probe here. suspend()/resume() ------------------ The machine driver has pre and post versions of suspend and resume to take care -of any machine audio tasks that have to be done before or after the codec, DAI's +of any machine audio tasks that have to be done before or after the codec, DAIs and DMA is suspended and resumed. Optional. @@ -49,10 +49,10 @@ The machine specific audio operations can be set here. Again this is optional. Machine DAI Configuration ------------------------- -The machine DAI configuration glues all the codec and CPU DAI's together. It can +The machine DAI configuration glues all the codec and CPU DAIs together. It can also be used to set up the DAI system clock and for any machine related DAI initialisation e.g. the machine audio map can be connected to the codec audio -map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c +map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c for examples. struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. @@ -67,7 +67,7 @@ static struct snd_soc_dai_link corgi_dai = { .ops = &corgi_ops, }; -struct snd_soc_machine then sets up the machine with it's DAI's. e.g. +struct snd_soc_machine then sets up the machine with it's DAIs. e.g. /* corgi audio machine driver */ static struct snd_soc_machine snd_soc_machine_corgi = { @@ -110,4 +110,4 @@ details. Machine Controls ---------------- -Machine specific audio mixer controls can be added in the dai init function. \ No newline at end of file +Machine specific audio mixer controls can be added in the DAI init function. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index c47ce95..1e4c6d3 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -1,25 +1,26 @@ ALSA SoC Layer ============== -The overall project goal of the ALSA System on Chip (ASoC) layer is to provide -better ALSA support for embedded system-on-chip processors (e.g. pxa2xx, au1x00, -iMX, etc) and portable audio codecs. Currently there is some support in the -kernel for SoC audio, however it has some limitations:- +The overall project goal of the ALSA System on Chip (ASoC) layer is to +provide better ALSA support for embedded system-on-chip processors (e.g. +pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC +subsystem there was some support in the kernel for SoC audio, however it +had some limitations:- - * Currently, codec drivers are often tightly coupled to the underlying SoC - CPU. This is not ideal and leads to code duplication i.e. Linux now has 4 - different wm8731 drivers for 4 different SoC platforms. + * Codec drivers were often tightly coupled to the underlying SoC + CPU. This is not ideal and leads to code duplication - for example, + Linux had different wm8731 drivers for 4 different SoC platforms. - * There is no standard method to signal user initiated audio events (e.g. + * There was no standard method to signal user initiated audio events (e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion event). These are quite common events on portable devices and often require machine specific code to re-route audio, enable amps, etc., after such an event. - * Current drivers tend to power up the entire codec when playing - (or recording) audio. This is fine for a PC, but tends to waste a lot of - power on portable devices. There is also no support for saving power via - changing codec oversampling rates, bias currents, etc. + * Drivers tended to power up the entire codec when playing (or + recording) audio. This is fine for a PC, but tends to waste a lot of + power on portable devices. There was also no support for saving + power via changing codec oversampling rates, bias currents, etc. ASoC Design @@ -31,12 +32,13 @@ features :- * Codec independence. Allows reuse of codec drivers on other platforms and machines. - * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface - and codec registers it's audio interface capabilities with the core and are - subsequently matched and configured when the application hw params are known. + * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC + interface and codec registers it's audio interface capabilities with the + core and are subsequently matched and configured when the application + hardware parameters are known. * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to - it's minimum power state at all times. This includes powering up/down + its minimum power state at all times. This includes powering up/down internal power blocks depending on the internal codec audio routing and any active streams. @@ -45,16 +47,16 @@ features :- signals the codec when to change power states. * Machine specific controls: Allow machines to add controls to the sound card - (e.g. volume control for speaker amp). + (e.g. volume control for speaker amplifier). To achieve all this, ASoC basically splits an embedded audio system into 3 components :- * Codec driver: The codec driver is platform independent and contains audio - controls, audio interface capabilities, codec dapm definition and codec IO + controls, audio interface capabilities, codec DAPM definition and codec IO functions. - * Platform driver: The platform driver contains the audio dma engine and audio + * Platform driver: The platform driver contains the audio DMA engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. * Machine driver: The machine driver handles any machine specific controls and @@ -81,4 +83,4 @@ machine.txt: Machine driver internals. pop_clicks.txt: How to minimise audio artifacts. -clocking.txt: ASoC clocking for best power performance. \ No newline at end of file +clocking.txt: ASoC clocking for best power performance. diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index d4678b4..b681d17 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -8,7 +8,7 @@ specific code. Audio DMA ========= -The platform DMA driver optionally supports the following alsa operations:- +The platform DMA driver optionally supports the following ALSA operations:- /* SoC audio ops */ struct snd_soc_ops { @@ -38,7 +38,7 @@ struct snd_soc_platform { struct snd_pcm_ops *pcm_ops; }; -Please refer to the alsa driver documentation for details of audio DMA. +Please refer to the ALSA driver documentation for details of audio DMA. http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm An example DMA driver is soc/pxa/pxa2xx-pcm.c @@ -52,7 +52,7 @@ Each SoC DAI driver must provide the following features:- 1) Digital audio interface (DAI) description 2) Digital audio interface configuration 3) PCM's description - 4) Sysclk configuration + 4) SYSCLK configuration 5) Suspend and resume (optional) Please see codec.txt for a description of items 1 - 4. diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt index 3371bd9..e1e74da 100644 --- a/Documentation/sound/alsa/soc/pops_clicks.txt +++ b/Documentation/sound/alsa/soc/pops_clicks.txt @@ -15,11 +15,11 @@ click every time a component power state is changed. Minimising Playback Pops and Clicks =================================== -Playback pops in portable audio subsystems cannot be completely eliminated atm, -however future audio codec hardware will have better pop and click suppression. -Pops can be reduced within playback by powering the audio components in a -specific order. This order is different for startup and shutdown and follows -some basic rules:- +Playback pops in portable audio subsystems cannot be completely eliminated +currently, however future audio codec hardware will have better pop and click +suppression. Pops can be reduced within playback by powering the audio +components in a specific order. This order is different for startup and +shutdown and follows some basic rules:- Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute -- cgit v0.10.2 From 5be55be58c3255cf0b19c936353d4f6cebc38e4a Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 23 Jan 2008 11:52:06 +0100 Subject: [ALSA] fix cs5535 section mismatch snd_cs5535audio_mixer() is only called by __devinit snd_cs5535audio_probe(), so the mixer function can also be __devinit. WARNING: vmlinux.o(.text+0xfdbba0): Section mismatch: reference to .init.data:ac97_quirks (between 'snd_cs5535audio_mixer' and 'process_bm0_irq') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index a0c54f0..1d8b160 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -144,7 +144,7 @@ static unsigned short snd_cs5535audio_ac97_codec_read(struct snd_ac97 *ac97, return snd_cs5535audio_codec_read(cs5535au, reg); } -static int snd_cs5535audio_mixer(struct cs5535audio *cs5535au) +static int __devinit snd_cs5535audio_mixer(struct cs5535audio *cs5535au) { struct snd_card *card = cs5535au->card; struct snd_ac97_bus *pbus; -- cgit v0.10.2 From 9478bc3bed1e15208f8041b44d45505cb93e6cc8 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 23 Jan 2008 11:52:38 +0100 Subject: [ALSA] fix opti9xx/miro section mismatch snd_opti93x_mixer() is only called by __devinit snd_opti93x_probe(), so the former can also be __devinit. snd_miro_mixer() is only called by __devinit snd_miro_probe(), so the former can also be __devinit. sound/isa/opti9xx/opti92x-ad1848.c: WARNING: vmlinux.o(.text+0xf91cd7): Section mismatch: reference to .init.data:snd_opti93x_controls (between 'snd_opti93x_mixer' and 'snd_card_opti9xx_free') WARNING: vmlinux.o(.text+0xf91d66): Section mismatch: reference to .init.data:snd_miro_controls (between 'snd_opti93x_mixer' and 'snd_card_opti9xx_free') opti9xx/miro.c: WARNING: vmlinux.o(.text+0xf926c2): Section mismatch: reference to .init.data:snd_miro_controls (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') WARNING: vmlinux.o(.text+0xf926e5): Section mismatch: reference to .init.data:snd_miro_eq_controls (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') WARNING: vmlinux.o(.text+0xf926f9): Section mismatch: reference to .init.data:snd_miro_line_control (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') WARNING: vmlinux.o(.text+0xf92716): Section mismatch: reference to .init.data:snd_miro_amp_control (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') WARNING: vmlinux.o(.text+0xf9273e): Section mismatch: reference to .init.data:snd_miro_preamp_control (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') WARNING: vmlinux.o(.text+0xf92764): Section mismatch: reference to .init.data:snd_miro_capture_control (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') WARNING: vmlinux.o(.text+0xf92783): Section mismatch: reference to .init.data:snd_miro_radio_control (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') WARNING: vmlinux.o(.text+0xf9279a): Section mismatch: reference to .init.data:snd_miro_eq_controls (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') WARNING: vmlinux.o(.text+0xf927b9): Section mismatch: reference to .init.data:snd_miro_radio_control (between 'snd_miro_mixer' and 'snd_legacy_find_free_ioport') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b18d14f..2a1e2f5 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -669,7 +669,7 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) return 0; } -static int snd_miro_mixer(struct snd_miro *miro) +static int __devinit snd_miro_mixer(struct snd_miro *miro) { struct snd_card *card; unsigned int idx; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 1f9c557..fe1afc1 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -1594,7 +1594,7 @@ OPTi93X_DOUBLE("Capture Volume", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 0 } }; -static int snd_opti93x_mixer(struct snd_opti93x *chip) +static int __devinit snd_opti93x_mixer(struct snd_opti93x *chip) { struct snd_card *card; struct snd_kcontrol_new knew; -- cgit v0.10.2 From 33c646e4ffb1b48d67598fadf3323158f0cfd4b9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 24 Jan 2008 08:43:16 +0100 Subject: [ALSA] oxygen: fix SPDIF input rates Fix up SPDIF input sample rates again: 32 kHz and 64 kHz are not supported. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 272ef08..2785660 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -119,6 +119,11 @@ static int oxygen_open(struct snd_pcm_substream *substream, runtime->private_data = (void *)(uintptr_t)channel; runtime->hw = *oxygen_hardware[channel]; + if (channel == PCM_C) { + runtime->hw.rates &= ~(SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_64000); + runtime->hw.rate_min = 44100; + } if (chip->model->pcm_hardware_filter) chip->model->pcm_hardware_filter(channel, &runtime->hw); err = snd_pcm_hw_constraint_step(runtime, 0, -- cgit v0.10.2 From cd93dc8ccad8680f104c9134ae73888feb14e946 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 24 Jan 2008 08:43:39 +0100 Subject: [ALSA] oxygen: remove MIDI for generic cards None of the reference design models have MIDI, only the X-Meridian allows to connect a MIDI adapter. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index af6e802..4f809bd 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -386,7 +386,7 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { static int dev; - const struct oxygen_model *model; + int is_meridian; int err; if (dev >= SNDRV_CARDS) @@ -395,8 +395,9 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - model = pci_id->driver_data ? &model_meridian : &model_generic; - err = oxygen_pci_probe(pci, index[dev], id[dev], 1, model); + is_meridian = pci_id->driver_data; + err = oxygen_pci_probe(pci, index[dev], id[dev], is_meridian, + is_meridian ? &model_meridian : &model_generic); if (err >= 0) ++dev; return err; -- cgit v0.10.2 From 90da78bf6aaabd4d31c6663b7c1d1b9c5a8c023f Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 24 Jan 2008 11:48:01 +0100 Subject: [ALSA] hda: Added mono_out_pin to autoconfig Added a mono_out_pin field to autocfg struct, and code to parse for the mono_out_line. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f6a9a5d..df927be 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2708,6 +2708,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_LINE_OUT: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); + + if (!(wid_caps & AC_WCAP_STEREO)) + if (!cfg->mono_out_pin) + cfg->mono_out_pin = nid; if (!assoc) continue; if (!assoc_line_out) @@ -2856,6 +2860,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_outs, cfg->hp_pins[0], cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); + snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 448c4ce..8a96047d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -332,6 +332,7 @@ struct auto_pin_cfg { hda_nid_t input_pins[AUTO_PIN_LAST]; hda_nid_t dig_out_pin; hda_nid_t dig_in_pin; + hda_nid_t mono_out_pin; }; #define get_defcfg_connect(cfg) \ -- cgit v0.10.2 From 09a99959180d25f4e5070f902e3adc1b20439cd6 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 24 Jan 2008 11:49:21 +0100 Subject: [ALSA] hda: Add dynamic mono mixer support for STAC92xx codecs Allows for dynamically creating mono out mixer controls and well as mono mux controls. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index df927be..26812dc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -769,7 +769,7 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key) /* * query AMP capabilities for the given widget and direction */ -static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) +u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) { struct hda_amp_info *info; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 8a96047d..ad0014a 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -375,6 +375,7 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) return codec->wcaps[nid - codec->start_nid]; } +u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e014ae3..b8152ef 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -756,7 +756,6 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), - STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -768,15 +767,12 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), - - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -785,8 +781,6 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; @@ -1157,7 +1151,7 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { static unsigned int ref925x_pin_configs[8] = { 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021, - 0x90a70320, 0x02214210, 0x400003f1, 0x9033032e, + 0x90a70320, 0x02214210, 0x01019020, 0x9033032e, }; static unsigned int stac925x_MA6_pin_configs[8] = { @@ -1561,7 +1555,7 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { static unsigned int ref9205_pin_configs[12] = { 0x40000100, 0x40000100, 0x01016011, 0x01014010, - 0x01813122, 0x01a19021, 0x40000100, 0x40000100, + 0x01813122, 0x01a19021, 0x01019020, 0x40000100, 0x90a000f0, 0x90a000f0, 0x01441030, 0x01c41030 }; @@ -2018,6 +2012,7 @@ static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol, enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, + STAC_CTL_WIDGET_MONO_MUX, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH }; @@ -2025,6 +2020,7 @@ enum { static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + STAC_MONO_MUX, STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2388,7 +2384,9 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) mono_mux->items[mono_mux->num_items].index = i; mono_mux->num_items++; } - return 0; + + return stac92xx_add_control(spec, STAC_CTL_WIDGET_MONO_MUX, + "Mono Mux", spec->mono_nid); } /* labels for dmic mux inputs */ @@ -2570,6 +2568,50 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->autocfg.line_outs = spec->autocfg.hp_outs; hp_speaker_swap = 1; } + if (spec->autocfg.mono_out_pin) { + int dir = (get_wcaps(codec, spec->autocfg.mono_out_pin) + & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; + u32 caps = query_amp_caps(codec, + spec->autocfg.mono_out_pin, dir); + hda_nid_t conn_list[1]; + + /* get the mixer node and then the mono mux if it exists */ + if (snd_hda_get_connections(codec, + spec->autocfg.mono_out_pin, conn_list, 1) && + snd_hda_get_connections(codec, conn_list[0], + conn_list, 1)) { + + int wcaps = get_wcaps(codec, conn_list[0]); + int wid_type = (wcaps & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + /* LR swap check, some stac925x have a mux that + * changes the DACs output path instead of the + * mono-mux path. + */ + if (wid_type == AC_WID_AUD_SEL && + !(wcaps & AC_WCAP_LR_SWAP)) + spec->mono_nid = conn_list[0]; + } + /* all mono outs have a least a mute/unmute switch */ + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, + "Mono Playback Switch", + HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, + 1, 0, dir)); + if (err < 0) + return err; + /* check to see if there is volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, + "Mono Playback Volume", + HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, + 1, 0, dir)); + if (err < 0) + return err; + } + + stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin, + AC_PINCTL_OUT_EN); + } if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) return err; @@ -3317,7 +3359,6 @@ again: spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ - spec->mono_nid = 0x15; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; -- cgit v0.10.2 From 6330079fc6df4a0829f952b73c4d4999e56034f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Jan 2008 15:31:36 +0100 Subject: [ALSA] hda-codec - Fix handling of multiple capture streams Fixed the bug that multiple capture streams conflict on Realtek codec routines. Also, this adds a framework to enable the alternative playback stream, e.g. for VoIP. It's not fully implemented yet, though. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f652af..c08852a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -220,6 +220,8 @@ struct alc_spec { char *stream_name_analog; /* analog PCM stream */ struct hda_pcm_stream *stream_analog_playback; struct hda_pcm_stream *stream_analog_capture; + struct hda_pcm_stream *stream_analog_alt_playback; + struct hda_pcm_stream *stream_analog_alt_capture; char *stream_name_digital; /* digital PCM stream */ struct hda_pcm_stream *stream_digital_playback; @@ -230,6 +232,7 @@ struct alc_spec { * max_channels, dacs must be set * dig_out_nid and hp_nid are optional */ + hda_nid_t alt_dac_nid; /* capture */ unsigned int num_adc_nids; @@ -2370,7 +2373,7 @@ static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, /* * Analog capture */ -static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int alc880_alt_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, @@ -2378,18 +2381,18 @@ static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, { struct alc_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], + snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number + 1], stream_tag, 0, format); return 0; } -static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, +static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], + snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number + 1], 0, 0, 0); return 0; } @@ -2410,13 +2413,27 @@ static struct hda_pcm_stream alc880_pcm_analog_playback = { }; static struct hda_pcm_stream alc880_pcm_analog_capture = { - .substreams = 2, + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in alc_build_pcms */ +}; + +static struct hda_pcm_stream alc880_pcm_analog_alt_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in alc_build_pcms */ +}; + +static struct hda_pcm_stream alc880_pcm_analog_alt_capture = { + .substreams = 2, /* can be overridden */ .channels_min = 2, .channels_max = 2, /* NID is set in alc_build_pcms */ .ops = { - .prepare = alc880_capture_pcm_prepare, - .cleanup = alc880_capture_pcm_cleanup + .prepare = alc880_alt_capture_pcm_prepare, + .cleanup = alc880_alt_capture_pcm_cleanup }, }; @@ -2440,7 +2457,7 @@ static struct hda_pcm_stream alc880_pcm_digital_capture = { }; /* Used by alc_build_pcms to flag that a PCM has no playback stream */ -static struct hda_pcm_stream alc_pcm_null_playback = { +static struct hda_pcm_stream alc_pcm_null_stream = { .substreams = 0, .channels_min = 0, .channels_max = 0, @@ -2497,17 +2514,32 @@ static int alc_build_pcms(struct hda_codec *codec) * model, configure a second analog capture-only PCM. */ /* Additional Analaog capture for index #2 */ - if (spec->num_adc_nids > 1 && spec->stream_analog_capture && - spec->adc_nids) { + if ((spec->alt_dac_nid && spec->stream_analog_alt_playback) || + (spec->num_adc_nids > 1 && spec->stream_analog_alt_capture)) { codec->num_pcms = 3; info = spec->pcm_rec + 2; info->name = spec->stream_name_analog; - /* No playback stream for second PCM */ - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; - if (spec->stream_analog_capture) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1]; + if (spec->alt_dac_nid) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *spec->stream_analog_alt_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->alt_dac_nid; + } else { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + alc_pcm_null_stream; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; + } + if (spec->num_adc_nids > 1) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + *spec->stream_analog_alt_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = + spec->adc_nids[1]; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = + spec->num_adc_nids - 1; + } else { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + alc_pcm_null_stream; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = 0; } } @@ -3615,6 +3647,7 @@ static int patch_alc880(struct hda_codec *codec) spec->stream_name_analog = "ALC880 Analog"; spec->stream_analog_playback = &alc880_pcm_analog_playback; spec->stream_analog_capture = &alc880_pcm_analog_capture; + spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; spec->stream_name_digital = "ALC880 Digital"; spec->stream_digital_playback = &alc880_pcm_digital_playback; @@ -4527,17 +4560,8 @@ static struct hda_verb alc260_test_init_verbs[] = { }; #endif -static struct hda_pcm_stream alc260_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, -}; - -static struct hda_pcm_stream alc260_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, -}; +#define alc260_pcm_analog_playback alc880_pcm_analog_alt_playback +#define alc260_pcm_analog_capture alc880_pcm_analog_capture #define alc260_pcm_digital_playback alc880_pcm_digital_playback #define alc260_pcm_digital_capture alc880_pcm_digital_capture @@ -6204,6 +6228,9 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_name_analog = "ALC882 Analog"; spec->stream_analog_playback = &alc882_pcm_analog_playback; spec->stream_analog_capture = &alc882_pcm_analog_capture; + /* FIXME: setup DAC5 */ + /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/ + spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; spec->stream_name_digital = "ALC882 Digital"; spec->stream_digital_playback = &alc882_pcm_digital_playback; @@ -7409,6 +7436,7 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = { /* pcm configuration: identiacal with ALC880 */ #define alc883_pcm_analog_playback alc880_pcm_analog_playback #define alc883_pcm_analog_capture alc880_pcm_analog_capture +#define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture #define alc883_pcm_digital_playback alc880_pcm_digital_playback #define alc883_pcm_digital_capture alc880_pcm_digital_capture @@ -7894,6 +7922,7 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_name_analog = "ALC883 Analog"; spec->stream_analog_playback = &alc883_pcm_analog_playback; spec->stream_analog_capture = &alc883_pcm_analog_capture; + spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture; spec->stream_name_digital = "ALC883 Digital"; spec->stream_digital_playback = &alc883_pcm_digital_playback; @@ -9821,6 +9850,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) /* pcm configuration: identiacal with ALC880 */ #define alc268_pcm_analog_playback alc880_pcm_analog_playback #define alc268_pcm_analog_capture alc880_pcm_analog_capture +#define alc268_pcm_analog_alt_capture alc880_pcm_analog_alt_capture #define alc268_pcm_digital_playback alc880_pcm_digital_playback /* @@ -10022,6 +10052,7 @@ static int patch_alc268(struct hda_codec *codec) spec->stream_name_analog = "ALC268 Analog"; spec->stream_analog_playback = &alc268_pcm_analog_playback; spec->stream_analog_capture = &alc268_pcm_analog_capture; + spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture; spec->stream_name_digital = "ALC268 Digital"; spec->stream_digital_playback = &alc268_pcm_digital_playback; -- cgit v0.10.2 From e2e7d624ad3519809c5ff5ff53171e07aeeb7f68 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 24 Jan 2008 15:32:15 +0100 Subject: [ALSA] hda: STAC9205 GPIO line fix Fixed issue that the incorrect GPIO line was being pulled high for some STAC9205 based laptops. