From 45c1de8e20cec40b6846def0aeca09cb1bfb839b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 2 Nov 2010 17:08:37 +0100 Subject: ALSA: oxygen: merge HiFier driver into snd-oxygen The snd-hifier driver contains more duplicated code than model-specific code, so it does not make sense for it to be a separate driver. Handling the two-channel output restriction can be easily done in the generic driver. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index d0eb696..f1a1787 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -974,13 +974,6 @@ 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. - Module snd-ice1712 ------------------ @@ -1531,7 +1524,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-oxygen ----------------- - Module for sound cards based on the C-Media CMI8788 chip: + Module for sound cards based on the C-Media CMI8787/8788 chip: * Asound A-8788 * AuzenTech X-Meridian * Bgears b-Enspirer @@ -1540,6 +1533,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * HT-Omega Claro halo (XT) * Razer Barracuda AC-1 * Sondigo Inferno + * TempoTec HiFier Fantasia This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 12e3465..dfe406d 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -209,7 +209,7 @@ config SND_OXYGEN_LIB tristate config SND_OXYGEN - tristate "C-Media 8788 (Oxygen)" + tristate "C-Media 8787, 8788 (Oxygen)" select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -224,6 +224,7 @@ config SND_OXYGEN * HT-Omega Claro halo (XT) * Razer Barracuda AC-1 * Sondigo Inferno + * TempoTec/MediaTek HiFier Fantasia To compile this driver as a module, choose M here: the module will be called snd-oxygen. @@ -578,18 +579,6 @@ 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" - select SND_OXYGEN_LIB - select SND_PCM - select SND_MPU401_UART - 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)" select SND_MPU401_UART diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index acd8f15..bd67c0d 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -1,10 +1,8 @@ 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 xonar_lib.o \ xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.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/hifier.c b/sound/pci/oxygen/hifier.c deleted file mode 100644 index 5a87d68..0000000 --- a/sound/pci/oxygen/hifier.c +++ /dev/null @@ -1,239 +0,0 @@ -/* - * 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 - */ - -/* - * CMI8788: - * - * SPI 0 -> AK4396 - */ - -#include -#include -#include -#include -#include -#include -#include -#include "oxygen.h" -#include "ak4396.h" - -MODULE_AUTHOR("Clemens Ladisch "); -MODULE_DESCRIPTION("TempoTec HiFier driver"); -MODULE_LICENSE("GPL v2"); - -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 DEFINE_PCI_DEVICE_TABLE(hifier_ids) = { - { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, - { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, - { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, - { } -}; -MODULE_DEVICE_TABLE(pci, hifier_ids); - -struct hifier_data { - u8 ak4396_regs[5]; -}; - -static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) -{ - struct hifier_data *data = chip->model_data; - - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | - OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_160 | - (0 << OXYGEN_SPI_CODEC_SHIFT) | - OXYGEN_SPI_CEN_LATCH_CLOCK_HI, - AK4396_WRITE | (reg << 8) | value); - data->ak4396_regs[reg] = value; -} - -static void ak4396_write_cached(struct oxygen *chip, u8 reg, u8 value) -{ - struct hifier_data *data = chip->model_data; - - if (value != data->ak4396_regs[reg]) - ak4396_write(chip, reg, value); -} - -static void hifier_registers_init(struct oxygen *chip) -{ - struct hifier_data *data = chip->model_data; - - ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); - ak4396_write(chip, AK4396_CONTROL_2, - data->ak4396_regs[AK4396_CONTROL_2]); - ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); -} - -static void hifier_init(struct oxygen *chip) -{ - struct hifier_data *data = chip->model_data; - - data->ak4396_regs[AK4396_CONTROL_2] = - AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; - hifier_registers_init(chip); - - snd_component_add(chip->card, "AK4396"); - snd_component_add(chip->card, "CS5340"); -} - -static void hifier_cleanup(struct oxygen *chip) -{ -} - -static void hifier_resume(struct oxygen *chip) -{ - hifier_registers_init(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_regs[AK4396_CONTROL_2] & ~AK4396_DFS_MASK; - if (params_rate(params) <= 54000) - value |= AK4396_DFS_NORMAL; - else if (params_rate(params) <= 108000) - value |= AK4396_DFS_DOUBLE; - else - value |= AK4396_DFS_QUAD; - - msleep(1); /* wait for the new MCLK to become stable */ - - if (value != data->ak4396_regs[AK4396_CONTROL_2]) { - 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_cached(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write_cached(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_regs[AK4396_CONTROL_2] & ~AK4396_SMUTE; - if (chip->dac_mute) - value |= AK4396_SMUTE; - ak4396_write_cached(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 const struct oxygen_model model_hifier = { - .shortname = "C-Media CMI8787", - .longname = "C-Media Oxygen HD Audio", - .chip = "CMI8788", - .init = hifier_init, - .cleanup = hifier_cleanup, - .resume = hifier_resume, - .get_i2s_mclk = oxygen_default_i2s_mclk, - .set_dac_params = set_ak4396_params, - .set_adc_params = set_cs5340_params, - .update_dac_volume = update_ak4396_volume, - .update_dac_mute = update_ak4396_mute, - .dac_tlv = ak4396_db_scale, - .model_data_size = sizeof(struct hifier_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_1, - .dac_channels = 2, - .dac_volume_min = 0, - .dac_volume_max = 255, - .function_flags = OXYGEN_FUNCTION_SPI, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - -static int __devinit get_hifier_model(struct oxygen *chip, - const struct pci_device_id *id) -{ - chip->model = model_hifier; - return 0; -} - -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], THIS_MODULE, - hifier_ids, get_hifier_model); - 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), -#ifdef CONFIG_PM - .suspend = oxygen_pci_suspend, - .resume = oxygen_pci_resume, -#endif -}; - -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 98a8eb3..5e258b2 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -70,6 +70,7 @@ enum { MODEL_MERIDIAN, /* AuzenTech X-Meridian */ MODEL_CLARO, /* HT-Omega Claro */ MODEL_CLARO_HALO, /* HT-Omega Claro halo */ + MODEL_HIFIER, /* TempoTec HiFier Fantasia */ }; static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { @@ -81,6 +82,8 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x14c3, 0x1710), .driver_data = MODEL_HIFIER }, + { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_HIFIER }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO }, @@ -98,6 +101,7 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_CLARO_HP 0x0100 struct generic_data { + unsigned int dacs; u8 ak4396_regs[4][5]; u16 wm8785_regs[3]; }; @@ -148,7 +152,7 @@ static void ak4396_registers_init(struct oxygen *chip) struct generic_data *data = chip->model_data; unsigned int i; - for (i = 0; i < 4; ++i) { + for (i = 0; i < data->dacs; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); ak4396_write(chip, i, AK4396_CONTROL_2, @@ -166,6 +170,7 @@ static void ak4396_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; + data->dacs = chip->model.dac_channels / 2; data->ak4396_regs[0][AK4396_CONTROL_2] = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; ak4396_registers_init(chip); @@ -232,6 +237,12 @@ static void claro_halo_init(struct oxygen *chip) claro_enable_hp(chip); } +static void hifier_init(struct oxygen *chip) +{ + ak4396_init(chip); + snd_component_add(chip->card, "CS5340"); +} + static void generic_cleanup(struct oxygen *chip) { } @@ -268,6 +279,11 @@ static void claro_resume(struct oxygen *chip) claro_enable_hp(chip); } +static void stereo_resume(struct oxygen *chip) +{ + ak4396_registers_init(chip); +} + static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -286,7 +302,7 @@ static void set_ak4396_params(struct oxygen *chip, msleep(1); /* wait for the new MCLK to become stable */ if (value != data->ak4396_regs[0][AK4396_CONTROL_2]) { - for (i = 0; i < 4; ++i) { + for (i = 0; i < data->dacs; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB); ak4396_write(chip, i, AK4396_CONTROL_2, value); @@ -298,9 +314,10 @@ static void set_ak4396_params(struct oxygen *chip, static void update_ak4396_volume(struct oxygen *chip) { + struct generic_data *data = chip->model_data; unsigned int i; - for (i = 0; i < 4; ++i) { + for (i = 0; i < data->dacs; ++i) { ak4396_write_cached(chip, i, AK4396_LCH_ATT, chip->dac_volume[i * 2]); ak4396_write_cached(chip, i, AK4396_RCH_ATT, @@ -317,7 +334,7 @@ static void update_ak4396_mute(struct oxygen *chip) value = data->ak4396_regs[0][AK4396_CONTROL_2] & ~AK4396_SMUTE; if (chip->dac_mute) value |= AK4396_SMUTE; - for (i = 0; i < 4; ++i) + for (i = 0; i < data->dacs; ++i) ak4396_write_cached(chip, i, AK4396_CONTROL_2, value); } @@ -356,6 +373,10 @@ static void set_ak5385_params(struct oxygen *chip, value, GPIO_AK5385_DFS_MASK); } +static void set_no_params(struct oxygen *chip, struct snd_pcm_hw_params *params) +{ +} + static int rolloff_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -400,7 +421,7 @@ static int rolloff_put(struct snd_kcontrol *ctl, reg &= ~AK4396_SLOW; changed = reg != data->ak4396_regs[0][AK4396_CONTROL_2]; if (changed) { - for (i = 0; i < 4; ++i) + for (i = 0; i < data->dacs; ++i) ak4396_write(chip, i, AK4396_CONTROL_2, reg); } mutex_unlock(&chip->mutex); @@ -550,6 +571,18 @@ static int __devinit get_oxygen_model(struct oxygen *chip, CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; break; + case MODEL_HIFIER: + chip->model.shortname = "C-Media CMI8787"; + chip->model.chip = "CMI8787"; + chip->model.init = hifier_init; + chip->model.resume = stereo_resume; + chip->model.mixer_init = generic_mixer_init; + chip->model.set_adc_params = set_no_params; + chip->model.device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1; + chip->model.dac_channels = 2; + break; } if (id->driver_data == MODEL_MERIDIAN || id->driver_data == MODEL_CLARO_HALO) { -- cgit v0.10.2 From 31f86bacfc9c8f6a3f25fa991c1f373374a9f25b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 2 Nov 2010 17:18:23 +0100 Subject: ALSA: oxygen: add Kuroutoshikou CMI8787-HG2PCI support Add support for the Kuroutoshikou CMI8787-HG2PCI sound card. [replaced non-latin letters in the patch by tiwai] Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index f1a1787..fdd388d 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1531,6 +1531,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Club3D Theatron DTS * HT-Omega Claro (plus) * HT-Omega Claro halo (XT) + * Kuroutoshikou CMI8787-HG2PCI * Razer Barracuda AC-1 * Sondigo Inferno * TempoTec HiFier Fantasia diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index dfe406d..f7139d0 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -222,6 +222,7 @@ config SND_OXYGEN * Club3D Theatron DTS * HT-Omega Claro (plus) * HT-Omega Claro halo (XT) + * Kuroutoshikou CMI8787-HG2PCI * Razer Barracuda AC-1 * Sondigo Inferno * TempoTec/MediaTek HiFier Fantasia diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 5e258b2..dd0f3f4 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -71,6 +71,7 @@ enum { MODEL_CLARO, /* HT-Omega Claro */ MODEL_CLARO_HALO, /* HT-Omega Claro halo */ MODEL_HIFIER, /* TempoTec HiFier Fantasia */ + MODEL_HG2PCI, /* Kuroutoshikou CMI8787-HG2PCI */ }; static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { @@ -80,7 +81,7 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x13f6, 0x0001), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x0010), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF }, - { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_HG2PCI }, { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x14c3, 0x1710), .driver_data = MODEL_HIFIER }, { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_HIFIER }, @@ -243,6 +244,11 @@ static void hifier_init(struct oxygen *chip) snd_component_add(chip->card, "CS5340"); } +static void hg2pci_init(struct oxygen *chip) +{ + ak4396_init(chip); +} + static void generic_cleanup(struct oxygen *chip) { } @@ -572,15 +578,20 @@ static int __devinit get_oxygen_model(struct oxygen *chip, CAPTURE_1_FROM_SPDIF; break; case MODEL_HIFIER: + case MODEL_HG2PCI: chip->model.shortname = "C-Media CMI8787"; chip->model.chip = "CMI8787"; - chip->model.init = hifier_init; + if (id->driver_data == MODEL_HIFIER) + chip->model.init = hifier_init; + else + chip->model.init = hg2pci_init; chip->model.resume = stereo_resume; chip->model.mixer_init = generic_mixer_init; chip->model.set_adc_params = set_no_params; chip->model.device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_1; + PLAYBACK_1_TO_SPDIF; + if (id->driver_data == MODEL_HIFIER) + chip->model.device_config |= CAPTURE_0_FROM_I2S_1; chip->model.dac_channels = 2; break; } diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index e5ebe56..2e65799 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -262,7 +262,7 @@ oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[]) */ subdevice = oxygen_read_eeprom(chip, 2); /* use default ID if EEPROM is missing */ - if (subdevice == 0xffff) + if (subdevice == 0xffff && oxygen_read_eeprom(chip, 1) == 0xffff) subdevice = 0x8788; /* * We use only the subsystem device ID for searching because it is -- cgit v0.10.2 From 18f24839f18f1934c1e37e86ce8f3fecbb0328c9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 3 Nov 2010 11:36:33 +0100 Subject: ALSA: oxygen: reorganize PCI IDs Sort the PCI IDs so that they make logical sense. Also move the card name comments into this list because the model symbols should be (more) self-explanationary. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index dd0f3f4..13f39e5 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -66,28 +66,34 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); enum { - MODEL_CMEDIA_REF, /* C-Media's reference design */ - MODEL_MERIDIAN, /* AuzenTech X-Meridian */ - MODEL_CLARO, /* HT-Omega Claro */ - MODEL_CLARO_HALO, /* HT-Omega Claro halo */ - MODEL_HIFIER, /* TempoTec HiFier Fantasia */ - MODEL_HG2PCI, /* Kuroutoshikou CMI8787-HG2PCI */ + MODEL_CMEDIA_REF, + MODEL_MERIDIAN, + MODEL_CLARO, + MODEL_CLARO_HALO, + MODEL_HIFIER, + MODEL_HG2PCI, }; static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { + /* C-Media's reference design */ { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0218), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x0001), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x0010), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF }, - { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_HG2PCI }, { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, + /* Kuroutoshikou CMI8787-HG2PCI */ + { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_HG2PCI }, + /* TempoTec HiFier Fantasia */ { OXYGEN_PCI_SUBID(0x14c3, 0x1710), .driver_data = MODEL_HIFIER }, { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_HIFIER }, - { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, + /* AuzenTech X-Meridian */ { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, + /* HT-Omega Claro */ { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO }, + /* HT-Omega Claro halo */ { OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_CLARO_HALO }, { } }; -- cgit v0.10.2 From 2146dcfd15ad55cfdd18b45e1e6601d6a86f0cbe Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 3 Nov 2010 12:26:35 +0100 Subject: ALSA: oxygen: add HiFier Serenade support Add support for the TempoTec/MediaTek HiFier Serenade sound card. The PCI ID was already there, but the driver handled it like the Fantasia model, which resulted in a dummy recording device. As a stereo output-only card, this model is to be handled exactly like the HG2PCI. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index fdd388d..7124340 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1535,6 +1535,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Razer Barracuda AC-1 * Sondigo Inferno * TempoTec HiFier Fantasia + * TempoTec HiFier Serenade This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index f7139d0..5add96b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -226,6 +226,7 @@ config SND_OXYGEN * Razer Barracuda AC-1 * Sondigo Inferno * TempoTec/MediaTek HiFier Fantasia + * TempoTec/MediaTek HiFier Serenade To compile this driver as a module, choose M here: the module will be called snd-oxygen. diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 13f39e5..ea8fffe 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -70,8 +70,8 @@ enum { MODEL_MERIDIAN, MODEL_CLARO, MODEL_CLARO_HALO, - MODEL_HIFIER, - MODEL_HG2PCI, + MODEL_FANTASIA, + MODEL_2CH_OUTPUT, }; static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { @@ -85,10 +85,11 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, /* Kuroutoshikou CMI8787-HG2PCI */ - { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_HG2PCI }, + { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_2CH_OUTPUT }, /* TempoTec HiFier Fantasia */ - { OXYGEN_PCI_SUBID(0x14c3, 0x1710), .driver_data = MODEL_HIFIER }, - { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_HIFIER }, + { OXYGEN_PCI_SUBID(0x14c3, 0x1710), .driver_data = MODEL_FANTASIA }, + /* TempoTec HiFier Serenade */ + { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_2CH_OUTPUT }, /* AuzenTech X-Meridian */ { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, /* HT-Omega Claro */ @@ -244,13 +245,13 @@ static void claro_halo_init(struct oxygen *chip) claro_enable_hp(chip); } -static void hifier_init(struct oxygen *chip) +static void fantasia_init(struct oxygen *chip) { ak4396_init(chip); snd_component_add(chip->card, "CS5340"); } -static void hg2pci_init(struct oxygen *chip) +static void stereo_output_init(struct oxygen *chip) { ak4396_init(chip); } @@ -583,20 +584,20 @@ static int __devinit get_oxygen_model(struct oxygen *chip, CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; break; - case MODEL_HIFIER: - case MODEL_HG2PCI: + case MODEL_FANTASIA: + case MODEL_2CH_OUTPUT: chip->model.shortname = "C-Media CMI8787"; chip->model.chip = "CMI8787"; - if (id->driver_data == MODEL_HIFIER) - chip->model.init = hifier_init; + if (id->driver_data == MODEL_FANTASIA) + chip->model.init = fantasia_init; else - chip->model.init = hg2pci_init; + chip->model.init = stereo_output_init; chip->model.resume = stereo_resume; chip->model.mixer_init = generic_mixer_init; chip->model.set_adc_params = set_no_params; chip->model.device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF; - if (id->driver_data == MODEL_HIFIER) + if (id->driver_data == MODEL_FANTASIA) chip->model.device_config |= CAPTURE_0_FROM_I2S_1; chip->model.dac_channels = 2; break; -- cgit v0.10.2 From 03b7a1ab557efe34e8f79b78660e514bd7374248 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 9 Nov 2010 14:35:30 +0100 Subject: ALSA: HDA: Create mixers on ALC887 BugLink: http://launchpad.net/bugs/669092 ALC887 does not have any volume control ability on the mixer NIDs, so put the volume controls on the dac NIDs instead. Without this patch, ALC887 users cannot use alsamixer at all. Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5f00589..74029b5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10816,6 +10816,9 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) return 0; } +static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg); + /* almost identical with ALC880 parser... */ static int alc882_parse_auto_config(struct hda_codec *codec) { @@ -10833,7 +10836,10 @@ static int alc882_parse_auto_config(struct hda_codec *codec) err = alc880_auto_fill_dac_nids(spec, &spec->autocfg); if (err < 0) return err; - err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (codec->vendor_id == 0x10ec0887) + err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg); + else + err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0], @@ -16963,7 +16969,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) #define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c) /* add playback controls from the parsed DAC table */ -/* Based on ALC880 version. But ALC861VD has separate, +/* Based on ALC880 version. But ALC861VD and ALC887 have separate, * different NIDs for mute/unmute switch and volume control */ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) -- cgit v0.10.2 From 86cbbad2b6712fbd25c07a17e86b4345cee82c6d Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sat, 20 Nov 2010 10:20:35 -0500 Subject: ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup BugLink: https://launchpad.net/bugs/677830 The original reporter states that the subwoofer does not mute when inserting headphones. We need an entry for his machine's SSID in the subwoofer pin fixup list, so add it there (verified using hda_analyzer). Reported-and-tested-by: i-NoD Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 74029b5..b7e2348 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19304,6 +19304,7 @@ static const struct alc_fixup alc662_fixups[] = { static struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), {} -- cgit v0.10.2 From 7974150c8524423f878e8269010e911c3cc7ddb8 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Sun, 21 Nov 2010 12:09:32 +0100 Subject: ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer . Fix PulseAudio "ALSA driver bug" issue (if we have two alternated areas within a 64k DMA buffer, then max period size should obviously be 32k only). Back references: http://pulseaudio.org/wiki/AlsaIssues http://fedoraproject.org/wiki/Features/GlitchFreeAudio . In stop timer function, need to supply ACK in the timer control byte. . Minor log output correction When I did my first PA testing recently, the period size bug resulted in quite precisely observeable half-period-based playback distortion. PA-based operation is quite a bit more underrun-prone (despite its zero-copy optimizations etc.) than raw ALSA with this rather spartan sound hardware implementation on my puny Athlon. Note that even with this patch, azt3328 still doesn't work for both cases yet, PA tsched=0 and tsched (on tsched=0 it will playback tiny fragments of periods, leading to tiny stuttering sounds with some pauses in between, whereas with timer-scheduled operation playback works fine - minus some quite increased underrun trouble on PA vs. ALSA, that is). Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 4679ed8..2f3cacb 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1129,10 +1129,11 @@ snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip, count_areas = size/2; addr_area2 = addr+count_areas; - count_areas--; /* max. index */ snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n", addr, count_areas, addr_area2, count_areas); + count_areas--; /* max. index */ + /* build combined I/O buffer length word */ lengths = (count_areas << 16) | (count_areas); spin_lock_irqsave(&chip->reg_lock, flags); @@ -1740,11 +1741,15 @@ static const struct snd_pcm_hardware snd_azf3328_hardware = .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, - .buffer_bytes_max = 65536, - .period_bytes_min = 64, - .period_bytes_max = 65536, - .periods_min = 1, - .periods_max = 1024, + .buffer_bytes_max = (64*1024), + .period_bytes_min = 1024, + .period_bytes_max = (32*1024), + /* We simply have two DMA areas (instead of a list of descriptors + such as other cards); I believe that this is a fixed hardware + attribute and there isn't much driver magic to be done to expand it. + Thus indicate that we have at least and at most 2 periods. */ + .periods_min = 2, + .periods_max = 2, /* FIXME: maybe that card actually has a FIFO? * Hmm, it seems newer revisions do have one, but we still don't know * its size... */ @@ -1980,8 +1985,13 @@ snd_azf3328_timer_stop(struct snd_timer *timer) chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->reg_lock, flags); /* disable timer countdown and interrupt */ - /* FIXME: should we write TIMER_IRQ_ACK here? */ - snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); + /* Hmm, should we write TIMER_IRQ_ACK here? + YES indeed, otherwise a rogue timer operation - which prompts + ALSA(?) to call repeated stop() in vain, but NOT start() - + will never end (value 0x03 is kept shown in control byte). + Simply manually poking 0x04 _once_ immediately successfully stops + the hardware/ALSA interrupt activity. */ + snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x04); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); return 0; -- cgit v0.10.2 From 5ad57d20c91bdaf743bd8e3015df5a388314df8d Mon Sep 17 00:00:00 2001 From: Vasiliy Kulikov Date: Sun, 21 Nov 2010 20:40:07 +0300 Subject: ALSA: snd-atmel-abdac: test wrong variable After clk_get() pclk is checked second time instead of sample_clk check. Signed-off-by: Vasiliy Kulikov Signed-off-by: Takashi Iwai diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index f2f41c8..4e47baad 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -420,7 +420,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) return PTR_ERR(pclk); } sample_clk = clk_get(&pdev->dev, "sample_clk"); - if (IS_ERR(pclk)) { + if (IS_ERR(sample_clk)) { dev_dbg(&pdev->dev, "no sample clock\n"); retval = PTR_ERR(pclk); goto out_put_pclk; -- cgit v0.10.2 From a1d71a2c91239ecc1c1f9c97a081d71ebd30bfe5 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 21 Nov 2010 14:01:14 -0500 Subject: ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J BugLink: https://launchpad.net/bugs/677652 The original reporter states that, in 2.6.35, headphones do not appear to work, nor does inserting them mute the A52J's onboard speakers. Upon inspecting the codec dump, it appears that the newly committed hp-laptop quirk will suffice to enable this basic functionality. Testing was done with an alsa-driver build from 2010-11-21. Reported-and-tested-by: Joan Creus Cc: [2.6.35+] Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6361f75..3cfb31e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3100,6 +3100,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), -- cgit v0.10.2 From ab69a4904b5dd4d7cd6996587ba066bca8d13838 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 15 Nov 2010 10:46:23 +0100 Subject: ALSA: pcm: support for period wakeup disabling This patch allows to disable period interrupts which are not needed when the application relies on a system timer to wake-up and refill the ring buffer. The behavior of the driver is left unchanged, and interrupts are only disabled if the application requests this configuration. The behavior in case of underruns is slightly different, instead of being detected during the period interrupts the underruns are detected when the application calls snd_pcm_update_avail, which in turns forces a refresh of the hw pointer and shows the buffer is empty. More specifically this patch makes a lot of sense when PulseAudio relies on timer-based scheduling to access audio devices such as HDAudio or Intel SST. Disabling interrupts removes two unwanted wake-ups due to period elapsed events in low-power playback modes. It also simplifies PulseAudio voice modules used for speech calls. To quote Lennart "This patch looks very interesting and desirable. This is something have long been waiting for." Support for this in hardware drivers is optional. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/include/sound/asound.h b/include/sound/asound.h index a1803ec..5d6074f 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -259,6 +259,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_HALF_DUPLEX 0x00100000 /* only half duplex */ #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ +#define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; @@ -334,6 +335,8 @@ typedef int snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_TICK_TIME #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ +#define SNDRV_PCM_HW_PARAMS_EXPORT_BUFFER (1<<1) /* export buffer */ +#define SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP (1<<2) /* disable period wakeups */ struct snd_interval { unsigned int min, max; diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dfd9b76..e731f8d7 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -297,6 +297,7 @@ struct snd_pcm_runtime { unsigned int info; unsigned int rate_num; unsigned int rate_den; + unsigned int no_period_wakeup: 1; /* -- SW params -- */ int tstamp_mode; /* mmap timestamp is updated */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b75db8e..bc57501 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -373,6 +373,11 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (unsigned long)new_hw_ptr, (unsigned long)runtime->hw_ptr_base); } + + /* without period interrupts, there are no regular pointer updates */ + if (runtime->no_period_wakeup) + goto no_delta_check; + /* something must be really wrong */ if (delta >= runtime->buffer_size + runtime->period_size) { hw_ptr_error(substream, @@ -442,6 +447,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (long)old_hw_ptr); } + no_delta_check: if (runtime->status->hw_ptr == new_hw_ptr) return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 8bc7cb3..f91a439 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -423,6 +423,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->info = params->info; runtime->rate_num = params->rate_num; runtime->rate_den = params->rate_den; + runtime->no_period_wakeup = + (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && + (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; -- cgit v0.10.2 From 7bb8fb70c491bd6f5ec99728db8d1b5f43b95471 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 15 Nov 2010 10:49:47 +0100 Subject: ALSA: hda-intel: support for period wakeup disabling Allow disabling period wakeup interrupts for HDA PCM streams. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21aa9b0..a78ea34 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1235,7 +1235,8 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; } else { ofs = setup_bdle(substream, azx_dev, - &bdl, ofs, pos_adj, 1); + &bdl, ofs, pos_adj, + !substream->runtime->no_period_wakeup); if (ofs < 0) goto error; } @@ -1247,7 +1248,8 @@ static int azx_setup_periods(struct azx *chip, period_bytes - pos_adj, 0); else ofs = setup_bdle(substream, azx_dev, &bdl, ofs, - period_bytes, 1); + period_bytes, + !substream->runtime->no_period_wakeup); if (ofs < 0) goto error; } @@ -1515,7 +1517,8 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* No full-resume yet implemented */ /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START), + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, -- cgit v0.10.2 From 075140ea8bf1405057c072a84ccf4e0d3f2c76f5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 15 Nov 2010 10:50:37 +0100 Subject: ALSA: oxygen: support for period wakeup disabling Allow disabling period wakeup interrupts for all PCM streams. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 8146674..60e4aa0 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -39,7 +39,8 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_32000 | @@ -65,7 +66,8 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_32000 | @@ -91,7 +93,8 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -530,7 +533,10 @@ static int oxygen_prepare(struct snd_pcm_substream *substream) oxygen_set_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); oxygen_clear_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); - chip->interrupt_mask |= channel_mask; + if (substream->runtime->no_period_wakeup) + chip->interrupt_mask &= ~channel_mask; + else + chip->interrupt_mask |= channel_mask; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); spin_unlock_irq(&chip->reg_lock); return 0; -- cgit v0.10.2 From 59ff878ffb26bc0be812ca8295799164f413ae88 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 18 Nov 2010 09:43:52 +0100 Subject: ALSA: pcm: detect xruns in no-period-wakeup mode When period wakeups are disabled, successive calls to the pointer update function do not have a maximum allowed distance, so xruns cannot be detected with the pointer value only. To detect xruns, compare the actually elapsed time with the time that should have theoretically elapsed since the last update. When the hardware pointer has wrapped around due to an xrun, the actually elapsed time will be too big by about hw_ptr_buffer_jiffies. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index bc57501..e9debaa 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -374,9 +374,23 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (unsigned long)runtime->hw_ptr_base); } - /* without period interrupts, there are no regular pointer updates */ - if (runtime->no_period_wakeup) + if (runtime->no_period_wakeup) { + /* + * Without regular period interrupts, we have to check + * the elapsed time to detect xruns. + */ + jdelta = jiffies - runtime->hw_ptr_jiffies; + hdelta = jdelta - delta * HZ / runtime->rate; + while (hdelta > runtime->hw_ptr_buffer_jiffies / 2 + 1) { + delta += runtime->buffer_size; + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + hdelta -= runtime->hw_ptr_buffer_jiffies; + } goto no_delta_check; + } /* something must be really wrong */ if (delta >= runtime->buffer_size + runtime->period_size) { -- cgit v0.10.2 From 47228e48aecdbec423a1275a5e27697d47f1f912 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 18 Nov 2010 09:53:07 +0100 Subject: ALSA: pcm: optimize xrun detection in no-period-wakeup mode Add a lightweight condition on top of the xrun checking so that we can avoid the division when the application is calling the update function often enough. Suggested-by: Jaroslav Kysela Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e9debaa..fd18c3c 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -380,6 +380,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, * the elapsed time to detect xruns. */ jdelta = jiffies - runtime->hw_ptr_jiffies; + if (jdelta < runtime->hw_ptr_buffer_jiffies / 2) + goto no_delta_check; hdelta = jdelta - delta * HZ / runtime->rate; while (hdelta > runtime->hw_ptr_buffer_jiffies / 2 + 1) { delta += runtime->buffer_size; -- cgit v0.10.2 From 109fef9edcc100952eec980acbc2e1295627fbab Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 18 Nov 2010 09:53:54 +0100 Subject: ALSA: timer: automatically load the high-resolution timer Increase the default timer limit so that snd-hrtimer.ko can be automatically loaded when needed, e.g., when used as the default sequencer timer. This replaces the check for the obsolete CONFIG_SND_HPET. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/core/timer.c b/sound/core/timer.c index 13afb60..b3aaa60 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -34,8 +34,8 @@ #include #include -#if defined(CONFIG_SND_HPET) || defined(CONFIG_SND_HPET_MODULE) -#define DEFAULT_TIMER_LIMIT 3 +#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE) +#define DEFAULT_TIMER_LIMIT 4 #elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE) #define DEFAULT_TIMER_LIMIT 2 #else -- cgit v0.10.2 From 03cfe6f57dc5c13ccdd235c23c80e3fa170f03d1 Mon Sep 17 00:00:00 2001 From: Kay Sievers Date: Tue, 23 Nov 2010 17:43:19 +0100 Subject: ALSA: support module on-demand loading for seq and timer If CONFIG_SND_DYNAMIC_MINORS is used, assign /dev/snd/seq and /dev/snd/timer the usual static minors, and export specific module aliases to generate udev module on-demand loading instructions: $ cat /lib/modules/2.6.33.4-smp/modules.devname # Device nodes to trigger on-demand module loading. microcode cpu/microcode c10:184 fuse fuse c10:229 ppp_generic ppp c108:0 tun net/tun c10:200 uinput uinput c10:223 dm_mod mapper/control c10:236 snd_timer snd/timer c116:33 snd_seq snd/seq c116:1 The last two lines instruct udev to create device nodes, even when the modules are not loaded at that time. As soon as userspace accesses any of these nodes, the in-kernel module-loader will load the module, and the device can be used. The header file minor calculation needed to be simplified to make __stringify() (supports only two indirections) in the MODULE_ALIAS macro work. This is part of systemd's effort to get rid of unconditional module load instructions and needless init scripts. Cc: Lennart Poettering Signed-off-by: Kay Sievers Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/include/sound/minors.h b/include/sound/minors.h index a81798a..8f76420 100644 --- a/include/sound/minors.h +++ b/include/sound/minors.h @@ -31,8 +31,8 @@ /* these minors can still be used for autoloading devices (/dev/aload*) */ #define SNDRV_MINOR_CONTROL 0 /* 0 */ #define SNDRV_MINOR_GLOBAL 1 /* 1 */ -#define SNDRV_MINOR_SEQUENCER (SNDRV_MINOR_GLOBAL + 0 * 32) -#define SNDRV_MINOR_TIMER (SNDRV_MINOR_GLOBAL + 1 * 32) +#define SNDRV_MINOR_SEQUENCER 1 /* SNDRV_MINOR_GLOBAL + 0 * 32 */ +#define SNDRV_MINOR_TIMER 33 /* SNDRV_MINOR_GLOBAL + 1 * 32 */ #ifndef CONFIG_SND_DYNAMIC_MINORS /* 2 - 3 (reserved) */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index bf09a5a..119fddb6 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -32,6 +32,7 @@ #include "seq_timer.h" #include "seq_system.h" #include "seq_info.h" +#include #include #if defined(CONFIG_SND_SEQ_DUMMY_MODULE) @@ -73,6 +74,9 @@ MODULE_PARM_DESC(seq_default_timer_subdevice, "The default timer subdevice numbe module_param(seq_default_timer_resolution, int, 0644); MODULE_PARM_DESC(seq_default_timer_resolution, "The default timer resolution in Hz."); +MODULE_ALIAS_CHARDEV(CONFIG_SND_MAJOR, SNDRV_MINOR_SEQUENCER); +MODULE_ALIAS("devname:snd/seq"); + /* * INIT PART */ diff --git a/sound/core/sound.c b/sound/core/sound.c index 62a093e..345caea 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -189,14 +189,22 @@ static const struct file_operations snd_fops = }; #ifdef CONFIG_SND_DYNAMIC_MINORS -static int snd_find_free_minor(void) +static int snd_find_free_minor(int type) { int minor; + /* static minors for module auto loading */ + if (type == SNDRV_DEVICE_TYPE_SEQUENCER) + return SNDRV_MINOR_SEQUENCER; + if (type == SNDRV_DEVICE_TYPE_TIMER) + return SNDRV_MINOR_TIMER; + for (minor = 0; minor < ARRAY_SIZE(snd_minors); ++minor) { - /* skip minors still used statically for autoloading devices */ - if (SNDRV_MINOR_DEVICE(minor) == SNDRV_MINOR_CONTROL || - minor == SNDRV_MINOR_SEQUENCER) + /* skip static minors still used for module auto loading */ + if (SNDRV_MINOR_DEVICE(minor) == SNDRV_MINOR_CONTROL) + continue; + if (minor == SNDRV_MINOR_SEQUENCER || + minor == SNDRV_MINOR_TIMER) continue; if (!snd_minors[minor]) return minor; @@ -270,7 +278,7 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, preg->private_data = private_data; mutex_lock(&sound_mutex); #ifdef CONFIG_SND_DYNAMIC_MINORS - minor = snd_find_free_minor(); + minor = snd_find_free_minor(type); #else minor = snd_kernel_minor(type, card, dev); if (minor >= 0 && snd_minors[minor]) diff --git a/sound/core/timer.c b/sound/core/timer.c index b3aaa60..ed01632 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -52,6 +52,9 @@ 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)."); +MODULE_ALIAS_CHARDEV(CONFIG_SND_MAJOR, SNDRV_MINOR_TIMER); +MODULE_ALIAS("devname:snd/timer"); + struct snd_timer_user { struct snd_timer_instance *timeri; int tread; /* enhanced read with timestamps and events */ -- cgit v0.10.2 From 28b26e15533e60970a014582d812d471ad63bcd0 Mon Sep 17 00:00:00 2001 From: Florian Faber Date: Wed, 1 Dec 2010 12:14:47 +0100 Subject: ALSA: hdsp - Add support for RPM io box Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added. Signed-off-by: Florian Faber Signed-off-by: Takashi Iwai diff --git a/include/sound/hdsp.h b/include/sound/hdsp.h index d98a78d..0909a38 100644 --- a/include/sound/hdsp.h +++ b/include/sound/hdsp.h @@ -28,6 +28,7 @@ enum HDSP_IO_Type { Multiface, H9652, H9632, + RPM, Undefined, }; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 0b720cf..2d83324 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -60,6 +60,7 @@ MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP}," "{RME HDSP-9652}," "{RME HDSP-9632}}"); #ifdef HDSP_FW_LOADER +MODULE_FIRMWARE("rpm_firmware.bin"); MODULE_FIRMWARE("multiface_firmware.bin"); MODULE_FIRMWARE("multiface_firmware_rev11.bin"); MODULE_FIRMWARE("digiface_firmware.bin"); @@ -81,6 +82,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define H9632_SS_CHANNELS 12 #define H9632_DS_CHANNELS 8 #define H9632_QS_CHANNELS 4 +#define RPM_CHANNELS 6 /* Write registers. These are defined as byte-offsets from the iobase value. */ @@ -191,6 +193,25 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_PhoneGain1 (1<<30) #define HDSP_QuadSpeed (1<<31) +/* RPM uses some of the registers for special purposes */ +#define HDSP_RPM_Inp12 0x04A00 +#define HDSP_RPM_Inp12_Phon_6dB 0x00800 /* Dolby */ +#define HDSP_RPM_Inp12_Phon_0dB 0x00000 /* .. */ +#define HDSP_RPM_Inp12_Phon_n6dB 0x04000 /* inp_0 */ +#define HDSP_RPM_Inp12_Line_0dB 0x04200 /* Dolby+PRO */ +#define HDSP_RPM_Inp12_Line_n6dB 0x00200 /* PRO */ + +#define HDSP_RPM_Inp34 0x32000 +#define HDSP_RPM_Inp34_Phon_6dB 0x20000 /* SyncRef1 */ +#define HDSP_RPM_Inp34_Phon_0dB 0x00000 /* .. */ +#define HDSP_RPM_Inp34_Phon_n6dB 0x02000 /* SyncRef2 */ +#define HDSP_RPM_Inp34_Line_0dB 0x30000 /* SyncRef1+SyncRef0 */ +#define HDSP_RPM_Inp34_Line_n6dB 0x10000 /* SyncRef0 */ + +#define HDSP_RPM_Bypass 0x01000 + +#define HDSP_RPM_Disconnect 0x00001 + #define HDSP_ADGainMask (HDSP_ADGain0|HDSP_ADGain1) #define HDSP_ADGainMinus10dBV HDSP_ADGainMask #define HDSP_ADGainPlus4dBu (HDSP_ADGain0) @@ -450,7 +471,7 @@ struct hdsp { u32 creg_spdif; u32 creg_spdif_stream; int clock_source_locked; - char *card_name; /* digiface/multiface */ + char *card_name; /* digiface/multiface/rpm */ enum HDSP_IO_Type io_type; /* ditto, but for code use */ unsigned short firmware_rev; unsigned short state; /* stores state bits */ @@ -612,6 +633,7 @@ static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) switch (hdsp->io_type) { case Multiface: case Digiface: + case RPM: default: if (hdsp->firmware_rev == 0xa) return (64 * out) + (32 + (in)); @@ -629,6 +651,7 @@ static int hdsp_input_to_output_key (struct hdsp *hdsp, int in, int out) switch (hdsp->io_type) { case Multiface: case Digiface: + case RPM: default: if (hdsp->firmware_rev == 0xa) return (64 * out) + in; @@ -655,7 +678,7 @@ static int hdsp_check_for_iobox (struct hdsp *hdsp) { if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { - snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n"); + snd_printk("Hammerfall-DSP: no IO box connected!\n"); hdsp->state &= ~HDSP_FirmwareLoaded; return -EIO; } @@ -680,7 +703,7 @@ static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, } } - snd_printk("Hammerfall-DSP: no Digiface or Multiface connected!\n"); + snd_printk("Hammerfall-DSP: no IO box connected!\n"); hdsp->state &= ~HDSP_FirmwareLoaded; return -EIO; } @@ -752,17 +775,21 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); hdsp_write (hdsp, HDSP_fifoData, 0); - if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT)) { - hdsp->io_type = Multiface; - hdsp_write (hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); - hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); - hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { + hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); + hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) + hdsp->io_type = RPM; + else + hdsp->io_type = Multiface; } else { hdsp->io_type = Digiface; } } else { /* firmware was already loaded, get iobox type */ - if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) + if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version2) + hdsp->io_type = RPM; + else if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; else hdsp->io_type = Digiface; @@ -1184,6 +1211,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) hdsp->channel_map = channel_map_ds; } else { switch (hdsp->io_type) { + case RPM: case Multiface: hdsp->channel_map = channel_map_mf_ss; break; @@ -3231,6 +3259,318 @@ HDSP_PRECISE_POINTER("Precise Pointer", 0), HDSP_USE_MIDI_TASKLET("Use Midi Tasklet", 0), }; + +static int hdsp_rpm_input12(struct hdsp *hdsp) +{ + switch (hdsp->control_register & HDSP_RPM_Inp12) { + case HDSP_RPM_Inp12_Phon_6dB: + return 0; + case HDSP_RPM_Inp12_Phon_n6dB: + return 2; + case HDSP_RPM_Inp12_Line_0dB: + return 3; + case HDSP_RPM_Inp12_Line_n6dB: + return 4; + } + return 1; +} + + +static int snd_hdsp_get_rpm_input12(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = hdsp_rpm_input12(hdsp); + return 0; +} + + +static int hdsp_set_rpm_input12(struct hdsp *hdsp, int mode) +{ + hdsp->control_register &= ~HDSP_RPM_Inp12; + switch (mode) { + case 0: + hdsp->control_register |= HDSP_RPM_Inp12_Phon_6dB; + break; + case 1: + break; + case 2: + hdsp->control_register |= HDSP_RPM_Inp12_Phon_n6dB; + break; + case 3: + hdsp->control_register |= HDSP_RPM_Inp12_Line_0dB; + break; + case 4: + hdsp->control_register |= HDSP_RPM_Inp12_Line_n6dB; + break; + default: + return -1; + } + + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); + return 0; +} + + +static int snd_hdsp_put_rpm_input12(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + int change; + int val; + + if (!snd_hdsp_use_is_exclusive(hdsp)) + return -EBUSY; + val = ucontrol->value.enumerated.item[0]; + if (val < 0) + val = 0; + if (val > 4) + val = 4; + spin_lock_irq(&hdsp->lock); + if (val != hdsp_rpm_input12(hdsp)) + change = (hdsp_set_rpm_input12(hdsp, val) == 0) ? 