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b8152ef..3b8b860 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3661,11 +3661,11 @@ static int patch_stac9205(struct hda_codec *codec) stac92xx_set_config_reg(codec, 0x1f, 0x01441030); stac92xx_set_config_reg(codec, 0x20, 0x1c410030); - spec->gpio_mask = 0x00000007; /* GPIO0-2 */ - /* GPIO0 High = EAPD, GPIO1 Low = DRM, - * GPIO2 High = Headphone Mute + spec->gpio_mask = 0x0000000b; + /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute, + * GPIO3 High = DRM */ - spec->gpio_data = 0x00000005; + spec->gpio_data = 0x00000009; break; default: /* GPIO0 High = EAPD */ -- cgit v0.10.2 From 29a52c242d76deee155cb94756bcf7ebf58de4fe Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Thu, 24 Jan 2008 17:29:00 +0100 Subject: [ALSA] hda-codec - Add model for Acer Aspire 5315 Simplify usage of the Acer Aspire 5315 laptop with the ALC268 based codec sound card via add correct PCI SSID. Signed-off-by: Andy Shevchenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c08852a..99e2c66 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9926,6 +9926,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), + SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), -- cgit v0.10.2 From 69252128ec628e9d19739db0101e1826d993aecb Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Thu, 24 Jan 2008 18:11:53 +0100 Subject: [ALSA] fm801 - Add mute support for FM-only card with FM801 PCI to tuner bridge This is improvement of the early support of the FM-only cards where the fm801 chip represents the PCI to tuner bridge. The tuner initialization isn't included the mute on as well as mute support via V4L request. Proposed patch should fix this at least for 64-PCR model. Signed-off-by: Andy Shevchenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/tea575x-tuner.h b/include/sound/tea575x-tuner.h index e8eeb3a..b62ce3e 100644 --- a/include/sound/tea575x-tuner.h +++ b/include/sound/tea575x-tuner.h @@ -30,6 +30,7 @@ struct snd_tea575x; struct snd_tea575x_ops { void (*write)(struct snd_tea575x *tea, unsigned int val); unsigned int (*read)(struct snd_tea575x *tea); + void (*mute)(struct snd_tea575x *tea, unsigned int mute); }; struct snd_tea575x { diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 28a4af7..87e3aef 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -158,6 +158,10 @@ static int snd_tea575x_ioctl(struct inode *inode, struct file *file, struct video_audio v; if(copy_from_user(&v, arg, sizeof(v))) return -EFAULT; + if (tea->ops->mute) + tea->ops->mute(tea, + (v.flags & + VIDEO_AUDIO_MUTE) ? 1 : 0); if(v.audio) return -EINVAL; return 0; @@ -205,6 +209,10 @@ void snd_tea575x_init(struct snd_tea575x *tea) tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); + + /* mute on init */ + if (tea->ops->mute) + tea->ops->mute(tea, 1); } void snd_tea575x_exit(struct snd_tea575x *tea) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 25c1087..4c300e6 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -978,6 +978,27 @@ static unsigned int snd_fm801_tea575x_64pcr_read(struct snd_tea575x *tea) return val; } +static void snd_fm801_tea575x_64pcr_mute(struct snd_tea575x *tea, + unsigned int mute) +{ + struct fm801 *chip = tea->private_data; + unsigned short reg; + + spin_lock_irq(&chip->reg_lock); + + reg = inw(FM801_REG(chip, GPIO_CTRL)); + if (mute) + /* 0xf800 (mute) */ + reg &= ~FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); + else + /* 0xf802 (unmute) */ + reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + + spin_unlock_irq(&chip->reg_lock); +} + static struct snd_tea575x_ops snd_fm801_tea_ops[3] = { { /* 1 = MediaForte 256-PCS */ @@ -993,6 +1014,7 @@ static struct snd_tea575x_ops snd_fm801_tea_ops[3] = { /* 3 = MediaForte 64-PCR */ .write = snd_fm801_tea575x_64pcr_write, .read = snd_fm801_tea575x_64pcr_read, + .mute = snd_fm801_tea575x_64pcr_mute, } }; #endif -- cgit v0.10.2 From 09189ac793d7fef7d0f058815043e3d4bf7097c0 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 24 Jan 2008 18:46:42 +0100 Subject: [ALSA] usb/caiaq: decrease period_bytes_min This patch decreases the snd_pcm_hardware->period_bytes_min field in the caiaq/usb audio driver. The hardware can actually handle as few as 128 bytes, depending on the system. So it makes no sense to keep applications from actually using such values. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index 2f9e9b2..9cc4cd8 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -57,7 +57,7 @@ static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { .channels_min = CHANNELS_PER_STREAM, .channels_max = CHANNELS_PER_STREAM, .buffer_bytes_max = MAX_BUFFER_SIZE, - .period_bytes_min = 4096, + .period_bytes_min = 128, .period_bytes_max = MAX_BUFFER_SIZE, .periods_min = 1, .periods_max = 1024, diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 5cd92ae..58d25e4 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.1"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.2"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- cgit v0.10.2 From 976cd62700ae378df330ec82112da3d17e33a0fe Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 25 Jan 2008 08:37:49 +0100 Subject: [ALSA] oxygen: make the number of analog output configurable Add a field to struct oxygen_model to allow model drivers for cards with less than eight output channels. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 4f809bd..60af3a7 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -350,6 +350,7 @@ static const struct oxygen_model model_generic = { .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, .model_data_size = sizeof(struct generic_data), + .dac_channels = 8, .used_channels = OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_C | OXYGEN_CHANNEL_SPDIF | @@ -372,6 +373,7 @@ static const struct oxygen_model model_meridian = { .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, .model_data_size = sizeof(struct generic_data), + .dac_channels = 8, .used_channels = OXYGEN_CHANNEL_B | OXYGEN_CHANNEL_C | OXYGEN_CHANNEL_SPDIF | diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 4894dbd..9ec9e6b 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -85,6 +85,7 @@ struct oxygen_model { void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); size_t model_data_size; + u8 dac_channels; u8 used_channels; u8 function_flags; u16 dac_i2s_format; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 6b7420f..21b227a 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -28,8 +28,10 @@ static int dac_volume_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { + struct oxygen *chip = ctl->private_data; + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = 8; + info->count = chip->model->dac_channels; info->value.integer.min = 0; info->value.integer.max = 0xff; return 0; @@ -42,7 +44,7 @@ static int dac_volume_get(struct snd_kcontrol *ctl, unsigned int i; mutex_lock(&chip->mutex); - for (i = 0; i < 8; ++i) + for (i = 0; i < chip->model->dac_channels; ++i) value->value.integer.value[i] = chip->dac_volume[i]; mutex_unlock(&chip->mutex); return 0; @@ -57,7 +59,7 @@ static int dac_volume_put(struct snd_kcontrol *ctl, changed = 0; mutex_lock(&chip->mutex); - for (i = 0; i < 8; ++i) + for (i = 0; i < chip->model->dac_channels; ++i) if (value->value.integer.value[i] != chip->dac_volume[i]) { chip->dac_volume[i] = value->value.integer.value[i]; changed = 1; @@ -100,11 +102,14 @@ static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) static const char *const names[3] = { "Front", "Front+Surround", "Front+Surround+Back" }; + struct oxygen *chip = ctl->private_data; + unsigned int count = 2 + (chip->model->dac_channels == 8); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item > 2) - info->value.enumerated.item = 2; + info->value.enumerated.items = count; + if (info->value.enumerated.item >= count) + info->value.enumerated.item = count - 1; strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); return 0; } @@ -167,12 +172,14 @@ void oxygen_update_dac_routing(struct oxygen *chip) static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; + unsigned int count = 2 + (chip->model->dac_channels == 8); int changed; mutex_lock(&chip->mutex); changed = value->value.enumerated.item[0] != chip->dac_routing; if (changed) { - chip->dac_routing = min(value->value.enumerated.item[0], 2u); + chip->dac_routing = min(value->value.enumerated.item[0], + count - 1); spin_lock_irq(&chip->reg_lock); oxygen_update_dac_routing(chip); spin_unlock_irq(&chip->reg_lock); diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 2785660..df1d0ca 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -119,10 +119,15 @@ static int oxygen_open(struct snd_pcm_substream *substream, runtime->private_data = (void *)(uintptr_t)channel; runtime->hw = *oxygen_hardware[channel]; - if (channel == PCM_C) { + switch (channel) { + case PCM_C: runtime->hw.rates &= ~(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000); runtime->hw.rate_min = 44100; + break; + case PCM_MULTICH: + runtime->hw.channels_max = chip->model->dac_channels; + break; } if (chip->model->pcm_hardware_filter) chip->model->pcm_hardware_filter(channel, &runtime->hw); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 7b240c7..665115d 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -310,6 +310,7 @@ static const struct oxygen_model model_xonar = { .set_adc_params = set_cs5381_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, + .dac_channels = 8, .used_channels = OXYGEN_CHANNEL_B | OXYGEN_CHANNEL_C | OXYGEN_CHANNEL_SPDIF | -- cgit v0.10.2 From b78e3dbb04ab4cbe3b94ef5426bcd5b167b6fc75 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 25 Jan 2008 08:39:26 +0100 Subject: [ALSA] oxygen: more initialization Initialize more registers of the controller and the second AC97 codec. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 49eabda..06394e4 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -231,6 +231,29 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC | chip->model->function_flags); + oxygen_write8_masked(chip, OXYGEN_FUNCTION, + OXYGEN_FUNCTION_SPI, + OXYGEN_FUNCTION_2WIRE_SPI_MASK); + oxygen_write8(chip, OXYGEN_DMA_STATUS, 0); + oxygen_write8(chip, OXYGEN_DMA_PAUSE, 0); + oxygen_write8(chip, OXYGEN_PLAY_CHANNELS, + OXYGEN_PLAY_CHANNELS_2 | + OXYGEN_DMA_A_BURST_8 | + OXYGEN_DMA_MULTICH_BURST_8); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + oxygen_write8_masked(chip, OXYGEN_MISC, 0, + OXYGEN_MISC_WRITE_PCI_SUBID | + OXYGEN_MISC_REC_C_FROM_SPDIF | + OXYGEN_MISC_REC_B_FROM_AC97 | + OXYGEN_MISC_REC_A_FROM_MULTICH); + oxygen_write8(chip, OXYGEN_REC_FORMAT, + (OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_A_SHIFT) | + (OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_B_SHIFT) | + (OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_C_SHIFT)); + oxygen_write8(chip, OXYGEN_PLAY_FORMAT, + (OXYGEN_FORMAT_16 << OXYGEN_SPDIF_FORMAT_SHIFT) | + (OXYGEN_FORMAT_16 << OXYGEN_MULTICH_FORMAT_SHIFT)); + oxygen_write8(chip, OXYGEN_REC_CHANNELS, OXYGEN_REC_CHANNELS_2_2_2); oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT, OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST | OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | @@ -262,15 +285,19 @@ static void __devinit oxygen_init(struct oxygen *chip) OXYGEN_SPDIF_LOCK_PAR | OXYGEN_SPDIF_IN_CLOCK_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); + oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK); + oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0); + oxygen_write16(chip, OXYGEN_GPIO_INTERRUPT_MASK, 0); oxygen_write16(chip, OXYGEN_PLAY_ROUTING, - OXYGEN_PLAY_MULTICH_I2S_DAC | OXYGEN_PLAY_SPDIF_SPDIF | + OXYGEN_PLAY_MULTICH_I2S_DAC | + OXYGEN_PLAY_SPDIF_SPDIF | (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT)); oxygen_write8(chip, OXYGEN_REC_ROUTING, OXYGEN_REC_A_ROUTE_I2S_ADC_1 | - OXYGEN_REC_B_ROUTE_AC97_1 | + OXYGEN_REC_B_ROUTE_I2S_ADC_2 | OXYGEN_REC_C_ROUTE_SPDIF); oxygen_write8(chip, OXYGEN_ADC_MONITOR, 0); oxygen_write8(chip, OXYGEN_A_MONITOR_ROUTING, @@ -279,23 +306,16 @@ static void __devinit oxygen_init(struct oxygen *chip) (2 << OXYGEN_A_MONITOR_ROUTE_2_SHIFT) | (3 << OXYGEN_A_MONITOR_ROUTE_3_SHIFT)); - oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); - oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); - oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0); - if (chip->has_ac97_0) { - oxygen_clear_bits16(chip, OXYGEN_AC97_OUT_CONFIG, - OXYGEN_AC97_CODEC0_FRONTL | - OXYGEN_AC97_CODEC0_FRONTR | - OXYGEN_AC97_CODEC0_SIDEL | - OXYGEN_AC97_CODEC0_SIDER | - OXYGEN_AC97_CODEC0_CENTER | - OXYGEN_AC97_CODEC0_BASE | - OXYGEN_AC97_CODEC0_REARL | - OXYGEN_AC97_CODEC0_REARR); - oxygen_set_bits16(chip, OXYGEN_AC97_IN_CONFIG, - OXYGEN_AC97_CODEC0_LINEL | - OXYGEN_AC97_CODEC0_LINER); + oxygen_write32(chip, OXYGEN_AC97_OUT_CONFIG, 0); + oxygen_write32(chip, OXYGEN_AC97_IN_CONFIG, 0); + if (!(chip->has_ac97_0 | chip->has_ac97_1)) + oxygen_set_bits16(chip, OXYGEN_AC97_CONTROL, + OXYGEN_AC97_CLOCK_DISABLE); + if (!chip->has_ac97_0) { + oxygen_set_bits16(chip, OXYGEN_AC97_CONTROL, + OXYGEN_AC97_NO_CODEC_0); + } else { oxygen_write_ac97(chip, 0, AC97_RESET, 0); msleep(1); oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_SETUP, @@ -325,6 +345,26 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_ac97_set_bits(chip, 0, AC97_EXTENDED_STATUS, AC97_EA_PRI | AC97_EA_PRJ | AC97_EA_PRK); } + if (chip->has_ac97_1) { + oxygen_set_bits32(chip, OXYGEN_AC97_OUT_CONFIG, + OXYGEN_AC97_CODEC1_SLOT3 | + OXYGEN_AC97_CODEC1_SLOT4); + oxygen_write_ac97(chip, 1, AC97_RESET, 0); + msleep(1); + oxygen_write_ac97(chip, 1, AC97_MASTER, 0x0000); + oxygen_write_ac97(chip, 1, AC97_HEADPHONE, 0x8000); + oxygen_write_ac97(chip, 1, AC97_PC_BEEP, 0x8000); + oxygen_write_ac97(chip, 1, AC97_MIC, 0x8808); + oxygen_write_ac97(chip, 1, AC97_LINE, 0x8808); + oxygen_write_ac97(chip, 1, AC97_CD, 0x8808); + oxygen_write_ac97(chip, 1, AC97_VIDEO, 0x8808); + oxygen_write_ac97(chip, 1, AC97_AUX, 0x8808); + oxygen_write_ac97(chip, 1, AC97_PCM, 0x0808); + oxygen_write_ac97(chip, 1, AC97_REC_SEL, 0x0000); + oxygen_write_ac97(chip, 1, AC97_REC_GAIN, 0x8000); + oxygen_ac97_clear_bits(chip, 1, AC97_REC_GAIN, 0x1c00); + oxygen_ac97_set_bits(chip, 1, 0x6a, 0x0040); + } } static void oxygen_card_free(struct snd_card *card) diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index df1d0ca..5b89c72 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -253,8 +253,10 @@ static unsigned int oxygen_rate(struct snd_pcm_hw_params *hw_params) static unsigned int oxygen_i2s_mclk(struct snd_pcm_hw_params *hw_params) { - return params_rate(hw_params) <= 96000 - ? OXYGEN_I2S_MCLK_256 : OXYGEN_I2S_MCLK_128; + if (params_rate(hw_params) <= 96000) + return OXYGEN_I2S_MCLK_256; + else + return OXYGEN_I2S_MCLK_128; } static unsigned int oxygen_i2s_bits(struct snd_pcm_hw_params *hw_params) @@ -339,9 +341,6 @@ static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, OXYGEN_I2S_FORMAT_MASK | OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); - oxygen_write8_masked(chip, OXYGEN_REC_ROUTING, - OXYGEN_REC_A_ROUTE_I2S_ADC_1, - OXYGEN_REC_A_ROUTE_MASK); spin_unlock_irq(&chip->reg_lock); mutex_lock(&chip->mutex); @@ -373,9 +372,6 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, OXYGEN_I2S_FORMAT_MASK | OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); - oxygen_write8_masked(chip, OXYGEN_REC_ROUTING, - OXYGEN_REC_B_ROUTE_I2S_ADC_2, - OXYGEN_REC_B_ROUTE_MASK); spin_unlock_irq(&chip->reg_lock); mutex_lock(&chip->mutex); @@ -398,9 +394,6 @@ static int oxygen_rec_c_hw_params(struct snd_pcm_substream *substream, oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, oxygen_format(hw_params) << OXYGEN_REC_FORMAT_C_SHIFT, OXYGEN_REC_FORMAT_C_MASK); - oxygen_write8_masked(chip, OXYGEN_REC_ROUTING, - OXYGEN_REC_C_ROUTE_SPDIF, - OXYGEN_REC_C_ROUTE_MASK); spin_unlock_irq(&chip->reg_lock); return 0; } @@ -453,11 +446,6 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | OXYGEN_I2S_BITS_MASK); - oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, - OXYGEN_PLAY_MULTICH_I2S_DAC, - OXYGEN_PLAY_MUTE01 | OXYGEN_PLAY_MUTE23 | - OXYGEN_PLAY_MUTE45 | OXYGEN_PLAY_MUTE67 | - OXYGEN_PLAY_MULTICH_MASK); oxygen_update_dac_routing(chip); oxygen_update_spdif_source(chip); spin_unlock_irq(&chip->reg_lock); -- cgit v0.10.2 From c626026dd72ec8363aaa862178adeacfa7ac09c5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 25 Jan 2008 08:41:52 +0100 Subject: [ALSA] add TempoTec HiFier driver Add a driver for the MediaTek/TempoTec HiFier Fantasia sound card. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 07317da..67256bd 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1092,6 +1092,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. See hdspm.txt for details. + Module snd-hifier + ----------------- + + Module for the MediaTek/TempoTec HiFier Fantasia sound card. + + This module supports autoprobe and multiple cards. + + Power management is _not_ supported. + Module snd-ice1712 ------------------ diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 10c3694..812085d 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -647,6 +647,17 @@ config SND_HDSPM To compile this driver as a module, choose M here: the module will be called snd-hdspm. +config SND_HIFIER + tristate "TempoTec HiFier Fantasia" + depends on SND + select SND_OXYGEN_LIB + help + Say Y here to include support for the MediaTek/TempoTec HiFier + Fantasia sound card. + + To compile this driver as a module, choose M here: the module + will be called snd-hifier. + config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" depends on SND diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index 98d5bc0..4ba07d4 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -1,7 +1,9 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o +snd-hifier-objs := hifier.o snd-oxygen-objs := oxygen.o snd-virtuoso-objs := virtuoso.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o +obj-$(CONFIG_SND_HIFIER) += snd-hifier.o obj-$(CONFIG_SND_OXYGEN) += snd-oxygen.o obj-$(CONFIG_SND_VIRTUOSO) += snd-virtuoso.o diff --git a/sound/pci/oxygen/ak4396.h b/sound/pci/oxygen/ak4396.h new file mode 100644 index 0000000..551c1cf --- /dev/null +++ b/sound/pci/oxygen/ak4396.h @@ -0,0 +1,44 @@ +#ifndef AK4396_H_INCLUDED +#define AK4396_H_INCLUDED + +#define AK4396_WRITE 0x2000 + +#define AK4396_CONTROL_1 0 +#define AK4396_CONTROL_2 1 +#define AK4396_CONTROL_3 2 +#define AK4396_LCH_ATT 3 +#define AK4396_RCH_ATT 4 + +/* control 1 */ +#define AK4396_RSTN 0x01 +#define AK4396_DIF_MASK 0x0e +#define AK4396_DIF_16_LSB 0x00 +#define AK4396_DIF_20_LSB 0x02 +#define AK4396_DIF_24_MSB 0x04 +#define AK4396_DIF_24_I2S 0x06 +#define AK4396_DIF_24_LSB 0x08 +#define AK4396_ACKS 0x80 +/* control 2 */ +#define AK4396_SMUTE 0x01 +#define AK4396_DEM_MASK 0x06 +#define AK4396_DEM_441 0x00 +#define AK4396_DEM_OFF 0x02 +#define AK4396_DEM_48 0x04 +#define AK4396_DEM_32 0x06 +#define AK4396_DFS_MASK 0x18 +#define AK4396_DFS_NORMAL 0x00 +#define AK4396_DFS_DOUBLE 0x08 +#define AK4396_DFS_QUAD 0x10 +#define AK4396_SLOW 0x20 +#define AK4396_DZFM 0x40 +#define AK4396_DZFE 0x80 +/* control 3 */ +#define AK4396_DZFB 0x04 +#define AK4396_DCKB 0x10 +#define AK4396_DCKS 0x20 +#define AK4396_DSDM 0x40 +#define AK4396_D_P_MASK 0x80 +#define AK4396_PCM 0x00 +#define AK4396_DSD 0x80 + +#endif diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c new file mode 100644 index 0000000..05ea2b6 --- /dev/null +++ b/sound/pci/oxygen/hifier.c @@ -0,0 +1,207 @@ +/* + * C-Media CMI8788 driver for the MediaTek/TempoTec HiFier Fantasia + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include +#include "oxygen.h" +#include "ak4396.h" + +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_DESCRIPTION("TempoTec HiFier driver"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "card index"); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string"); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "enable card"); + +static struct pci_device_id hifier_ids[] __devinitdata = { + { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, + { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, + { } +}; +MODULE_DEVICE_TABLE(pci, hifier_ids); + +struct hifier_data { + u8 ak4396_ctl2; +}; + +static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) +{ + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_320 | + (0 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI, + AK4396_WRITE | (reg << 8) | value); +} + +static void hifier_init(struct oxygen *chip) +{ + struct hifier_data *data = chip->model_data; + + data->ak4396_ctl2 = AK4396_DEM_OFF | AK4396_DFS_NORMAL; + ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); + ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); + ak4396_write(chip, AK4396_LCH_ATT, 0xff); + ak4396_write(chip, AK4396_RCH_ATT, 0xff); + + snd_component_add(chip->card, "AK4396"); + snd_component_add(chip->card, "CS5340"); +} + +static void hifier_cleanup(struct oxygen *chip) +{ +} + +static void set_ak4396_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct hifier_data *data = chip->model_data; + u8 value; + + value = data->ak4396_ctl2 & ~AK4396_DFS_MASK; + if (params_rate(params) <= 54000) + value |= AK4396_DFS_NORMAL; + else if (params_rate(params) < 120000) + value |= AK4396_DFS_DOUBLE; + else + value |= AK4396_DFS_QUAD; + data->ak4396_ctl2 = value; + ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB); + ak4396_write(chip, AK4396_CONTROL_2, value); + ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); +} + +static void update_ak4396_volume(struct oxygen *chip) +{ + ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); + ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); +} + +static void update_ak4396_mute(struct oxygen *chip) +{ + struct hifier_data *data = chip->model_data; + u8 value; + + value = data->ak4396_ctl2 & ~AK4396_SMUTE; + if (chip->dac_mute) + value |= AK4396_SMUTE; + data->ak4396_ctl2 = value; + ak4396_write(chip, AK4396_CONTROL_2, value); +} + +static void set_cs5340_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ +} + +static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); + +static int hifier_control_filter(struct snd_kcontrol_new *template) +{ + if (!strcmp(template->name, "Master Playback Volume")) { + template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + template->tlv.p = ak4396_db_scale; + } else if (!strcmp(template->name, "Stereo Upmixing")) { + return 1; /* stereo only - we don't need upmixing */ + } else if (!strcmp(template->name, + SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK)) || + !strcmp(template->name, + SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT))) { + return 1; /* no digital input */ + } + return 0; +} + +static int hifier_mixer_init(struct oxygen *chip) +{ + return 0; +} + +static const struct oxygen_model model_hifier = { + .shortname = "C-Media CMI8787", + .longname = "C-Media Oxygen HD Audio", + .chip = "CMI8788", + .init = hifier_init, + .control_filter = hifier_control_filter, + .mixer_init = hifier_mixer_init, + .cleanup = hifier_cleanup, + .set_dac_params = set_ak4396_params, + .set_adc_params = set_cs5340_params, + .update_dac_volume = update_ak4396_volume, + .update_dac_mute = update_ak4396_mute, + .model_data_size = sizeof(struct hifier_data), + .dac_channels = 2, + .used_channels = OXYGEN_CHANNEL_A | + OXYGEN_CHANNEL_SPDIF | + OXYGEN_CHANNEL_MULTICH, + .function_flags = 0, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +static int __devinit hifier_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + ++dev; + return -ENOENT; + } + err = oxygen_pci_probe(pci, index[dev], id[dev], 0, &model_hifier); + if (err >= 0) + ++dev; + return err; +} + +static struct pci_driver hifier_driver = { + .name = "CMI8787HiFier", + .id_table = hifier_ids, + .probe = hifier_probe, + .remove = __devexit_p(oxygen_pci_remove), +}; + +static int __init alsa_card_hifier_init(void) +{ + return pci_register_driver(&hifier_driver); +} + +static void __exit alsa_card_hifier_exit(void) +{ + pci_unregister_driver(&hifier_driver); +} + +module_init(alsa_card_hifier_init) +module_exit(alsa_card_hifier_exit) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 60af3a7..a0457a5 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -36,6 +36,7 @@ #include #include #include "oxygen.h" +#include "ak4396.h" MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 driver"); @@ -61,8 +62,6 @@ static struct pci_device_id oxygen_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x13f6, 0x0010) }, { OXYGEN_PCI_SUBID(0x13f6, 0x8788) }, { OXYGEN_PCI_SUBID(0x147a, 0xa017) }, - { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, - { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910) }, { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = 1 }, { OXYGEN_PCI_SUBID(0x7284, 0x9761) }, @@ -76,46 +75,6 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_AK5385_DFS_DOUBLE 0x0001 #define GPIO_AK5385_DFS_QUAD 0x0002 -#define AK4396_WRITE 0x2000 - -#define AK4396_CONTROL_1 0 -#define AK4396_CONTROL_2 1 -#define AK4396_CONTROL_3 2 -#define AK4396_LCH_ATT 3 -#define AK4396_RCH_ATT 4 - -/* control 1 */ -#define AK4396_RSTN 0x01 -#define AK4396_DIF_MASK 0x0e -#define AK4396_DIF_16_LSB 0x00 -#define AK4396_DIF_20_LSB 0x02 -#define AK4396_DIF_24_MSB 0x04 -#define AK4396_DIF_24_I2S 0x06 -#define AK4396_DIF_24_LSB 0x08 -#define AK4396_ACKS 0x80 -/* control 2 */ -#define AK4396_SMUTE 0x01 -#define AK4396_DEM_MASK 0x06 -#define AK4396_DEM_441 0x00 -#define AK4396_DEM_OFF 0x02 -#define AK4396_DEM_48 0x04 -#define AK4396_DEM_32 0x06 -#define AK4396_DFS_MASK 0x18 -#define AK4396_DFS_NORMAL 0x00 -#define AK4396_DFS_DOUBLE 0x08 -#define AK4396_DFS_QUAD 0x10 -#define AK4396_SLOW 0x20 -#define AK4396_DZFM 0x40 -#define AK4396_DZFE 0x80 -/* control 3 */ -#define AK4396_DZFB 0x04 -#define AK4396_DCKB 0x10 -#define AK4396_DCKS 0x20 -#define AK4396_DSDM 0x40 -#define AK4396_D_P_MASK 0x80 -#define AK4396_PCM 0x00 -#define AK4396_DSD 0x80 - #define WM8785_R0 0 #define WM8785_R1 1 #define WM8785_R2 2 diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 21b227a..fe53318 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -712,6 +712,8 @@ static int add_controls(struct oxygen *chip, err = chip->model->control_filter(&template); if (err < 0) return err; + if (err == 1) + continue; ctl = snd_ctl_new1(&template, chip); if (!ctl) return -ENOMEM; -- cgit v0.10.2 From 461e2c78b153e38f284d09721c50c0cd3c47e073 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Jan 2008 11:35:17 +0100 Subject: [ALSA] hda-codec - Add Conexant 5051 codec support Added the support for Conexant 5051 audio codec. Right now there are two preset models, laptop and hp. The whole patch is based on the information from the base patch by Linuxant. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 67256bd..73f5012 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -961,6 +961,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + Conexant 5051 + laptop Basic Laptop config (default) + hp HP Spartan laptop + STAC9200 ref Reference board dell-d21 Dell (unknown) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1ed0f075..5f6de34 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -64,6 +64,11 @@ struct conexant_spec { hda_nid_t *adc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ + unsigned int cur_adc_idx; + hda_nid_t cur_adc; + unsigned int cur_adc_stream_tag; + unsigned int cur_adc_format; + /* capture source */ const struct hda_input_mux *input_mux; hda_nid_t *capsrc_nids; @@ -217,6 +222,41 @@ static struct hda_pcm_stream conexant_pcm_digital_capture = { /* NID is set in alc_build_pcms */ }; +static int cx5051_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct conexant_spec *spec = codec->spec; + spec->cur_adc = spec->adc_nids[spec->cur_adc_idx]; + spec->cur_adc_stream_tag = stream_tag; + spec->cur_adc_format = format; + snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + return 0; +} + +static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct conexant_spec *spec = codec->spec; + snd_hda_codec_setup_stream(codec, spec->cur_adc, 0, 0, 0); + spec->cur_adc = 0; + return 0; +} + +static struct hda_pcm_stream cx5051_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0, /* fill later */ + .ops = { + .prepare = cx5051_capture_pcm_prepare, + .cleanup = cx5051_capture_pcm_cleanup + }, +}; + static int conexant_build_pcms(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -231,7 +271,12 @@ static int conexant_build_pcms(struct hda_codec *codec) spec->multiout.max_channels; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = conexant_pcm_analog_capture; + if (codec->vendor_id == 0x14f15051) + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + cx5051_pcm_analog_capture; + else + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + conexant_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; @@ -1421,10 +1466,260 @@ static int patch_cxt5047(struct hda_codec *codec) return 0; } +/* Conexant 5051 specific */ +static hda_nid_t cxt5051_dac_nids[1] = { 0x10 }; +static hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 }; +#define CXT5051_SPDIF_OUT 0x1C +#define CXT5051_PORTB_EVENT 0x38 +#define CXT5051_PORTC_EVENT 0x39 + +static struct hda_channel_mode cxt5051_modes[1] = { + { 2, NULL }, +}; + +static void cxt5051_update_speaker(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int pinctl; + pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); +} + +/* turn on/off EAPD (+ mute HP) as a master switch */ +static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + if (!cxt_eapd_put(kcontrol, ucontrol)) + return 0; + cxt5051_update_speaker(codec); + return 1; +} + +/* toggle input of built-in and mic jack appropriately */ +static void cxt5051_portb_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_CONNECT_SEL, + present ? 0x01 : 0x00); +} + +/* switch the current ADC according to the jack state */ +static void cxt5051_portc_automic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int present; + hda_nid_t new_adc; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE; + if (present) + spec->cur_adc_idx = 1; + else + spec->cur_adc_idx = 0; + new_adc = spec->adc_nids[spec->cur_adc_idx]; + if (spec->cur_adc && spec->cur_adc != new_adc) { + /* stream is running, let's swap the current ADC */ + snd_hda_codec_setup_stream(codec, spec->cur_adc, 0, 0, 0); + spec->cur_adc = new_adc; + snd_hda_codec_setup_stream(codec, new_adc, + spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); + } +} + +/* mute internal speaker if HP is plugged */ +static void cxt5051_hp_automute(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + + spec->hp_present = snd_hda_codec_read(codec, 0x16, 0, + AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE; + cxt5051_update_speaker(codec); +} + +/* unsolicited event for HP jack sensing */ +static void cxt5051_hp_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5051_hp_automute(codec); + break; + case CXT5051_PORTB_EVENT: + cxt5051_portb_automic(codec); + break; + case CXT5051_PORTC_EVENT: + cxt5051_portc_automic(codec); + break; + } +} + +static struct snd_kcontrol_new cxt5051_mixers[] = { + HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + +static struct snd_kcontrol_new cxt5051_hp_mixers[] = { + HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("External Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("External Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + +static struct hda_verb cxt5051_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, + { } /* end */ +}; + +/* initialize jack-sensing, too */ +static int cxt5051_init(struct hda_codec *codec) +{ + conexant_init(codec); + if (codec->patch_ops.unsol_event) { + cxt5051_hp_automute(codec); + cxt5051_portb_automic(codec); + cxt5051_portc_automic(codec); + } + return 0; +} + + +enum { + CXT5051_LAPTOP, /* Laptops w/ EAPD support */ + CXT5051_HP, /* no docking */ + CXT5051_MODELS +}; + +static const char *cxt5051_models[CXT5051_MODELS] = { + [CXT5051_LAPTOP] = "laptop", + [CXT5051_HP] = "hp", +}; + +static struct snd_pci_quirk cxt5051_cfg_tbl[] = { + SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", + CXT5051_LAPTOP), + SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), + {} +}; + +static int patch_cxt5051(struct hda_codec *codec) +{ + struct conexant_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + codec->patch_ops = conexant_patch_ops; + codec->patch_ops.init = cxt5051_init; + + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(cxt5051_dac_nids); + spec->multiout.dac_nids = cxt5051_dac_nids; + spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT; + spec->num_adc_nids = 1; /* not 2; via auto-mic switch */ + spec->adc_nids = cxt5051_adc_nids; + spec->num_mixers = 1; + spec->mixers[0] = cxt5051_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = cxt5051_init_verbs; + spec->spdif_route = 0; + spec->num_channel_mode = ARRAY_SIZE(cxt5051_modes); + spec->channel_mode = cxt5051_modes; + spec->cur_adc = 0; + spec->cur_adc_idx = 0; + + board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, + cxt5051_models, + cxt5051_cfg_tbl); + switch (board_config) { + case CXT5051_HP: + codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; + spec->mixers[0] = cxt5051_hp_mixers; + break; + default: + case CXT5051_LAPTOP: + codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; + break; + } + + return 0; +} + + +/* + */ + struct hda_codec_preset snd_hda_preset_conexant[] = { { .id = 0x14f15045, .name = "CX20549 (Venice)", .patch = patch_cxt5045 }, { .id = 0x14f15047, .name = "CX20551 (Waikiki)", .patch = patch_cxt5047 }, + { .id = 0x14f15051, .name = "CX20561 (Hermosa)", + .patch = patch_cxt5051 }, {} /* terminator */ }; -- cgit v0.10.2 From dafc83578d1633d7faf3e9de67fd922286c1b38d Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Fri, 25 Jan 2008 11:53:50 +0100 Subject: [ALSA] hda-codec - Add model for Acer Aspire 5310 Simplify usage of the Acer Aspire 5310 laptop with the ALC268 based codec sound card via add correct PCI SSID. Signed-off-by: Andy Shevchenko Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 99e2c66..4df4cb9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9925,6 +9925,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), + SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), -- cgit v0.10.2 From 628ed1333a9d2a7c7d3dc53641ea59af511e5fe5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Jan 2008 11:56:57 +0100 Subject: [ALSA] hda-codec - Add missing slave for AD1884 master switch The Speaker switch is missing in the slave list. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1295931..3a8f00e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -138,6 +138,7 @@ static const char *ad_slave_vols[] = { "Side Playback Volume", "Headphone Playback Volume", "Mono Playback Volume", + "Speaker Playback Volume", NULL }; @@ -149,6 +150,7 @@ static const char *ad_slave_sws[] = { "Side Playback Switch", "Headphone Playback Switch", "Mono Playback Switch", + "Speaker Playback Switch", NULL }; -- cgit v0.10.2 From b5895dc8b467a2457f4d33bef51f9879cf780a07 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 25 Jan 2008 15:24:50 +0100 Subject: [ALSA] hda: STAC92xx Line In/Mic as output check This patch checks to see the Line In/Mic port have the ability to do output before creating the the control switches. The 92hd71bxx series of codecs has this issue with the port 0xe, which only allows input. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3b8b860..34890c5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2226,7 +2226,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, int i, err; struct sigmatel_spec *spec = codec->spec; - unsigned int wid_caps; + unsigned int wid_caps, pincap; for (i = 0; i < cfg->line_outs; i++) { @@ -2262,13 +2262,31 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } - if (spec->line_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Line In as Output Switch", cfg->input_pins[AUTO_PIN_LINE] << 8)) < 0) - return err; + if (spec->line_switch) { + nid = cfg->input_pins[AUTO_PIN_LINE]; + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_IO_SWITCH, + "Line In as Output Switch", nid << 8); + if (err < 0) + return err; + } + } - if (spec->mic_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (cfg->input_pins[AUTO_PIN_MIC] << 8) | 1)) < 0) - return err; + if (spec->mic_switch) { + nid = cfg->input_pins[AUTO_PIN_MIC]; + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_IO_SWITCH, + "Mic as Output Switch", (nid << 8) | 1); + if (err < 0) + return err; + } + } return 0; } -- cgit v0.10.2 From 4806ef0cf481a4e41f5046aaf1032a7601049ab6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 26 Jan 2008 09:58:13 +0100 Subject: [ALSA] hda-codec - Add SPDIF controls as slave on AD codecs The AD codecs have hardware SPDIF volume/switch controls but they are not assigned to the slave list for virtual master controls. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3a8f00e..4914032 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -139,6 +139,7 @@ static const char *ad_slave_vols[] = { "Headphone Playback Volume", "Mono Playback Volume", "Speaker Playback Volume", + "IEC958 Playback Volume", NULL }; @@ -151,6 +152,7 @@ static const char *ad_slave_sws[] = { "Headphone Playback Switch", "Mono Playback Switch", "Speaker Playback Switch", + "IEC958 Playback Switch", NULL }; @@ -3078,6 +3080,7 @@ static const char *ad1884_slave_vols[] = { "Internal Mic Playback Volume", "Docking Mic Playback Volume" "Beep Playback Volume", + "IEC958 Playback Volume", NULL }; -- cgit v0.10.2 From 236c4920ea825acbffa7bbba24c182ec6a9a8245 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Jan 2008 08:32:58 +0100 Subject: [ALSA] oxygen: fix AK4396 double rate upper limit Fix the upper sample rate limit for the double rate mode of the AK4396 to the value from the datasheet. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 05ea2b6..51c4d1a 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -90,7 +90,7 @@ static void set_ak4396_params(struct oxygen *chip, value = data->ak4396_ctl2 & ~AK4396_DFS_MASK; if (params_rate(params) <= 54000) value |= AK4396_DFS_NORMAL; - else if (params_rate(params) < 120000) + else if (params_rate(params) <= 108000) value |= AK4396_DFS_DOUBLE; else value |= AK4396_DFS_QUAD; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index a0457a5..840e4a6 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -206,7 +206,7 @@ static void set_ak4396_params(struct oxygen *chip, value = data->ak4396_ctl2 & ~AK4396_DFS_MASK; if (params_rate(params) <= 54000) value |= AK4396_DFS_NORMAL; - else if (params_rate(params) < 120000) + else if (params_rate(params) <= 108000) value |= AK4396_DFS_DOUBLE; else value |= AK4396_DFS_QUAD; -- cgit v0.10.2 From 911b499af45e879ccf4b8db234278a7136d056c9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Jan 2008 08:33:44 +0100 Subject: [ALSA] oxygen: make line-in exclusive only on Xonar Move the line input switching code to the Virtuoso driver because only the Xonar cards bypass the analog mixer for line input. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 9ec9e6b..8fc9e7c 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -84,6 +84,8 @@ struct oxygen_model { struct snd_pcm_hw_params *params); void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); + void (*ac97_switch_hook)(struct oxygen *chip, + unsigned int reg, int mute); size_t model_data_size; u8 dac_channels; u8 used_channels; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 06394e4..d98867c 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -337,8 +337,6 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_write_ac97(chip, 0, AC97_REC_GAIN, 0x8000); oxygen_write_ac97(chip, 0, AC97_CENTER_LFE_MASTER, 0x8080); oxygen_write_ac97(chip, 0, AC97_SURROUND_MASTER, 0x8080); - oxygen_ac97_clear_bits(chip, 0, - CM9780_GPIO_STATUS, CM9780_GPO0); /* power down unused ADCs and DACs */ oxygen_ac97_set_bits(chip, 0, AC97_POWERDOWN, AC97_PD_PR0 | AC97_PD_PR1); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index fe53318..cf34b12 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -465,19 +465,6 @@ static int ac97_switch_get(struct snd_kcontrol *ctl, return 0; } -static void ac97_mute_ctl(struct oxygen *chip, unsigned int control) -{ - unsigned int index = chip->controls[control]->private_value & 0xff; - u16 value; - - value = oxygen_read_ac97(chip, 0, index); - if (!(value & 0x8000)) { - oxygen_write_ac97(chip, 0, index, value | 0x8000); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->controls[control]->id); - } -} - static int ac97_switch_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -498,22 +485,9 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, change = newreg != oldreg; if (change) { oxygen_write_ac97(chip, 0, index, newreg); - if (index == AC97_LINE) { - oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, - newreg & 0x8000 ? - CM9780_GPO0 : 0, CM9780_GPO0); - if (!(newreg & 0x8000)) { - ac97_mute_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH); - ac97_mute_ctl(chip, CONTROL_CD_CAPTURE_SWITCH); - ac97_mute_ctl(chip, CONTROL_AUX_CAPTURE_SWITCH); - } - } else if ((index == AC97_MIC || index == AC97_CD || - index == AC97_VIDEO || index == AC97_AUX) && - bitnr == 15 && !(newreg & 0x8000)) { - ac97_mute_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH); - oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, - CM9780_GPO0, CM9780_GPO0); - } + if (bitnr == 15 && chip->model->ac97_switch_hook) + chip->model->ac97_switch_hook(chip, index, + newreg & 0x8000); } mutex_unlock(&chip->mutex); return change; @@ -671,6 +645,7 @@ static const struct snd_kcontrol_new ac97_controls[] = { AC97_VOLUME("Mic Capture Volume", AC97_MIC), AC97_SWITCH("Mic Capture Switch", AC97_MIC, 15, 1), AC97_SWITCH("Mic Boost (+20dB)", AC97_MIC, 6, 0), + AC97_VOLUME("Line Capture Volume", AC97_LINE), AC97_SWITCH("Line Capture Switch", AC97_LINE, 15, 1), AC97_VOLUME("CD Capture Volume", AC97_CD), AC97_SWITCH("CD Capture Switch", AC97_CD, 15, 1), diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 665115d..23bfab4 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -80,6 +80,8 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_ALT 0x0080 #define GPIO_OUTPUT_ENABLE 0x0100 +#define GPIO_LINE_MUTE CM9780_GPO0 + /* register 16 */ #define PCM1796_ATL_MASK 0xff /* register 17 */ @@ -168,6 +170,7 @@ static void xonar_init(struct oxygen *chip) GPIO_CS5381_M_SINGLE, GPIO_CS5381_M_MASK | GPIO_ALT); oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); + oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS, GPIO_LINE_MUTE); msleep(300); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE); oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); @@ -231,6 +234,47 @@ static void set_cs5381_params(struct oxygen *chip, value, GPIO_CS5381_M_MASK); } +static void mute_ac97_ctl(struct oxygen *chip, unsigned int control) +{ + unsigned int index = chip->controls[control]->private_value & 0xff; + u16 value; + + value = oxygen_read_ac97(chip, 0, index); + if (!