1 : 0; + else + change = 0; + spin_unlock_irq(&hdsp->lock); + return change; +} + + +static int snd_hdsp_info_rpm_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = {"Phono +6dB", "Phono 0dB", "Phono -6dB", "Line 0dB", "Line -6dB"}; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 5; + 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 hdsp_rpm_input34(struct hdsp *hdsp) +{ + switch (hdsp->control_register & HDSP_RPM_Inp34) { + case HDSP_RPM_Inp34_Phon_6dB: + return 0; + case HDSP_RPM_Inp34_Phon_n6dB: + return 2; + case HDSP_RPM_Inp34_Line_0dB: + return 3; + case HDSP_RPM_Inp34_Line_n6dB: + return 4; + } + return 1; +} + + +static int snd_hdsp_get_rpm_input34(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = hdsp_rpm_input34(hdsp); + return 0; +} + + +static int hdsp_set_rpm_input34(struct hdsp *hdsp, int mode) +{ + hdsp->control_register &= ~HDSP_RPM_Inp34; + switch (mode) { + case 0: + hdsp->control_register |= HDSP_RPM_Inp34_Phon_6dB; + break; + case 1: + break; + case 2: + hdsp->control_register |= HDSP_RPM_Inp34_Phon_n6dB; + break; + case 3: + hdsp->control_register |= HDSP_RPM_Inp34_Line_0dB; + break; + case 4: + hdsp->control_register |= HDSP_RPM_Inp34_Line_n6dB; + break; + default: + return -1; + } + + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); + return 0; +} + + +static int snd_hdsp_put_rpm_input34(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + int change; + int val; + + if (!snd_hdsp_use_is_exclusive(hdsp)) + return -EBUSY; + val = ucontrol->value.enumerated.item[0]; + if (val < 0) + val = 0; + if (val > 4) + val = 4; + spin_lock_irq(&hdsp->lock); + if (val != hdsp_rpm_input34(hdsp)) + change = (hdsp_set_rpm_input34(hdsp, val) == 0) ? 1 : 0; + else + change = 0; + spin_unlock_irq(&hdsp->lock); + return change; +} + + +/* RPM Bypass switch */ +static int hdsp_rpm_bypass(struct hdsp *hdsp) +{ + return (hdsp->control_register & HDSP_RPM_Bypass) ? 1 : 0; +} + + +static int snd_hdsp_get_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = hdsp_rpm_bypass(hdsp); + return 0; +} + + +static int hdsp_set_rpm_bypass(struct hdsp *hdsp, int on) +{ + if (on) + hdsp->control_register |= HDSP_RPM_Bypass; + else + hdsp->control_register &= ~HDSP_RPM_Bypass; + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); + return 0; +} + + +static int snd_hdsp_put_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + int change; + unsigned int val; + + if (!snd_hdsp_use_is_exclusive(hdsp)) + return -EBUSY; + val = ucontrol->value.integer.value[0] & 1; + spin_lock_irq(&hdsp->lock); + change = (int)val != hdsp_rpm_bypass(hdsp); + hdsp_set_rpm_bypass(hdsp, val); + spin_unlock_irq(&hdsp->lock); + return change; +} + + +static int snd_hdsp_info_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = {"On", "Off"}; + + 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; +} + + +/* RPM Disconnect switch */ +static int hdsp_rpm_disconnect(struct hdsp *hdsp) +{ + return (hdsp->control_register & HDSP_RPM_Disconnect) ? 1 : 0; +} + + +static int snd_hdsp_get_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = hdsp_rpm_disconnect(hdsp); + return 0; +} + + +static int hdsp_set_rpm_disconnect(struct hdsp *hdsp, int on) +{ + if (on) + hdsp->control_register |= HDSP_RPM_Disconnect; + else + hdsp->control_register &= ~HDSP_RPM_Disconnect; + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); + return 0; +} + + +static int snd_hdsp_put_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + int change; + unsigned int val; + + if (!snd_hdsp_use_is_exclusive(hdsp)) + return -EBUSY; + val = ucontrol->value.integer.value[0] & 1; + spin_lock_irq(&hdsp->lock); + change = (int)val != hdsp_rpm_disconnect(hdsp); + hdsp_set_rpm_disconnect(hdsp, val); + spin_unlock_irq(&hdsp->lock); + return change; +} + +static int snd_hdsp_info_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = {"On", "Off"}; + + 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 struct snd_kcontrol_new snd_hdsp_rpm_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "RPM Bypass", + .get = snd_hdsp_get_rpm_bypass, + .put = snd_hdsp_put_rpm_bypass, + .info = snd_hdsp_info_rpm_bypass + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "RPM Disconnect", + .get = snd_hdsp_get_rpm_disconnect, + .put = snd_hdsp_put_rpm_disconnect, + .info = snd_hdsp_info_rpm_disconnect + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 1/2", + .get = snd_hdsp_get_rpm_input12, + .put = snd_hdsp_put_rpm_input12, + .info = snd_hdsp_info_rpm_input + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 3/4", + .get = snd_hdsp_get_rpm_input34, + .put = snd_hdsp_put_rpm_input34, + .info = snd_hdsp_info_rpm_input + }, + HDSP_SYSTEM_SAMPLE_RATE("System Sample Rate", 0), + HDSP_MIXER("Mixer", 0) +}; + static struct snd_kcontrol_new snd_hdsp_96xx_aeb = HDSP_AEB("Analog Extension Board", 0); static struct snd_kcontrol_new snd_hdsp_adat_sync_check = HDSP_ADAT_SYNC_CHECK; @@ -3240,6 +3580,16 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) int err; struct snd_kcontrol *kctl; + if (hdsp->io_type == RPM) { + /* RPM Bypass, Disconnect and Input switches */ + for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_rpm_controls); idx++) { + err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_rpm_controls[idx], hdsp)); + if (err < 0) + return err; + } + return 0; + } + for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_controls); idx++) { if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_controls[idx], hdsp))) < 0) return err; @@ -3459,48 +3809,102 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "\n"); - switch (hdsp_spdif_in(hdsp)) { - case HDSP_SPDIFIN_OPTICAL: - snd_iprintf(buffer, "IEC958 input: Optical\n"); - break; - case HDSP_SPDIFIN_COAXIAL: - snd_iprintf(buffer, "IEC958 input: Coaxial\n"); - break; - case HDSP_SPDIFIN_INTERNAL: - snd_iprintf(buffer, "IEC958 input: Internal\n"); - break; - case HDSP_SPDIFIN_AES: - snd_iprintf(buffer, "IEC958 input: AES\n"); - break; - default: - snd_iprintf(buffer, "IEC958 input: ???\n"); - break; + if (hdsp->io_type != RPM) { + switch (hdsp_spdif_in(hdsp)) { + case HDSP_SPDIFIN_OPTICAL: + snd_iprintf(buffer, "IEC958 input: Optical\n"); + break; + case HDSP_SPDIFIN_COAXIAL: + snd_iprintf(buffer, "IEC958 input: Coaxial\n"); + break; + case HDSP_SPDIFIN_INTERNAL: + snd_iprintf(buffer, "IEC958 input: Internal\n"); + break; + case HDSP_SPDIFIN_AES: + snd_iprintf(buffer, "IEC958 input: AES\n"); + break; + default: + snd_iprintf(buffer, "IEC958 input: ???\n"); + break; + } } - if (hdsp->control_register & HDSP_SPDIFOpticalOut) - snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); - else - snd_iprintf(buffer, "IEC958 output: Coaxial only\n"); + if (RPM == hdsp->io_type) { + if (hdsp->control_register & HDSP_RPM_Bypass) + snd_iprintf(buffer, "RPM Bypass: disabled\n"); + else + snd_iprintf(buffer, "RPM Bypass: enabled\n"); + if (hdsp->control_register & HDSP_RPM_Disconnect) + snd_iprintf(buffer, "RPM disconnected\n"); + else + snd_iprintf(buffer, "RPM connected\n"); - if (hdsp->control_register & HDSP_SPDIFProfessional) - snd_iprintf(buffer, "IEC958 quality: Professional\n"); - else - snd_iprintf(buffer, "IEC958 quality: Consumer\n"); + switch (hdsp->control_register & HDSP_RPM_Inp12) { + case HDSP_RPM_Inp12_Phon_6dB: + snd_iprintf(buffer, "Input 1/2: Phono, 6dB\n"); + break; + case HDSP_RPM_Inp12_Phon_0dB: + snd_iprintf(buffer, "Input 1/2: Phono, 0dB\n"); + break; + case HDSP_RPM_Inp12_Phon_n6dB: + snd_iprintf(buffer, "Input 1/2: Phono, -6dB\n"); + break; + case HDSP_RPM_Inp12_Line_0dB: + snd_iprintf(buffer, "Input 1/2: Line, 0dB\n"); + break; + case HDSP_RPM_Inp12_Line_n6dB: + snd_iprintf(buffer, "Input 1/2: Line, -6dB\n"); + break; + default: + snd_iprintf(buffer, "Input 1/2: ???\n"); + } - if (hdsp->control_register & HDSP_SPDIFEmphasis) - snd_iprintf(buffer, "IEC958 emphasis: on\n"); - else - snd_iprintf(buffer, "IEC958 emphasis: off\n"); + switch (hdsp->control_register & HDSP_RPM_Inp34) { + case HDSP_RPM_Inp34_Phon_6dB: + snd_iprintf(buffer, "Input 3/4: Phono, 6dB\n"); + break; + case HDSP_RPM_Inp34_Phon_0dB: + snd_iprintf(buffer, "Input 3/4: Phono, 0dB\n"); + break; + case HDSP_RPM_Inp34_Phon_n6dB: + snd_iprintf(buffer, "Input 3/4: Phono, -6dB\n"); + break; + case HDSP_RPM_Inp34_Line_0dB: + snd_iprintf(buffer, "Input 3/4: Line, 0dB\n"); + break; + case HDSP_RPM_Inp34_Line_n6dB: + snd_iprintf(buffer, "Input 3/4: Line, -6dB\n"); + break; + default: + snd_iprintf(buffer, "Input 3/4: ???\n"); + } - if (hdsp->control_register & HDSP_SPDIFNonAudio) - snd_iprintf(buffer, "IEC958 NonAudio: on\n"); - else - snd_iprintf(buffer, "IEC958 NonAudio: off\n"); - if ((x = hdsp_spdif_sample_rate (hdsp)) != 0) - snd_iprintf (buffer, "IEC958 sample rate: %d\n", x); - else - snd_iprintf (buffer, "IEC958 sample rate: Error flag set\n"); + } else { + if (hdsp->control_register & HDSP_SPDIFOpticalOut) + snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); + else + snd_iprintf(buffer, "IEC958 output: Coaxial only\n"); + + if (hdsp->control_register & HDSP_SPDIFProfessional) + snd_iprintf(buffer, "IEC958 quality: Professional\n"); + else + snd_iprintf(buffer, "IEC958 quality: Consumer\n"); + + if (hdsp->control_register & HDSP_SPDIFEmphasis) + snd_iprintf(buffer, "IEC958 emphasis: on\n"); + else + snd_iprintf(buffer, "IEC958 emphasis: off\n"); + if (hdsp->control_register & HDSP_SPDIFNonAudio) + snd_iprintf(buffer, "IEC958 NonAudio: on\n"); + else + snd_iprintf(buffer, "IEC958 NonAudio: off\n"); + x = hdsp_spdif_sample_rate(hdsp); + if (x != 0) + snd_iprintf(buffer, "IEC958 sample rate: %d\n", x); + else + snd_iprintf(buffer, "IEC958 sample rate: Error flag set\n"); + } snd_iprintf(buffer, "\n"); /* Sync Check */ @@ -3765,7 +4169,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) snd_hdsp_midi_input_read (&hdsp->midi[0]); } } - if (hdsp->io_type != Multiface && hdsp->io_type != H9632 && midi1 && midi1status) { + if (hdsp->io_type != Multiface && hdsp->io_type != RPM && hdsp->io_type != H9632 && midi1 && midi1status) { if (hdsp->use_midi_tasklet) { /* we disable interrupts for this input until processing is done */ hdsp->control_register &= ~HDSP_Midi1InterruptEnable; @@ -4093,7 +4497,7 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo = SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, - .channels_min = 14, + .channels_min = 6, .channels_max = HDSP_MAX_CHANNELS, .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS, .period_bytes_min = (64 * 4) * 10, @@ -4122,7 +4526,7 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo = SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, - .channels_min = 14, + .channels_min = 5, .channels_max = HDSP_MAX_CHANNELS, .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS, .period_bytes_min = (64 * 4) * 10, @@ -4357,10 +4761,12 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) snd_hdsp_hw_rule_rate_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - hdsp->creg_spdif_stream = hdsp->creg_spdif; - hdsp->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; - snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE | - SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id); + if (RPM != hdsp->io_type) { + hdsp->creg_spdif_stream = hdsp->creg_spdif; + hdsp->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id); + } return 0; } @@ -4375,9 +4781,11 @@ static int snd_hdsp_playback_release(struct snd_pcm_substream *substream) spin_unlock_irq(&hdsp->lock); - hdsp->spdif_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; - snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE | - SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id); + if (RPM != hdsp->io_type) { + hdsp->spdif_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id); + } return 0; } @@ -4616,7 +5024,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (hdsp->io_type != H9632) info.adatsync_sync_check = (unsigned char)hdsp_adatsync_sync_check(hdsp); info.spdif_sync_check = (unsigned char)hdsp_spdif_sync_check(hdsp); - for (i = 0; i < ((hdsp->io_type != Multiface && hdsp->io_type != H9632) ? 3 : 1); ++i) + for (i = 0; i < ((hdsp->io_type != Multiface && hdsp->io_type != RPM && hdsp->io_type != H9632) ? 3 : 1); ++i) info.adat_sync_check[i] = (unsigned char)hdsp_adat_sync_check(hdsp, i); info.spdif_in = (unsigned char)hdsp_spdif_in(hdsp); info.spdif_out = (unsigned char)hdsp_spdif_out(hdsp); @@ -4636,6 +5044,9 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp); info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp); + } else if (hdsp->io_type == RPM) { + info.da_gain = (unsigned char) hdsp_rpm_input12(hdsp); + info.ad_gain = (unsigned char) hdsp_rpm_input34(hdsp); } if (hdsp->io_type == H9632 || hdsp->io_type == H9652) info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp); @@ -4844,6 +5255,14 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS; break; + case RPM: + hdsp->card_name = "RME Hammerfall DSP + RPM"; + hdsp->ss_in_channels = RPM_CHANNELS-1; + hdsp->ss_out_channels = RPM_CHANNELS; + hdsp->ds_in_channels = RPM_CHANNELS-1; + hdsp->ds_out_channels = RPM_CHANNELS; + break; + default: /* should never get here */ break; @@ -4930,6 +5349,9 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) /* caution: max length of firmware filename is 30! */ switch (hdsp->io_type) { + case RPM: + fwfile = "rpm_firmware.bin"; + break; case Multiface: if (hdsp->firmware_rev == 0xa) fwfile = "multiface_firmware.bin"; @@ -5100,7 +5522,9 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return 0; } else { snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); - if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) + if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version2) + hdsp->io_type = RPM; + else if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; else hdsp->io_type = Digiface; -- cgit v0.10.2 From f7e4bad74e1b18aaff6e89cf2bc4a3868a6ba56e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Dec 2010 11:36:51 +0100 Subject: ALSA: virtuoso: initialize unknown GPIO bits Initialize the configuration of some unknown GPIO output bits (that might not be used at all) to be the same as in the Windows driver, just to be sure. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index aa27c310..ae4e5b5 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -63,6 +63,7 @@ #define GPI_EXT_POWER 0x01 #define GPIO_D1_OUTPUT_ENABLE 0x0001 #define GPIO_D1_FRONT_PANEL 0x0002 +#define GPIO_D1_MAGIC 0x00c0 #define GPIO_D1_INPUT_ROUTE 0x0100 #define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ @@ -169,7 +170,9 @@ static void xonar_d1_init(struct oxygen *chip) cs43xx_registers_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); + GPIO_D1_FRONT_PANEL | + GPIO_D1_MAGIC | + GPIO_D1_INPUT_ROUTE); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index d491fd6..fe4b265 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -167,12 +167,14 @@ #define GPIO_INPUT_ROUTE 0x0100 #define GPIO_HDAV_OUTPUT_ENABLE 0x0001 +#define GPIO_HDAV_MAGIC 0x00c0 #define GPIO_DB_MASK 0x0030 #define GPIO_DB_H6 0x0000 #define GPIO_ST_OUTPUT_ENABLE 0x0001 #define GPIO_ST_HP_REAR 0x0002 +#define GPIO_ST_MAGIC 0x0040 #define GPIO_ST_HP 0x0080 #define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ii, /W=0 */ @@ -350,7 +352,8 @@ static void xonar_hdav_init(struct oxygen *chip) pcm1796_init(chip); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_HDAV_MAGIC | GPIO_INPUT_ROUTE); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_INPUT_ROUTE); xonar_init_cs53x1(chip); @@ -381,7 +384,8 @@ static void xonar_st_init_common(struct oxygen *chip) pcm1796_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | + GPIO_ST_MAGIC | GPIO_ST_HP); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); -- cgit v0.10.2 From 2509ec623d44320419d44d4060dbedf91b8d192d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Dec 2010 11:38:06 +0100 Subject: ALSA: virtuoso: add HDMI enable switch for HDAV1.3 The GPIO bit that enables analog output on the Xonar HDAV1.3 also disables the HDMI audio output, so we better add a switch for it. Hopefully, this is sufficient to make the HDMI output work. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 5add96b..7b2678a 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -817,7 +817,7 @@ config SND_VIRTUOSO Say Y here to include support for sound cards based on the Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), and Essence STX. - Support for the HDAV1.3 (Deluxe) is incomplete; for the + Support for the HDAV1.3 (Deluxe) is experimental; for the HDAV1.3 Slim and Xense, missing. To compile this driver as a module, choose M here: the module diff --git a/sound/pci/oxygen/xonar.h b/sound/pci/oxygen/xonar.h index b35343b..0434c20 100644 --- a/sound/pci/oxygen/xonar.h +++ b/sound/pci/oxygen/xonar.h @@ -24,6 +24,8 @@ void xonar_init_ext_power(struct oxygen *chip); void xonar_init_cs53x1(struct oxygen *chip); void xonar_set_cs53x1_params(struct oxygen *chip, struct snd_pcm_hw_params *params); + +#define XONAR_GPIO_BIT_INVERT (1 << 16) int xonar_gpio_bit_switch_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value); int xonar_gpio_bit_switch_put(struct snd_kcontrol *ctl, diff --git a/sound/pci/oxygen/xonar_lib.c b/sound/pci/oxygen/xonar_lib.c index b3ff713..0ebe7f5 100644 --- a/sound/pci/oxygen/xonar_lib.c +++ b/sound/pci/oxygen/xonar_lib.c @@ -104,9 +104,10 @@ int xonar_gpio_bit_switch_get(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; u16 bit = ctl->private_value; + bool invert = ctl->private_value & XONAR_GPIO_BIT_INVERT; value->value.integer.value[0] = - !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit); + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit) ^ invert; return 0; } @@ -115,12 +116,13 @@ int xonar_gpio_bit_switch_put(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; u16 bit = ctl->private_value; + bool invert = ctl->private_value & XONAR_GPIO_BIT_INVERT; 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]) + if (!!value->value.integer.value[0] ^ invert) new_bits = old_bits | bit; else new_bits = old_bits & ~bit; diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index fe4b265..3850834 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -776,6 +776,15 @@ static const struct snd_kcontrol_new os_128_control = { .put = os_128_put, }; +static const struct snd_kcontrol_new hdav_hdmi_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "HDMI Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = xonar_gpio_bit_switch_get, + .put = xonar_gpio_bit_switch_put, + .private_value = GPIO_HDAV_OUTPUT_ENABLE | XONAR_GPIO_BIT_INVERT, +}; + static int st_output_switch_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -960,7 +969,15 @@ static int xonar_d2_mixer_init(struct oxygen *chip) static int xonar_hdav_mixer_init(struct oxygen *chip) { - return add_pcm1796_controls(chip); + int err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&hdav_hdmi_control, chip)); + if (err < 0) + return err; + err = add_pcm1796_controls(chip); + if (err < 0) + return err; + return 0; } static int xonar_st_mixer_init(struct oxygen *chip) -- cgit v0.10.2 From e96f38f732d24515792296b3738842934c985539 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Dec 2010 11:39:34 +0100 Subject: ALSA: virtuoso: fix front panel routing for D1/DX/ST(X) The "Front Panel" switch on the Xonar D1/DX actually switches only the output direction, so mark it appropriately. The front panel microphone is controlled by the FMIC2MIC bit of the CM9780. It was unconditionally enabled on the D1/DX and never set on the ST(X); add a control for it. Selecting the front panel microphone as source does not actually disable the microphone jack, but this is bug-compatible with the Windows driver, and users rely on it. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 7d5222c..cf9054e 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -35,6 +35,7 @@ #define MIDI_OUTPUT 0x0800 #define MIDI_INPUT 0x1000 #define AC97_CD_INPUT 0x2000 +#define AC97_FMIC_SWITCH 0x4000 enum { CONTROL_SPDIF_PCM, diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 2849b36..605e84b 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -644,6 +644,51 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, return change; } +static int mic_fmic_source_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[] = { "Mic Jack", "Front Panel" }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + info->value.enumerated.item &= 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int mic_fmic_source_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] = + !!(oxygen_read_ac97(chip, 0, CM9780_JACK) & CM9780_FMIC2MIC); + mutex_unlock(&chip->mutex); + return 0; +} + +static int mic_fmic_source_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, 0, CM9780_JACK); + if (value->value.enumerated.item[0]) + newreg = oldreg | CM9780_FMIC2MIC; + else + newreg = oldreg & ~CM9780_FMIC2MIC; + change = newreg != oldreg; + if (change) + oxygen_write_ac97(chip, 0, CM9780_JACK, newreg); + mutex_unlock(&chip->mutex); + return change; +} + static int ac97_fp_rec_volume_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -908,6 +953,13 @@ static const struct snd_kcontrol_new ac97_controls[] = { AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC, 0), AC97_SWITCH("Mic Capture Switch", 0, AC97_MIC, 15, 1), AC97_SWITCH("Mic Boost (+20dB)", 0, AC97_MIC, 6, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Source Capture Enum", + .info = mic_fmic_source_info, + .get = mic_fmic_source_get, + .put = mic_fmic_source_put, + }, AC97_SWITCH("Line Capture Switch", 0, AC97_LINE, 15, 1), AC97_VOLUME("CD Capture Volume", 0, AC97_CD, 1), AC97_SWITCH("CD Capture Switch", 0, AC97_CD, 15, 1), @@ -972,6 +1024,9 @@ static int add_controls(struct oxygen *chip, if (!strcmp(template.name, "Stereo Upmixing") && chip->model.dac_channels == 2) continue; + if (!strcmp(template.name, "Mic Source Capture Enum") && + !(chip->model.device_config & AC97_FMIC_SWITCH)) + continue; if (!strncmp(template.name, "CD Capture ", 11) && !(chip->model.device_config & AC97_CD_INPUT)) continue; diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index ae4e5b5..501fe45 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -28,7 +28,7 @@ * GPI 0 <- external power present (DX only) * * GPIO 0 -> enable output to speakers - * GPIO 1 -> enable front panel I/O + * GPIO 1 -> route output to front panel * GPIO 2 -> M0 of CS5361 * GPIO 3 -> M1 of CS5361 * GPIO 8 -> route input jack to line-in (0) or mic-in (1) @@ -176,8 +176,6 @@ static void xonar_d1_init(struct oxygen *chip) oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); - oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - xonar_init_cs53x1(chip); xonar_enable_output(chip); @@ -287,7 +285,7 @@ static void update_cs43xx_center_lfe_mix(struct oxygen *chip, bool mixed) static const struct snd_kcontrol_new front_panel_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Front Panel Switch", + .name = "Front Panel Playback Switch", .info = snd_ctl_boolean_mono_info, .get = xonar_gpio_bit_switch_get, .put = xonar_gpio_bit_switch_put, @@ -402,7 +400,8 @@ static const struct oxygen_model model_xonar_d1 = { .model_data_size = sizeof(struct xonar_cs43xx), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, + CAPTURE_0_FROM_I2S_2 | + AC97_FMIC_SWITCH, .dac_channels = 8, .dac_volume_min = 127 - 60, .dac_volume_max = 127, diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 3850834..5193d73 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -1079,7 +1079,8 @@ static const struct oxygen_model model_xonar_st = { .model_data_size = sizeof(struct xonar_pcm179x), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, + CAPTURE_0_FROM_I2S_2 | + AC97_FMIC_SWITCH, .dac_channels = 2, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, -- cgit v0.10.2 From 9719fcaa6a82be59a2d7767725e5cd8233c6a387 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Dec 2010 11:41:10 +0100 Subject: ALSA: oxygen: allow to dump codec registers To help with debugging, add the registers of the model-specific codecs to the controller and AC97 register dump in the proc file. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ea8fffe..a58e448 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -41,6 +41,7 @@ #include #include #include +#include #include #include #include @@ -518,6 +519,39 @@ static int generic_wm8785_mixer_init(struct oxygen *chip) return 0; } +static void dump_ak4396_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct generic_data *data = chip->model_data; + unsigned int dac, i; + + for (dac = 0; dac < data->dacs; ++dac) { + snd_iprintf(buffer, "\nAK4396 %u:", dac + 1); + for (i = 0; i < 5; ++i) + snd_iprintf(buffer, " %02x", data->ak4396_regs[dac][i]); + } + snd_iprintf(buffer, "\n"); +} + +static void dump_wm8785_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct generic_data *data = chip->model_data; + unsigned int i; + + snd_iprintf(buffer, "\nWM8785:"); + for (i = 0; i < 3; ++i) + snd_iprintf(buffer, " %03x", data->wm8785_regs[i]); + snd_iprintf(buffer, "\n"); +} + +static void dump_oxygen_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + dump_ak4396_registers(chip, buffer); + dump_wm8785_registers(chip, buffer); +} + static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static const struct oxygen_model model_generic = { @@ -533,6 +567,7 @@ 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, + .dump_registers = dump_oxygen_registers, .dac_tlv = ak4396_db_scale, .model_data_size = sizeof(struct generic_data), .device_config = PLAYBACK_0_TO_I2S | @@ -561,6 +596,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.mixer_init = generic_mixer_init; chip->model.resume = meridian_resume; chip->model.set_adc_params = set_ak5385_params; + chip->model.dump_registers = dump_ak4396_registers; chip->model.device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | @@ -579,6 +615,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; chip->model.set_adc_params = set_ak5385_params; + chip->model.dump_registers = dump_ak4396_registers; chip->model.device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | @@ -595,6 +632,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.resume = stereo_resume; chip->model.mixer_init = generic_mixer_init; chip->model.set_adc_params = set_no_params; + chip->model.dump_registers = dump_ak4396_registers; chip->model.device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF; if (id->driver_data == MODEL_FANTASIA) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index cf9054e..b8fbc15 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -66,6 +66,7 @@ struct snd_pcm_hardware; struct snd_pcm_hw_params; struct snd_kcontrol_new; struct snd_rawmidi; +struct snd_info_buffer; struct oxygen; struct oxygen_model { @@ -93,6 +94,8 @@ struct oxygen_model { void (*uart_input)(struct oxygen *chip); void (*ac97_switch)(struct oxygen *chip, unsigned int reg, unsigned int mute); + void (*dump_registers)(struct oxygen *chip, + struct snd_info_buffer *buffer); const unsigned int *dac_tlv; unsigned long private_data; size_t model_data_size; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 2e65799..e581e7a 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -202,7 +202,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry, struct oxygen *chip = entry->private_data; int i, j; - snd_iprintf(buffer, "CMI8788\n\n"); + snd_iprintf(buffer, "CMI8788:\n"); for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; ++j) @@ -212,7 +212,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry, if (mutex_lock_interruptible(&chip->mutex) < 0) return; if (chip->has_ac97_0) { - snd_iprintf(buffer, "\nAC97\n"); + snd_iprintf(buffer, "\nAC97:\n"); for (i = 0; i < 0x80; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; j += 2) @@ -222,7 +222,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry, } } if (chip->has_ac97_1) { - snd_iprintf(buffer, "\nAC97 2\n"); + 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) @@ -232,13 +232,15 @@ static void oxygen_proc_read(struct snd_info_entry *entry, } } mutex_unlock(&chip->mutex); + if (chip->model.dump_registers) + chip->model.dump_registers(chip, buffer); } static void oxygen_proc_init(struct oxygen *chip) { struct snd_info_entry *entry; - if (!snd_card_proc_new(chip->card, "cmi8788", &entry)) + if (!snd_card_proc_new(chip->card, "oxygen", &entry)) snd_info_set_text_ops(entry, chip, oxygen_proc_read); } #else diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 501fe45..092addb 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -381,6 +381,30 @@ static int xonar_d1_mixer_init(struct oxygen *chip) return 0; } +static void dump_cs4362a_registers(struct xonar_cs43xx *data, + struct snd_info_buffer *buffer) +{ + unsigned int i; + + snd_iprintf(buffer, "\nCS4362A:"); + for (i = 1; i <= 14; ++i) + snd_iprintf(buffer, " %02x", data->cs4362a_regs[i]); + snd_iprintf(buffer, "\n"); +} + +static void dump_d1_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_cs43xx *data = chip->model_data; + unsigned int i; + + snd_iprintf(buffer, "\nCS4398: 7?"); + for (i = 2; i <= 8; ++i) + snd_iprintf(buffer, " %02x", data->cs4398_regs[i]); + snd_iprintf(buffer, "\n"); + dump_cs4362a_registers(data, buffer); +} + static const struct oxygen_model model_xonar_d1 = { .longname = "Asus Virtuoso 100", .chip = "AV200", @@ -396,6 +420,7 @@ static const struct oxygen_model model_xonar_d1 = { .update_dac_mute = update_cs43xx_mute, .update_center_lfe_mix = update_cs43xx_center_lfe_mix, .ac97_switch = xonar_d1_line_mic_ac97_switch, + .dump_registers = dump_d1_registers, .dac_tlv = cs4362a_db_scale, .model_data_size = sizeof(struct xonar_cs43xx), .device_config = PLAYBACK_0_TO_I2S | diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 5193d73..dc69fdd 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -150,6 +150,7 @@ #include #include #include +#include #include #include #include @@ -192,7 +193,7 @@ struct xonar_pcm179x { bool hp_active; s8 hp_gain_offset; bool has_cs2000; - u8 cs2000_fun_cfg_1; + u8 cs2000_regs[0x1f]; }; struct xonar_hdav { @@ -251,16 +252,14 @@ static void cs2000_write(struct oxygen *chip, u8 reg, u8 value) struct xonar_pcm179x *data = chip->model_data; oxygen_write_i2c(chip, I2C_DEVICE_CS2000, reg, value); - if (reg == CS2000_FUN_CFG_1) - data->cs2000_fun_cfg_1 = value; + data->cs2000_regs[reg] = value; } static void cs2000_write_cached(struct oxygen *chip, u8 reg, u8 value) { struct xonar_pcm179x *data = chip->model_data; - if (reg != CS2000_FUN_CFG_1 || - value != data->cs2000_fun_cfg_1) + if (value != data->cs2000_regs[reg]) cs2000_write(chip, reg, value); } @@ -414,7 +413,8 @@ static void cs2000_registers_init(struct oxygen *chip) cs2000_write(chip, CS2000_RATIO_0 + 1, 0x10); cs2000_write(chip, CS2000_RATIO_0 + 2, 0x00); cs2000_write(chip, CS2000_RATIO_0 + 3, 0x00); - cs2000_write(chip, CS2000_FUN_CFG_1, data->cs2000_fun_cfg_1); + cs2000_write(chip, CS2000_FUN_CFG_1, + data->cs2000_regs[CS2000_FUN_CFG_1]); cs2000_write(chip, CS2000_FUN_CFG_2, 0); cs2000_write(chip, CS2000_GLOBAL_CFG, CS2000_EN_DEV_CFG_2); } @@ -425,7 +425,7 @@ static void xonar_st_init(struct oxygen *chip) data->generic.anti_pop_delay = 100; data->has_cs2000 = 1; - data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; + data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_I2S | @@ -997,6 +997,45 @@ static int xonar_st_mixer_init(struct oxygen *chip) return 0; } +static void dump_pcm1796_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int dac, i; + + for (dac = 0; dac < data->dacs; ++dac) { + snd_iprintf(buffer, "\nPCM1796 %u:", dac + 1); + for (i = 0; i < 5; ++i) + snd_iprintf(buffer, " %02x", + data->pcm1796_regs[dac][i]); + } + snd_iprintf(buffer, "\n"); +} + +static void dump_cs2000_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + + if (data->has_cs2000) { + snd_iprintf(buffer, "\nCS2000:\n00: "); + for (i = 1; i < 0x10; ++i) + snd_iprintf(buffer, " %02x", data->cs2000_regs[i]); + snd_iprintf(buffer, "\n10:"); + for (i = 0x10; i < 0x1f; ++i) + snd_iprintf(buffer, " %02x", data->cs2000_regs[i]); + snd_iprintf(buffer, "\n"); + } +} + +static void dump_st_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + dump_pcm1796_registers(chip, buffer); + dump_cs2000_registers(chip, buffer); +} + static const struct oxygen_model model_xonar_d2 = { .longname = "Asus Virtuoso 200", .chip = "AV200", @@ -1011,6 +1050,7 @@ static const struct oxygen_model model_xonar_d2 = { .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, + .dump_registers = dump_pcm1796_registers, .dac_tlv = pcm1796_db_scale, .model_data_size = sizeof(struct xonar_pcm179x), .device_config = PLAYBACK_0_TO_I2S | @@ -1046,6 +1086,7 @@ static const struct oxygen_model model_xonar_hdav = { .update_dac_mute = update_pcm1796_mute, .uart_input = xonar_hdmi_uart_input, .ac97_switch = xonar_line_mic_ac97_switch, + .dump_registers = dump_pcm1796_registers, .dac_tlv = pcm1796_db_scale, .model_data_size = sizeof(struct xonar_hdav), .device_config = PLAYBACK_0_TO_I2S | @@ -1075,6 +1116,7 @@ static const struct oxygen_model model_xonar_st = { .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, .ac97_switch = xonar_line_mic_ac97_switch, + .dump_registers = dump_st_registers, .dac_tlv = pcm1796_db_scale, .model_data_size = sizeof(struct xonar_pcm179x), .device_config = PLAYBACK_0_TO_I2S | diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 200f760..2b5e69b 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -42,6 +42,7 @@ #include #include #include +#include #include #include #include @@ -1062,6 +1063,34 @@ static int xonar_ds_mixer_init(struct oxygen *chip) return 0; } +static void dump_wm8776_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + + snd_iprintf(buffer, "\nWM8776:\n00:"); + for (i = 0; i < 0x10; ++i) + snd_iprintf(buffer, " %03x", data->wm8776_regs[i]); + snd_iprintf(buffer, "\n10:"); + for (i = 0x10; i < 0x17; ++i) + snd_iprintf(buffer, " %03x", data->wm8776_regs[i]); + snd_iprintf(buffer, "\n"); +} + +static void dump_wm87x6_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + + dump_wm8776_registers(chip, buffer); + snd_iprintf(buffer, "\nWM8766:\n00:"); + for (i = 0; i < 0x10; ++i) + snd_iprintf(buffer, " %03x", data->wm8766_regs[i]); + snd_iprintf(buffer, "\n"); +} + static const struct oxygen_model model_xonar_ds = { .shortname = "Xonar DS", .longname = "Asus Virtuoso 66", @@ -1079,6 +1108,7 @@ static const struct oxygen_model model_xonar_ds = { .update_dac_mute = update_wm87x6_mute, .update_center_lfe_mix = update_wm8766_center_lfe_mix, .gpio_changed = xonar_ds_gpio_changed, + .dump_registers = dump_wm87x6_registers, .dac_tlv = wm87x6_dac_db_scale, .model_data_size = sizeof(struct xonar_wm87x6), .device_config = PLAYBACK_0_TO_I2S | -- cgit v0.10.2 From e2943efa4fda376903974e33298b29091fc596b3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Dec 2010 11:42:00 +0100 Subject: ALSA: oxygen: show correct package ID Instead of the hardcoded "CMI8788", show the actual chip name. Note: This is neither what the chip is (it's always the same), nor what the chip is actually called. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index e581e7a..3078ed6 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -202,7 +202,13 @@ static void oxygen_proc_read(struct snd_info_entry *entry, struct oxygen *chip = entry->private_data; int i, j; - snd_iprintf(buffer, "CMI8788:\n"); + switch (oxygen_read8(chip, OXYGEN_REVISION) & OXYGEN_PACKAGE_ID_MASK) { + case OXYGEN_PACKAGE_ID_8786: i = '6'; break; + case OXYGEN_PACKAGE_ID_8787: i = '7'; break; + case OXYGEN_PACKAGE_ID_8788: i = '8'; break; + default: i = '?'; break; + } + snd_iprintf(buffer, "CMI878%c:\n", i); for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; ++j) -- cgit v0.10.2 From de664936930dae5469170f7eed24bcff7e91ef82 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Dec 2010 11:42:48 +0100 Subject: ALSA: oxygen: update hardware comments Reformat and update the comments that describe the hardware connections on the various models. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index a58e448..dc47977 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -20,19 +20,25 @@ /* * CMI8788: * - * SPI 0 -> 1st AK4396 (front) - * SPI 1 -> 2nd AK4396 (surround) - * SPI 2 -> 3rd AK4396 (center/LFE) - * SPI 3 -> WM8785 - * SPI 4 -> 4th AK4396 (back) + * SPI 0 -> 1st AK4396 (front) + * SPI 1 -> 2nd AK4396 (surround) + * SPI 2 -> 3rd AK4396 (center/LFE) + * SPI 3 -> WM8785 + * SPI 4 -> 4th AK4396 (back) * - * GPIO 0 -> DFS0 of AK5385 - * GPIO 1 -> DFS1 of AK5385 - * GPIO 8 -> enable headphone amplifier on HT-Omega models + * GPIO 0 -> DFS0 of AK5385 + * GPIO 1 -> DFS1 of AK5385 + * GPIO 8 -> enable headphone amplifier on HT-Omega models * * CM9780: * - * GPO 0 -> route line-in (0) or AC97 output (1) to ADC input + * LINE_OUT -> input of ADC + * + * AUX_IN <- aux + * CD_IN <- CD + * MIC_IN <- mic + * + * GPO 0 -> route line-in (0) or AC97 output (1) to ADC input */ #include diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 092addb..de32895 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -22,29 +22,28 @@ * * CMI8788: * - * I²C <-> CS4398 (front) - * <-> CS4362A (surround, center/LFE, back) + * I²C <-> CS4398 (addr 1001111) (front) + * <-> CS4362A (addr 0011000) (surround, center/LFE, back) * - * GPI 0 <- external power present (DX only) + * GPI 0 <- external power present (DX only) * - * GPIO 0 -> enable output to speakers - * GPIO 1 -> route output to front panel - * GPIO 2 -> M0 of CS5361 - * GPIO 3 -> M1 of CS5361 - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * GPIO 0 -> enable output to speakers + * GPIO 1 -> route output to front panel + * GPIO 2 -> M0 of CS5361 + * GPIO 3 -> M1 of CS5361 + * GPIO 6 -> ? + * GPIO 7 -> ? + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) * - * CS4398: - * - * AD0 <- 1 - * AD1 <- 1 - * - * CS4362A: + * CM9780: * - * AD0 <- 0 + * LINE_OUT -> input of ADC * - * CM9780: + * AUX_IN <- aux + * MIC_IN <- mic + * FMIC_IN <- front mic * - * GPO 0 -> route line-in (0) or AC97 output (1) to CS5361 input + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5361 input */ #include diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index dc69fdd..bf357c0 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -22,20 +22,26 @@ * * CMI8788: * - * SPI 0 -> 1st PCM1796 (front) - * SPI 1 -> 2nd PCM1796 (surround) - * SPI 2 -> 3rd PCM1796 (center/LFE) - * SPI 4 -> 4th PCM1796 (back) + * SPI 0 -> 1st PCM1796 (front) + * SPI 1 -> 2nd PCM1796 (surround) + * SPI 2 -> 3rd PCM1796 (center/LFE) + * SPI 4 -> 4th PCM1796 (back) * - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 5 <- external power present (D2X only) - * GPIO 7 -> ALT - * GPIO 8 -> enable output to speakers + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 5 <- external power present (D2X only) + * GPIO 7 -> ALT + * GPIO 8 -> enable output to speakers * * CM9780: * - * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * LINE_OUT -> input of ADC + * + * AUX_IN <- aux + * VIDEO_IN <- CD + * FMIC_IN <- mic + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input */ /* @@ -44,52 +50,53 @@ * * CMI8788: * - * I²C <-> PCM1796 (front) - * - * GPI 0 <- external power present + * I²C <-> PCM1796 (addr 1001100) (front) * - * GPIO 0 -> enable output to speakers - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * GPI 0 <- external power present * - * TXD -> HDMI controller - * RXD <- HDMI controller + * GPIO 0 -> enable HDMI (0) or speaker (1) output + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 4 <- daughterboard detection + * GPIO 5 <- daughterboard detection + * GPIO 6 -> ? + * GPIO 7 -> ? + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) * - * PCM1796 front: AD1,0 <- 0,0 + * UART <-> HDMI controller * * CM9780: * - * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * LINE_OUT -> input of ADC + * + * AUX_IN <- aux + * CD_IN <- CD + * MIC_IN <- mic + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input * * no daughterboard * ---------------- * - * GPIO 4 <- 1 + * GPIO 4 <- 1 * * H6 daughterboard * ---------------- * - * GPIO 4 <- 0 - * GPIO 5 <- 0 - * - * I²C <-> PCM1796 (surround) - * <-> PCM1796 (center/LFE) - * <-> PCM1796 (back) + * GPIO 4 <- 0 + * GPIO 5 <- 0 * - * PCM1796 surround: AD1,0 <- 0,1 - * PCM1796 center/LFE: AD1,0 <- 1,0 - * PCM1796 back: AD1,0 <- 1,1 + * I²C <-> PCM1796 (addr 1001101) (surround) + * <-> PCM1796 (addr 1001110) (center/LFE) + * <-> PCM1796 (addr 1001111) (back) * * unknown daughterboard * --------------------- * - * GPIO 4 <- 0 - * GPIO 5 <- 1 - * - * I²C <-> CS4362A (surround, center/LFE, back) + * GPIO 4 <- 0 + * GPIO 5 <- 1 * - * CS4362A: AD0 <- 0 + * I²C <-> CS4362A (addr 0011000) (surround, center/LFE, back) */ /* @@ -98,32 +105,35 @@ * * CMI8788: * - * I²C <-> PCM1792A - * <-> CS2000 (ST only) + * I²C <-> PCM1792A (addr 1001100) + * <-> CS2000 (addr 1001110) (ST only) * - * ADC1 MCLK -> REF_CLK of CS2000 (ST only) + * ADC1 MCLK -> REF_CLK of CS2000 (ST only) * - * GPI 0 <- external power present (STX only) + * GPI 0 <- external power present (STX only) * - * GPIO 0 -> enable output to speakers - * GPIO 1 -> route HP to front panel (0) or rear jack (1) - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 7 -> route output to speaker jacks (0) or HP (1) - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * GPIO 0 -> enable output to speakers + * GPIO 1 -> route HP to front panel (0) or rear jack (1) + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 4 <- daughterboard detection + * GPIO 5 <- daughterboard detection + * GPIO 6 -> ? + * GPIO 7 -> route output to speaker jacks (0) or HP (1) + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) * * PCM1792A: * - * AD1,0 <- 0,0 - * SCK <- CLK_OUT of CS2000 (ST only) + * SCK <- CLK_OUT of CS2000 (ST only) * - * CS2000: + * CM9780: * - * AD0 <- 0 + * LINE_OUT -> input of ADC * - * CM9780: + * AUX_IN <- aux + * MIC_IN <- mic * - * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input * * H6 daughterboard * ---------------- @@ -133,15 +143,39 @@ */ /* - * Xonar HDAV1.3 Slim - * ------------------ + * Xonar Xense + * ----------- * * CMI8788: * - * GPIO 1 -> enable output + * I²C <-> PCM1796 (addr 1001100) (front) + * <-> CS4362A (addr 0011000) (surround, center/LFE, back) + * <-> CS2000 (addr 1001110) + * + * ADC1 MCLK -> REF_CLK of CS2000 + * + * GPI 0 <- external power present + * + * GPIO 0 -> enable output + * GPIO 1 -> route HP to front panel (0) or rear jack (1) + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 4 -> enable output + * GPIO 5 -> enable output + * GPIO 6 -> ? + * GPIO 7 -> route output to HP (0) or speaker (1) + * GPIO 8 -> route input jack to mic-in (0) or line-in (1) + * + * CM9780: + * + * LINE_OUT -> input of ADC + * + * AUX_IN <- aux + * VIDEO_IN <- ? + * FMIC_IN <- mic * - * TXD -> HDMI controller - * RXD <- HDMI controller + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * GPO 1 -> route mic-in from input jack (0) or front panel header (1) */ #include diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 2b5e69b..1705d1e 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -22,20 +22,41 @@ * * CMI8788: * - * SPI 0 -> WM8766 (surround, center/LFE, back) - * SPI 1 -> WM8776 (front, input) + * SPI 0 -> WM8766 (surround, center/LFE, back) + * SPI 1 -> WM8776 (front, input) * - * GPIO 4 <- headphone detect, 0 = plugged - * GPIO 6 -> route input jack to mic-in (0) or line-in (1) - * GPIO 7 -> enable output to front L/R speaker channels - * GPIO 8 -> enable output to other speaker channels and front panel headphone + * GPIO 4 <- headphone detect, 0 = plugged + * GPIO 6 -> route input jack to mic-in (0) or line-in (1) + * GPIO 7 -> enable output to front L/R speaker channels + * GPIO 8 -> enable output to other speaker channels and front panel headphone * - * WM8766: + * WM8776: * - * input 1 <- line - * input 2 <- mic - * input 3 <- front mic - * input 4 <- aux + * input 1 <- line + * input 2 <- mic + * input 3 <- front mic + * input 4 <- aux + */ + +/* + * Xonar HDAV1.3 Slim + * ------------------ + * + * CMI8788: + * + * I²C <-> WM8776 (addr 0011010) + * + * GPIO 0 -> disable HDMI output + * GPIO 1 -> enable HP output + * GPIO 6 -> firmware EEPROM I²C clock + * GPIO 7 <-> firmware EEPROM I²C data + * + * UART <-> HDMI controller + * + * WM8776: + * + * input 1 <- mic + * input 2 <- aux */ #include -- cgit v0.10.2 From d0359c6fac18588da04fbefdad096453442653ee Mon Sep 17 00:00:00 2001 From: Jeffrin Jose Date: Mon, 6 Dec 2010 19:27:53 +0530 Subject: sound: Fixed line limit issue in sound/ac97_bus.c This is a patch to the sound/ac97_bus.c file that fixes up a 80 character line limit issue found by the checkpatch.pl tool. Signed-off-by: Jeffrin Jose Signed-off-by: Takashi Iwai diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index a351dd0..2b50cbe 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -19,8 +19,8 @@ /* * Let drivers decide whether they want to support given codec from their - * probe method. Drivers have direct access to the struct snd_ac97 structure and may - * decide based on the id field amongst other things. + * probe method. Drivers have direct access to the struct snd_ac97 + * structure and may decide based on the id field amongst other things. */ static int ac97_bus_match(struct device *dev, struct device_driver *drv) { -- cgit v0.10.2 From 93430096f9d757104080f40f51afb2dada8877b5 Mon Sep 17 00:00:00 2001 From: Brian Bloniarz Date: Wed, 8 Dec 2010 12:45:20 -0800 Subject: ALSA: ice1712 - working M-Audio Delta 66E support Rev. E of the M-Audio Delta 66 is partially supported (commit ef2cd2ccad66b4aba518eca7514eface267ee0f3), but the layout of the GPIO pins was still unclear. This patch adds the GPIO definitions so that communication to the CS8247 & 2x AK4524 works correctly. ALSA bug#3327 has more details; users cap & jhunt report there that the GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 = CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of conflicting information