(value & 0x8000)) { + oxygen_write_ac97(chip, 0, index, value | 0x8000); + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->controls[control]->id); + } +} + +static void xonar_ac97_switch_hook(struct oxygen *chip, + unsigned int reg, int mute) +{ + /* line-in is exclusive */ + switch (reg) { + case AC97_LINE: + oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS, + mute ? GPIO_LINE_MUTE : 0, + GPIO_LINE_MUTE); + if (!mute) { + mute_ac97_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH); + mute_ac97_ctl(chip, CONTROL_CD_CAPTURE_SWITCH); + mute_ac97_ctl(chip, CONTROL_AUX_CAPTURE_SWITCH); + } + break; + case AC97_MIC: + case AC97_CD: + case AC97_VIDEO: + case AC97_AUX: + if (!mute) { + oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_STATUS, + GPIO_LINE_MUTE); + mute_ac97_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH); + } + break; + } +} + static int pcm1796_volume_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -288,7 +332,10 @@ static int xonar_control_filter(struct snd_kcontrol_new *template) template->info = pcm1796_volume_info, template->tlv.p = pcm1796_db_scale; } else if (!strncmp(template->name, "CD Capture ", 11)) { + /* CD in is actually connected to the video in pin */ template->private_value ^= AC97_CD ^ AC97_VIDEO; + } else if (!strcmp(template->name, "Line Capture Volume")) { + return 1; /* line-in bypasses the AC'97 mixer */ } return 0; } @@ -310,6 +357,7 @@ static const struct oxygen_model model_xonar = { .set_adc_params = set_cs5381_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, + .ac97_switch_hook = xonar_ac97_switch_hook, .dac_channels = 8, .used_channels = OXYGEN_CHANNEL_B | OXYGEN_CHANNEL_C | -- cgit v0.10.2 From 1e821dd2763c97df1a0a451e553d218cb8886cd7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Jan 2008 08:34:21 +0100 Subject: [ALSA] oxygen: use AC97 interrupt After an AC97 register read or write, use the AC97 interrupt instead of polling to wait for the access to be completed. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 8fc9e7c..55ce244 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -3,6 +3,7 @@ #include #include +#include #include #include "oxygen_regs.h" @@ -65,6 +66,7 @@ struct oxygen { struct snd_pcm_substream *streams[PCM_COUNT]; struct snd_kcontrol *controls[CONTROL_COUNT]; struct work_struct spdif_input_bits_work; + wait_queue_head_t ac97_waitqueue; }; struct oxygen_model { diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index d0cdce0..74e23ef 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -85,14 +85,22 @@ EXPORT_SYMBOL(oxygen_write32_masked); static int oxygen_ac97_wait(struct oxygen *chip, unsigned int mask) { - unsigned long timeout = jiffies + msecs_to_jiffies(1); - do { - udelay(5); - cond_resched(); - if (oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS) & mask) - return 0; - } while (time_after_eq(timeout, jiffies)); - return -EIO; + u8 status = 0; + + /* + * Reading the status register also clears the bits, so we have to save + * the read bits in status. + */ + wait_event_timeout(chip->ac97_waitqueue, + ({ status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS); + status & mask; }), + msecs_to_jiffies(1) + 1); + /* + * Check even after a timeout because this function should not require + * the AC'97 interrupt to be enabled. + */ + status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS); + return status & mask ? 0 : -EIO; } /* diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index d98867c..de6bf41 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -53,7 +53,8 @@ static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) OXYGEN_CHANNEL_MULTICH | OXYGEN_CHANNEL_AC97 | OXYGEN_INT_SPDIF_IN_DETECT | - OXYGEN_INT_GPIO); + OXYGEN_INT_GPIO | + OXYGEN_INT_AC97); if (clear) { if (clear & OXYGEN_INT_SPDIF_IN_DETECT) chip->interrupt_mask &= ~OXYGEN_INT_SPDIF_IN_DETECT; @@ -89,6 +90,9 @@ static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) if ((status & OXYGEN_INT_MIDI) && chip->midi) snd_mpu401_uart_interrupt(0, chip->midi->private_data); + if (status & OXYGEN_INT_AC97) + wake_up(&chip->ac97_waitqueue); + return IRQ_HANDLED; } @@ -306,7 +310,9 @@ static void __devinit oxygen_init(struct oxygen *chip) (2 << OXYGEN_A_MONITOR_ROUTE_2_SHIFT) | (3 << OXYGEN_A_MONITOR_ROUTE_3_SHIFT)); - oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0); + oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, + OXYGEN_AC97_INT_READ_DONE | + OXYGEN_AC97_INT_WRITE_DONE); oxygen_write32(chip, OXYGEN_AC97_OUT_CONFIG, 0); oxygen_write32(chip, OXYGEN_AC97_IN_CONFIG, 0); if (!(chip->has_ac97_0 | chip->has_ac97_1)) @@ -408,6 +414,7 @@ int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, mutex_init(&chip->mutex); INIT_WORK(&chip->spdif_input_bits_work, oxygen_spdif_input_bits_changed); + init_waitqueue_head(&chip->ac97_waitqueue); err = pci_enable_device(pci); if (err < 0) @@ -471,7 +478,7 @@ int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, oxygen_proc_init(chip); spin_lock_irq(&chip->reg_lock); - chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT; + chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT | OXYGEN_INT_AC97; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); spin_unlock_irq(&chip->reg_lock); -- cgit v0.10.2 From a3601560496d7b46d2d1187169824d11570ff63a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Jan 2008 08:35:20 +0100 Subject: [ALSA] oxygen: add front panel controls Add mixer controls for the front panel AC97 codec. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 55ce244..e71c534 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -86,7 +86,7 @@ struct oxygen_model { struct snd_pcm_hw_params *params); void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); - void (*ac97_switch_hook)(struct oxygen *chip, + void (*ac97_switch_hook)(struct oxygen *chip, unsigned int codec, unsigned int reg, int mute); size_t model_data_size; u8 dac_channels; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index de6bf41..540e56b 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -365,8 +365,7 @@ static void __devinit oxygen_init(struct oxygen *chip) oxygen_write_ac97(chip, 1, AC97_AUX, 0x8808); oxygen_write_ac97(chip, 1, AC97_PCM, 0x0808); oxygen_write_ac97(chip, 1, AC97_REC_SEL, 0x0000); - oxygen_write_ac97(chip, 1, AC97_REC_GAIN, 0x8000); - oxygen_ac97_clear_bits(chip, 1, AC97_REC_GAIN, 0x1c00); + oxygen_write_ac97(chip, 1, AC97_REC_GAIN, 0x0000); oxygen_ac97_set_bits(chip, 1, 0x6a, 0x0040); } } diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index cf34b12..a8e4623 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -450,13 +450,14 @@ static int ac97_switch_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; + unsigned int codec = (ctl->private_value >> 24) & 1; unsigned int index = ctl->private_value & 0xff; unsigned int bitnr = (ctl->private_value >> 8) & 0xff; int invert = ctl->private_value & (1 << 16); u16 reg; mutex_lock(&chip->mutex); - reg = oxygen_read_ac97(chip, 0, index); + reg = oxygen_read_ac97(chip, codec, index); mutex_unlock(&chip->mutex); if (!(reg & (1 << bitnr)) ^ !invert) value->value.integer.value[0] = 1; @@ -469,6 +470,7 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; + unsigned int codec = (ctl->private_value >> 24) & 1; unsigned int index = ctl->private_value & 0xff; unsigned int bitnr = (ctl->private_value >> 8) & 0xff; int invert = ctl->private_value & (1 << 16); @@ -476,7 +478,7 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, int change; mutex_lock(&chip->mutex); - oldreg = oxygen_read_ac97(chip, 0, index); + oldreg = oxygen_read_ac97(chip, codec, index); newreg = oldreg; if (!value->value.integer.value[0] ^ !invert) newreg |= 1 << bitnr; @@ -484,9 +486,9 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, newreg &= ~(1 << bitnr); change = newreg != oldreg; if (change) { - oxygen_write_ac97(chip, 0, index, newreg); + oxygen_write_ac97(chip, codec, index, newreg); if (bitnr == 15 && chip->model->ac97_switch_hook) - chip->model->ac97_switch_hook(chip, index, + chip->model->ac97_switch_hook(chip, codec, index, newreg & 0x8000); } mutex_unlock(&chip->mutex); @@ -507,11 +509,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; - unsigned int index = ctl->private_value; + unsigned int codec = (ctl->private_value >> 24) & 1; + unsigned int index = ctl->private_value & 0xff; u16 reg; mutex_lock(&chip->mutex); - reg = oxygen_read_ac97(chip, 0, index); + reg = oxygen_read_ac97(chip, codec, index); mutex_unlock(&chip->mutex); value->value.integer.value[0] = 31 - (reg & 0x1f); value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); @@ -522,12 +525,13 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; - unsigned int index = ctl->private_value; + unsigned int codec = (ctl->private_value >> 24) & 1; + unsigned int index = ctl->private_value & 0xff; u16 oldreg, newreg; int change; mutex_lock(&chip->mutex); - oldreg = oxygen_read_ac97(chip, 0, index); + oldreg = oxygen_read_ac97(chip, codec, index); newreg = oldreg; newreg = (newreg & ~0x1f) | (31 - (value->value.integer.value[0] & 0x1f)); @@ -535,30 +539,77 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, ((31 - (value->value.integer.value[0] & 0x1f)) << 8); change = newreg != oldreg; if (change) - oxygen_write_ac97(chip, 0, index, newreg); + oxygen_write_ac97(chip, codec, index, newreg); mutex_unlock(&chip->mutex); return change; } -#define AC97_SWITCH(xname, index, bitnr, invert) { \ +static int ac97_fp_rec_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0; + info->value.integer.max = 7; + return 0; +} + +static int ac97_fp_rec_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 reg; + + mutex_lock(&chip->mutex); + reg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN); + mutex_unlock(&chip->mutex); + value->value.integer.value[0] = reg & 7; + value->value.integer.value[1] = (reg >> 8) & 7; + return 0; +} + +static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 oldreg, newreg; + int change; + + mutex_lock(&chip->mutex); + oldreg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN); + newreg = oldreg & ~0x0707; + newreg = newreg | (value->value.integer.value[0] & 7); + newreg = newreg | ((value->value.integer.value[0] & 7) << 8); + change = newreg != oldreg; + if (change) + oxygen_write_ac97(chip, 1, AC97_REC_GAIN, newreg); + mutex_unlock(&chip->mutex); + return change; +} + +#define AC97_SWITCH(xname, codec, index, bitnr, invert) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .info = snd_ctl_boolean_mono_info, \ .get = ac97_switch_get, \ .put = ac97_switch_put, \ - .private_value = ((invert) << 16) | ((bitnr) << 8) | (index), \ + .private_value = ((codec) << 24) | ((invert) << 16) | \ + ((bitnr) << 8) | (index), \ } -#define AC97_VOLUME(xname, index) { \ +#define AC97_VOLUME(xname, codec, index) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = ac97_volume_info, \ .get = ac97_volume_get, \ .put = ac97_volume_put, \ .tlv = { .p = ac97_db_scale, }, \ - .private_value = (index), \ + .private_value = ((codec) << 24) | (index), \ } static DECLARE_TLV_DB_SCALE(ac97_db_scale, -3450, 150, 0); +static DECLARE_TLV_DB_SCALE(ac97_rec_db_scale, 0, 150, 0); static const struct snd_kcontrol_new controls[] = { { @@ -642,15 +693,31 @@ static const struct snd_kcontrol_new controls[] = { }; static const struct snd_kcontrol_new ac97_controls[] = { - AC97_VOLUME("Mic Capture Volume", AC97_MIC), - AC97_SWITCH("Mic Capture Switch", AC97_MIC, 15, 1), - AC97_SWITCH("Mic Boost (+20dB)", AC97_MIC, 6, 0), - AC97_VOLUME("Line Capture Volume", AC97_LINE), - AC97_SWITCH("Line Capture Switch", AC97_LINE, 15, 1), - AC97_VOLUME("CD Capture Volume", AC97_CD), - AC97_SWITCH("CD Capture Switch", AC97_CD, 15, 1), - AC97_VOLUME("Aux Capture Volume", AC97_AUX), - AC97_SWITCH("Aux Capture Switch", AC97_AUX, 15, 1), + AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC), + AC97_SWITCH("Mic Capture Switch", 0, AC97_MIC, 15, 1), + AC97_SWITCH("Mic Boost (+20dB)", 0, AC97_MIC, 6, 0), + AC97_VOLUME("Line Capture Volume", 0, AC97_LINE), + AC97_SWITCH("Line Capture Switch", 0, AC97_LINE, 15, 1), + AC97_VOLUME("CD Capture Volume", 0, AC97_CD), + AC97_SWITCH("CD Capture Switch", 0, AC97_CD, 15, 1), + AC97_VOLUME("Aux Capture Volume", 0, AC97_AUX), + AC97_SWITCH("Aux Capture Switch", 0, AC97_AUX, 15, 1), +}; + +static const struct snd_kcontrol_new ac97_fp_controls[] = { + AC97_VOLUME("Front Panel Playback Volume", 1, AC97_HEADPHONE), + AC97_SWITCH("Front Panel Playback Switch", 1, AC97_HEADPHONE, 15, 1), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Front Panel Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = ac97_fp_rec_volume_info, + .get = ac97_fp_rec_volume_get, + .put = ac97_fp_rec_volume_put, + .tlv = { .p = ac97_rec_db_scale, }, + }, + AC97_SWITCH("Front Panel Capture Switch", 1, AC97_REC_GAIN, 15, 1), }; static void oxygen_any_ctl_free(struct snd_kcontrol *ctl) @@ -717,5 +784,11 @@ int oxygen_mixer_init(struct oxygen *chip) if (err < 0) return err; } + if (chip->has_ac97_1) { + err = add_controls(chip, ac97_fp_controls, + ARRAY_SIZE(ac97_fp_controls)); + if (err < 0) + return err; + } return chip->model->mixer_init ? chip->model->mixer_init(chip) : 0; } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 23bfab4..2e1a699 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -247,9 +247,11 @@ static void mute_ac97_ctl(struct oxygen *chip, unsigned int control) } } -static void xonar_ac97_switch_hook(struct oxygen *chip, +static void xonar_ac97_switch_hook(struct oxygen *chip, unsigned int codec, unsigned int reg, int mute) { + if (codec != 0) + return; /* line-in is exclusive */ switch (reg) { case AC97_LINE: -- cgit v0.10.2 From 5f7b9b457751efc9f3ad120d0ebdb19fe753e9d0 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Jan 2008 08:35:47 +0100 Subject: [ALSA] oxygen: add front panel capture When a second AC97 codec is present, add a PCM device for capturing from the front panel. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 5b89c72..dfad3db 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -118,7 +118,11 @@ static int oxygen_open(struct snd_pcm_substream *substream, int err; runtime->private_data = (void *)(uintptr_t)channel; - runtime->hw = *oxygen_hardware[channel]; + if (channel == PCM_B && chip->has_ac97_1 && + (chip->model->used_channels & OXYGEN_CHANNEL_AC97)) + runtime->hw = oxygen_ac97_hardware; + else + runtime->hw = *oxygen_hardware[channel]; switch (channel) { case PCM_C: runtime->hw.rates &= ~(SNDRV_PCM_RATE_32000 | @@ -353,30 +357,37 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct oxygen *chip = snd_pcm_substream_chip(substream); + int is_ac97; int err; err = oxygen_hw_params(substream, hw_params); if (err < 0) return err; + is_ac97 = chip->has_ac97_1 && + (chip->model->used_channels & OXYGEN_CHANNEL_AC97); + spin_lock_irq(&chip->reg_lock); oxygen_write8_masked(chip, OXYGEN_REC_FORMAT, oxygen_format(hw_params) << OXYGEN_REC_FORMAT_B_SHIFT, OXYGEN_REC_FORMAT_B_MASK); - oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT, - oxygen_rate(hw_params) | - oxygen_i2s_mclk(hw_params) | - chip->model->adc_i2s_format | - oxygen_i2s_bits(hw_params), - OXYGEN_I2S_RATE_MASK | - OXYGEN_I2S_FORMAT_MASK | - OXYGEN_I2S_MCLK_MASK | - OXYGEN_I2S_BITS_MASK); + if (!is_ac97) + oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT, + oxygen_rate(hw_params) | + oxygen_i2s_mclk(hw_params) | + chip->model->adc_i2s_format | + oxygen_i2s_bits(hw_params), + OXYGEN_I2S_RATE_MASK | + OXYGEN_I2S_FORMAT_MASK | + OXYGEN_I2S_MCLK_MASK | + OXYGEN_I2S_BITS_MASK); spin_unlock_irq(&chip->reg_lock); - mutex_lock(&chip->mutex); - chip->model->set_adc_params(chip, hw_params); - mutex_unlock(&chip->mutex); + if (!is_ac97) { + mutex_lock(&chip->mutex); + chip->model->set_adc_params(chip, hw_params); + mutex_unlock(&chip->mutex); + } return 0; } @@ -677,23 +688,28 @@ int __devinit oxygen_pcm_init(struct oxygen *chip) outs = chip->has_ac97_1 && (chip->model->used_channels & OXYGEN_CHANNEL_AC97); - ins = (chip->model->used_channels & (OXYGEN_CHANNEL_A | - OXYGEN_CHANNEL_B)) + ins = outs || + (chip->model->used_channels & (OXYGEN_CHANNEL_A | + OXYGEN_CHANNEL_B)) == (OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_B); if (outs | ins) { - err = snd_pcm_new(chip->card, ins ? "Analog2" : "AC97", + err = snd_pcm_new(chip->card, outs ? "AC97" : "Analog2", 2, outs, ins, &pcm); if (err < 0) return err; - if (outs) + if (outs) { snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &oxygen_ac97_ops); + oxygen_write8_masked(chip, OXYGEN_REC_ROUTING, + OXYGEN_REC_B_ROUTE_AC97_1, + OXYGEN_REC_B_ROUTE_MASK); + } if (ins) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &oxygen_rec_b_ops); pcm->private_data = chip; pcm->private_free = oxygen_pcm_free; - strcpy(pcm->name, ins ? "Analog 2" : "Front Panel"); + strcpy(pcm->name, outs ? "Front Panel" : "Analog 2"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 128 * 1024, 256 * 1024); -- cgit v0.10.2 From 7c0141591fcf92ddc96a4ee04e35783a15bd68c8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Jan 2008 08:36:55 +0100 Subject: [ALSA] virtuoso: monitor external power on D2X On the Xonar D2X, monitor the GPIO pin that indicates whether external power is present. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index e71c534..ad50fb8 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -66,6 +66,7 @@ struct oxygen { struct snd_pcm_substream *streams[PCM_COUNT]; struct snd_kcontrol *controls[CONTROL_COUNT]; struct work_struct spdif_input_bits_work; + struct work_struct gpio_work; wait_queue_head_t ac97_waitqueue; }; @@ -88,6 +89,7 @@ struct oxygen_model { void (*update_dac_mute)(struct oxygen *chip); void (*ac97_switch_hook)(struct oxygen *chip, unsigned int codec, unsigned int reg, int mute); + void (*gpio_changed)(struct oxygen *chip); size_t model_data_size; u8 dac_channels; u8 used_channels; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 540e56b..6eb36dd 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -85,7 +85,7 @@ static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) } if (status & OXYGEN_INT_GPIO) - ; + schedule_work(&chip->gpio_work); if ((status & OXYGEN_INT_MIDI) && chip->midi) snd_mpu401_uart_interrupt(0, chip->midi->private_data); @@ -157,6 +157,14 @@ static void oxygen_spdif_input_bits_changed(struct work_struct *work) } } +static void oxygen_gpio_changed(struct work_struct *work) +{ + struct oxygen *chip = container_of(work, struct oxygen, gpio_work); + + if (chip->model->gpio_changed) + chip->model->gpio_changed(chip); +} + #ifdef CONFIG_PROC_FS static void oxygen_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) @@ -413,6 +421,7 @@ int __devinit oxygen_pci_probe(struct pci_dev *pci, int index, char *id, mutex_init(&chip->mutex); INIT_WORK(&chip->spdif_input_bits_work, oxygen_spdif_input_bits_changed); + INIT_WORK(&chip->gpio_work, oxygen_gpio_changed); init_waitqueue_head(&chip->ac97_waitqueue); err = pci_enable_device(pci); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 2e1a699..40e92f5 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -136,6 +136,11 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); /* register 23 */ #define PCM1796_ID_MASK 0x1f +struct xonar_data { + u8 is_d2x; + u8 has_power; +}; + static void pcm1796_write(struct oxygen *chip, unsigned int codec, u8 reg, u8 value) { @@ -153,8 +158,11 @@ static void pcm1796_write(struct oxygen *chip, unsigned int codec, static void xonar_init(struct oxygen *chip) { + struct xonar_data *data = chip->model_data; unsigned int i; + data->is_d2x = chip->pci->subsystem_device == 0x82b7; + for (i = 0; i < 4; ++i) { pcm1796_write(chip, i, 18, PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); @@ -169,6 +177,15 @@ static void xonar_init(struct oxygen *chip) oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, GPIO_CS5381_M_SINGLE, GPIO_CS5381_M_MASK | GPIO_ALT); + if (data->is_d2x) { + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_EXT_POWER); + oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, + GPIO_EXT_POWER); + chip->interrupt_mask |= OXYGEN_INT_GPIO; + data->has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) + & GPIO_EXT_POWER); + } oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS, GPIO_LINE_MUTE); msleep(300); @@ -234,6 +251,27 @@ static void set_cs5381_params(struct oxygen *chip, value, GPIO_CS5381_M_MASK); } +static void xonar_gpio_changed(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + u8 has_power; + + if (!data->is_d2x) + return; + has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) + & GPIO_EXT_POWER); + if (has_power != data->has_power) { + data->has_power = has_power; + if (has_power) { + snd_printk(KERN_NOTICE "power restored\n"); + } else { + snd_printk(KERN_CRIT + "Hey! Don't unplug the power cable!\n"); + /* TODO: stop PCMs */ + } + } +} + static void mute_ac97_ctl(struct oxygen *chip, unsigned int control) { unsigned int index = chip->controls[control]->private_value & 0xff; @@ -360,6 +398,8 @@ static const struct oxygen_model model_xonar = { .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, .ac97_switch_hook = xonar_ac97_switch_hook, + .gpio_changed = xonar_gpio_changed, + .model_data_size = sizeof(struct xonar_data), .dac_channels = 8, .used_channels = OXYGEN_CHANNEL_B | OXYGEN_CHANNEL_C | -- cgit v0.10.2 From 4444704cdcd44bb3534ead01d3318f0db858cb9f Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Mon, 28 Jan 2008 12:28:24 +0100 Subject: [ALSA] hda-codec - Add model for Gigabyte P35 DS3R Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4df4cb9..3dd7ac3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5884,6 +5884,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), -- cgit v0.10.2 From 8c0d9649babcf7379dc94cb148bbd2a6d8792984 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Jan 2008 12:30:17 +0100 Subject: [ALSA] hda-codec - Add SPDIF output support to AD1986a laptop-eapd model The SPDIF output on AD1986A laptop-eapd model is disabled although some Samsung laptops have SPDIF output. Enable it after checking the pin default config. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4914032..845e949 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -925,6 +925,13 @@ static struct hda_amp_list ad1986a_loopbacks[] = { }; #endif +static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + return get_defcfg_connect(conf) != AC_JACK_PORT_NONE; +} + static int patch_ad1986a(struct hda_codec *codec) { struct ad198x_spec *spec; @@ -984,7 +991,8 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - spec->multiout.dig_out_nid = 0; + if (!is_jack_available(codec, 0x25)) + spec->multiout.dig_out_nid = 0; spec->input_mux = &ad1986a_laptop_eapd_capture_source; break; case AD1986A_LAPTOP_AUTOMUTE: @@ -995,7 +1003,8 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - spec->multiout.dig_out_nid = 0; + if (!is_jack_available(codec, 0x25)) + spec->multiout.dig_out_nid = 0; spec->input_mux = &ad1986a_laptop_eapd_capture_source; codec->patch_ops.unsol_event = ad1986a_hp_unsol_event; codec->patch_ops.init = ad1986a_hp_init; -- cgit v0.10.2 From 86cd92983a78d20d16ed77754829b5ee00c78be6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Jan 2008 18:09:56 +0100 Subject: [ALSA] hda-codec - Rename non-standard 'iSpeaker' Renamed the non-standard mixer elements 'iSpeaker' to 'Speaker' in Realtek codecs. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3dd7ac3..79eb0c1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1478,7 +1478,6 @@ static const char *alc_slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", "Mono Playback Volume", - "iSpeaker Playback Volume", "Line-Out Playback Volume", NULL, }; @@ -1492,7 +1491,6 @@ static const char *alc_slave_sws[] = { "Headphone Playback Switch", "Speaker Playback Switch", "Mono Playback Switch", - "iSpeaker Playback Switch", NULL, }; @@ -3837,8 +3835,8 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("iSpeaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("iSpeaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -6679,8 +6677,8 @@ static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = { static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), @@ -8110,8 +8108,8 @@ static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { HDA_BIND_SW("PCM Playback Switch",&alc262_hp_t5735_bind_front_sw), HDA_CODEC_VOLUME("LineOut Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("LineOut Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -8151,8 +8149,8 @@ static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = { HDA_BIND_SW("PCM Playback Switch", &alc262_hp_rp5700_bind_front_sw), HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), { } /* end */ @@ -12795,8 +12793,8 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), - HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("iSpeaker Playback Switch", 0x03, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), @@ -12806,7 +12804,7 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { }; static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("LineOut Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -12830,7 +12828,7 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), -- cgit v0.10.2 From 4bb261302b34998e8c3c8d8f0493ee9e58d57cb2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Jan 2008 18:12:42 +0100 Subject: [ALSA] hda-codec - Fix mixer controls with ALC262 HP T5735 model The PCM mixer elements in HP T5735 model of ALC262 codec conflict with Speaker and Headphone volumes. They should be removed. Ditto for LineOut that is identical with Speaker. Also, fixed/cleaned up the auto-mute callback to use the amp cache correctly. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79eb0c1..e484f6092 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8051,43 +8051,20 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { { } /* end */ }; -static struct hda_bind_ctls alc262_hp_t5735_bind_front_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static struct hda_bind_ctls alc262_hp_t5735_bind_front_sw = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - 0 - }, -}; - /* mute/unmute internal speaker according to the hp jack and mute state */ static void alc262_hp_t5735_automute(struct hda_codec *codec, int force) { struct alc_spec *spec = codec->spec; - unsigned int mute; if (force || !spec->sense_updated) { unsigned int present; present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; spec->sense_updated = 1; } - if (spec->jack_present) - mute = (0x7080 | ((0)<<8)); /* mute internal speaker */ - else /* unmute internal speaker if necessary */ - mute = (0x7000 | ((0)<<8)); - snd_hda_codec_write(codec, 0x0c, 0, - AC_VERB_SET_AMP_GAIN_MUTE, mute ); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, HDA_AMP_MUTE, + spec->jack_present ? HDA_AMP_MUTE : 0); } static void alc262_hp_t5735_unsol_event(struct hda_codec *codec, @@ -8104,10 +8081,6 @@ static void alc262_hp_t5735_init_hook(struct hda_codec *codec) } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { - HDA_BIND_VOL("PCM Playback Volume", &alc262_hp_t5735_bind_front_vol), - HDA_BIND_SW("PCM Playback Switch",&alc262_hp_t5735_bind_front_sw), - HDA_CODEC_VOLUME("LineOut Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("LineOut Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), -- cgit v0.10.2 From f2f48e1859167c936af957a828da1e9ec76848cd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Jan 2008 18:14:43 +0100 Subject: [ALSA] hda-codec - Fix ALC262 HP-RP5700 model Removed the PCM mixer elements conflicting with others. Also renamed Master control to Headphone, which isn't a real master. (The Master control is still created as a virtual master even after this rename.) Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e484f6092..dca8ba0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8099,29 +8099,9 @@ static struct hda_verb alc262_hp_t5735_verbs[] = { { } }; -static struct hda_bind_ctls alc262_hp_rp5700_bind_front_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x0e, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static struct hda_bind_ctls alc262_hp_rp5700_bind_front_sw = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT), - 0 - }, -}; - static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = { - HDA_BIND_VOL("PCM Playback Volume", &alc262_hp_rp5700_bind_front_vol), - HDA_BIND_SW("PCM Playback Switch", &alc262_hp_rp5700_bind_front_sw), - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), -- cgit v0.10.2 From bec15c3a5a1814019424228cd8127e9c82965ae1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Jan 2008 18:16:30 +0100 Subject: [ALSA] hda-codec - Add speaker automute to ALC260 HP models Added the speaker-automute function to ALC260 HP models. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dca8ba0..764b32b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -266,6 +266,7 @@ struct alc_spec { /* for pin sensing */ unsigned int sense_updated: 1; unsigned int jack_present: 1; + unsigned int master_sw: 1; /* for virtual master */ hda_nid_t vmaster_nid; @@ -3828,7 +3829,101 @@ static struct snd_kcontrol_new alc260_pc_beep_mixer[] = { { } /* end */ }; +/* update HP, line and mono out pins according to the master switch */ +static void alc260_hp_master_update(struct hda_codec *codec, + hda_nid_t hp, hda_nid_t line, + hda_nid_t mono) +{ + struct alc_spec *spec = codec->spec; + unsigned int val = spec->master_sw ? PIN_HP : 0; + /* change HP and line-out pins */ + snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + val); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + val); + /* mono (speaker) depending on the HP jack sense */ + val = (val && !spec->jack_present) ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + val); +} + +static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + *ucontrol->value.integer.value = spec->master_sw; + return 0; +} + +static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int val = !!*ucontrol->value.integer.value; + hda_nid_t hp, line, mono; + + if (val == spec->master_sw) + return 0; + spec->master_sw = val; + hp = (kcontrol->private_value >> 16) & 0xff; + line = (kcontrol->private_value >> 8) & 0xff; + mono = kcontrol->private_value & 0xff; + alc260_hp_master_update(codec, hp, line, mono); + return 1; +} + +static struct snd_kcontrol_new alc260_hp_output_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = alc260_hp_master_sw_get, + .put = alc260_hp_master_sw_put, + .private_value = (0x0f << 16) | (0x10 << 8) | 0x11 + }, + HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, + HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc260_hp_unsol_verbs[] = { + {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {}, +}; + +static void alc260_hp_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x10, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + alc260_hp_master_update(codec, 0x0f, 0x10, 0x11); +} + +static void alc260_hp_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc260_hp_automute(codec); +} + static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = alc260_hp_master_sw_get, + .put = alc260_hp_master_sw_put, + .private_value = (0x10 << 16) | (0x15 << 8) | 0x11 + }, HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT), @@ -3840,6 +3935,29 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { { } /* end */ }; +static struct hda_verb alc260_hp_3013_unsol_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {}, +}; + +static void alc260_hp_3013_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + alc260_hp_master_update(codec, 0x10, 0x15, 0x11); +} + +static void alc260_hp_3013_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc260_hp_3013_automute(codec); +} + /* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. */ @@ -4897,10 +5015,11 @@ static struct alc_config_preset alc260_presets[] = { .input_mux = &alc260_capture_source, }, [ALC260_HP] = { - .mixers = { alc260_base_output_mixer, + .mixers = { alc260_hp_output_mixer, alc260_input_mixer, alc260_capture_alt_mixer }, - .init_verbs = { alc260_init_verbs }, + .init_verbs = { alc260_init_verbs, + alc260_hp_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), @@ -4908,12 +5027,15 @@ static struct alc_config_preset alc260_presets[] = { .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, + .unsol_event = alc260_hp_unsol_event, + .init_hook = alc260_hp_automute, }, [ALC260_HP_3013] = { .mixers = { alc260_hp_3013_mixer, alc260_input_mixer, alc260_capture_alt_mixer }, - .init_verbs = { alc260_hp_3013_init_verbs }, + .init_verbs = { alc260_hp_3013_init_verbs, + alc260_hp_3013_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), @@ -4921,6 +5043,8 @@ static struct alc_config_preset alc260_presets[] = { .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, + .unsol_event = alc260_hp_3013_unsol_event, + .init_hook = alc260_hp_3013_automute, }, [ALC260_FUJITSU_S702X] = { .mixers = { alc260_fujitsu_mixer, -- cgit v0.10.2 From ce875f079efcfdcf693de89c7ab0ca7f71a9bdce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Jan 2008 18:17:43 +0100 Subject: [ALSA] hda-codec - Add speaker automute to ALC262 HP models Added the speaker-automute function to ALC262 HP models. Also, 'Mono' mixer elements are renamed as more intuitive 'Speaker'. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 764b32b..8621c9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8125,13 +8125,99 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = { { } /* end */ }; +/* update HP, line and mono-out pins according to the master switch */ +static void alc262_hp_master_update(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int val = spec->master_sw; + + /* HP & line-out */ + snd_hda_codec_write_cache(codec, 0x1b, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + val ? PIN_HP : 0); + snd_hda_codec_write_cache(codec, 0x15, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + val ? PIN_HP : 0); + /* mono (speaker) depending on the HP jack sense */ + val = val && !spec->jack_present; + snd_hda_codec_write_cache(codec, 0x16, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + val ? PIN_OUT : 0); +} + +static void alc262_hp_bpc_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int presence; + presence = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE); + alc262_hp_master_update(codec); +} + +static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_hp_bpc_automute(codec); +} + +static void alc262_hp_wildwest_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int presence; + presence = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE); + alc262_hp_master_update(codec); +} + +static void alc262_hp_wildwest_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_hp_wildwest_automute(codec); +} + +static int alc262_hp_master_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + *ucontrol->value.integer.value = spec->master_sw; + return 0; +} + +static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int val = !!*ucontrol->value.integer.value; + + if (val == spec->master_sw) + return 0; + spec->master_sw = val; + alc262_hp_master_update(codec); + return 1; +} + static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = alc262_hp_master_sw_get, + .put = alc262_hp_master_sw_put, + }, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), - + HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0, + HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0, + HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), @@ -8150,12 +8236,21 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { }; static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = alc262_hp_master_sw_get, + .put = alc262_hp_master_sw_put, + }, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0, + HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0, + HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x1a, 0, HDA_INPUT), @@ -8880,7 +8975,7 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = { {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -8922,6 +9017,8 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } }; @@ -9016,6 +9113,8 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = { /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } }; @@ -9200,6 +9299,8 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_HP_capture_source, + .unsol_event = alc262_hp_bpc_unsol_event, + .init_hook = alc262_hp_bpc_automute, }, [ALC262_HP_BPC_D7000_WF] = { .mixers = { alc262_HP_BPC_WildWest_mixer }, @@ -9210,6 +9311,8 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_HP_D7000_capture_source, + .unsol_event = alc262_hp_wildwest_unsol_event, + .init_hook = alc262_hp_wildwest_automute, }, [ALC262_HP_BPC_D7000_WL] = { .mixers = { alc262_HP_BPC_WildWest_mixer, @@ -9221,6 +9324,8 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_HP_D7000_capture_source, + .unsol_event = alc262_hp_wildwest_unsol_event, + .init_hook = alc262_hp_wildwest_automute, }, [ALC262_HP_TC_T5735] = { .mixers = { alc262_hp_t5735_mixer }, -- cgit v0.10.2 From 4939c660341d1439b8c70c3e006e7fc182f5111e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Jan 2008 23:53:41 +0100 Subject: [ALSA] Fix Oops with PCM OSS sync The PCM OSS emulation can cause Oops at sync operation due to the wrong data size calculation. Typically happening on Sparc64: http://lkml.org/lkml/2008/1/24/426 Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 092c2d8..4c601b1 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1621,6 +1621,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) snd_pcm_format_set_silence(runtime->format, runtime->oss.buffer, size1); + size1 /= runtime->channels; /* frames */ fs = snd_enter_user(); snd_pcm_lib_write(substream, (void __user *)runtime->oss.buffer, size1); snd_leave_user(fs); -- cgit v0.10.2 From c34f5a0469c64a3e4e84b04a691247b72175402d Mon Sep 17 00:00:00 2001 From: Jason Gaston Date: Tue, 29 Jan 2008 12:38:49 +0100 Subject: [ALSA] hda_intel: ALSA HD Audio patch for Intel ICH10 DeviceID's This patch adds the Intel ICH10 HD Audio Controller DeviceID's. Signed-off-by: Jason Gaston Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0003021..e5e63fa 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -96,6 +96,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ESB2}," "{Intel, ICH8}," "{Intel, ICH9}," + "{Intel, ICH10}," "{ATI, SB450}," "{ATI, SB600}," "{ATI, RS600}," @@ -1975,6 +1976,8 @@ static struct pci_device_id azx_ids[] = { { 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */ { 0x8086, 0x293e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ { 0x8086, 0x293f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ + { 0x8086, 0x3a3e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ + { 0x8086, 0x3a6e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */ { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */ { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */ -- cgit v0.10.2 From edb54a55d2e36de2183a89efa97fadede33b1166 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Jan 2008 12:47:02 +0100 Subject: [ALSA] hda-codec - Control SPDIF as slave Add SPDIF playback switch to the slave element list so that it can be toggled via the master control as well. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8621c9f..586d98f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1492,6 +1492,7 @@ static const char *alc_slave_sws[] = { "Headphone Playback Switch", "Speaker Playback Switch", "Mono Playback Switch", + "IEC958 Playback Switch", NULL, }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 34890c5..46c237b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -873,6 +873,7 @@ static const char *slave_sws[] = { "Speaker Playback Switch", "External Speaker Playback Switch", "Speaker2 Playback Switch", + "IEC958 Playback Switch", NULL }; -- cgit v0.10.2 From 0f6a5156dee091466b743c163800708383c15bdb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Jan 2008 13:26:04 +0100 Subject: [ALSA] hda-codec - Add model for HP DV9553EG laptop Added the proper model for HP DV9553EG laptop with Cxt5045. ALSA bug#3534 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3534 Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5f6de34..f6dd51c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -853,6 +853,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), -- cgit v0.10.2 From 0aaa22e554c0934625faf79deea42bfecbdbc23d Mon Sep 17 00:00:00 2001 From: Douglas Kosovic Date: Tue, 29 Jan 2008 15:02:50 +0100 Subject: [ALSA] hda-codec - Add Dell T3400 support Added the support for Dell T3400 with AD1984 codec chip. ALSA bug#3699: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3699 Signed-off-by: Douglas Kosovic Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 73f5012..e985cf5 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -926,6 +926,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. AD1984 basic default configuration thinkpad Lenovo Thinkpad T61/X61 + dell Dell T3400 AD1986A 6stack 6-jack, separate surrounds (default) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 845e949..19f0884 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3141,6 +3141,20 @@ static struct hda_input_mux ad1984_thinkpad_capture_source = { }, }; + +/* + * Dell Precision T3400 + */ +static struct hda_input_mux ad1984_dell_desktop_capture_source = { + .num_items = 3, + .items = { + { "Front Mic", 0x0 }, + { "Line-In", 0x1 }, + { "Mix", 0x3 }, + }, +}; + + static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ @@ -3200,6 +3214,44 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = { { } /* end */ }; +/* + * Dell Precision T3400 + */ +static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT), + /* + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), + */ + HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + /* Digial MIC ADC NID 0x05 + 0x06 */ static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, @@ -3253,17 +3305,20 @@ static int ad1984_build_pcms(struct hda_codec *codec) enum { AD1984_BASIC, AD1984_THINKPAD, + AD1984_DELL_DESKTOP, AD1984_MODELS }; static const char *ad1984_models[AD1984_MODELS] = { [AD1984_BASIC] = "basic", [AD1984_THINKPAD] = "thinkpad", + [AD1984_DELL_DESKTOP] = "dell_desktop", }; static struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD), + SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), {} }; @@ -3290,6 +3345,11 @@ static int patch_ad1984(struct hda_codec *codec) spec->mixers[0] = ad1984_thinkpad_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; break; + case AD1984_DELL_DESKTOP: + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984_dell_desktop_capture_source; + spec->mixers[0] = ad1984_dell_desktop_mixers; + break; } return 0; } -- cgit v0.10.2 From 4fe5195c46f4303ca295f7e4350fb45a2aa6cff2 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 29 Jan 2008 15:28:44 +0100 Subject: [ALSA] hda: Add GPIO mute support to STAC9205 Support added for detecting HP jack presence via GPIO on several laptop docks. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 46c237b..7c8cd59 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -122,7 +122,13 @@ struct sigmatel_spec { unsigned int alt_switch: 1; unsigned int hp_detect: 1; - unsigned int gpio_mask, gpio_data; + /* gpio lines */ + unsigned int gpio_mask; + unsigned int gpio_dir; + unsigned int gpio_data; + unsigned int gpio_mute; + + /* analog loopback */ unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -2803,13 +2809,13 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) */ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, - unsigned int data) + unsigned int dir_mask, unsigned int data) { unsigned int gpiostate, gpiomask, gpiodir; gpiostate = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DATA, 0); - gpiostate = (gpiostate & ~mask) | (data & mask); + gpiostate = (gpiostate & ~dir_mask) | (data & dir_mask); gpiomask = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_MASK, 0); @@ -2817,7 +2823,7 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, gpiodir = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DIRECTION, 0); - gpiodir |= mask; + gpiodir |= dir_mask; /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); @@ -2912,7 +2918,8 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_data); + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); return 0; } @@ -3002,10 +3009,14 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) int i, presence; presence = 0; + if (spec->gpio_mute) + presence = !(snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0) & spec->gpio_mute); + for (i = 0; i < cfg->hp_outs; i++) { - presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); if (presence) break; + presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } if (presence) { @@ -3068,7 +3079,8 @@ static int stac92xx_resume(struct hda_codec *codec) stac92xx_set_config_regs(codec); snd_hda_sequence_write(codec, spec->init); - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_data); + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); /* invoke unsolicited event to reset the HP state */ @@ -3302,7 +3314,8 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x000001; + spec->gpio_mask = spec->gpio_dir = 0x1; + spec->gpio_data = 0x01; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; @@ -3376,7 +3389,8 @@ again: spec->aloopback_mask = 0x20; spec->aloopback_shift = 0; - spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + /* GPIO0 High = EAPD */ + spec->gpio_mask = spec->gpio_dir = spec->gpio_data = 0x1; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -3432,7 +3446,8 @@ static int patch_stac922x(struct hda_codec *codec) stac922x_models, stac922x_cfg_tbl); if (spec->board_config == STAC_INTEL_MAC_V3) { - spec->gpio_mask = spec->gpio_data = 0x03; + spec->gpio_mask = spec->gpio_dir = 0x03; + spec->gpio_data = 0x03; /* Intel Macs have all same PCI SSID, so we need to check * codec SSID to distinguish the exact models */ @@ -3560,7 +3575,8 @@ static int patch_stac927x(struct hda_codec *codec) case STAC_D965_3ST: case STAC_D965_5ST: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->gpio_mask = spec->gpio_dir = 0x01; + spec->gpio_data = 0x01; spec->num_dmics = 0; spec->init = d965_core_init; @@ -3574,7 +3590,8 @@ static int patch_stac927x(struct hda_codec *codec) /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000004; + spec->gpio_mask = spec->gpio_dir = 0x04; + spec->gpio_data = 0x04; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; @@ -3585,7 +3602,8 @@ static int patch_stac927x(struct hda_codec *codec) break; default: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->gpio_mask = spec->gpio_dir = 0x1; + spec->gpio_data = 0x01; spec->num_dmics = 0; spec->init = stac927x_core_init; @@ -3680,15 +3698,25 @@ static int patch_stac9205(struct hda_codec *codec) stac92xx_set_config_reg(codec, 0x1f, 0x01441030); stac92xx_set_config_reg(codec, 0x20, 0x1c410030); - spec->gpio_mask = 0x0000000b; + /* Enable unsol response for GPIO4/Dock HP connection */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_HP_EVENT)); + + spec->gpio_dir = 0x0b; + spec->gpio_mask = 0x1b; + spec->gpio_mute = 0x10; /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute, - * GPIO3 High = DRM + * GPIO3 Low = DRM */ - spec->gpio_data = 0x00000009; + spec->gpio_data = 0x01; break; default: /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->gpio_mask = spec->gpio_dir = 0x1; + spec->gpio_data = 0x01; break; } -- cgit v0.10.2 From 4979bca9dcfe4c21c26f378ce446c912fc583ac1 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Wed, 30 Jan 2008 08:13:55 +0100 Subject: [ALSA] HDA-Intel - Add support for Intel SCH This patch adds support for Intel's SCH mobile chipset. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e5e63fa..56f8a30 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -97,6 +97,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH8}," "{Intel, ICH9}," "{Intel, ICH10}," + "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," "{ATI, RS600}," @@ -375,6 +376,7 @@ struct azx { /* driver types */ enum { AZX_DRIVER_ICH, + AZX_DRIVER_SCH, AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_VIA, @@ -385,6 +387,7 @@ enum { static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ICH] = "HDA Intel", + [AZX_DRIVER_SCH] = "HDA Intel MID", [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", @@ -1978,6 +1981,7 @@ static struct pci_device_id azx_ids[] = { { 0x8086, 0x293f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ { 0x8086, 0x3a3e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ { 0x8086, 0x3a6e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ + { 0x8086, 0x811b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SCH }, /* SCH*/ { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */ { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */ { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */ -- cgit v0.10.2 From 19e2e3c30485ba78a653dc521ed9e1f2b6a8bee1 Mon Sep 17 00:00:00 2001 From: Hermann Lauer Date: Wed, 30 Jan 2008 08:25:13 +0100 Subject: [ALSA] es1938 - improve capture hw pointer reads With the Solo1 (es1938) I got a lot of xrun's during capture on my machine. Tracing that down it seems to be comming from reading ocassionaly bad hw pointers from the chip. This patch uses more checking to avoid that false pointer reads. Failed reads are giving back the last good value read instead of spinning in a tight loop, which seems more appropriate to me in an interrupt. I think I saw this trick used in another driver Signed-off-by: Hermann Lauer Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fbe3da7..1a314fa 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -226,6 +226,7 @@ struct es1938 { unsigned int dma2_start; unsigned int dma1_shift; unsigned int dma2_shift; + unsigned int last_capture_dmaaddr; unsigned int active; spinlock_t reg_lock; @@ -528,6 +529,7 @@ static void snd_es1938_capture_setdma(struct es1938 *chip) outb(1, SLDM_REG(chip, DMAMASK)); outb(0x14, SLDM_REG(chip, DMAMODE)); outl(chip->dma1_start, SLDM_REG(chip, DMAADDR)); + chip->last_capture_dmaaddr = chip->dma1_start; outw(chip->dma1_size - 1, SLDM_REG(chip, DMACOUNT)); /* 3. Unmask DMA */ outb(0, SLDM_REG(chip, DMAMASK)); @@ -769,19 +771,40 @@ static int snd_es1938_playback_prepare(struct snd_pcm_substream *substream) return -EINVAL; } +/* during the incrementing of dma counters the DMA register reads sometimes + returns garbage. To ensure a valid hw pointer, the following checks which + should be very unlikely to fail are used: + - is the current DMA address in the valid DMA range ? + - is the sum of DMA address and DMA counter pointing to the last DMA byte ? + One can argue this could differ by one byte depending on which register is + updated first, so the implementation below allows for that. +*/ static snd_pcm_uframes_t snd_es1938_capture_pointer(struct snd_pcm_substream *substream) { struct es1938 *chip = snd_pcm_substream_chip(substream); size_t ptr; +#if 0 size_t old, new; -#if 1 /* This stuff is *needed*, don't ask why - AB */ old = inw(SLDM_REG(chip, DMACOUNT)); while ((new = inw(SLDM_REG(chip, DMACOUNT))) != old) old = new; ptr = chip->dma1_size - 1 - new; #else - ptr = inl(SLDM_REG(chip, DMAADDR)) - chip->dma1_start; + size_t count; + unsigned int diff; + + ptr = inl(SLDM_REG(chip, DMAADDR)); + count = inw(SLDM_REG(chip, DMACOUNT)); + diff = chip->dma1_start + chip->dma1_size - ptr - count; + + if (diff > 3 || ptr < chip->dma1_start + || ptr >= chip->dma1_start+chip->dma1_size) + ptr = chip->last_capture_dmaaddr; /* bad, use last saved */ + else + chip->last_capture_dmaaddr = ptr; /* good, remember it */ + + ptr -= chip->dma1_start; #endif return ptr >> chip->dma1_shift; } -- cgit v0.10.2 From 2ea85986dd1fa96ccf0a2f37877d58a57ff6f583 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 30 Jan 2008 08:38:30 +0100 Subject: [ALSA] oxygen: revert SPI clock frequency change for AK4396/WM8785 While the AK4396 and WM8785 datasheets say that the SPI clock cycle length must be at least 200 ns, 320 ns seems not to work reliably with the controller, so we better use 160 ns. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 51c4d1a..3ea1f05 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -56,7 +56,7 @@ static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) { oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_320 | + OXYGEN_SPI_CLOCK_160 | (0 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, AK4396_WRITE | (reg << 8) | value); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 840e4a6..f31a0eb 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -129,7 +129,7 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, }; oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_320 | + OXYGEN_SPI_CLOCK_160 | (codec_spi_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, AK4396_WRITE | (reg << 8) | value); @@ -139,7 +139,7 @@ static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) { oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_320 | + OXYGEN_SPI_CLOCK_160 | (3 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); -- cgit v0.10.2 From 44893a36ba49349e0f9d4bc3b7bbf9b54469ae1a Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Mon, 12 Nov 2007 12:11:53 +0000 Subject: [ALSA] emu10k1: Add comments regarding E-Mu ins and outs. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 441aa06..fb66a94 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1120,6 +1120,99 @@ /************************************************************************************************/ /* EMU1010m HANA Destinations */ /************************************************************************************************/ +/* Hana, original 1010,1212,1820 using Alice2 + * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2 + * 0x01, 0x10-0x1f: 32 Elink channels to Audio Dock + * 0x01, 0x00: Dock DAC 1 Left + * 0x01, 0x04: Dock DAC 1 Right + * 0x01, 0x08: Dock DAC 2 Left + * 0x01, 0x0c: Dock DAC 2 Right + * 0x01, 0x10: Dock DAC 3 Left + * 0x01, 0x12: PHONES Left + * 0x01, 0x14: Dock DAC 3 Right + * 0x01, 0x16: PHONES Right + * 0x01, 0x18: Dock DAC 4 Left + * 0x01, 0x1a: S/PDIF Left + * 0x01, 0x1c: Dock DAC 4 Right + * 0x01, 0x1e: S/PDIF Right + * 0x02, 0x00: Hana S/PDIF Left + * 0x02, 0x01: Hana S/PDIF Right + * 0x03, 0x00: Hanoa DAC Left + * 0x03, 0x01: Hanoa DAC Right + * 0x04, 0x00-0x07: Hana ADAT + * 0x05, 0x00: I2S0 Left to Alice2 + * 0x05, 0x01: I2S0 Right to Alice2 + * 0x06, 0x00: I2S0 Left to Alice2 + * 0x06, 0x01: I2S0 Right to Alice2 + * 0x07, 0x00: I2S0 Left to Alice2 + * 0x07, 0x01: I2S0 Right to Alice2 + * + * Hana2 never released, but used Tina + * Not needed. + * + * Hana3, rev2 1010,1212,1616 using Tina + * Destinations for SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00, 0x00-0x0f: 16 EMU32A channels to Tina + * 0x01, 0x10-0x1f: 32 EDI channels to Micro Dock + * 0x01, 0x00: Dock DAC 1 Left + * 0x01, 0x04: Dock DAC 1 Right + * 0x01, 0x08: Dock DAC 2 Left + * 0x01, 0x0c: Dock DAC 2 Right + * 0x01, 0x10: Dock DAC 3 Left + * 0x01, 0x12: Dock S/PDIF Left + * 0x01, 0x14: Dock DAC 3 Right + * 0x01, 0x16: Dock S/PDIF Right + * 0x01, 0x18-0x1f: Dock ADAT 0-7 + * 0x02, 0x00: Hana3 S/PDIF Left + * 0x02, 0x01: Hana3 S/PDIF Right + * 0x03, 0x00: Hanoa DAC Left + * 0x03, 0x01: Hanoa DAC Right + * 0x04, 0x00-0x07: Hana3 ADAT 0-7 + * 0x05, 0x00-0x0f: 16 EMU32B channels to Tina + * 0x06-0x07: Not used + * + * HanaLite, rev1 0404 using Alice2 + * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2 + * 0x01: Not used + * 0x02, 0x00: S/PDIF Left + * 0x02, 0x01: S/PDIF Right + * 0x03, 0x00: DAC Left + * 0x03, 0x01: DAC Right + * 0x04-0x07: Not used + * + * HanaLiteLite, rev2 0404 using Alice2 + * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2 + * 0x01: Not used + * 0x02, 0x00: S/PDIF Left + * 0x02, 0x01: S/PDIF Right + * 0x03, 0x00: DAC Left + * 0x03, 0x01: DAC Right + * 0x04-0x07: Not used + * + * Mana, Cardbus 1616 using Tina2 + * Destinations for SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00, 0x00-0x0f: 16 EMU32A channels to Tina2 + * 0x01, 0x10-0x1f: 32 EDI channels to Micro Dock + * 0x01, 0x00: Dock DAC 1 Left + * 0x01, 0x04: Dock DAC 1 Right + * 0x01, 0x08: Dock DAC 2 Left + * 0x01, 0x0c: Dock DAC 2 Right + * 0x01, 0x10: Dock DAC 3 Left + * 0x01, 0x12: Dock S/PDIF Left + * 0x01, 0x14: Dock DAC 3 Right + * 0x01, 0x16: Dock S/PDIF Right + * 0x01, 0x18-0x1f: Dock ADAT 0-7 + * 0x02: Not used + * 0x03, 0x00: Mana DAC Left + * 0x03, 0x01: Mana DAC Right + * 0x04, 0x00-0x0f: 16 EMU32B channels to Tina2 + * 0x05-0x07: Not used + * + * + */ /* 32-bit destinations of signal in the Hana FPGA. Destinations are either * physical outputs of Hana, or outputs going to Alice2 (audigy) for capture * - 16 x EMU_DST_ALICE2_EMU32_X. @@ -1209,6 +1302,102 @@ /************************************************************************************************/ /* EMU1010m HANA Sources */ /************************************************************************************************/ +/* Hana, original 1010,1212,1820 using Alice2 + * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00,0x00-0x1f: Silence + * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock + * 0x01, 0x00: Dock Mic A + * 0x01, 0x04: Dock Mic B + * 0x01, 0x08: Dock ADC 1 Left + * 0x01, 0x0c: Dock ADC 1 Right + * 0x01, 0x10: Dock ADC 2 Left + * 0x01, 0x14: Dock ADC 2 Right + * 0x01, 0x18: Dock ADC 3 Left + * 0x01, 0x1c: Dock ADC 3 Right + * 0x02, 0x00: Hana ADC Left + * 0x02, 0x01: Hana ADC Right + * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output + * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output + * 0x04, 0x00-0x07: Hana ADAT + * 0x05, 0x00: Hana S/PDIF Left + * 0x05, 0x01: Hana S/PDIF Right + * 0x06-0x07: Not used + * + * Hana2 never released, but used Tina + * Not needed. + * + * Hana3, rev2 1010,1212,1616 using Tina + * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00,0x00-0x1f: Silence + * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock + * 0x01, 0x00: Dock Mic A + * 0x01, 0x04: Dock Mic B + * 0x01, 0x08: Dock ADC 1 Left + * 0x01, 0x0c: Dock ADC 1 Right + * 0x01, 0x10: Dock ADC 2 Left + * 0x01, 0x12: Dock S/PDIF Left + * 0x01, 0x14: Dock ADC 2 Right + * 0x01, 0x16: Dock S/PDIF Right + * 0x01, 0x18-0x1f: Dock ADAT 0-7 + * 0x01, 0x18: Dock ADC 3 Left + * 0x01, 0x1c: Dock ADC 3 Right + * 0x02, 0x00: Hanoa ADC Left + * 0x02, 0x01: Hanoa ADC Right + * 0x03, 0x00-0x0f: 16 inputs from Tina Emu32A output + * 0x03, 0x10-0x1f: 16 inputs from Tina Emu32B output + * 0x04, 0x00-0x07: Hana3 ADAT + * 0x05, 0x00: Hana3 S/PDIF Left + * 0x05, 0x01: Hana3 S/PDIF Right + * 0x06-0x07: Not used + * + * HanaLite, rev1 0404 using Alice2 + * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00,0x00-0x1f: Silence + * 0x01: Not used + * 0x02, 0x00: ADC Left + * 0x02, 0x01: ADC Right + * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output + * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output + * 0x04: Not used + * 0x05, 0x00: S/PDIF Left + * 0x05, 0x01: S/PDIF Right + * 0x06-0x07: Not used + * + * HanaLiteLite, rev2 0404 using Alice2 + * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00,0x00-0x1f: Silence + * 0x01: Not used + * 0x02, 0x00: ADC Left + * 0x02, 0x01: ADC Right + * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output + * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output + * 0x04: Not used + * 0x05, 0x00: S/PDIF Left + * 0x05, 0x01: S/PDIF Right + * 0x06-0x07: Not used + * + * Mana, Cardbus 1616 using Tina2 + * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz + * 0x00,0x00-0x1f: Silence + * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock + * 0x01, 0x00: Dock Mic A + * 0x01, 0x04: Dock Mic B + * 0x01, 0x08: Dock ADC 1 Left + * 0x01, 0x0c: Dock ADC 1 Right + * 0x01, 0x10: Dock ADC 2 Left + * 0x01, 0x12: Dock S/PDIF Left + * 0x01, 0x14: Dock ADC 2 Right + * 0x01, 0x16: Dock S/PDIF Right + * 0x01, 0x18-0x1f: Dock ADAT 0-7 + * 0x01, 0x18: Dock ADC 3 Left + * 0x01, 0x1c: Dock ADC 3 Right + * 0x02: Not used + * 0x03, 0x00-0x0f: 16 inputs from Tina Emu32A output + * 0x03, 0x10-0x1f: 16 inputs from Tina Emu32B output + * 0x04-0x07: Not used + * + */ + /* 32-bit sources of signal in the Hana FPGA. The sources are routed to * destinations using mixer control for each destination - see emumixer.c * Sources are either physical inputs of FPGA, -- cgit v0.10.2 From 190d2c46e52592ba092e8bf8acd4427c920f2d69 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 4 Nov 2007 14:08:26 +0000 Subject: [ALSA] snd:emu10k1: E-Mu updates. Fixes to firmware loading and support for 0404. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index fb66a94..8a0c3c1 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1628,7 +1628,7 @@ struct snd_emu_chip_details { unsigned char spdif_bug; /* Has Spdif phasing bug */ unsigned char ac97_chip; /* Has an AC97 chip: 1 = mandatory, 2 = optional */ unsigned char ecard; /* APS EEPROM */ - unsigned char emu1010; /* EMU 1010m card */ + unsigned char emu_model; /* EMU model type */ unsigned char spi_dac; /* SPI interface for DAC */ unsigned char i2c_adc; /* I2C interface for ADC */ unsigned char adc_1361t; /* Use Philips 1361T ADC */ diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 69f342c..da66067 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -54,12 +54,14 @@ #define DOCK_FILENAME "emu/audio_dock.fw" #define EMU1010B_FILENAME "emu/emu1010b.fw" #define MICRO_DOCK_FILENAME "emu/micro_dock.fw" +#define EMU0404_FILENAME "emu/emu0404.fw" #define EMU1010_NOTEBOOK_FILENAME "emu/emu1010_notebook.fw" MODULE_FIRMWARE(HANA_FILENAME); MODULE_FIRMWARE(DOCK_FILENAME); MODULE_FIRMWARE(EMU1010B_FILENAME); MODULE_FIRMWARE(MICRO_DOCK_FILENAME); +MODULE_FIRMWARE(EMU0404_FILENAME); MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); @@ -287,7 +289,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) snd_emu10k1_ptr_write(emu, MAPB, ch, silent_page); } - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { outl(HCFG_AUTOMUTE_ASYNC | HCFG_EMU32_SLAVE | HCFG_AUDIOENABLE, emu->port + HCFG); @@ -317,7 +319,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); if (enable_ir) { /* enable IR for SB Live */ - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { ; /* Disable all access to A_IOCFG for the emu1010 */ } else if (emu->card_capabilities->i2c_adc) { ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ @@ -338,7 +340,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) } } - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { ; /* Disable all access to A_IOCFG for the emu1010 */ } else if (emu->card_capabilities->i2c_adc) { ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ @@ -358,7 +360,7 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) outl(inl(emu->port + HCFG) | HCFG_AUDIOENABLE, emu->port + HCFG); /* Enable analog/digital outs on audigy */ - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { ; /* Disable all access to A_IOCFG for the emu1010 */ } else if (emu->card_capabilities->i2c_adc) { ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ @@ -660,6 +662,8 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file int n, i; int reg; int value; + unsigned int write_post; + unsigned long flags; const struct firmware *fw_entry; if ((err = request_firmware(&fw_entry, filename, &emu->pci->dev)) != 0) { @@ -667,12 +671,6 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return err; } snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size); -#if 0 - if (fw_entry->size != 0x133a4) { - snd_printk(KERN_ERR "firmware: %s wrong size.\n",filename); - return -EINVAL; - } -#endif /* The FPGA is a Xilinx Spartan IIE XC2S50E */ /* GPIO7 -> FPGA PGMN @@ -680,9 +678,12 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file * GPIO5 -> FPGA DIN * FPGA CONFIG OFF -> FPGA PGMN */ + spin_lock_irqsave(&emu->emu_lock, flags); outl(0x00, emu->port + A_IOCFG); /* Set PGMN low for 1uS. */ - udelay(1); + write_post = inl(emu->port + A_IOCFG); + udelay(100); outl(0x80, emu->port + A_IOCFG); /* Leave bit 7 set during netlist setup. */ + write_post = inl(emu->port + A_IOCFG); udelay(100); /* Allow FPGA memory to clean */ for(n = 0; n < fw_entry->size; n++) { value=fw_entry->data[n]; @@ -692,12 +693,15 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file reg = reg | 0x20; value = value >> 1; outl(reg, emu->port + A_IOCFG); + write_post = inl(emu->port + A_IOCFG); outl(reg | 0x40, emu->port + A_IOCFG); + write_post = inl(emu->port + A_IOCFG); } } /* After programming, set GPIO bit 4 high again. */ outl(0x10, emu->port + A_IOCFG); - + write_post = inl(emu->port + A_IOCFG); + spin_unlock_irqrestore(&emu->emu_lock, flags); release_firmware(fw_entry); return 0; @@ -711,7 +715,7 @@ int emu1010_firmware_thread(void *data) { for (;;) { /* Delay to allow Audio Dock to settle */ - msleep(1000); + msleep_interruptible(1000); if (kthread_should_stop()) break; snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ @@ -721,17 +725,17 @@ int emu1010_firmware_thread(void *data) { /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); - if (emu->card_capabilities->emu1010 == 1) { + if (emu->card_capabilities->emu_model == 1) { if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { - return err; + continue; } - } else if (emu->card_capabilities->emu1010 == 2) { + } else if (emu->card_capabilities->emu_model == 2) { if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { - return err; + continue; } - } else if (emu->card_capabilities->emu1010 == 3) { + } else if (emu->card_capabilities->emu_model == 3) { if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { - return err; + continue; } } @@ -744,8 +748,7 @@ int emu1010_firmware_thread(void *data) { if ((reg & 0x1f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); - return 0; - return -ENODEV; + continue; } snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); @@ -756,9 +759,9 @@ int emu1010_firmware_thread(void *data) { msleep(10); /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); - break; } } + snd_printk(KERN_INFO "emu1010: firmware thread stopping\n"); return 0; } @@ -799,6 +802,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) int tmp,tmp2; int reg; int err; + const char *filename = NULL; snd_printk(KERN_INFO "emu1010: Special config.\n"); /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, @@ -840,21 +844,31 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) return -ENODEV; } snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); - if (emu->card_capabilities->emu1010 == 1) { - if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { - snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); - return err; - } - } else if (emu->card_capabilities->emu1010 == 2) { - if ((err = snd_emu1010_load_firmware(emu, EMU1010B_FILENAME)) != 0) { - snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME); - return err; - } - } else if (emu->card_capabilities->emu1010 == 3) { - if ((err = snd_emu1010_load_firmware(emu, EMU1010_NOTEBOOK_FILENAME)) != 0) { - snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010_NOTEBOOK_FILENAME); - return err; - } + switch (emu->card_capabilities->emu_model) { + case 1: + filename = HANA_FILENAME; + break; + case 2: + filename = EMU1010B_FILENAME; + break; + case 3: + filename = EMU1010_NOTEBOOK_FILENAME; + break; + case 4: + filename = EMU0404_FILENAME; + break; + default: + filename = NULL; + return -ENODEV; + break; + } + snd_printk(KERN_INFO "emu1010: filename %s testing\n", filename); + err = snd_emu1010_load_firmware(emu, filename); + if (err != 0) { + snd_printk( + KERN_INFO "emu1010: Loading Firmware file %s failed\n", + filename); + return err; } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ @@ -1201,11 +1215,12 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) } snd_emu10k1_free_efx(emu); } - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model == 1) { /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); - kthread_stop(emu->emu1010.firmware_thread); } + if (emu->card_capabilities->emu_model) + kthread_stop(emu->emu1010.firmware_thread); if (emu->memhdr) snd_util_memhdr_free(emu->memhdr); if (emu->silent_page.area) @@ -1337,6 +1352,15 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spi_dac = 1, .i2c_adc = 1, .spk71 = 1} , + /* Tested by James@superbug.co.uk 20-3-2007. */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, + .driver = "Audigy2", .name = "E-mu 0404 [4002]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .spk71 = 1, + .emu_model = 4} , /* EMU 0404 */ + /* Tested by James@superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", .id = "EMU1010", @@ -1344,28 +1368,30 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .ca_cardbus_chip = 1, .spk71 = 1 , - .emu1010 = 3} , + .emu_model = 3} , + /* Tested by James@superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, - .spk71 = 1 , - .emu1010 = 2} , + .spk71 = 1, + .emu_model = 2} , + /* Tested by James@superbug.co.uk 8th July 2005. */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, + .driver = "Audigy2", .name = "E-mu 1010 [4001]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .spk71 = 1, + .emu_model = 1} , /* Emu 1010 */ + /* Audigy4 (Not PRO) SB0610 */ {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, .ac97_chip = 1} , - /* Tested by James@superbug.co.uk 8th July 2005. No sound available yet. */ - {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, - .driver = "Audigy2", .name = "E-mu 1010 [4001]", - .id = "EMU1010", - .emu10k2_chip = 1, - .ca0102_chip = 1, - .spk71 = 1, - .emu1010 = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102, .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", @@ -1793,7 +1819,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, if (emu->card_capabilities->ecard) { if ((err = snd_emu10k1_ecard_init(emu)) < 0) goto error; - } else if (emu->card_capabilities->emu1010) { + } else if (emu->card_capabilities->emu_model) { if ((err = snd_emu10k1_emu1010_init(emu)) < 0) { snd_emu10k1_free(emu); return err; @@ -1942,7 +1968,7 @@ void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) snd_emu10k1_cardbus_init(emu); if (emu->card_capabilities->ecard) snd_emu10k1_ecard_init(emu); - else if (emu->card_capabilities->emu1010) + else if (emu->card_capabilities->emu_model) snd_emu10k1_emu1010_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 354a892..71dc4c8 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1261,7 +1261,7 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* emu1212 DSP 0 and DSP 1 Capture */ - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { if (emu->card_capabilities->ca0108_chip) { /* Note:JCD:No longer bit shift lower 16bits to upper 16bits of 32bit value. */ A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x0), A_C_00000001); @@ -1515,7 +1515,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* digital outputs */ /* A_PUT_STEREO_OUTPUT(A_EXTOUT_FRONT_L, A_EXTOUT_FRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS); */ - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { /* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */ snd_printk("EMU outputs on\n"); for (z = 0; z < 8; z++) { @@ -1563,7 +1563,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) A_PUT_OUTPUT(A_EXTOUT_ADC_CAP_R, capture+1); #endif - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { if (emu->card_capabilities->ca0108_chip) { snd_printk("EMU2 inputs on\n"); for (z = 0; z < 0x10; z++) { diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 5a38fe7..83acfa6 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1792,7 +1792,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, return err; } - if ( emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { ; /* Disable the snd_audigy_spdif_shared_spdif */ } else if (emu->audigy) { if ((kctl = snd_ctl_new1(&snd_audigy_shared_spdif, emu)) == NULL) @@ -1817,7 +1817,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, return err; } - if ( emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { int i; for (i = 0; i < ARRAY_SIZE(snd_emu1010_output_enum_ctls); i++) { diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index cf4e3ec..cf9276d 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -357,7 +357,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]); snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24)); snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24)); - if (emu->card_capabilities->emu1010) + if (emu->card_capabilities->emu_model) pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ else pitch_target = emu10k1_calc_pitch_target(runtime->rate); @@ -700,7 +700,7 @@ static void snd_emu10k1_playback_trigger_voice(struct snd_emu10k1 *emu, struct s voice = evoice->number; pitch = snd_emu10k1_rate_to_pitch(runtime->rate) >> 8; - if (emu->card_capabilities->emu1010) + if (emu->card_capabilities->emu_model) pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ else pitch_target = emu10k1_calc_pitch_target(runtime->rate); @@ -1231,7 +1231,7 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) runtime->hw.rates = SNDRV_PCM_RATE_48000; runtime->hw.rate_min = runtime->hw.rate_max = 48000; spin_lock_irq(&emu->reg_lock); - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { /* Nb. of channels has been increased to 16 */ /* TODO * SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE @@ -1790,7 +1790,7 @@ int __devinit snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct s /* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */ if (emu->audigy) { emu->efx_voices_mask[0] = 0; - if (emu->card_capabilities->emu1010) + if (emu->card_capabilities->emu_model) /* Pavel Hofman - 32 voices will be used for * capture (write mode) - * each bit = corresponding voice diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index bd8a471..f3caa3f 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -244,7 +244,7 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, unsigned long flags; u32 rate; - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x38, &value); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -584,7 +584,7 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu) { struct snd_info_entry *entry; #ifdef CONFIG_SND_DEBUG - if (emu->card_capabilities->emu1010) { + if (emu->card_capabilities->emu_model) { if (! snd_card_proc_new(emu->card, "emu1010_regs", &entry)) snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read); } diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 2862e17..a026383 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -227,11 +227,14 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) { + unsigned long flags; + if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */ return 1; + spin_lock_irqsave(&emu->emu_lock, flags); outl(reg, emu->port + A_IOCFG); udelay(10); outl(reg | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */ @@ -239,20 +242,24 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) outl(value, emu->port + A_IOCFG); udelay(10); outl(value | 0x80 , emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */ + spin_unlock_irqrestore(&emu->emu_lock, flags); return 0; } int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value) { + unsigned long flags; if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ + spin_lock_irqsave(&emu->emu_lock, flags); outl(reg, emu->port + A_IOCFG); udelay(10); outl(reg | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */ udelay(10); *value = ((inl(emu->port + A_IOCFG) >> 8) & 0x7f); + spin_unlock_irqrestore(&emu->emu_lock, flags); return 0; } -- cgit v0.10.2 From 1c02e36681ae20a796204e8d629d13fa9d5e20b5 Mon Sep 17 00:00:00 2001 From: Ctirad Fertr Date: Thu, 13 Dec 2007 16:27:13 +0100 Subject: [ALSA] emu10k1 - 1616(M) cardbus improvements This patch improves E-Mu 1616(M) cardbus support. It adds definitions of the new Microdock and 1010 cardbus registers (thanks again for descriptions James) and improves mixer for this card. Now you can use S/PDIF and ADAT on Mirodock and also use headpohone output on host cardbus card as another independent output. Signed-off-by: Ctirad Fertr Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 8a0c3c1..4474b4e 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1299,6 +1299,23 @@ #define EMU_DST_ALICE_I2S2_LEFT 0x0700 /* Alice2 I2S2 Left */ #define EMU_DST_ALICE_I2S2_RIGHT 0x0701 /* Alice2 I2S2 Right */ +/* Additional destinations for 1616(M)/Microdock */ +/* Microdock S/PDIF OUT Left, 1st or 48kHz only */ +#define EMU_DST_MDOCK_SPDIF_LEFT1 0x0112 +/* Microdock S/PDIF OUT Left, 2nd or 96kHz */ +#define EMU_DST_MDOCK_SPDIF_LEFT2 0x0113 +/* Microdock S/PDIF OUT Right, 1st or 48kHz only */ +#define EMU_DST_MDOCK_SPDIF_RIGHT1 0x0116 +/* Microdock S/PDIF OUT Right, 2nd or 96kHz */ +#define EMU_DST_MDOCK_SPDIF_RIGHT2 0x0117 +/* Microdock S/PDIF ADAT 8 channel out +8 to +f */ +#define EMU_DST_MDOCK_ADAT 0x0118 + +/* Headphone jack on 1010 cardbus? 44.1/48kHz only? */ +#define EMU_DST_MANA_DAC_LEFT 0x0300 +/* Headphone jack on 1010 cardbus? 44.1/48kHz only? */ +#define EMU_DST_MANA_DAC_RIGHT 0x0301 + /************************************************************************************************/ /* EMU1010m HANA Sources */ /************************************************************************************************/ @@ -1452,6 +1469,19 @@ #define EMU_SRC_HANA_SPDIF_LEFT2 0x0502 /* Hana SPDIF Left, 2nd or 96kHz */ #define EMU_SRC_HANA_SPDIF_RIGHT1 0x0501 /* Hana SPDIF Right, 1st or 48kHz only */ #define EMU_SRC_HANA_SPDIF_RIGHT2 0x0503 /* Hana SPDIF Right, 2nd or 96kHz */ + +/* Additional inputs for 1616(M)/Microdock */ +/* Microdock S/PDIF Left, 1st or 48kHz only */ +#define EMU_SRC_MDOCK_SPDIF_LEFT1 0x0112 +/* Microdock S/PDIF Left, 2nd or 96kHz */ +#define EMU_SRC_MDOCK_SPDIF_LEFT2 0x0113 +/* Microdock S/PDIF Right, 1st or 48kHz only */ +#define EMU_SRC_MDOCK_SPDIF_RIGHT1 0x0116 +/* Microdock S/PDIF Right, 2nd or 96kHz */ +#define EMU_SRC_MDOCK_SPDIF_RIGHT2 0x0117 +/* Microdock ADAT 8 channel in +8 to +f */ +#define EMU_SRC_MDOCK_ADAT 0x0118 + /* 0x600 and 0x700 no used */ /* ------------------- STRUCTURES -------------------- */ diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index da66067..54b978e 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1103,79 +1103,114 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) EMU_DST_HANA_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 3); /* ALICE2 bus 0xb3 */ #endif /* Default outputs */ - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_DAC1_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ - emu->emu1010.output_source[0] = 21; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_DAC1_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); - emu->emu1010.output_source[1] = 22; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_DAC2_LEFT1, EMU_SRC_ALICE_EMU32A + 2); - emu->emu1010.output_source[2] = 23; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_DAC2_RIGHT1, EMU_SRC_ALICE_EMU32A + 3); - emu->emu1010.output_source[3] = 24; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_DAC3_LEFT1, EMU_SRC_ALICE_EMU32A + 4); - emu->emu1010.output_source[4] = 25; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_DAC3_RIGHT1, EMU_SRC_ALICE_EMU32A + 5); - emu->emu1010.output_source[5] = 26; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_DAC4_LEFT1, EMU_SRC_ALICE_EMU32A + 6); - emu->emu1010.output_source[6] = 27; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_DAC4_RIGHT1, EMU_SRC_ALICE_EMU32A + 7); - emu->emu1010.output_source[7] = 28; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_PHONES_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ - emu->emu1010.output_source[8] = 21; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_PHONES_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); - emu->emu1010.output_source[9] = 22; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ - emu->emu1010.output_source[10] = 21; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_DOCK_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); - emu->emu1010.output_source[11] = 22; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ - emu->emu1010.output_source[12] = 21; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); - emu->emu1010.output_source[13] = 22; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ - emu->emu1010.output_source[14] = 21; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HAMOA_DAC_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); - emu->emu1010.output_source[15] = 22; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_ADAT, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ - emu->emu1010.output_source[16] = 21; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_ADAT + 1, EMU_SRC_ALICE_EMU32A + 1); - emu->emu1010.output_source[17] = 22; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_ADAT + 2, EMU_SRC_ALICE_EMU32A + 2); - emu->emu1010.output_source[18] = 23; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_ADAT + 3, EMU_SRC_ALICE_EMU32A + 3); - emu->emu1010.output_source[19] = 24; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_ADAT + 4, EMU_SRC_ALICE_EMU32A + 4); - emu->emu1010.output_source[20] = 25; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_ADAT + 5, EMU_SRC_ALICE_EMU32A + 5); - emu->emu1010.output_source[21] = 26; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_ADAT + 6, EMU_SRC_ALICE_EMU32A + 6); - emu->emu1010.output_source[22] = 27; - snd_emu1010_fpga_link_dst_src_write(emu, - EMU_DST_HANA_ADAT + 7, EMU_SRC_ALICE_EMU32A + 7); - emu->emu1010.output_source[23] = 28; - + if (emu->card_capabilities->emu_model == 3) { + /* 1616(M) cardbus default outputs */ + /* ALICE2 bus 0xa0 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC1_LEFT1, EMU_SRC_ALICE_EMU32A + 0); + emu->emu1010.output_source[0] = 17; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC1_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[1] = 18; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC2_LEFT1, EMU_SRC_ALICE_EMU32A + 2); + emu->emu1010.output_source[2] = 19; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC2_RIGHT1, EMU_SRC_ALICE_EMU32A + 3); + emu->emu1010.output_source[3] = 20; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC3_LEFT1, EMU_SRC_ALICE_EMU32A + 4); + emu->emu1010.output_source[4] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC3_RIGHT1, EMU_SRC_ALICE_EMU32A + 5); + emu->emu1010.output_source[5] = 22; + /* ALICE2 bus 0xa0 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_MANA_DAC_LEFT, EMU_SRC_ALICE_EMU32A + 0); + emu->emu1010.output_source[16] = 17; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_MANA_DAC_RIGHT, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[17] = 18; + } else { + /* ALICE2 bus 0xa0 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC1_LEFT1, EMU_SRC_ALICE_EMU32A + 0); + emu->emu1010.output_source[0] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC1_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[1] = 22; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC2_LEFT1, EMU_SRC_ALICE_EMU32A + 2); + emu->emu1010.output_source[2] = 23; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC2_RIGHT1, EMU_SRC_ALICE_EMU32A + 3); + emu->emu1010.output_source[3] = 24; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC3_LEFT1, EMU_SRC_ALICE_EMU32A + 4); + emu->emu1010.output_source[4] = 25; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC3_RIGHT1, EMU_SRC_ALICE_EMU32A + 5); + emu->emu1010.output_source[5] = 26; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC4_LEFT1, EMU_SRC_ALICE_EMU32A + 6); + emu->emu1010.output_source[6] = 27; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC4_RIGHT1, EMU_SRC_ALICE_EMU32A + 7); + emu->emu1010.output_source[7] = 28; + /* ALICE2 bus 0xa0 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_PHONES_LEFT1, EMU_SRC_ALICE_EMU32A + 0); + emu->emu1010.output_source[8] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_PHONES_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[9] = 22; + /* ALICE2 bus 0xa0 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 0); + emu->emu1010.output_source[10] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[11] = 22; + /* ALICE2 bus 0xa0 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 0); + emu->emu1010.output_source[12] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[13] = 22; + /* ALICE2 bus 0xa0 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32A + 0); + emu->emu1010.output_source[14] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HAMOA_DAC_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[15] = 22; + /* ALICE2 bus 0xa0 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT, EMU_SRC_ALICE_EMU32A + 0); + emu->emu1010.output_source[16] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[17] = 22; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 2, EMU_SRC_ALICE_EMU32A + 2); + emu->emu1010.output_source[18] = 23; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 3, EMU_SRC_ALICE_EMU32A + 3); + emu->emu1010.output_source[19] = 24; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 4, EMU_SRC_ALICE_EMU32A + 4); + emu->emu1010.output_source[20] = 25; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 5, EMU_SRC_ALICE_EMU32A + 5); + emu->emu1010.output_source[21] = 26; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 6, EMU_SRC_ALICE_EMU32A + 6); + emu->emu1010.output_source[22] = 27; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 7, EMU_SRC_ALICE_EMU32A + 7); + emu->emu1010.output_source[23] = 28; + } /* TEMP: Select SPDIF in/out */ //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 83acfa6..9b5883b 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -139,6 +139,61 @@ static char *emu1010_src_texts[] = { "DSP 31", }; +/* 1616(m) cardbus */ + +static char *emu1616_src_texts[] = { + "Silence", + "Dock Mic A", + "Dock Mic B", + "Dock ADC1 Left", + "Dock ADC1 Right", + "Dock ADC2 Left", + "Dock ADC2 Right", + "Dock SPDIF Left", + "Dock SPDIF Right", + "ADAT 0", + "ADAT 1", + "ADAT 2", + "ADAT 3", + "ADAT 4", + "ADAT 5", + "ADAT 6", + "ADAT 7", + "DSP 0", + "DSP 1", + "DSP 2", + "DSP 3", + "DSP 4", + "DSP 5", + "DSP 6", + "DSP 7", + "DSP 8", + "DSP 9", + "DSP 10", + "DSP 11", + "DSP 12", + "DSP 13", + "DSP 14", + "DSP 15", + "DSP 16", + "DSP 17", + "DSP 18", + "DSP 19", + "DSP 20", + "DSP 21", + "DSP 22", + "DSP 23", + "DSP 24", + "DSP 25", + "DSP 26", + "DSP 27", + "DSP 28", + "DSP 29", + "DSP 30", + "DSP 31", +}; + + /* * List of data sources available for each destination */ @@ -198,6 +253,59 @@ static unsigned int emu1010_src_regs[] = { EMU_SRC_ALICE_EMU32B+0xf, /* 52 */ }; +/* 1616(m) cardbus */ +static unsigned int emu1616_src_regs[] = { + EMU_SRC_SILENCE, + EMU_SRC_DOCK_MIC_A1, + EMU_SRC_DOCK_MIC_B1, + EMU_SRC_DOCK_ADC1_LEFT1, + EMU_SRC_DOCK_ADC1_RIGHT1, + EMU_SRC_DOCK_ADC2_LEFT1, + EMU_SRC_DOCK_ADC2_RIGHT1, + EMU_SRC_MDOCK_SPDIF_LEFT1, + EMU_SRC_MDOCK_SPDIF_RIGHT1, + EMU_SRC_MDOCK_ADAT, + EMU_SRC_MDOCK_ADAT+1, + EMU_SRC_MDOCK_ADAT+2, + EMU_SRC_MDOCK_ADAT+3, + EMU_SRC_MDOCK_ADAT+4, + EMU_SRC_MDOCK_ADAT+5, + EMU_SRC_MDOCK_ADAT+6, + EMU_SRC_MDOCK_ADAT+7, + EMU_SRC_ALICE_EMU32A, + EMU_SRC_ALICE_EMU32A+1, + EMU_SRC_ALICE_EMU32A+2, + EMU_SRC_ALICE_EMU32A+3, + EMU_SRC_ALICE_EMU32A+4, + EMU_SRC_ALICE_EMU32A+5, + EMU_SRC_ALICE_EMU32A+6, + EMU_SRC_ALICE_EMU32A+7, + EMU_SRC_ALICE_EMU32A+8, + EMU_SRC_ALICE_EMU32A+9, + EMU_SRC_ALICE_EMU32A+0xa, + EMU_SRC_ALICE_EMU32A+0xb, + EMU_SRC_ALICE_EMU32A+0xc, + EMU_SRC_ALICE_EMU32A+0xd, + EMU_SRC_ALICE_EMU32A+0xe, + EMU_SRC_ALICE_EMU32A+0xf, + EMU_SRC_ALICE_EMU32B, + EMU_SRC_ALICE_EMU32B+1, + EMU_SRC_ALICE_EMU32B+2, + EMU_SRC_ALICE_EMU32B+3, + EMU_SRC_ALICE_EMU32B+4, + EMU_SRC_ALICE_EMU32B+5, + EMU_SRC_ALICE_EMU32B+6, + EMU_SRC_ALICE_EMU32B+7, + EMU_SRC_ALICE_EMU32B+8, + EMU_SRC_ALICE_EMU32B+9, + EMU_SRC_ALICE_EMU32B+0xa, + EMU_SRC_ALICE_EMU32B+0xb, + EMU_SRC_ALICE_EMU32B+0xc, + EMU_SRC_ALICE_EMU32B+0xd, + EMU_SRC_ALICE_EMU32B+0xe, + EMU_SRC_ALICE_EMU32B+0xf, +}; + /* * Data destinations - physical EMU outputs. * Each destination has an enum mixer control to choose a data source @@ -229,6 +337,28 @@ static unsigned int emu1010_output_dst[] = { EMU_DST_HANA_ADAT+7, /* 23 */ }; +/* 1616(m) cardbus */ +static unsigned int emu1616_output_dst[] = { + EMU_DST_DOCK_DAC1_LEFT1, + EMU_DST_DOCK_DAC1_RIGHT1, + EMU_DST_DOCK_DAC2_LEFT1, + EMU_DST_DOCK_DAC2_RIGHT1, + EMU_DST_DOCK_DAC3_LEFT1, + EMU_DST_DOCK_DAC3_RIGHT1, + EMU_DST_MDOCK_SPDIF_LEFT1, + EMU_DST_MDOCK_SPDIF_RIGHT1, + EMU_DST_MDOCK_ADAT, + EMU_DST_MDOCK_ADAT+1, + EMU_DST_MDOCK_ADAT+2, + EMU_DST_MDOCK_ADAT+3, + EMU_DST_MDOCK_ADAT+4, + EMU_DST_MDOCK_ADAT+5, + EMU_DST_MDOCK_ADAT+6, + EMU_DST_MDOCK_ADAT+7, + EMU_DST_MANA_DAC_LEFT, + EMU_DST_MANA_DAC_RIGHT, +}; + /* * Data destinations - HANA outputs going to Alice2 (audigy) for * capture (EMU32 + I2S links) @@ -259,14 +389,26 @@ static unsigned int emu1010_input_dst[] = { EMU_DST_ALICE_I2S2_RIGHT, }; -static int snd_emu1010_input_output_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int snd_emu1010_input_output_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + char **items; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = 53; + if (emu->card_capabilities->emu_model == 3) { /* 1616(m) cardbus */ + uinfo->value.enumerated.items = 49; + items = emu1616_src_texts; + } else { + uinfo->value.enumerated.items = 53; + items = emu1010_src_texts; + } if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, emu1010_src_texts[uinfo->value.enumerated.item]); + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + items[uinfo->value.enumerated.item]); return 0; } @@ -278,7 +420,8 @@ static int snd_emu1010_output_source_get(struct snd_kcontrol *kcontrol, channel = (kcontrol->private_value) & 0xff; /* Limit: emu1010_output_dst, emu->emu1010.output_source */ - if (channel >= 24) + if (channel >= 24 || + (emu->card_capabilities->emu_model == 3 && channel >= 18)) return -EINVAL; ucontrol->value.enumerated.item[0] = emu->emu1010.output_source[channel]; return 0; @@ -288,24 +431,28 @@ static int snd_emu1010_output_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - int change = 0; unsigned int val; unsigned int channel; val = ucontrol->value.enumerated.item[0]; - if (val >= 53) + if (val >= 53 || + (emu->card_capabilities->emu_model == 3 && val >= 49)) return -EINVAL; channel = (kcontrol->private_value) & 0xff; /* Limit: emu1010_output_dst, emu->emu1010.output_source */ - if (channel >= 24) + if (channel >= 24 || + (emu->card_capabilities->emu_model == 3 && channel >= 18)) return -EINVAL; - if (emu->emu1010.output_source[channel] != val) { - emu->emu1010.output_source[channel] = val; - change = 1; + if (emu->emu1010.output_source[channel] == val) + return 0; + emu->emu1010.output_source[channel] = val; + if (emu->card_capabilities->emu_model == 3) /* 1616(m) cardbus */ + snd_emu1010_fpga_link_dst_src_write(emu, + emu1616_output_dst[channel], emu1616_src_regs[val]); + else snd_emu1010_fpga_link_dst_src_write(emu, emu1010_output_dst[channel], emu1010_src_regs[val]); - } - return change; + return 1; } static int snd_emu1010_input_source_get(struct snd_kcontrol *kcontrol, @@ -326,24 +473,27 @@ static int snd_emu1010_input_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - int change = 0; unsigned int val; unsigned int channel; val = ucontrol->value.enumerated.item[0]; - if (val >= 53) + if (val >= 53 || + (emu->card_capabilities->emu_model == 3 && val >= 49)) return -EINVAL; channel = (kcontrol->private_value) & 0xff; /* Limit: emu1010_input_dst, emu->emu1010.input_source */ if (channel >= 22) return -EINVAL; - if (emu->emu1010.input_source[channel] != val) { - emu->emu1010.input_source[channel] = val; - change = 1; + if (emu->emu1010.input_source[channel] == val) + return 0; + emu->emu1010.input_source[channel] = val; + if (emu->card_capabilities->emu_model == 3) /* 1616(m) cardbus */ + snd_emu1010_fpga_link_dst_src_write(emu, + emu1010_input_dst[channel], emu1616_src_regs[val]); + else snd_emu1010_fpga_link_dst_src_write(emu, emu1010_input_dst[channel], emu1010_src_regs[val]); - } - return change; + return 1; } #define EMU1010_SOURCE_OUTPUT(xname,chid) \ @@ -383,6 +533,30 @@ static struct snd_kcontrol_new snd_emu1010_output_enum_ctls[] __devinitdata = { EMU1010_SOURCE_OUTPUT("1010 ADAT 7 Playback Enum", 0x17), }; + +/* 1616(m) cardbus */ +static struct snd_kcontrol_new snd_emu1616_output_enum_ctls[] __devinitdata = { + EMU1010_SOURCE_OUTPUT("Dock DAC1 Left Playback Enum", 0), + EMU1010_SOURCE_OUTPUT("Dock DAC1 Right Playback Enum", 1), + EMU1010_SOURCE_OUTPUT("Dock DAC2 Left Playback Enum", 2), + EMU1010_SOURCE_OUTPUT("Dock DAC2 Right Playback Enum", 3), + EMU1010_SOURCE_OUTPUT("Dock DAC3 Left Playback Enum", 4), + EMU1010_SOURCE_OUTPUT("Dock DAC3 Right Playback Enum", 5), + EMU1010_SOURCE_OUTPUT("Dock SPDIF Left Playback Enum", 6), + EMU1010_SOURCE_OUTPUT("Dock SPDIF Right Playback Enum", 7), + EMU1010_SOURCE_OUTPUT("Dock ADAT 0 Playback Enum", 8), + EMU1010_SOURCE_OUTPUT("Dock ADAT 1 Playback Enum", 9), + EMU1010_SOURCE_OUTPUT("Dock ADAT 2 Playback Enum", 0xa), + EMU1010_SOURCE_OUTPUT("Dock ADAT 3 Playback Enum", 0xb), + EMU1010_SOURCE_OUTPUT("Dock ADAT 4 Playback Enum", 0xc), + EMU1010_SOURCE_OUTPUT("Dock ADAT 5 Playback Enum", 0xd), + EMU1010_SOURCE_OUTPUT("Dock ADAT 6 Playback Enum", 0xe), + EMU1010_SOURCE_OUTPUT("Dock ADAT 7 Playback Enum", 0xf), + EMU1010_SOURCE_OUTPUT("Mana DAC Left Playback Enum", 0x10), + EMU1010_SOURCE_OUTPUT("Mana DAC Right Playback Enum", 0x11), +}; + + #define EMU1010_SOURCE_INPUT(xname,chid) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -1817,30 +1991,73 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, return err; } - if (emu->card_capabilities->emu_model) { + if (emu->card_capabilities->emu_model == 3) { + /* 1616(m) cardbus */ + int i; + + for (i = 0; i < ARRAY_SIZE(snd_emu1616_output_enum_ctls); i++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1616_output_enum_ctls[i], + emu)); + if (err < 0) + return err; + } + for (i = 0; i < ARRAY_SIZE(snd_emu1010_input_enum_ctls); i++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_input_enum_ctls[i], + emu)); + if (err < 0) + return err; + } + for (i = 0; i < ARRAY_SIZE(snd_emu1010_adc_pads) - 2; i++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_adc_pads[i], emu)); + if (err < 0) + return err; + } + for (i = 0; i < ARRAY_SIZE(snd_emu1010_dac_pads) - 2; i++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_dac_pads[i], emu)); + if (err < 0) + return err; + } + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_internal_clock, emu)); + if (err < 0) + return err; + + } else { + /* all other e-mu cards for now */ int i; for (i = 0; i < ARRAY_SIZE(snd_emu1010_output_enum_ctls); i++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_emu1010_output_enum_ctls[i], emu)); + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_output_enum_ctls[i], + emu)); if (err < 0) return err; } for (i = 0; i < ARRAY_SIZE(snd_emu1010_input_enum_ctls); i++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_emu1010_input_enum_ctls[i], emu)); + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_input_enum_ctls[i], + emu)); if (err < 0) return err; } for (i = 0; i < ARRAY_SIZE(snd_emu1010_adc_pads); i++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_emu1010_adc_pads[i], emu)); + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_adc_pads[i], emu)); if (err < 0) return err; } for (i = 0; i < ARRAY_SIZE(snd_emu1010_dac_pads); i++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_emu1010_dac_pads[i], emu)); + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_dac_pads[i], emu)); if (err < 0) return err; } - err = snd_ctl_add(card, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; } -- cgit v0.10.2 From 88aa139057f2740c5dd55e2a542b2425186e4d3c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 21 Dec 2007 16:20:12 +0100 Subject: [ALSA] emu10k1 - Don't create emu1010 controls for non-emu boards The last change for emu1616 introduced a bug that the driver creates emu1010-related controls even on non-emu boards. Fixed now. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 9b5883b..7ebf035 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -2026,7 +2026,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, if (err < 0) return err; - } else { + } else if (emu->card_capabilities->emu_model) { /* all other e-mu cards for now */ int i; -- cgit v0.10.2 From 3839e4f136d6da3dc85d237aa9569ee94bfea763 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 21 Dec 2007 16:33:32 +0100 Subject: [ALSA] emu10k1 - Use enum for emu_model types Use enum instead of digits for emu_model types. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 4474b4e..494648d 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1642,6 +1642,14 @@ struct snd_emu10k1_midi { void (*interrupt)(struct snd_emu10k1 *emu, unsigned int status); }; +enum { + EMU_MODEL_SB, + EMU_MODEL_EMU1010, + EMU_MODEL_EMU1010B, + EMU_MODEL_EMU1616, + EMU_MODEL_EMU0404, +}; + struct snd_emu_chip_details { u32 vendor; u32 device; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 54b978e..7e46325 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -725,15 +725,18 @@ int emu1010_firmware_thread(void *data) { /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); - if (emu->card_capabilities->emu_model == 1) { + if (emu->card_capabilities->emu_model == + EMU_MODEL_EMU1010) { if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { continue; } - } else if (emu->card_capabilities->emu_model == 2) { + } else if (emu->card_capabilities->emu_model == + EMU_MODEL_EMU1010B) { if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { continue; } - } else if (emu->card_capabilities->emu_model == 3) { + } else if (emu->card_capabilities->emu_model == + EMU_MODEL_EMU1616) { if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { continue; } @@ -845,16 +848,16 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); switch (emu->card_capabilities->emu_model) { - case 1: + case EMU_MODEL_EMU1010: filename = HANA_FILENAME; break; - case 2: + case EMU_MODEL_EMU1010B: filename = EMU1010B_FILENAME; break; - case 3: + case EMU_MODEL_EMU1616: filename = EMU1010_NOTEBOOK_FILENAME; break; - case 4: + case EMU_MODEL_EMU0404: filename = EMU0404_FILENAME; break; default: @@ -1103,7 +1106,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) EMU_DST_HANA_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 3); /* ALICE2 bus 0xb3 */ #endif /* Default outputs */ - if (emu->card_capabilities->emu_model == 3) { + if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) { /* 1616(M) cardbus default outputs */ /* ALICE2 bus 0xa0 */ snd_emu1010_fpga_link_dst_src_write(emu, @@ -1250,7 +1253,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) } snd_emu10k1_free_efx(emu); } - if (emu->card_capabilities->emu_model == 1) { + if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); } @@ -1394,7 +1397,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, - .emu_model = 4} , /* EMU 0404 */ + .emu_model = EMU_MODEL_EMU0404} , /* EMU 0404 */ /* Tested by James@superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", @@ -1403,7 +1406,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .ca_cardbus_chip = 1, .spk71 = 1 , - .emu_model = 3} , + .emu_model = EMU_MODEL_EMU1616}, /* Tested by James@superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", @@ -1411,7 +1414,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, - .emu_model = 2} , + .emu_model = EMU_MODEL_EMU1010B}, /* Tested by James@superbug.co.uk 8th July 2005. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, .driver = "Audigy2", .name = "E-mu 1010 [4001]", @@ -1419,7 +1422,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, - .emu_model = 1} , /* Emu 1010 */ + .emu_model = EMU_MODEL_EMU1010} , /* Emu 1010 */ /* Audigy4 (Not PRO) SB0610 */ {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 7ebf035..fd22120 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -397,7 +397,7 @@ static int snd_emu1010_input_output_source_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - if (emu->card_capabilities->emu_model == 3) { /* 1616(m) cardbus */ + if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) { uinfo->value.enumerated.items = 49; items = emu1616_src_texts; } else { @@ -421,7 +421,8 @@ static int snd_emu1010_output_source_get(struct snd_kcontrol *kcontrol, channel = (kcontrol->private_value) & 0xff; /* Limit: emu1010_output_dst, emu->emu1010.output_source */ if (channel >= 24 || - (emu->card_capabilities->emu_model == 3 && channel >= 18)) + (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616 && + channel >= 18)) return -EINVAL; ucontrol->value.enumerated.item[0] = emu->emu1010.output_source[channel]; return 0; @@ -436,17 +437,19 @@ static int snd_emu1010_output_source_put(struct snd_kcontrol *kcontrol, val = ucontrol->value.enumerated.item[0]; if (val >= 53 || - (emu->card_capabilities->emu_model == 3 && val >= 49)) + (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616 && + val >= 49)) return -EINVAL; channel = (kcontrol->private_value) & 0xff; /* Limit: emu1010_output_dst, emu->emu1010.output_source */ if (channel >= 24 || - (emu->card_capabilities->emu_model == 3 && channel >= 18)) + (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616 && + channel >= 18)) return -EINVAL; if (emu->emu1010.output_source[channel] == val) return 0; emu->emu1010.output_source[channel] = val; - if (emu->card_capabilities->emu_model == 3) /* 1616(m) cardbus */ + if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) snd_emu1010_fpga_link_dst_src_write(emu, emu1616_output_dst[channel], emu1616_src_regs[val]); else @@ -478,7 +481,8 @@ static int snd_emu1010_input_source_put(struct snd_kcontrol *kcontrol, val = ucontrol->value.enumerated.item[0]; if (val >= 53 || - (emu->card_capabilities->emu_model == 3 && val >= 49)) + (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616 && + val >= 49)) return -EINVAL; channel = (kcontrol->private_value) & 0xff; /* Limit: emu1010_input_dst, emu->emu1010.input_source */ @@ -487,7 +491,7 @@ static int snd_emu1010_input_source_put(struct snd_kcontrol *kcontrol, if (emu->emu1010.input_source[channel] == val) return 0; emu->emu1010.input_source[channel] = val; - if (emu->card_capabilities->emu_model == 3) /* 1616(m) cardbus */ + if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) snd_emu1010_fpga_link_dst_src_write(emu, emu1010_input_dst[channel], emu1616_src_regs[val]); else @@ -1991,7 +1995,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, return err; } - if (emu->card_capabilities->emu_model == 3) { + if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) { /* 1616(m) cardbus */ int i; -- cgit v0.10.2 From c94fa4c9168e51a8dab8e72cb9f0d89673fc8d8c Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 10 Nov 2007 17:55:14 +0000 Subject: [ALSA] emu10k1: General cleanup, add new locks, fix alsa bug#3501, kernel bug#9304. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 494648d..7b7b9b1 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1742,6 +1742,8 @@ struct snd_emu10k1 { spinlock_t reg_lock; spinlock_t emu_lock; spinlock_t voice_lock; + spinlock_t spi_lock; /* serialises access to spi port */ + spinlock_t i2c_lock; /* serialises access to i2c port */ struct snd_emu10k1_voice voices[NUM_G]; struct snd_emu10k1_voice p16v_voices[4]; diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 01965bd..45088eb 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -35,9 +35,9 @@ struct best_voice { /* * prototypes */ -static void lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, +static void lookup_voices(struct snd_emux *emux, struct snd_emu10k1 *hw, struct best_voice *best, int active_only); -static struct snd_emux_voice *get_voice(struct snd_emux *emu, +static struct snd_emux_voice *get_voice(struct snd_emux *emux, struct snd_emux_port *port); static int start_voice(struct snd_emux_voice *vp); static void trigger_voice(struct snd_emux_voice *vp); @@ -45,7 +45,6 @@ static void release_voice(struct snd_emux_voice *vp); static void update_voice(struct snd_emux_voice *vp, int update); static void terminate_voice(struct snd_emux_voice *vp); static void free_voice(struct snd_emux_voice *vp); - static void set_fmmod(struct snd_emu10k1 *hw, struct snd_emux_voice *vp); static void set_fm2frq2(struct snd_emu10k1 *hw, struct snd_emux_voice *vp); static void set_filterQ(struct snd_emu10k1 *hw, struct snd_emux_voice *vp); @@ -75,9 +74,9 @@ static struct snd_emux_operators emu10k1_ops = { }; void -snd_emu10k1_ops_setup(struct snd_emux *emu) +snd_emu10k1_ops_setup(struct snd_emux *emux) { - emu->ops = emu10k1_ops; + emux->ops = emu10k1_ops; } @@ -166,7 +165,11 @@ free_voice(struct snd_emux_voice *vp) struct snd_emu10k1 *hw; hw = vp->hw; - if (vp->ch >= 0) { + /* FIXME: emu10k1_synth is broken. */ + /* This can get called with hw == 0 */ + /* Problem apparent on plug, unplug then plug */ + /* on the Audigy 2 ZS Notebook. */ + if (hw && (vp->ch >= 0)) { snd_emu10k1_ptr_write(hw, IFATN, vp->ch, 0xff00); snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0x807f | DCYSUSV_CHANNELENABLE_MASK); // snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 7e46325..f29caf1 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -259,7 +259,6 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) * GPIO7: Unknown */ outl(0x76, emu->port + A_IOCFG); /* Windows uses 0x3f76 */ - } if (emu->card_capabilities->i2c_adc) { /* Audigy 2 ZS Notebook with ADC Wolfson WM8775 */ int size, n; @@ -275,7 +274,6 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) emu->i2c_capture_volume[n][0]= 0xcf; emu->i2c_capture_volume[n][1]= 0xcf; } - } @@ -653,6 +651,8 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu) value = inl(special_port); snd_emu10k1_ptr20_write(emu, TINA2_VOLUME, 0, 0xfefefefe); /* Defaults to 0x30303030 */ + /* Delay to give time for ADC chip to switch on. It needs 113ms */ + msleep(200); return 0; } @@ -1717,6 +1717,8 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->card = card; spin_lock_init(&emu->reg_lock); spin_lock_init(&emu->emu_lock); + spin_lock_init(&emu->spi_lock); + spin_lock_init(&emu->i2c_lock); spin_lock_init(&emu->voice_lock); spin_lock_init(&emu->synth_lock); spin_lock_init(&emu->memblk_lock); diff --git a/sound/pci/emu10k1/emu10k1_synth.c b/sound/pci/emu10k1/emu10k1_synth.c index 204995a..ad7b714 100644 --- a/sound/pci/emu10k1/emu10k1_synth.c +++ b/sound/pci/emu10k1/emu10k1_synth.c @@ -30,7 +30,7 @@ MODULE_LICENSE("GPL"); */ static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) { - struct snd_emux *emu; + struct snd_emux *emux; struct snd_emu10k1 *hw; struct snd_emu10k1_synth_arg *arg; unsigned long flags; @@ -46,53 +46,56 @@ static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) else if (arg->max_voices > 64) arg->max_voices = 64; - if (snd_emux_new(&emu) < 0) + if (snd_emux_new(&emux) < 0) return -ENOMEM; - snd_emu10k1_ops_setup(emu); - emu->hw = hw = arg->hwptr; - emu->max_voices = arg->max_voices; - emu->num_ports = arg->seq_ports; - emu->pitch_shift = -501; - emu->memhdr = hw->memhdr; - emu->midi_ports = arg->seq_ports < 2 ? arg->seq_ports : 2; /* maximum two ports */ - emu->midi_devidx = hw->audigy ? 2 : 1; /* audigy has two external midis */ - emu->linear_panning = 0; - emu->hwdep_idx = 2; /* FIXED */ - - if (snd_emux_register(emu, dev->card, arg->index, "Emu10k1") < 0) { - snd_emux_free(emu); + snd_emu10k1_ops_setup(emux); + hw = arg->hwptr; + emux->hw = hw; + emux->max_voices = arg->max_voices; + emux->num_ports = arg->seq_ports; + emux->pitch_shift = -501; + emux->memhdr = hw->memhdr; + /* maximum two ports */ + emux->midi_ports = arg->seq_ports < 2 ? arg->seq_ports : 2; + /* audigy has two external midis */ + emux->midi_devidx = hw->audigy ? 2 : 1; + emux->linear_panning = 0; + emux->hwdep_idx = 2; /* FIXED */ + + if (snd_emux_register(emux, dev->card, arg->index, "Emu10k1") < 0) { + snd_emux_free(emux); return -ENOMEM; } spin_lock_irqsave(&hw->voice_lock, flags); - hw->synth = emu; + hw->synth = emux; hw->get_synth_voice = snd_emu10k1_synth_get_voice; spin_unlock_irqrestore(&hw->voice_lock, flags); - dev->driver_data = emu; + dev->driver_data = emux; return 0; } static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev) { - struct snd_emux *emu; + struct snd_emux *emux; struct snd_emu10k1 *hw; unsigned long flags; if (dev->driver_data == NULL) return 0; /* not registered actually */ - emu = dev->driver_data; + emux = dev->driver_data; - hw = emu->hw; + hw = emux->hw; spin_lock_irqsave(&hw->voice_lock, flags); hw->synth = NULL; hw->get_synth_voice = NULL; spin_unlock_irqrestore(&hw->voice_lock, flags); - snd_emux_free(emu); + snd_emux_free(emux); return 0; } diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index a026383..b5a802b 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -70,6 +70,11 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i unsigned long flags; unsigned int mask; + if (!emu) { + snd_printk(KERN_ERR "ptr_write: emu is null!\n"); + dump_stack(); + return; + } mask = emu->audigy ? A_PTR_ADDRESS_MASK : PTR_ADDRESS_MASK; regptr = ((reg << 16) & mask) | (chn & PTR_CHANNELNUM_MASK); @@ -134,15 +139,23 @@ int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int reset, set; unsigned int reg, tmp; int n, result; + int err = 0; + + /* This function is not re-entrant, so protect against it. */ + spin_lock(&emu->spi_lock); if (emu->card_capabilities->ca0108_chip) reg = 0x3c; /* PTR20, reg 0x3c */ else { /* For other chip types the SPI register * is currently unknown. */ - return 1; + err = 1; + goto spi_write_exit; + } + if (data > 0xffff) { + /* Only 16bit values allowed */ + err = 1; + goto spi_write_exit; } - if (data > 0xffff) /* Only 16bit values allowed */ - return 1; tmp = snd_emu10k1_ptr20_read(emu, reg, 0); reset = (tmp & ~0x3ffff) | 0x20000; /* Set xxx20000 */ @@ -160,11 +173,17 @@ int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, break; } } - if (result) /* Timed out */ - return 1; + if (result) { + /* Timed out */ + err = 1; + goto spi_write_exit; + } snd_emu10k1_ptr20_write(emu, reg, 0, reset | data); tmp = snd_emu10k1_ptr20_read(emu, reg, 0); /* Write post */ - return 0; + err = 0; +spi_write_exit: + spin_unlock(&emu->spi_lock); + return err; } /* The ADC does not support i2c read, so only write is implemented */ @@ -176,15 +195,17 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, int timeout = 0; int status; int retry; + int err = 0; + if ((reg > 0x7f) || (value > 0x1ff)) { snd_printk(KERN_ERR "i2c_write: invalid values.\n"); return -EINVAL; } + /* This function is not re-entrant, so protect against it. */ + spin_lock(&emu->i2c_lock); + tmp = reg << 25 | value << 16; - // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); - /* Not sure what this I2C channel controls. */ - /* snd_emu10k1_ptr_write(emu, P17V_I2C_0, 0, tmp); */ /* This controls the I2C connected to the WM8775 ADC Codec */ snd_emu10k1_ptr20_write(emu, P17V_I2C_1, 0, tmp); @@ -192,17 +213,14 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, for (retry = 0; retry < 10; retry++) { /* Send the data to i2c */ - //tmp = snd_emu10k1_ptr_read(emu, P17V_I2C_ADDR, 0); - //tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); tmp = 0; tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD); snd_emu10k1_ptr20_write(emu, P17V_I2C_ADDR, 0, tmp); /* Wait till the transaction ends */ while (1) { - udelay(10); + mdelay(1); status = snd_emu10k1_ptr20_read(emu, P17V_I2C_ADDR, 0); - // snd_printk("I2C:status=0x%x\n", status); timeout++; if ((status & I2C_A_ADC_START) == 0) break; @@ -219,10 +237,14 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, if (retry == 10) { snd_printk(KERN_ERR "Writing to ADC failed!\n"); - return -EINVAL; + snd_printk(KERN_ERR "status=0x%x, reg=%d, value=%d\n", + status, reg, value); + /* dump_stack(); */ + err = -EINVAL; } - return 0; + spin_unlock(&emu->i2c_lock); + return err; } int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index 276d08c..30bfed6 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -34,9 +34,10 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) struct snd_emu10k1 *emu = dev_id; unsigned int status, status2, orig_status, orig_status2; int handled = 0; + int timeout = 0; - while ((status = inl(emu->port + IPR)) != 0) { - //snd_printk(KERN_INFO "emu10k1 irq - status = 0x%x\n", status); + while (((status = inl(emu->port + IPR)) != 0) && (timeout < 1000)) { + timeout++; orig_status = status; handled = 1; if ((status & 0xffffffff) == 0xffffffff) { @@ -200,5 +201,8 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) } outl(orig_status, emu->port + IPR); /* ack all */ } + if (timeout == 1000) + snd_printk(KERN_INFO "emu10k1 irq routine failure\n"); + return IRQ_RETVAL(handled); } -- cgit v0.10.2 From bd3d1c2096ff99311b8524273ba0f53502365136 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Dec 2007 12:43:00 +0100 Subject: [ALSA] emu10k1 - Fix kthread handling at resume Don't create emu1010 kthread again at resume if it's already created. Also make the thread function static. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index f29caf1..560bc50 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -707,7 +707,8 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return 0; } -int emu1010_firmware_thread(void *data) { +static int emu1010_firmware_thread(void *data) +{ struct snd_emu10k1 * emu = data; int tmp,tmp2; int reg; @@ -1090,10 +1091,12 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ /* Start Micro/Audio Dock firmware loader thread */ - emu->emu1010.firmware_thread = kthread_create(&emu1010_firmware_thread, - emu, - "emu1010_firmware"); - wake_up_process(emu->emu1010.firmware_thread); + if (!emu->emu1010.firmware_thread) { + emu->emu1010.firmware_thread = + kthread_create(emu1010_firmware_thread, emu, + "emu1010_firmware"); + wake_up_process(emu->emu1010.firmware_thread); + } #if 0 snd_emu1010_fpga_link_dst_src_write(emu, @@ -1257,7 +1260,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); } - if (emu->card_capabilities->emu_model) + if (emu->emu1010.firmware_thread) kthread_stop(emu->emu1010.firmware_thread); if (emu->memhdr) snd_util_memhdr_free(emu->memhdr); -- cgit v0.10.2 From 493b4acb9ed3193b19d45d62e0e5740d20f47adc Mon Sep 17 00:00:00 2001 From: Veli-Matti Valtonen Date: Mon, 7 Jan 2008 12:36:56 +0100 Subject: [ALSA] emu10k1 - Another EMU0404 Board ID This is based on pseudo-random playing around with the capabilities. With ca0102 this card gives no output atall, ca0108 appears to work fine, so it rather looks similar to the EMU1010b/EMU1010 changes. Some other people seem to have succeeded in using this aswell: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3496 From: Veli-Matti Valtonen Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 560bc50..9a9b977 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1393,14 +1393,6 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spi_dac = 1, .i2c_adc = 1, .spk71 = 1} , - /* Tested by James@superbug.co.uk 20-3-2007. */ - {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, - .driver = "Audigy2", .name = "E-mu 0404 [4002]", - .id = "EMU0404", - .emu10k2_chip = 1, - .ca0102_chip = 1, - .spk71 = 1, - .emu_model = EMU_MODEL_EMU0404} , /* EMU 0404 */ /* Tested by James@superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", @@ -1425,7 +1417,23 @@ static struct snd_emu_chip_details emu_chip_details[] = { .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, - .emu_model = EMU_MODEL_EMU1010} , /* Emu 1010 */ + .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */ + /* EMU0404b */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, + .driver = "Audigy2", .name = "E-mu 0404b [4002]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ + /* Tested by James@superbug.co.uk 20-3-2007. */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, + .driver = "Audigy2", .name = "E-mu 0404 [4002]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ /* Audigy4 (Not PRO) SB0610 */ {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", -- cgit v0.10.2 From cace16f174d971a80f81e68ed04f1124a50dd800 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 30 Jan 2008 14:58:38 +0100 Subject: [ALSA] hda: fix Mic in as output Some laptop has an internal analog microphone that is 'fixed'. This patch prevents creating a 'Mic In as Output' switch for ports that can't be outputs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7c8cd59..caf48ed 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2283,15 +2283,23 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } if (spec->mic_switch) { + unsigned int def_conf; nid = cfg->input_pins[AUTO_PIN_MIC]; - pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); - if (pincap & AC_PINCAP_OUT) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_IO_SWITCH, - "Mic as Output Switch", (nid << 8) | 1); - if (err < 0) - return err; + def_conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + + /* some laptops have an internal analog microphone + * which can't be used as a output */ + if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_IO_SWITCH, + "Mic as Output Switch", (nid << 8) | 1); + if (err < 0) + return err; + } } } -- cgit v0.10.2 From 2ecba4ffbbc6c85fce8c3878514be415edace413 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 31 Jan 2008 17:40:18 +0100 Subject: [ALSA] version 1.0.16rc2 Signed-off-by: Jaroslav Kysela diff --git a/include/sound/version.h b/include/sound/version.h index a9781eb..fac66c4 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h. Generated by alsa/ksync script. */ -#define CONFIG_SND_VERSION "1.0.15" -#define CONFIG_SND_DATE " (Tue Nov 20 19:16:42 2007 UTC)" +#define CONFIG_SND_VERSION "1.0.16rc2" +#define CONFIG_SND_DATE " (Thu Jan 31 16:40:16 2008 UTC)" -- cgit v0.10.2