in the bug, but given these definitions, my Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz. Signed-off-by: Brian Bloniarz Signed-off-by: Takashi Iwai diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 712c171..7b62de0 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -96,6 +96,11 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice) tmp |= ICE1712_DELTA_AP_CCLK | ICE1712_DELTA_AP_CS_CODEC; tmp &= ~ICE1712_DELTA_AP_CS_DIGITAL; break; + case ICE1712_SUBDEVICE_DELTA66E: + tmp |= ICE1712_DELTA_66E_CCLK | ICE1712_DELTA_66E_CS_CHIP_A | + ICE1712_DELTA_66E_CS_CHIP_B; + tmp &= ~ICE1712_DELTA_66E_CS_CS8427; + break; case ICE1712_SUBDEVICE_VX442: tmp |= ICE1712_VX442_CCLK | ICE1712_VX442_CODEC_CHIP_A | ICE1712_VX442_CODEC_CHIP_B; tmp &= ~ICE1712_VX442_CS_DIGITAL; @@ -119,6 +124,9 @@ static void ap_cs8427_codec_deassert(struct snd_ice1712 *ice, unsigned char tmp) case ICE1712_SUBDEVICE_DELTA410: tmp |= ICE1712_DELTA_AP_CS_DIGITAL; break; + case ICE1712_SUBDEVICE_DELTA66E: + tmp |= ICE1712_DELTA_66E_CS_CS8427; + break; case ICE1712_SUBDEVICE_VX442: tmp |= ICE1712_VX442_CS_DIGITAL; break; @@ -276,6 +284,20 @@ static void delta1010lt_ak4524_lock(struct snd_akm4xxx *ak, int chip) } /* + * AK4524 on Delta66 rev E to choose the chip address + */ +static void delta66e_ak4524_lock(struct snd_akm4xxx *ak, int chip) +{ + struct snd_ak4xxx_private *priv = (void *)ak->private_value[0]; + struct snd_ice1712 *ice = ak->private_data[0]; + + snd_ice1712_save_gpio_status(ice); + priv->cs_mask = + priv->cs_addr = chip == 0 ? ICE1712_DELTA_66E_CS_CHIP_A : + ICE1712_DELTA_66E_CS_CHIP_B; +} + +/* * AK4528 on VX442 to choose the chip mask */ static void vx442_ak4524_lock(struct snd_akm4xxx *ak, int chip) @@ -487,6 +509,29 @@ static struct snd_ak4xxx_private akm_delta1010lt_priv __devinitdata = { .mask_flags = 0, }; +static struct snd_akm4xxx akm_delta66e __devinitdata = { + .type = SND_AK4524, + .num_adcs = 4, + .num_dacs = 4, + .ops = { + .lock = delta66e_ak4524_lock, + .set_rate_val = delta_ak4524_set_rate_val + } +}; + +static struct snd_ak4xxx_private akm_delta66e_priv __devinitdata = { + .caddr = 2, + .cif = 0, /* the default level of the CIF pin from AK4524 */ + .data_mask = ICE1712_DELTA_66E_DOUT, + .clk_mask = ICE1712_DELTA_66E_CCLK, + .cs_mask = 0, + .cs_addr = 0, /* set later */ + .cs_none = 0, + .add_flags = 0, + .mask_flags = 0, +}; + + static struct snd_akm4xxx akm_delta44 __devinitdata = { .type = SND_AK4524, .num_adcs = 4, @@ -644,9 +689,11 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_init(ak, &akm_delta44, &akm_delta44_priv, ice); break; case ICE1712_SUBDEVICE_VX442: - case ICE1712_SUBDEVICE_DELTA66E: err = snd_ice1712_akm4xxx_init(ak, &akm_vx442, &akm_vx442_priv, ice); break; + case ICE1712_SUBDEVICE_DELTA66E: + err = snd_ice1712_akm4xxx_init(ak, &akm_delta66e, &akm_delta66e_priv, ice); + break; default: snd_BUG(); return -EINVAL; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index 1a0ac6c..11a9c3a 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -144,6 +144,17 @@ extern struct snd_ice1712_card_info snd_ice1712_delta_cards[]; #define ICE1712_DELTA_1010LT_CS_NONE 0x50 /* nothing */ #define ICE1712_DELTA_1010LT_WORDCLOCK 0x80 /* sample clock source: 0 = Word Clock Input, 1 = S/PDIF Input ??? */ +/* M-Audio Delta 66 rev. E definitions. + * Newer revisions of Delta 66 have CS8427 over SPI for + * S/PDIF transceiver instead of CS8404/CS8414. */ +/* 0x01 = DFS */ +#define ICE1712_DELTA_66E_CCLK 0x02 /* SPI clock */ +#define ICE1712_DELTA_66E_DIN 0x04 /* data input */ +#define ICE1712_DELTA_66E_DOUT 0x08 /* data output */ +#define ICE1712_DELTA_66E_CS_CS8427 0x10 /* chip select, low = CS8427 */ +#define ICE1712_DELTA_66E_CS_CHIP_A 0x20 /* AK4524 #0 */ +#define ICE1712_DELTA_66E_CS_CHIP_B 0x40 /* AK4524 #1 */ + /* Digigram VX442 definitions */ #define ICE1712_VX442_CCLK 0x02 /* SPI clock */ #define ICE1712_VX442_DIN 0x04 /* data input */ -- cgit v0.10.2 From 5144c534d16529bc469396211131e8935589f833 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 10 Dec 2010 17:34:26 +0800 Subject: ALSA: aoa: Remove wrong i2c_set_clientdata in onyx_i2c_remove() It does not make sense to set clientdata to onyx in onyx_i2c_remove() as we are going to kfree onyx. What we really want here is i2c_set_clientdata(client, NULL); Since the i2c core will take care of it now, so this patch just removes it. Signed-off-by: Axel Lin Signed-off-by: Takashi Iwai diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 91852e4..3687a6c 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1114,7 +1114,6 @@ static int onyx_i2c_remove(struct i2c_client *client) of_node_put(onyx->codec.node); if (onyx->codec_info) kfree(onyx->codec_info); - i2c_set_clientdata(client, onyx); kfree(onyx); return 0; } -- cgit v0.10.2 From 5b84ba26a9672e615897234fa5efd3eea2d6b295 Mon Sep 17 00:00:00 2001 From: Tejun Heo Date: Sat, 11 Dec 2010 17:51:26 +0100 Subject: sound: don't use flush_scheduled_work() flush_scheduled_work() is deprecated and scheduled to be removed. * cancel[_delayed]_work() + flush_scheduled_work() -> cancel[_delayed]_work_sync(). * wm8350, wm8753 and soc-core use custom code to cancel a delayed work, execute it immediately if it was pending and wait for its completion. This is equivalent to flush_delayed_work_sync(). Use it instead. Signed-off-by: Tejun Heo Acked-by: Mark Brown Signed-off-by: Takashi Iwai diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index de8e03a..faa3174 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -287,10 +287,9 @@ static void ftr_gpio_exit(struct gpio_runtime *rt) free_irq(linein_detect_irq, &rt->line_in_notify); if (rt->line_out_notify.gpio_private) free_irq(lineout_detect_irq, &rt->line_out_notify); - cancel_delayed_work(&rt->headphone_notify.work); - cancel_delayed_work(&rt->line_in_notify.work); - cancel_delayed_work(&rt->line_out_notify.work); - flush_scheduled_work(); + cancel_delayed_work_sync(&rt->headphone_notify.work); + cancel_delayed_work_sync(&rt->line_in_notify.work); + cancel_delayed_work_sync(&rt->line_out_notify.work); mutex_destroy(&rt->headphone_notify.mutex); mutex_destroy(&rt->line_in_notify.mutex); mutex_destroy(&rt->line_out_notify.mutex); diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c index 7e267c9..c8d8a1a 100644 --- a/sound/aoa/core/gpio-pmf.c +++ b/sound/aoa/core/gpio-pmf.c @@ -107,10 +107,9 @@ static void pmf_gpio_exit(struct gpio_runtime *rt) /* make sure no work is pending before freeing * all things */ - cancel_delayed_work(&rt->headphone_notify.work); - cancel_delayed_work(&rt->line_in_notify.work); - cancel_delayed_work(&rt->line_out_notify.work); - flush_scheduled_work(); + cancel_delayed_work_sync(&rt->headphone_notify.work); + cancel_delayed_work_sync(&rt->line_in_notify.work); + cancel_delayed_work_sync(&rt->line_out_notify.work); mutex_destroy(&rt->headphone_notify.mutex); mutex_destroy(&rt->line_in_notify.mutex); diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index 971a84a..c424d32 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -57,8 +57,7 @@ static void snd_ak4113_free(struct ak4113 *chip) { chip->init = 1; /* don't schedule new work */ mb(); - cancel_delayed_work(&chip->work); - flush_scheduled_work(); + cancel_delayed_work_sync(&chip->work); kfree(chip); } @@ -141,7 +140,7 @@ void snd_ak4113_reinit(struct ak4113 *chip) { chip->init = 1; mb(); - flush_scheduled_work(); + flush_delayed_work_sync(&chip->work); ak4113_init_regs(chip); /* bring up statistics / event queing */ chip->init = 0; diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 0341451..d9fb537 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -67,8 +67,7 @@ static void snd_ak4114_free(struct ak4114 *chip) { chip->init = 1; /* don't schedule new work */ mb(); - cancel_delayed_work(&chip->work); - flush_scheduled_work(); + cancel_delayed_work_sync(&chip->work); kfree(chip); } @@ -154,7 +153,7 @@ void snd_ak4114_reinit(struct ak4114 *chip) { chip->init = 1; mb(); - flush_scheduled_work(); + flush_delayed_work_sync(&chip->work); ak4114_init_regs(chip); /* bring up statistics / event queing */ chip->init = 0; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index a7630e9..0fc614c 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1014,8 +1014,7 @@ static int snd_ac97_free(struct snd_ac97 *ac97) { if (ac97) { #ifdef CONFIG_SND_AC97_POWER_SAVE - cancel_delayed_work(&ac97->power_work); - flush_scheduled_work(); + cancel_delayed_work_sync(&ac97->power_work); #endif snd_ac97_proc_done(ac97); if (ac97->bus) @@ -2456,8 +2455,7 @@ void snd_ac97_suspend(struct snd_ac97 *ac97) if (ac97->build_ops->suspend) ac97->build_ops->suspend(ac97); #ifdef CONFIG_SND_AC97_POWER_SAVE - cancel_delayed_work(&ac97->power_work); - flush_scheduled_work(); + cancel_delayed_work_sync(&ac97->power_work); #endif snd_ac97_powerdown(ac97); } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d1c3f8d..7f4852a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -263,8 +263,7 @@ static void vt1708_stop_hp_work(struct via_spec *spec) return; snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, !spec->vt1708_jack_detectect); - cancel_delayed_work(&spec->vt1708_hp_work); - flush_scheduled_work(); + cancel_delayed_work_sync(&spec->vt1708_hp_work); } diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index e5ebe56..969605f 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -557,7 +557,8 @@ static void oxygen_card_free(struct snd_card *card) oxygen_shutdown(chip); if (chip->irq >= 0) free_irq(chip->irq, chip); - flush_scheduled_work(); + flush_work_sync(&chip->spdif_input_bits_work); + flush_work_sync(&chip->gpio_work); chip->model.cleanup(chip); kfree(chip->model_data); mutex_destroy(&chip->mutex); @@ -733,7 +734,8 @@ int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state) spin_unlock_irq(&chip->reg_lock); synchronize_irq(chip->irq); - flush_scheduled_work(); + flush_work_sync(&chip->spdif_input_bits_work); + flush_work_sync(&chip->gpio_work); chip->interrupt_mask = saved_interrupt_mask; pci_disable_device(pci); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 7611add..b3e9fac 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1626,7 +1626,6 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); struct wm8350 *wm8350 = dev_get_platdata(codec->dev); - int ret; wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, WM8350_JDL_ENA | WM8350_JDR_ENA); @@ -1641,15 +1640,9 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) priv->hpr.jack = NULL; priv->mic.jack = NULL; - /* cancel any work waiting to be queued. */ - ret = cancel_delayed_work(&codec->delayed_work); - /* if there was any work waiting then we run it now and * wait for its completion */ - if (ret) { - schedule_delayed_work(&codec->delayed_work, 0); - flush_scheduled_work(); - } + flush_delayed_work_sync(&codec->delayed_work); wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8f679a1..84a2367 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1526,25 +1526,6 @@ static int wm8753_resume(struct snd_soc_codec *codec) return 0; } -/* - * This function forces any delayed work to be queued and run. - */ -static int run_delayed_work(struct delayed_work *dwork) -{ - int ret; - - /* cancel any work waiting to be queued. */ - ret = cancel_delayed_work(dwork); - - /* if there was any work waiting then we run it now and - * wait for it's completion */ - if (ret) { - schedule_delayed_work(dwork, 0); - flush_scheduled_work(); - } - return ret; -} - static int wm8753_probe(struct snd_soc_codec *codec) { struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); @@ -1604,7 +1585,7 @@ static int wm8753_probe(struct snd_soc_codec *codec) /* power down chip */ static int wm8753_remove(struct snd_soc_codec *codec) { - run_delayed_work(&codec->delayed_work); + flush_delayed_work_sync(&codec->delayed_work); wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 441285a..b54ea9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -67,25 +67,6 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); -/* - * This function forces any delayed work to be queued and run. - */ -static int run_delayed_work(struct delayed_work *dwork) -{ - int ret; - - /* cancel any work waiting to be queued. */ - ret = cancel_delayed_work(dwork); - - /* if there was any work waiting then we run it now and - * wait for it's completion */ - if (ret) { - schedule_delayed_work(dwork, 0); - flush_scheduled_work(); - } - return ret; -} - /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { @@ -1016,7 +997,7 @@ static int soc_suspend(struct device *dev) /* close any waiting streams and save state */ for (i = 0; i < card->num_rtd; i++) { - run_delayed_work(&card->rtd[i].delayed_work); + flush_delayed_work_sync(&card->rtd[i].delayed_work); card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level; } @@ -1687,7 +1668,7 @@ static int soc_remove(struct platform_device *pdev) /* make sure any delayed work runs */ for (i = 0; i < card->num_rtd; i++) { struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; - run_delayed_work(&rtd->delayed_work); + flush_delayed_work_sync(&rtd->delayed_work); } /* remove and free each DAI */ @@ -1718,7 +1699,7 @@ static int soc_poweroff(struct device *dev) * now, we're shutting down so no imminent restart. */ for (i = 0; i < card->num_rtd; i++) { struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; - run_delayed_work(&rtd->delayed_work); + flush_delayed_work_sync(&rtd->delayed_work); } snd_soc_dapm_shutdown(card); -- cgit v0.10.2 From a8cc0f421b37956262a92591f7397b200d232da6 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Mon, 13 Dec 2010 13:42:22 -0800 Subject: ALSA: ml403-ac97cr: Use vsprintf extension %pR for struct resource Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index a1282c1..5cfcb90 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1143,8 +1143,8 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, (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); + "unable to remap memory region (%pR)\n", + resource); snd_ml403_ac97cr_free(ml403_ac97cr); return -EBUSY; } -- cgit v0.10.2 From adf5931f8c412e90f47033ca6bc7a0bc8a930ba1 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Mon, 27 Dec 2010 21:16:43 +0100 Subject: ALSA: azt3328: cosmetics, minor updates - correct samples to be POSIX shell compatible - add logging of jiffies value in _pointer() - several comments - cleanup Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 2f3cacb..006f8c0 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005 - 2009 by Andreas Mohr + * Copyright (C) 2002, 2005 - 2010 by Andreas Mohr * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -201,14 +201,15 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); /* === Debug settings === Further diagnostic functionality than the settings below - does not need to be provided, since one can easily write a bash script + does not need to be provided, since one can easily write a POSIX shell script to dump the card's I/O ports (those listed in lspci -v -v): - function dump() + dump() { local descr=$1; local addr=$2; local count=$3 echo "${descr}: ${count} @ ${addr}:" - dd if=/dev/port skip=$[${addr}] count=${count} bs=1 2>/dev/null| hexdump -C + dd if=/dev/port skip=`printf %d ${addr}` count=${count} bs=1 \ + 2>/dev/null| hexdump -C } and then use something like "dump joy200 0x200 8", "dump mpu388 0x388 4", "dump joy 0xb400 8", @@ -216,14 +217,14 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); possibly within a "while true; do ... sleep 1; done" loop. Tweaking ports could be done using VALSTRING="`printf "%02x" $value`" - printf "\x""$VALSTRING"|dd of=/dev/port seek=$[${addr}] bs=1 2>/dev/null + printf "\x""$VALSTRING"|dd of=/dev/port seek=`printf %d ${addr}` bs=1 \ + 2>/dev/null */ #define DEBUG_MISC 0 #define DEBUG_CALLS 0 #define DEBUG_MIXER 0 #define DEBUG_CODEC 0 -#define DEBUG_IO 0 #define DEBUG_TIMER 0 #define DEBUG_GAME 0 #define DEBUG_PM 0 @@ -299,6 +300,7 @@ struct snd_azf3328_codec_data { }; enum snd_azf3328_codec_type { + /* warning: fixed indices (also used for bitmask checks!) */ AZF_CODEC_PLAYBACK = 0, AZF_CODEC_CAPTURE = 1, AZF_CODEC_I2S_OUT = 2, @@ -362,6 +364,9 @@ MODULE_DEVICE_TABLE(pci, snd_azf3328_ids); static int snd_azf3328_io_reg_setb(unsigned reg, u8 mask, bool do_set) { + /* Well, strictly spoken, the inb/outb sequence isn't atomic + and would need locking. However we currently don't care + since it potentially complicates matters. */ u8 prev = inb(reg), new; new = (do_set) ? (prev|mask) : (prev & ~mask); @@ -1004,7 +1009,8 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, * (FIXME: yes, it works, but what exactly am I doing here?? :) * FIXME: does this have some side effects for full-duplex * or other dramatic side effects? */ - if (codec_type == AZF_CODEC_PLAYBACK) /* only do it for playback */ + /* do it for non-capture codecs only */ + if (codec_type == AZF_CODEC_PLAYBACK) snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS) | DMA_RUN_SOMETHING1 | @@ -1368,8 +1374,8 @@ snd_azf3328_codec_pointer(struct snd_pcm_substream *substream, /* calculate offset */ result -= bufptr; frmres = bytes_to_frames( substream->runtime, result); - snd_azf3328_dbgcodec("%s @ 0x%8lx, frames %8ld\n", - codec->name, result, frmres); + snd_azf3328_dbgcodec("%08li %s @ 0x%8lx, frames %8ld\n", + jiffies, codec->name, result, frmres); return frmres; } @@ -1532,7 +1538,7 @@ snd_azf3328_gameport_cooked_read(struct gameport *gameport, } } - /* trigger next axes sampling, to be evaluated the next time we + /* trigger next sampling of axes, to be evaluated the next time we * enter this function */ /* for some very, very strange reason we cannot enable @@ -1966,7 +1972,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) snd_azf3328_dbgtimer("delay was too low (%d)!\n", delay); delay = 49; /* minimum time is 49 ticks */ } - snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); + snd_azf3328_dbgtimer("setting timer countdown value %d\n", delay); delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); snd_azf3328_ctrl_outl(chip, IDX_IO_TIMER_VALUE, delay); @@ -2257,7 +2263,7 @@ snd_azf3328_create(struct snd_card *card, struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; - /* shutdown codecs to save power */ + /* shutdown codecs to reduce power / noise */ /* have ...ctrl_codec_activity() act properly */ codec->running = 1; snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); @@ -2419,6 +2425,7 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* same pcm object for playback/capture */ snd_pcm_suspend_all(chip->pcm[AZF_CODEC_PLAYBACK]); snd_pcm_suspend_all(chip->pcm[AZF_CODEC_I2S_OUT]); -- cgit v0.10.2 From 8d9a114e6d4acabf6b23ca8bccf0e486c3bdf85c Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Mon, 27 Dec 2010 21:16:49 +0100 Subject: ALSA: azt3328: _setfmt() update - use a separate variable for the frequency part, don't always "or" it - use a "clever"(?) macro to shorten the code Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 006f8c0..3a9a4a1 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -958,28 +958,35 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, unsigned long flags; const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; u16 val = 0xff00; + u8 freq = 0; snd_azf3328_dbgcallenter(); switch (bitrate) { - case AZF_FREQ_4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; - case AZF_FREQ_4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; - case AZF_FREQ_5512: - /* the AZF3328 names it "5510" for some strange reason */ - val |= SOUNDFORMAT_FREQ_5510; break; - case AZF_FREQ_6620: val |= SOUNDFORMAT_FREQ_6620; break; - case AZF_FREQ_8000: val |= SOUNDFORMAT_FREQ_8000; break; - case AZF_FREQ_9600: val |= SOUNDFORMAT_FREQ_9600; break; - case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break; - case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; - case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break; - case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break; - case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break; +#define AZF_FMT_XLATE(in_freq, out_bits) \ + do { \ + case AZF_FREQ_ ## in_freq: \ + freq = SOUNDFORMAT_FREQ_ ## out_bits; \ + break; \ + } while (0); + AZF_FMT_XLATE(4000, SUSPECTED_4000) + AZF_FMT_XLATE(4800, SUSPECTED_4800) + /* the AZF3328 names it "5510" for some strange reason: */ + AZF_FMT_XLATE(5512, 5510) + AZF_FMT_XLATE(6620, 6620) + AZF_FMT_XLATE(8000, 8000) + AZF_FMT_XLATE(9600, 9600) + AZF_FMT_XLATE(11025, 11025) + AZF_FMT_XLATE(13240, SUSPECTED_13240) + AZF_FMT_XLATE(16000, 16000) + AZF_FMT_XLATE(22050, 22050) + AZF_FMT_XLATE(32000, 32000) default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); /* fall-through */ - case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break; - case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break; - case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; + AZF_FMT_XLATE(44100, 44100) + AZF_FMT_XLATE(48000, 48000) + AZF_FMT_XLATE(66200, SUSPECTED_66200) +#undef AZF_FMT_XLATE } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */ @@ -991,6 +998,8 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, /* val = 0xff0d; 41m23.135s (5523,600Hz; -> 5512Hz???) */ /* val = 0xff0e; 28m30.777s (8017Hz; -> 8000Hz???) */ + val |= freq; + if (channels == 2) val |= SOUNDFORMAT_FLAG_2CHANNELS; -- cgit v0.10.2 From 9fd8d36caabaf3102f14cf652d5ca012d775aaa8 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Mon, 27 Dec 2010 21:17:00 +0100 Subject: ALSA: azt3328: cosmetics: use a helper variable for codec setup Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 3a9a4a1..0cb5499 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2195,6 +2195,7 @@ snd_azf3328_create(struct snd_card *card, }; u8 dma_init; enum snd_azf3328_codec_type codec_type; + struct snd_azf3328_codec *codec_setup; *rchip = NULL; @@ -2232,15 +2233,17 @@ snd_azf3328_create(struct snd_card *card, chip->opl3_io = pci_resource_start(pci, 3); chip->mixer_io = pci_resource_start(pci, 4); - chip->codecs[AZF_CODEC_PLAYBACK].io_base = - chip->ctrl_io + AZF_IO_OFFS_CODEC_PLAYBACK; - chip->codecs[AZF_CODEC_PLAYBACK].name = "PLAYBACK"; - chip->codecs[AZF_CODEC_CAPTURE].io_base = - chip->ctrl_io + AZF_IO_OFFS_CODEC_CAPTURE; - chip->codecs[AZF_CODEC_CAPTURE].name = "CAPTURE"; - chip->codecs[AZF_CODEC_I2S_OUT].io_base = - chip->ctrl_io + AZF_IO_OFFS_CODEC_I2S_OUT; - chip->codecs[AZF_CODEC_I2S_OUT].name = "I2S_OUT"; + codec_setup = &chip->codecs[AZF_CODEC_PLAYBACK]; + codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_PLAYBACK; + codec_setup->name = "PLAYBACK"; + + codec_setup = &chip->codecs[AZF_CODEC_CAPTURE]; + codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_CAPTURE; + codec_setup->name = "CAPTURE"; + + codec_setup = &chip->codecs[AZF_CODEC_I2S_OUT]; + codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_I2S_OUT; + codec_setup->name = "I2S_OUT"; if (request_irq(pci->irq, snd_azf3328_interrupt, IRQF_SHARED, card->shortname, chip)) { -- cgit v0.10.2 From 345855951a7d36eed815fd129c49b7ee2b7a6864 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Mon, 27 Dec 2010 21:17:11 +0100 Subject: ALSA: azt3328: use a helper variable to remove one indirection in hotpath Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 0cb5499..b1fad46 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -294,6 +294,7 @@ MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor ( struct snd_azf3328_codec_data { unsigned long io_base; + unsigned int dma_base; /* helper to avoid an indirection in hotpath */ struct snd_pcm_substream *substream; bool running; const char *name; @@ -1165,14 +1166,17 @@ snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip, static int snd_azf3328_codec_prepare(struct snd_pcm_substream *substream) { -#if 0 - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_azf3328_codec *codec = runtime->private_data; +#if 0 unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); #endif snd_azf3328_dbgcallenter(); + + codec->dma_base = runtime->dma_addr; + #if 0 snd_azf3328_codec_setfmt(chip, AZF_CODEC_..., runtime->rate, @@ -1370,18 +1374,17 @@ snd_azf3328_codec_pointer(struct snd_pcm_substream *substream, { const struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; - unsigned long bufptr, result; + unsigned long result; snd_pcm_uframes_t frmres; -#ifdef QUERY_HARDWARE - bufptr = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_START_1); -#else - bufptr = substream->runtime->dma_addr; -#endif result = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_CURRPOS); /* calculate offset */ - result -= bufptr; +#ifdef QUERY_HARDWARE + result -= snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_START_1); +#else + result -= codec->dma_base; +#endif frmres = bytes_to_frames( substream->runtime, result); snd_azf3328_dbgcodec("%08li %s @ 0x%8lx, frames %8ld\n", jiffies, codec->name, result, frmres); -- cgit v0.10.2 From da237f35a8a503fb8893fb3b9d0622a991bcebef Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Mon, 27 Dec 2010 21:17:26 +0100 Subject: ALSA: azt3328: use proper private_data hookup for codec identification - much improved implementation due to clean codec hierarchy - preparation for potential per-codec spinlock change NOTE: additionally removes a chip->pcm[codec_type] NULL ptr check (due to it requiring access to external chip struct), however I believe this to be ok since this condition should not occur and most drivers don't check against that either. Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index b1fad46..76ff5fd 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -292,14 +292,6 @@ static int seqtimer_scaling = 128; module_param(seqtimer_scaling, int, 0444); MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128."); -struct snd_azf3328_codec_data { - unsigned long io_base; - unsigned int dma_base; /* helper to avoid an indirection in hotpath */ - struct snd_pcm_substream *substream; - bool running; - const char *name; -}; - enum snd_azf3328_codec_type { /* warning: fixed indices (also used for bitmask checks!) */ AZF_CODEC_PLAYBACK = 0, @@ -307,6 +299,16 @@ enum snd_azf3328_codec_type { AZF_CODEC_I2S_OUT = 2, }; +struct snd_azf3328_codec_data { + unsigned long io_base; /* keep first! (avoid offset calc) */ + unsigned int dma_base; /* helper to avoid an indirection in hotpath */ + spinlock_t *lock; /* TODO: convert to our own per-codec lock member */ + struct snd_pcm_substream *substream; + bool running; + enum snd_azf3328_codec_type type; + const char *name; +}; + struct snd_azf3328 { /* often-used fields towards beginning, then grouped */ @@ -949,15 +951,13 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) } static void -snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, - enum snd_azf3328_codec_type codec_type, +snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) { unsigned long flags; - const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; u16 val = 0xff00; u8 freq = 0; @@ -1007,7 +1007,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, if (format_width == 16) val |= SOUNDFORMAT_FLAG_16BIT; - spin_lock_irqsave(&chip->reg_lock, flags); + spin_lock_irqsave(codec->lock, flags); /* set bitrate/format */ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_SOUNDFORMAT, val); @@ -1020,7 +1020,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, * FIXME: does this have some side effects for full-duplex * or other dramatic side effects? */ /* do it for non-capture codecs only */ - if (codec_type == AZF_CODEC_PLAYBACK) + if (codec->type != AZF_CODEC_CAPTURE) snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS) | DMA_RUN_SOMETHING1 | @@ -1030,20 +1030,19 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, DMA_SOMETHING_ELSE ); - spin_unlock_irqrestore(&chip->reg_lock, flags); + spin_unlock_irqrestore(codec->lock, flags); snd_azf3328_dbgcallleave(); } static inline void -snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, - enum snd_azf3328_codec_type codec_type +snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328_codec_data *codec ) { /* choose lowest frequency for low power consumption. * While this will cause louder noise due to rather coarse frequency, * it should never matter since output should always * get disabled properly when idle anyway. */ - snd_azf3328_codec_setfmt(chip, codec_type, AZF_FREQ_4000, 8, 1); + snd_azf3328_codec_setfmt(codec, AZF_FREQ_4000, 8, 1); } static void @@ -1117,23 +1116,18 @@ snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, /* ...and adjust clock, too * (reduce noise and power consumption) */ if (!enable) - snd_azf3328_codec_setfmt_lowpower( - chip, - codec_type - ); + snd_azf3328_codec_setfmt_lowpower(codec); codec->running = enable; } } static void -snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip, - enum snd_azf3328_codec_type codec_type, +snd_azf3328_codec_setdmaa(struct snd_azf3328_codec_data *codec, unsigned long addr, unsigned int count, unsigned int size ) { - const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; snd_azf3328_dbgcallenter(); if (!codec->running) { /* AZF3328 uses a two buffer pointer DMA transfer approach */ @@ -1152,22 +1146,22 @@ snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip, /* build combined I/O buffer length word */ lengths = (count_areas << 16) | (count_areas); - spin_lock_irqsave(&chip->reg_lock, flags); + spin_lock_irqsave(codec->lock, flags); snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_1, addr); snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_2, addr_area2); snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_LENGTHS, lengths); - spin_unlock_irqrestore(&chip->reg_lock, flags); + spin_unlock_irqrestore(codec->lock, flags); } snd_azf3328_dbgcallleave(); } static int -snd_azf3328_codec_prepare(struct snd_pcm_substream *substream) +snd_azf3328_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_azf3328_codec *codec = runtime->private_data; + struct snd_azf3328_codec_data *codec = runtime->private_data; #if 0 unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); @@ -1178,11 +1172,11 @@ snd_azf3328_codec_prepare(struct snd_pcm_substream *substream) codec->dma_base = runtime->dma_addr; #if 0 - snd_azf3328_codec_setfmt(chip, AZF_CODEC_..., + snd_azf3328_codec_setfmt(codec, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_codec_setdmaa(chip, AZF_CODEC_..., + snd_azf3328_codec_setdmaa(codec, runtime->dma_addr, count, size); #endif snd_azf3328_dbgcallleave(); @@ -1190,24 +1184,23 @@ snd_azf3328_codec_prepare(struct snd_pcm_substream *substream) } static int -snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, - struct snd_pcm_substream *substream, int cmd) +snd_azf3328_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_azf3328_codec_data *codec = runtime->private_data; int result = 0; u16 flags1; bool previously_muted = 0; - bool is_playback_codec = (AZF_CODEC_PLAYBACK == codec_type); + bool is_main_mixer_playback_codec = (AZF_CODEC_PLAYBACK == codec->type); - snd_azf3328_dbgcalls("snd_azf3328_codec_trigger cmd %d\n", cmd); + snd_azf3328_dbgcalls("snd_azf3328_pcm_trigger cmd %d\n", cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_START: snd_azf3328_dbgcodec("START %s\n", codec->name); - if (is_playback_codec) { + if (is_main_mixer_playback_codec) { /* mute WaveOut (avoid clicking during setup) */ previously_muted = snd_azf3328_mixer_set_mute( @@ -1215,12 +1208,12 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, ); } - snd_azf3328_codec_setfmt(chip, codec_type, + snd_azf3328_codec_setfmt(codec, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); /* first, remember current value: */ flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS); @@ -1230,14 +1223,14 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_IRQTYPE, 0xffff); - spin_unlock(&chip->reg_lock); + spin_unlock(codec->lock); - snd_azf3328_codec_setdmaa(chip, codec_type, runtime->dma_addr, + snd_azf3328_codec_setdmaa(codec, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream) ); - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); #ifdef WIN9X /* FIXME: enable playback/recording??? */ flags1 |= DMA_RUN_SOMETHING1 | DMA_RUN_SOMETHING2; @@ -1261,10 +1254,10 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, DMA_EPILOGUE_SOMETHING | DMA_SOMETHING_ELSE); #endif - spin_unlock(&chip->reg_lock); - snd_azf3328_ctrl_codec_activity(chip, codec_type, 1); + spin_unlock(codec->lock); + snd_azf3328_ctrl_codec_activity(chip, codec->type, 1); - if (is_playback_codec) { + if (is_main_mixer_playback_codec) { /* now unmute WaveOut */ if (!previously_muted) snd_azf3328_mixer_set_mute( @@ -1277,19 +1270,19 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgcodec("RESUME %s\n", codec->name); /* resume codec if we were active */ - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); if (codec->running) snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, snd_azf3328_codec_inw( codec, IDX_IO_CODEC_DMA_FLAGS ) | DMA_RESUME ); - spin_unlock(&chip->reg_lock); + spin_unlock(codec->lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgcodec("STOP %s\n", codec->name); - if (is_playback_codec) { + if (is_main_mixer_playback_codec) { /* mute WaveOut (avoid clicking during setup) */ previously_muted = snd_azf3328_mixer_set_mute( @@ -1297,7 +1290,7 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, ); } - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); /* first, remember current value: */ flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS); @@ -1312,10 +1305,10 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, flags1 &= ~DMA_RUN_SOMETHING1; snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); - spin_unlock(&chip->reg_lock); - snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); + spin_unlock(codec->lock); + snd_azf3328_ctrl_codec_activity(chip, codec->type, 0); - if (is_playback_codec) { + if (is_main_mixer_playback_codec) { /* now unmute WaveOut */ if (!previously_muted) snd_azf3328_mixer_set_mute( @@ -1349,31 +1342,12 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, return result; } -static int -snd_azf3328_codec_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_azf3328_codec_trigger(AZF_CODEC_PLAYBACK, substream, cmd); -} - -static int -snd_azf3328_codec_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_azf3328_codec_trigger(AZF_CODEC_CAPTURE, substream, cmd); -} - -static int -snd_azf3328_codec_i2s_out_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_azf3328_codec_trigger(AZF_CODEC_I2S_OUT, substream, cmd); -} - static snd_pcm_uframes_t -snd_azf3328_codec_pointer(struct snd_pcm_substream *substream, - enum snd_azf3328_codec_type codec_type +snd_azf3328_pcm_pointer(struct snd_pcm_substream *substream ) { - const struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; + const struct snd_azf3328_codec_data *codec = + substream->runtime->private_data; unsigned long result; snd_pcm_uframes_t frmres; @@ -1391,24 +1365,6 @@ snd_azf3328_codec_pointer(struct snd_pcm_substream *substream, return frmres; } -static snd_pcm_uframes_t -snd_azf3328_codec_playback_pointer(struct snd_pcm_substream *substream) -{ - return snd_azf3328_codec_pointer(substream, AZF_CODEC_PLAYBACK); -} - -static snd_pcm_uframes_t -snd_azf3328_codec_capture_pointer(struct snd_pcm_substream *substream) -{ - return snd_azf3328_codec_pointer(substream, AZF_CODEC_CAPTURE); -} - -static snd_pcm_uframes_t -snd_azf3328_codec_i2s_out_pointer(struct snd_pcm_substream *substream) -{ - return snd_azf3328_codec_pointer(substream, AZF_CODEC_I2S_OUT); -} - /******************************************************************/ #ifdef SUPPORT_GAMEPORT @@ -1642,29 +1598,29 @@ snd_azf3328_irq_log_unknown_type(u8 which) } static inline void -snd_azf3328_codec_interrupt(struct snd_azf3328 *chip, u8 status) +snd_azf3328_pcm_interrupt(const struct snd_azf3328_codec_data *first_codec, + u8 status +) { u8 which; enum snd_azf3328_codec_type codec_type; - const struct snd_azf3328_codec_data *codec; + const struct snd_azf3328_codec_data *codec = first_codec; for (codec_type = AZF_CODEC_PLAYBACK; codec_type <= AZF_CODEC_I2S_OUT; - ++codec_type) { + ++codec_type, ++codec) { /* skip codec if there's no interrupt for it */ if (!(status & (1 << codec_type))) continue; - codec = &chip->codecs[codec_type]; - - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); which = snd_azf3328_codec_inb(codec, IDX_IO_CODEC_IRQTYPE); /* ack all IRQ types immediately */ snd_azf3328_codec_outb(codec, IDX_IO_CODEC_IRQTYPE, which); - spin_unlock(&chip->reg_lock); + spin_unlock(codec->lock); - if ((chip->pcm[codec_type]) && (codec->substream)) { + if (codec->substream) { snd_pcm_period_elapsed(codec->substream); snd_azf3328_dbgcodec("%s period done (#%x), @ %x\n", codec->name, @@ -1719,7 +1675,7 @@ snd_azf3328_interrupt(int irq, void *dev_id) } if (status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_I2S_OUT)) - snd_azf3328_codec_interrupt(chip, status); + snd_azf3328_pcm_interrupt(chip->codecs, status); if (status & IRQ_GAMEPORT) snd_azf3328_gameport_interrupt(chip); @@ -1807,101 +1763,85 @@ snd_azf3328_pcm_open(struct snd_pcm_substream *substream, { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; snd_azf3328_dbgcallenter(); - chip->codecs[codec_type].substream = substream; + codec->substream = substream; /* same parameters for all our codecs - at least we think so... */ runtime->hw = snd_azf3328_hardware; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); + runtime->private_data = codec; snd_azf3328_dbgcallleave(); return 0; } static int -snd_azf3328_playback_open(struct snd_pcm_substream *substream) +snd_azf3328_pcm_playback_open(struct snd_pcm_substream *substream) { return snd_azf3328_pcm_open(substream, AZF_CODEC_PLAYBACK); } static int -snd_azf3328_capture_open(struct snd_pcm_substream *substream) +snd_azf3328_pcm_capture_open(struct snd_pcm_substream *substream) { return snd_azf3328_pcm_open(substream, AZF_CODEC_CAPTURE); } static int -snd_azf3328_i2s_out_open(struct snd_pcm_substream *substream) +snd_azf3328_pcm_i2s_out_open(struct snd_pcm_substream *substream) { return snd_azf3328_pcm_open(substream, AZF_CODEC_I2S_OUT); } static int -snd_azf3328_pcm_close(struct snd_pcm_substream *substream, - enum snd_azf3328_codec_type codec_type +snd_azf3328_pcm_close(struct snd_pcm_substream *substream ) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + struct snd_azf3328_codec_data *codec = + substream->runtime->private_data; snd_azf3328_dbgcallenter(); - chip->codecs[codec_type].substream = NULL; + codec->substream = NULL; snd_azf3328_dbgcallleave(); return 0; } -static int -snd_azf3328_playback_close(struct snd_pcm_substream *substream) -{ - return snd_azf3328_pcm_close(substream, AZF_CODEC_PLAYBACK); -} - -static int -snd_azf3328_capture_close(struct snd_pcm_substream *substream) -{ - return snd_azf3328_pcm_close(substream, AZF_CODEC_CAPTURE); -} - -static int -snd_azf3328_i2s_out_close(struct snd_pcm_substream *substream) -{ - return snd_azf3328_pcm_close(substream, AZF_CODEC_I2S_OUT); -} - /******************************************************************/ static struct snd_pcm_ops snd_azf3328_playback_ops = { - .open = snd_azf3328_playback_open, - .close = snd_azf3328_playback_close, + .open = snd_azf3328_pcm_playback_open, + .close = snd_azf3328_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_codec_prepare, - .trigger = snd_azf3328_codec_playback_trigger, - .pointer = snd_azf3328_codec_playback_pointer + .prepare = snd_azf3328_pcm_prepare, + .trigger = snd_azf3328_pcm_trigger, + .pointer = snd_azf3328_pcm_pointer }; static struct snd_pcm_ops snd_azf3328_capture_ops = { - .open = snd_azf3328_capture_open, - .close = snd_azf3328_capture_close, + .open = snd_azf3328_pcm_capture_open, + .close = snd_azf3328_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_codec_prepare, - .trigger = snd_azf3328_codec_capture_trigger, - .pointer = snd_azf3328_codec_capture_pointer + .prepare = snd_azf3328_pcm_prepare, + .trigger = snd_azf3328_pcm_trigger, + .pointer = snd_azf3328_pcm_pointer }; static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { - .open = snd_azf3328_i2s_out_open, - .close = snd_azf3328_i2s_out_close, + .open = snd_azf3328_pcm_i2s_out_open, + .close = snd_azf3328_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_codec_prepare, - .trigger = snd_azf3328_codec_i2s_out_trigger, - .pointer = snd_azf3328_codec_i2s_out_pointer + .prepare = snd_azf3328_pcm_prepare, + .trigger = snd_azf3328_pcm_trigger, + .pointer = snd_azf3328_pcm_pointer }; static int __devinit @@ -2198,7 +2138,7 @@ snd_azf3328_create(struct snd_card *card, }; u8 dma_init; enum snd_azf3328_codec_type codec_type; - struct snd_azf3328_codec *codec_setup; + struct snd_azf3328_codec_data *codec_setup; *rchip = NULL; @@ -2238,14 +2178,20 @@ snd_azf3328_create(struct snd_card *card, codec_setup = &chip->codecs[AZF_CODEC_PLAYBACK]; codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_PLAYBACK; + codec_setup->lock = &chip->reg_lock; + codec_setup->type = AZF_CODEC_PLAYBACK; codec_setup->name = "PLAYBACK"; codec_setup = &chip->codecs[AZF_CODEC_CAPTURE]; codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_CAPTURE; + codec_setup->lock = &chip->reg_lock; + codec_setup->type = AZF_CODEC_CAPTURE; codec_setup->name = "CAPTURE"; codec_setup = &chip->codecs[AZF_CODEC_I2S_OUT]; codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_I2S_OUT; + codec_setup->lock = &chip->reg_lock; + codec_setup->type = AZF_CODEC_I2S_OUT; codec_setup->name = "I2S_OUT"; if (request_irq(pci->irq, snd_azf3328_interrupt, @@ -2283,10 +2229,10 @@ snd_azf3328_create(struct snd_card *card, codec->running = 1; snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); - spin_lock_irq(&chip->reg_lock); + spin_lock_irq(codec->lock); snd_azf3328_codec_outb(codec, IDX_IO_CODEC_DMA_FLAGS, dma_init); - spin_unlock_irq(&chip->reg_lock); + spin_unlock_irq(codec->lock); } snd_card_set_dev(card, &pci->dev); -- cgit v0.10.2 From 689c69120ea3c8db069e11a7065ceffee90d0460 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Mon, 27 Dec 2010 21:17:35 +0100 Subject: ALSA: azt3328: improve snd_azf3328_codec_setdmaa() - add some WARN_ONCE - add multi-I/O helper (and use helper struct) - fix off-by-1 DMA length bug - better variable naming Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 76ff5fd..6117595 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -175,6 +175,7 @@ #include #include +#include /* WARN_ONCE */ #include #include #include @@ -421,6 +422,21 @@ snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec, outl(value, codec->io_base + reg); } +static inline void +snd_azf3328_codec_outl_multi(const struct snd_azf3328_codec_data *codec, + unsigned reg, const void *buffer, int count +) +{ + unsigned long addr = codec->io_base + reg; + if (count) { + const u32 *buf = buffer; + do { + outl(*buf++, addr); + addr += 4; + } while (--count); + } +} + static inline u32 snd_azf3328_codec_inl(const struct snd_azf3328_codec_data *codec, unsigned reg) { @@ -1124,34 +1140,54 @@ snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, static void snd_azf3328_codec_setdmaa(struct snd_azf3328_codec_data *codec, unsigned long addr, - unsigned int count, - unsigned int size + unsigned int period_bytes, + unsigned int buffer_bytes ) { snd_azf3328_dbgcallenter(); + WARN_ONCE(period_bytes & 1, "odd period length!?\n"); + WARN_ONCE(buffer_bytes != 2 * period_bytes, + "missed our input expectations! %u vs. %u\n", + buffer_bytes, period_bytes); if (!codec->running) { /* AZF3328 uses a two buffer pointer DMA transfer approach */ - unsigned long flags, addr_area2; + unsigned long flags; /* width 32bit (prevent overflow): */ - u32 count_areas, lengths; + u32 area_length; + struct codec_setup_io { + u32 dma_start_1; + u32 dma_start_2; + u32 dma_lengths; + } __attribute__((packed)) setup_io; - count_areas = size/2; - addr_area2 = addr+count_areas; - snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n", - addr, count_areas, addr_area2, count_areas); + area_length = buffer_bytes/2; - count_areas--; /* max. index */ + setup_io.dma_start_1 = addr; + setup_io.dma_start_2 = addr+area_length; + + snd_azf3328_dbgcodec( + "setdma: buffers %08x[%u] / %08x[%u], %u, %u\n", + setup_io.dma_start_1, area_length, + setup_io.dma_start_2, area_length, + period_bytes, buffer_bytes); + + /* Hmm, are we really supposed to decrement this by 1?? + Most definitely certainly not: configuring full length does + work properly (i.e. likely better), and BTW we + violated possibly differing frame sizes with this... + + area_length--; |* max. index *| + */ /* build combined I/O buffer length word */ - lengths = (count_areas << 16) | (count_areas); + setup_io.dma_lengths = (area_length << 16) | (area_length); + spin_lock_irqsave(codec->lock, flags); - snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_1, addr); - snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_2, - addr_area2); - snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_LENGTHS, - lengths); + snd_azf3328_codec_outl_multi( + codec, IDX_IO_CODEC_DMA_START_1, &setup_io, 3 + ); spin_unlock_irqrestore(codec->lock, flags); } snd_azf3328_dbgcallleave(); -- cgit v0.10.2 From 3daa7ea650d59ae86926f0ced4c4ccd99ab33fda Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 6 Jan 2011 22:19:47 +0100 Subject: ALSA: Don't leak in sound/core/oss/pcm_oss.c::snd_pcm_hw_param_near() snd_pcm_hw_param_near() will leak the memory allocated to 'save' if the call to snd_pcm_hw_param_max() returns less than zero. This patch makes sure we never leak. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index b753ec6..a2e4eb3 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -453,8 +453,10 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, } else { *params = *save; max = snd_pcm_hw_param_max(pcm, params, var, max, &maxdir); - if (max < 0) + if (max < 0) { + kfree(save); return max; + } last = 1; } _end: -- cgit v0.10.2 From 1f4d7be7293aecd5f8469a46f606f62f0f05d84c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 15:59:38 +0100 Subject: ALSA: oxygen: allow different number of PCM and mixer channels For cards like the Xonar HDAV1.3, differentiate between the number of PCM channels that can be played and the number of channels whose volume can be adjusted. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index dc47977..fe7ed4f 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -185,7 +185,7 @@ static void ak4396_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; - data->dacs = chip->model.dac_channels / 2; + data->dacs = chip->model.dac_channels_pcm / 2; data->ak4396_regs[0][AK4396_CONTROL_2] = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; ak4396_registers_init(chip); @@ -583,7 +583,8 @@ static const struct oxygen_model model_generic = { CAPTURE_1_FROM_SPDIF | CAPTURE_2_FROM_AC97_1 | AC97_CD_INPUT, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 8, .dac_volume_min = 0, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI | @@ -643,7 +644,8 @@ static int __devinit get_oxygen_model(struct oxygen *chip, PLAYBACK_1_TO_SPDIF; if (id->driver_data == MODEL_FANTASIA) chip->model.device_config |= CAPTURE_0_FROM_I2S_1; - chip->model.dac_channels = 2; + chip->model.dac_channels_pcm = 2; + chip->model.dac_channels_mixer = 2; break; } if (id->driver_data == MODEL_MERIDIAN || diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index b8fbc15..3d9535c 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -100,7 +100,8 @@ struct oxygen_model { unsigned long private_data; size_t model_data_size; unsigned int device_config; - u8 dac_channels; + u8 dac_channels_pcm; + u8 dac_channels_mixer; u8 dac_volume_min; u8 dac_volume_max; u8 misc_flags; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 605e84b..242c1ca 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -31,7 +31,7 @@ static int dac_volume_info(struct snd_kcontrol *ctl, struct oxygen *chip = ctl->private_data; info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = chip->model.dac_channels; + info->count = chip->model.dac_channels_mixer; info->value.integer.min = chip->model.dac_volume_min; info->value.integer.max = chip->model.dac_volume_max; return 0; @@ -44,7 +44,7 @@ static int dac_volume_get(struct snd_kcontrol *ctl, unsigned int i; mutex_lock(&chip->mutex); - for (i = 0; i < chip->model.dac_channels; ++i) + for (i = 0; i < chip->model.dac_channels_mixer; ++i) value->value.integer.value[i] = chip->dac_volume[i]; mutex_unlock(&chip->mutex); return 0; @@ -59,7 +59,7 @@ static int dac_volume_put(struct snd_kcontrol *ctl, changed = 0; mutex_lock(&chip->mutex); - for (i = 0; i < chip->model.dac_channels; ++i) + for (i = 0; i < chip->model.dac_channels_mixer; ++i) if (value->value.integer.value[i] != chip->dac_volume[i]) { chip->dac_volume[i] = value->value.integer.value[i]; changed = 1; @@ -1022,7 +1022,7 @@ static int add_controls(struct oxygen *chip, continue; } if (!strcmp(template.name, "Stereo Upmixing") && - chip->model.dac_channels == 2) + chip->model.dac_channels_pcm == 2) continue; if (!strcmp(template.name, "Mic Source Capture Enum") && !(chip->model.device_config & AC97_FMIC_SWITCH)) diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 60e4aa0..dc3c68f 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -143,7 +143,7 @@ static int oxygen_open(struct snd_pcm_substream *substream, runtime->hw.rate_min = 44100; break; case PCM_MULTICH: - runtime->hw.channels_max = chip->model.dac_channels; + runtime->hw.channels_max = chip->model.dac_channels_pcm; break; } if (chip->model.pcm_hardware_filter) diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index de32895..a25197c 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -426,7 +426,8 @@ static const struct oxygen_model model_xonar_d1 = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | AC97_FMIC_SWITCH, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 8, .dac_volume_min = 127 - 60, .dac_volume_max = 127, .function_flags = OXYGEN_FUNCTION_2WIRE, diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index bf357c0..b55149e 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -1094,7 +1094,8 @@ static const struct oxygen_model model_xonar_d2 = { MIDI_OUTPUT | MIDI_INPUT | AC97_CD_INPUT, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 8, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, @@ -1127,7 +1128,8 @@ static const struct oxygen_model model_xonar_hdav = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 2, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, @@ -1157,7 +1159,8 @@ static const struct oxygen_model model_xonar_st = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | AC97_FMIC_SWITCH, - .dac_channels = 2, + .dac_channels_pcm = 2, + .dac_channels_mixer = 2, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_2WIRE, @@ -1187,6 +1190,7 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, break; case GPIO_DB_H6: chip->model.shortname = "Xonar HDAV1.3+H6"; + chip->model.dac_channels_mixer = 8; chip->model.private_data = 1; break; } @@ -1200,7 +1204,8 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, break; case GPIO_DB_H6: chip->model.shortname = "Xonar ST+H6"; - chip->model.dac_channels = 8; + chip->model.dac_channels_pcm = 8; + chip->model.dac_channels_mixer = 8; chip->model.private_data = 1; break; } diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 1705d1e..da92cc2 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -1135,7 +1135,8 @@ static const struct oxygen_model model_xonar_ds = { .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_1, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 8, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI, -- cgit v0.10.2 From d2119c05e9aee7e44055220726bb8814a2e242c3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:00:34 +0100 Subject: ALSA: oxygen: remove oxygen_model::private_data field The number of DACs can now be deduced from the dac_channels_mixer field, so the private_data field is no longer needed. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 3d9535c..70eff37 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -97,7 +97,6 @@ struct oxygen_model { void (*dump_registers)(struct oxygen *chip, struct snd_info_buffer *buffer); const unsigned int *dac_tlv; - unsigned long private_data; size_t model_data_size; unsigned int device_config; u8 dac_channels_pcm; diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index b55149e..5ec8be3 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -381,7 +381,7 @@ static void xonar_hdav_init(struct oxygen *chip) data->pcm179x.generic.ext_power_reg = OXYGEN_GPI_DATA; data->pcm179x.generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->pcm179x.generic.ext_power_bit = GPI_EXT_POWER; - data->pcm179x.dacs = chip->model.private_data ? 4 : 1; + data->pcm179x.dacs = chip->model.dac_channels_mixer / 2; pcm1796_init(chip); @@ -411,7 +411,7 @@ static void xonar_st_init_common(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; - data->dacs = chip->model.private_data ? 4 : 1; + data->dacs = chip->model.dac_channels_mixer / 2; data->hp_gain_offset = 2*-18; pcm1796_init(chip); @@ -1191,7 +1191,6 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, case GPIO_DB_H6: chip->model.shortname = "Xonar HDAV1.3+H6"; chip->model.dac_channels_mixer = 8; - chip->model.private_data = 1; break; } break; @@ -1206,7 +1205,6 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, chip->model.shortname = "Xonar ST+H6"; chip->model.dac_channels_pcm = 8; chip->model.dac_channels_mixer = 8; - chip->model.private_data = 1; break; } break; -- cgit v0.10.2 From 5ea310ff8d651246cf001ebc894d2f294123328a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:01:17 +0100 Subject: ALSA: oxygen: fix SPI clocks slower than 6.25 MHz Fix wrong register bits for SPI clock cycle times longer than 160 ns, and adjust the polling loop timeout for these speeds. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 09b2b2a..f5164b1 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -197,11 +197,11 @@ void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; - /* should not need more than 7.68 us (24 * 320 ns) */ + /* should not need more than 30.72 us (24 * 1.28 us) */ count = 10; while ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & OXYGEN_SPI_BUSY) && count > 0) { - udelay(1); + udelay(4); --count; } diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index 4dcd41b..331d3d9 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -238,11 +238,11 @@ #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_MASK 0x0c #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_CLOCK_320 0x04 +#define OXYGEN_SPI_CLOCK_640 0x08 +#define OXYGEN_SPI_CLOCK_1280 0x0c #define OXYGEN_SPI_CODEC_MASK 0x70 /* 0..5 */ #define OXYGEN_SPI_CODEC_SHIFT 4 #define OXYGEN_SPI_CEN_MASK 0x80 -- cgit v0.10.2 From 4106055cedea86596391f36deacd05616333fbb3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:01:57 +0100 Subject: ALSA: virtuoso: do not use fast I2C speed To make the I2C communication reliable when using the H6 daughterboard, reduce the I2C clock frequency. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 5ec8be3..49898d3 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -374,7 +374,7 @@ static void xonar_hdav_init(struct oxygen *chip) oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); + OXYGEN_2WIRE_SPEED_STANDARD); data->pcm179x.generic.anti_pop_delay = 100; data->pcm179x.generic.output_enable_bit = GPIO_HDAV_OUTPUT_ENABLE; @@ -403,7 +403,7 @@ static void xonar_st_init_i2c(struct oxygen *chip) oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); + OXYGEN_2WIRE_SPEED_STANDARD); } static void xonar_st_init_common(struct oxygen *chip) -- cgit v0.10.2 From 79815e004c75dcc2be6433737b5678cee80035db Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:02:32 +0100 Subject: ALSA: virtuoso: wait for PCM1796 clock to become stable The PCM1796 needs the master clock for I2C communication to work, so add delays after clock changes to ensure that the clock is stable when we try to write the DACs' registers. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 49898d3..5ec164a 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -303,6 +303,7 @@ static void pcm1796_registers_init(struct oxygen *chip) unsigned int i; s8 gain_offset; + msleep(1); gain_offset = data->hp_active ? data->hp_gain_offset : 0; for (i = 0; i < data->dacs; ++i) { /* set ATLD before ATL/ATR */ @@ -451,6 +452,7 @@ static void cs2000_registers_init(struct oxygen *chip) data->cs2000_regs[CS2000_FUN_CFG_1]); cs2000_write(chip, CS2000_FUN_CFG_2, 0); cs2000_write(chip, CS2000_GLOBAL_CFG, CS2000_EN_DEV_CFG_2); + msleep(3); /* PLL lock delay */ } static void xonar_st_init(struct oxygen *chip) @@ -592,6 +594,7 @@ static void set_pcm1796_params(struct oxygen *chip, { struct xonar_pcm179x *data = chip->model_data; + msleep(1); data->current_rate = params_rate(params); update_pcm1796_oversampling(chip); } @@ -669,6 +672,7 @@ static void update_cs2000_rate(struct oxygen *chip, unsigned int rate) else reg = CS2000_REF_CLK_DIV_2; cs2000_write_cached(chip, CS2000_FUN_CFG_1, reg); + msleep(3); /* PLL lock delay */ } static void set_st_params(struct oxygen *chip, @@ -796,6 +800,7 @@ static int os_128_put(struct snd_kcontrol *ctl, oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, mclk_from_rate(chip, data->current_rate), OXYGEN_I2S_MCLK_MASK); + msleep(1); update_pcm1796_oversampling(chip); } mutex_unlock(&chip->mutex); -- cgit v0.10.2 From 03ff959dd4e290ed909fd57dec79ccd0262c4095 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:03:17 +0100 Subject: ALSA: virtuoso: change PCM1796 format to I2S Change the PCM format used for the PCM1796 from left-justified to I2S to ensure that the correct format is used even for the Essence ST Deluxe's center/LFE DAC, where I2C does not work because of an address conflict with the CS2000 chip. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 5ec164a..051554e 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -326,7 +326,7 @@ static void pcm1796_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = PCM1796_MUTE | - PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; data->pcm1796_regs[0][19 - PCM1796_REG_BASE] = PCM1796_FLT_SHARP | PCM1796_ATS_1; data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = PCM1796_OS_64; @@ -620,7 +620,7 @@ static void update_pcm1796_mute(struct oxygen *chip) unsigned int i; u8 value; - value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; if (chip->dac_mute) value |= PCM1796_MUTE; for (i = 0; i < data->dacs; ++i) @@ -1106,7 +1106,7 @@ static const struct oxygen_model model_xonar_d2 = { .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1139,7 +1139,7 @@ static const struct oxygen_model model_xonar_hdav = { .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1169,7 +1169,7 @@ static const struct oxygen_model model_xonar_st = { .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; -- cgit v0.10.2 From dd203fa97bd5df18dbb0af5acf3e9a8beea33f74 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:05:38 +0100 Subject: ALSA: virtuoso: remove non-working controls on Essence ST Deluxe On the Xonar Essence ST Deluxe, remove all mixer controls that would require I2C communication with the third DAC, which does not work because of an addressing conflict with the CS2000 chip. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 051554e..2e31b81 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -228,6 +228,7 @@ struct xonar_pcm179x { s8 hp_gain_offset; bool has_cs2000; u8 cs2000_regs[0x1f]; + bool broken_i2c; }; struct xonar_hdav { @@ -462,6 +463,7 @@ static void xonar_st_init(struct oxygen *chip) data->generic.anti_pop_delay = 100; data->has_cs2000 = 1; data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; + data->broken_i2c = true; oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_I2S | @@ -980,16 +982,29 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template) return 0; } +static int xonar_st_h6_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "Master Playback ", 16)) + /* no volume/mute, as I²C to the third DAC does not work */ + return 1; + return 0; +} + static int add_pcm1796_controls(struct oxygen *chip) { + struct xonar_pcm179x *data = chip->model_data; int err; - err = snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); - if (err < 0) - return err; - err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); - if (err < 0) - return err; + if (!data->broken_i2c) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&rolloff_control, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&os_128_control, chip)); + if (err < 0) + return err; + } return 0; } @@ -1208,6 +1223,7 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, break; case GPIO_DB_H6: chip->model.shortname = "Xonar ST+H6"; + chip->model.control_filter = xonar_st_h6_control_filter; chip->model.dac_channels_pcm = 8; chip->model.dac_channels_mixer = 8; break; -- cgit v0.10.2 From d353eaa9a8133cdad8c1da23c84f9f529a23f0c2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:07:11 +0100 Subject: ALSA: virtuoso: configure correct master clock frequency on the CS2000 The clock output of the CS2000, which is used as master clock for the DACs, was using half the actual master clock frequency for some reason. Using the theoretically correct frequency seems also to work in practice. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 2e31b81..fce55fa 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -467,7 +467,7 @@ static void xonar_st_init(struct oxygen *chip) oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_I2S | - OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); xonar_st_init_i2c(chip); @@ -635,41 +635,40 @@ static void update_cs2000_rate(struct oxygen *chip, unsigned int rate) u8 rate_mclk, reg; switch (rate) { - /* XXX Why is the I2S A MCLK half the actual I2S MCLK? */ case 32000: - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_512; break; case 44100: if (data->os_128) - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; else - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_128; + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; break; default: /* 48000 */ if (data->os_128) - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; else - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_128; + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; break; case 64000: - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_512; break; case 88200: - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; break; case 96000: - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; break; case 176400: - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; break; case 192000: - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; break; } oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); - if ((rate_mclk & OXYGEN_I2S_MCLK_MASK) <= OXYGEN_I2S_MCLK_128) + if ((rate_mclk & OXYGEN_I2S_MCLK_MASK) <= OXYGEN_I2S_MCLK_256) reg = CS2000_REF_CLK_DIV_1; else reg = CS2000_REF_CLK_DIV_2; -- cgit v0.10.2 From 00b8dd7dd71ef129176731d5fa24f5e298797599 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:09:23 +0100 Subject: ALSA: virtuoso: use lower master clock with H6 daughterboard Because of the unshielded connector cable, it is important to use as low a master clock frequency as possible with the H6. For double rate modes (64-96 kHz), the MCLK rate is unconditionally lowered from 512x to 256x because the higher rate would not improve anything. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index fce55fa..2a50a55 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -224,6 +224,7 @@ struct xonar_pcm179x { u8 pcm1796_regs[4][5]; unsigned int current_rate; bool os_128; + bool h6; bool hp_active; s8 hp_gain_offset; bool has_cs2000; @@ -384,6 +385,7 @@ static void xonar_hdav_init(struct oxygen *chip) data->pcm179x.generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->pcm179x.generic.ext_power_bit = GPI_EXT_POWER; data->pcm179x.dacs = chip->model.dac_channels_mixer / 2; + data->pcm179x.h6 = chip->model.dac_channels_mixer > 2; pcm1796_init(chip); @@ -461,6 +463,7 @@ static void xonar_st_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->generic.anti_pop_delay = 100; + data->h6 = chip->model.dac_channels_mixer > 2; data->has_cs2000 = 1; data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; data->broken_i2c = true; @@ -554,11 +557,10 @@ static unsigned int mclk_from_rate(struct oxygen *chip, unsigned int rate) struct xonar_pcm179x *data = chip->model_data; if (rate <= 32000) - return OXYGEN_I2S_MCLK_512; - else if (rate <= 48000 && data->os_128) - return OXYGEN_I2S_MCLK_512; - else if (rate <= 96000) - return OXYGEN_I2S_MCLK_256; + return data->h6 ? OXYGEN_I2S_MCLK_256 : OXYGEN_I2S_MCLK_512; + else if (rate <= 48000) + return data->os_128 && !data->h6 + ? OXYGEN_I2S_MCLK_512 : OXYGEN_I2S_MCLK_256; else return OXYGEN_I2S_MCLK_128; } @@ -579,9 +581,9 @@ static void update_pcm1796_oversampling(struct oxygen *chip) unsigned int i; u8 reg; - if (data->current_rate <= 32000) + if (data->current_rate <= 32000 && !data->h6) reg = PCM1796_OS_128; - else if (data->current_rate <= 48000 && data->os_128) + else if (data->current_rate <= 48000 && data->os_128 && !data->h6) reg = PCM1796_OS_128; else if (data->current_rate <= 96000 || data->os_128) reg = PCM1796_OS_64; @@ -636,28 +638,31 @@ static void update_cs2000_rate(struct oxygen *chip, unsigned int rate) switch (rate) { case 32000: - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_512; + if (data->h6) + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + else + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_512; break; case 44100: - if (data->os_128) + if (data->os_128 && !data->h6) rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; else rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; break; default: /* 48000 */ - if (data->os_128) + if (data->os_128 && !data->h6) rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; else rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; break; case 64000: - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_512; + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; break; case 88200: - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; break; case 96000: - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; break; case 176400: rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; -- cgit v0.10.2 From c97e2dc48457642e2f1c6183b986549b7fa0113a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:11:05 +0100 Subject: ALSA: virtuoso: handle DAC oversampling automatically Remove the DAC Oversampling mixer control because this setting does not make much sense. For cards with the H6 daughterboard, 128x oversampling was disabled anyway because these high MCLK frequency would not be compatible with the connector cable. For cards without the H6 daughterboard, 128x gives a slightly higher output quality; there is no reason to reduce it to 64x except for saving power, but then these cards have not been designed to be power efficient anyway (the D2's blinkenlights cannot be disabled). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 2a50a55..2d0a634 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -223,7 +223,6 @@ struct xonar_pcm179x { unsigned int dacs; u8 pcm1796_regs[4][5]; unsigned int current_rate; - bool os_128; bool h6; bool hp_active; s8 hp_gain_offset; @@ -331,9 +330,14 @@ static void pcm1796_init(struct oxygen *chip) PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; data->pcm1796_regs[0][19 - PCM1796_REG_BASE] = PCM1796_FLT_SHARP | PCM1796_ATS_1; - data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = PCM1796_OS_64; + data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = + data->h6 ? PCM1796_OS_64 : PCM1796_OS_128; pcm1796_registers_init(chip); data->current_rate = 48000; + if (!data->h6) + oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, + OXYGEN_I2S_MCLK_512, + OXYGEN_I2S_MCLK_MASK); } static void xonar_d2_init(struct oxygen *chip) @@ -469,9 +473,12 @@ static void xonar_st_init(struct oxygen *chip) data->broken_i2c = true; oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, - OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_I2S | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_RATE_48000 | + OXYGEN_I2S_FORMAT_I2S | + (data->h6 ? OXYGEN_I2S_MCLK_256 : OXYGEN_I2S_MCLK_512) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); xonar_st_init_i2c(chip); cs2000_registers_init(chip); @@ -556,11 +563,8 @@ static unsigned int mclk_from_rate(struct oxygen *chip, unsigned int rate) { struct xonar_pcm179x *data = chip->model_data; - if (rate <= 32000) + if (rate <= 48000) return data->h6 ? OXYGEN_I2S_MCLK_256 : OXYGEN_I2S_MCLK_512; - else if (rate <= 48000) - return data->os_128 && !data->h6 - ? OXYGEN_I2S_MCLK_512 : OXYGEN_I2S_MCLK_256; else return OXYGEN_I2S_MCLK_128; } @@ -581,14 +585,10 @@ static void update_pcm1796_oversampling(struct oxygen *chip) unsigned int i; u8 reg; - if (data->current_rate <= 32000 && !data->h6) + if (data->current_rate <= 48000 && !data->h6) reg = PCM1796_OS_128; - else if (data->current_rate <= 48000 && data->os_128 && !data->h6) - reg = PCM1796_OS_128; - else if (data->current_rate <= 96000 || data->os_128) - reg = PCM1796_OS_64; else - reg = PCM1796_OS_32; + reg = PCM1796_OS_64; for (i = 0; i < data->dacs; ++i) pcm1796_write_cached(chip, i, 20, reg); } @@ -644,16 +644,16 @@ static void update_cs2000_rate(struct oxygen *chip, unsigned int rate) rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_512; break; case 44100: - if (data->os_128 && !data->h6) - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; - else + if (data->h6) rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + else + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; break; default: /* 48000 */ - if (data->os_128 && !data->h6) - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; - else + if (data->h6) rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + else + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; break; case 64000: rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; @@ -767,60 +767,6 @@ static const struct snd_kcontrol_new rolloff_control = { .put = rolloff_put, }; -static int os_128_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) -{ - static const char *const names[2] = { "64x", "128x" }; - - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; -} - -static int os_128_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct xonar_pcm179x *data = chip->model_data; - - value->value.enumerated.item[0] = data->os_128; - return 0; -} - -static int os_128_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct xonar_pcm179x *data = chip->model_data; - int changed; - - mutex_lock(&chip->mutex); - changed = value->value.enumerated.item[0] != data->os_128; - if (changed) { - data->os_128 = value->value.enumerated.item[0]; - if (data->has_cs2000) - update_cs2000_rate(chip, data->current_rate); - oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, - mclk_from_rate(chip, data->current_rate), - OXYGEN_I2S_MCLK_MASK); - msleep(1); - update_pcm1796_oversampling(chip); - } - mutex_unlock(&chip->mutex); - return changed; -} - -static const struct snd_kcontrol_new os_128_control = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "DAC Oversampling Playback Enum", - .info = os_128_info, - .get = os_128_get, - .put = os_128_put, -}; - static const struct snd_kcontrol_new hdav_hdmi_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "HDMI Playback Switch", @@ -1004,10 +950,6 @@ static int add_pcm1796_controls(struct oxygen *chip) snd_ctl_new1(&rolloff_control, chip)); if (err < 0) return err; - err = snd_ctl_add(chip->card, - snd_ctl_new1(&os_128_control, chip)); - if (err < 0) - return err; } return 0; } -- cgit v0.10.2 From bc29e262c3062682c6099bd455ae8544916f723e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:11:32 +0100 Subject: ALSA: virtuoso: use headphone gain setting only on front DAC Do not apply the headphone gain offset to any but the front DAC. These DACs would not be used in headphone mode, so this saves a few register writes. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 2d0a634..e15ecee 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -319,6 +319,7 @@ static void pcm1796_registers_init(struct oxygen *chip) pcm1796_write(chip, i, 20, data->pcm1796_regs[0][20 - PCM1796_REG_BASE]); pcm1796_write(chip, i, 21, 0); + gain_offset = 0; } } @@ -615,6 +616,7 @@ static void update_pcm1796_volume(struct oxygen *chip) + gain_offset); pcm1796_write_cached(chip, i, 17, chip->dac_volume[i * 2 + 1] + gain_offset); + gain_offset = 0; } } -- cgit v0.10.2 From 5b8bf2a54fb13e40519ee846ce27bc8a2d7a7878 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:14:52 +0100 Subject: ALSA: oxygen: simplify model-specific MCLK handling Replace the get_i2s_mclk callback with tables of MCLK values. This simplifies the MCLK-handling code in both the framework and the model- specific drivers. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index fe7ed4f..784d500 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -568,7 +568,6 @@ static const struct oxygen_model model_generic = { .mixer_init = generic_wm8785_mixer_init, .cleanup = generic_cleanup, .resume = generic_resume, - .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, @@ -589,6 +588,8 @@ static const struct oxygen_model model_generic = { .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 70eff37..2c5fb9e 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -16,6 +16,10 @@ #define PCM_AC97 5 #define PCM_COUNT 6 +#define OXYGEN_MCLKS(f_single, f_double, f_quad) ((MCLK_##f_single << 0) | \ + (MCLK_##f_double << 2) | \ + (MCLK_##f_quad << 4)) + #define OXYGEN_IO_SIZE 0x100 #define OXYGEN_EEPROM_ID 0x434d /* "CM" */ @@ -81,8 +85,6 @@ struct oxygen_model { void (*resume)(struct oxygen *chip); void (*pcm_hardware_filter)(unsigned int channel, struct snd_pcm_hardware *hardware); - unsigned int (*get_i2s_mclk)(struct oxygen *chip, unsigned int channel, - struct snd_pcm_hw_params *hw_params); void (*set_dac_params)(struct oxygen *chip, struct snd_pcm_hw_params *params); void (*set_adc_params)(struct oxygen *chip, @@ -105,6 +107,8 @@ struct oxygen_model { u8 dac_volume_max; u8 misc_flags; u8 function_flags; + u8 dac_mclks; + u8 adc_mclks; u16 dac_i2s_format; u16 adc_i2s_format; }; @@ -171,8 +175,6 @@ void oxygen_update_spdif_source(struct oxygen *chip); /* oxygen_pcm.c */ int oxygen_pcm_init(struct oxygen *chip); -unsigned int oxygen_default_i2s_mclk(struct oxygen *chip, unsigned int channel, - struct snd_pcm_hw_params *hw_params); /* oxygen_io.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index c44c91e..77e1f08 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -414,28 +414,40 @@ static void oxygen_init(struct oxygen *chip) (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 | chip->model.dac_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_RATE_48000 | + chip->model.dac_i2s_format | + OXYGEN_I2S_MCLK(chip->model.dac_mclks) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); if (chip->model.device_config & CAPTURE_0_FROM_I2S_1) oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, - OXYGEN_RATE_48000 | chip->model.adc_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_RATE_48000 | + chip->model.adc_i2s_format | + OXYGEN_I2S_MCLK(chip->model.adc_mclks) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); else oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, - OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); + OXYGEN_I2S_MASTER | + OXYGEN_I2S_MUTE_MCLK); if (chip->model.device_config & (CAPTURE_0_FROM_I2S_2 | CAPTURE_2_FROM_I2S_2)) oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, - OXYGEN_RATE_48000 | chip->model.adc_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_RATE_48000 | + chip->model.adc_i2s_format | + OXYGEN_I2S_MCLK(chip->model.adc_mclks) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); else oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, - OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); + OXYGEN_I2S_MASTER | + OXYGEN_I2S_MUTE_MCLK); oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, - OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); + OXYGEN_I2S_MASTER | + OXYGEN_I2S_MUTE_MCLK); oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_OUT_ENABLE | OXYGEN_SPDIF_LOOPBACK); diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index dc3c68f..d5533e3 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -274,17 +274,6 @@ static unsigned int oxygen_rate(struct snd_pcm_hw_params *hw_params) } } -unsigned int oxygen_default_i2s_mclk(struct oxygen *chip, - unsigned int channel, - struct snd_pcm_hw_params *hw_params) -{ - if (params_rate(hw_params) <= 96000) - return OXYGEN_I2S_MCLK_256; - else - return OXYGEN_I2S_MCLK_128; -} -EXPORT_SYMBOL(oxygen_default_i2s_mclk); - static unsigned int oxygen_i2s_bits(struct snd_pcm_hw_params *hw_params) { if (params_format(hw_params) == SNDRV_PCM_FORMAT_S32_LE) @@ -344,6 +333,26 @@ static int oxygen_hw_params(struct snd_pcm_substream *substream, return 0; } +static u16 get_mclk(struct oxygen *chip, unsigned int channel, + struct snd_pcm_hw_params *params) +{ + unsigned int mclks, shift; + + if (channel == PCM_MULTICH) + mclks = chip->model.dac_mclks; + else + mclks = chip->model.adc_mclks; + + if (params_rate(params) <= 48000) + shift = 0; + else if (params_rate(params) <= 96000) + shift = 2; + else + shift = 4; + + return OXYGEN_I2S_MCLK(mclks >> shift); +} + static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -360,8 +369,8 @@ 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) | - chip->model.get_i2s_mclk(chip, PCM_A, hw_params) | chip->model.adc_i2s_format | + get_mclk(chip, PCM_A, hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | @@ -396,9 +405,8 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, if (!is_ac97) oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT, oxygen_rate(hw_params) | - chip->model.get_i2s_mclk(chip, PCM_B, - hw_params) | chip->model.adc_i2s_format | + get_mclk(chip, PCM_B, hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | @@ -479,8 +487,7 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, oxygen_rate(hw_params) | chip->model.dac_i2s_format | - chip->model.get_i2s_mclk(chip, PCM_MULTICH, - hw_params) | + get_mclk(chip, PCM_MULTICH, hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index 331d3d9..63dc7a0 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -139,9 +139,11 @@ #define OXYGEN_I2S_FORMAT_I2S 0x0000 #define OXYGEN_I2S_FORMAT_LJUST 0x0008 #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_MCLK_SHIFT 4 +#define MCLK_128 0 +#define MCLK_256 1 +#define MCLK_512 2 +#define OXYGEN_I2S_MCLK(f) (((f) & 3) << OXYGEN_I2S_MCLK_SHIFT) #define OXYGEN_I2S_BITS_MASK 0x00c0 #define OXYGEN_I2S_BITS_16 0x0000 #define OXYGEN_I2S_BITS_20 0x0040 diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index a25197c..b651938 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -412,7 +412,6 @@ static const struct oxygen_model model_xonar_d1 = { .cleanup = xonar_d1_cleanup, .suspend = xonar_d1_suspend, .resume = xonar_d1_resume, - .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_cs43xx_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_cs43xx_volume, @@ -431,6 +430,8 @@ static const struct oxygen_model model_xonar_d1 = { .dac_volume_min = 127 - 60, .dac_volume_max = 127, .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index e15ecee..9787193 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -335,10 +335,6 @@ static void pcm1796_init(struct oxygen *chip) data->h6 ? PCM1796_OS_64 : PCM1796_OS_128; pcm1796_registers_init(chip); data->current_rate = 48000; - if (!data->h6) - oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, - OXYGEN_I2S_MCLK_512, - OXYGEN_I2S_MCLK_MASK); } static void xonar_d2_init(struct oxygen *chip) @@ -476,7 +472,7 @@ static void xonar_st_init(struct oxygen *chip) oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_I2S | - (data->h6 ? OXYGEN_I2S_MCLK_256 : OXYGEN_I2S_MCLK_512) | + OXYGEN_I2S_MCLK(data->h6 ? MCLK_256 : MCLK_512) | OXYGEN_I2S_BITS_16 | OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); @@ -560,26 +556,6 @@ static void xonar_st_resume(struct oxygen *chip) xonar_stx_resume(chip); } -static unsigned int mclk_from_rate(struct oxygen *chip, unsigned int rate) -{ - struct xonar_pcm179x *data = chip->model_data; - - if (rate <= 48000) - return data->h6 ? OXYGEN_I2S_MCLK_256 : OXYGEN_I2S_MCLK_512; - else - return OXYGEN_I2S_MCLK_128; -} - -static unsigned int get_pcm1796_i2s_mclk(struct oxygen *chip, - unsigned int channel, - struct snd_pcm_hw_params *params) -{ - if (channel == PCM_MULTICH) - return mclk_from_rate(chip, params_rate(params)); - else - return oxygen_default_i2s_mclk(chip, channel, params); -} - static void update_pcm1796_oversampling(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; @@ -640,45 +616,32 @@ static void update_cs2000_rate(struct oxygen *chip, unsigned int rate) switch (rate) { case 32000: - if (data->h6) - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; - else - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_512; - break; - case 44100: - if (data->h6) - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; - else - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; - break; - default: /* 48000 */ - if (data->h6) - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; - else - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; - break; case 64000: - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_32000; break; + case 44100: case 88200: - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; - break; - case 96000: - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; - break; case 176400: - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_512; + rate_mclk = OXYGEN_RATE_44100; break; + default: + case 48000: + case 96000: case 192000: - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_512; + rate_mclk = OXYGEN_RATE_48000; break; } - oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, - OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); - if ((rate_mclk & OXYGEN_I2S_MCLK_MASK) <= OXYGEN_I2S_MCLK_256) + + if (rate <= 96000 && (rate > 48000 || data->h6)) { + rate_mclk |= OXYGEN_I2S_MCLK(MCLK_256); reg = CS2000_REF_CLK_DIV_1; - else + } else { + rate_mclk |= OXYGEN_I2S_MCLK(MCLK_512); reg = CS2000_REF_CLK_DIV_2; + } + + oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, + OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); cs2000_write_cached(chip, CS2000_FUN_CFG_1, reg); msleep(3); /* PLL lock delay */ } @@ -1047,7 +1010,6 @@ static const struct oxygen_model model_xonar_d2 = { .cleanup = xonar_d2_cleanup, .suspend = xonar_d2_suspend, .resume = xonar_d2_resume, - .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_pcm1796_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -1069,6 +1031,8 @@ static const struct oxygen_model model_xonar_d2 = { .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_mclks = OXYGEN_MCLKS(512, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1082,7 +1046,6 @@ static const struct oxygen_model model_xonar_hdav = { .suspend = xonar_hdav_suspend, .resume = xonar_hdav_resume, .pcm_hardware_filter = xonar_hdmi_pcm_hardware_filter, - .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_hdav_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -1102,6 +1065,8 @@ static const struct oxygen_model model_xonar_hdav = { .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_mclks = OXYGEN_MCLKS(512, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1114,7 +1079,6 @@ static const struct oxygen_model model_xonar_st = { .cleanup = xonar_st_cleanup, .suspend = xonar_st_suspend, .resume = xonar_st_resume, - .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_st_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -1132,6 +1096,8 @@ static const struct oxygen_model model_xonar_st = { .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_mclks = OXYGEN_MCLKS(512, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1159,6 +1125,7 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, case GPIO_DB_H6: chip->model.shortname = "Xonar HDAV1.3+H6"; chip->model.dac_channels_mixer = 8; + chip->model.dac_mclks = OXYGEN_MCLKS(256, 128, 128); break; } break; @@ -1174,6 +1141,7 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, chip->model.control_filter = xonar_st_h6_control_filter; chip->model.dac_channels_pcm = 8; chip->model.dac_channels_mixer = 8; + chip->model.dac_mclks = OXYGEN_MCLKS(256, 128, 128); break; } break; diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index da92cc2..4f96570 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -1122,7 +1122,6 @@ static const struct oxygen_model model_xonar_ds = { .suspend = xonar_ds_suspend, .resume = xonar_ds_resume, .pcm_hardware_filter = wm8776_adc_hardware_filter, - .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_wm87x6_dac_params, .set_adc_params = set_wm8776_adc_params, .update_dac_volume = update_wm87x6_volume, @@ -1140,6 +1139,8 @@ static const struct oxygen_model model_xonar_ds = { .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI, + .dac_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; -- cgit v0.10.2 From ce2c492090aa55ff2764f473abdb3c5a76b4a7c4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:16:08 +0100 Subject: ALSA: virtuoso: reduce MCLK in double rate modes For the CSxxxx and AKxxxx DAC/ADC chips, the MCLK factor in double rate modes (64-96 kHz) can be reduced to 128x without reducing sound quality. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 784d500..2316884 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -588,7 +588,7 @@ static const struct oxygen_model model_generic = { .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, - .dac_mclks = OXYGEN_MCLKS(256, 256, 128), + .dac_mclks = OXYGEN_MCLKS(256, 128, 128), .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, @@ -643,8 +643,10 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.dump_registers = dump_ak4396_registers; chip->model.device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF; - if (id->driver_data == MODEL_FANTASIA) + if (id->driver_data == MODEL_FANTASIA) { chip->model.device_config |= CAPTURE_0_FROM_I2S_1; + chip->model.adc_mclks = OXYGEN_MCLKS(256, 128, 128); + } chip->model.dac_channels_pcm = 2; chip->model.dac_channels_mixer = 2; break; diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index b651938..55c52c7 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -430,8 +430,8 @@ static const struct oxygen_model model_xonar_d1 = { .dac_volume_min = 127 - 60, .dac_volume_max = 127, .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_mclks = OXYGEN_MCLKS(256, 256, 128), - .adc_mclks = OXYGEN_MCLKS(256, 256, 128), + .dac_mclks = OXYGEN_MCLKS(256, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 9787193..003c480 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -1032,7 +1032,7 @@ static const struct oxygen_model model_xonar_d2 = { .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_mclks = OXYGEN_MCLKS(512, 128, 128), - .adc_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1066,7 +1066,7 @@ static const struct oxygen_model model_xonar_hdav = { .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_mclks = OXYGEN_MCLKS(512, 128, 128), - .adc_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1097,7 +1097,7 @@ static const struct oxygen_model model_xonar_st = { .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_mclks = OXYGEN_MCLKS(512, 128, 128), - .adc_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; -- cgit v0.10.2 From 8c50b75979a198194a38d38d855f9d7e9cac2889 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:16:32 +0100 Subject: ALSA: oxygen: add more PCI IDs Add PCI IDs for some unknown models. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 2316884..e187c95 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -84,6 +84,7 @@ enum { static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { /* C-Media's reference design */ { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x10b0, 0x0217), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0218), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x0001), .driver_data = MODEL_CMEDIA_REF }, @@ -91,6 +92,8 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, + /* PCI 2.0 HD Audio */ + { OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT }, /* Kuroutoshikou CMI8787-HG2PCI */ { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_2CH_OUTPUT }, /* TempoTec HiFier Fantasia */ -- cgit v0.10.2 From 8443d2eb81e30dcc027e531eaa442cdb2477c5ab Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:17:26 +0100 Subject: ALSA: oxygen: add X-Meridian 2G support Add support for the AuzenTech X-Meridian 7.1 2G sound card. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7124340..5cd5c96 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1527,6 +1527,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on the C-Media CMI8787/8788 chip: * Asound A-8788 * AuzenTech X-Meridian + * AuzenTech X-Meridian 2G * Bgears b-Enspirer * Club3D Theatron DTS * HT-Omega Claro (plus) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7b2678a..ddb5e69 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -218,6 +218,7 @@ config SND_OXYGEN C-Media CMI8788 (Oxygen HD Audio) chip: * Asound A-8788 * AuzenTech X-Meridian + * AuzenTech X-Meridian 2G * Bgears b-Enspirer * Club3D Theatron DTS * HT-Omega Claro (plus) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index e187c95..304f1a5 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -102,6 +102,8 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_2CH_OUTPUT }, /* AuzenTech X-Meridian */ { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, + /* AuzenTech X-Meridian 2G */ + { OXYGEN_PCI_SUBID(0x5431, 0x017a), .driver_data = MODEL_MERIDIAN }, /* HT-Omega Claro */ { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO }, /* HT-Omega Claro halo */ -- cgit v0.10.2 From 66410bfdf14f7c2ad3b2d4a8adeab41d368b6f05 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:20:29 +0100 Subject: ALSA: oxygen: add Xonar DG support Add experimental support for the Asus Xonar DG sound card. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 5cd5c96..805ce91 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1524,8 +1524,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-oxygen ----------------- - Module for sound cards based on the C-Media CMI8787/8788 chip: + Module for sound cards based on the C-Media CMI8786/8787/8788 chip: * Asound A-8788 + * Asus Xonar DG * AuzenTech X-Meridian * AuzenTech X-Meridian 2G * Bgears b-Enspirer diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index ddb5e69..f0bcf68 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -209,7 +209,7 @@ config SND_OXYGEN_LIB tristate config SND_OXYGEN - tristate "C-Media 8787, 8788 (Oxygen)" + tristate "C-Media 8786, 8787, 8788 (Oxygen)" select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -217,6 +217,7 @@ config SND_OXYGEN Say Y here to include support for sound cards based on the C-Media CMI8788 (Oxygen HD Audio) chip: * Asound A-8788 + * Asus Xonar DG * AuzenTech X-Meridian * AuzenTech X-Meridian 2G * Bgears b-Enspirer diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index bd67c0d..0f87265 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -1,5 +1,5 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o -snd-oxygen-objs := oxygen.o +snd-oxygen-objs := oxygen.o xonar_dg.o snd-virtuoso-objs := virtuoso.o xonar_lib.o \ xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o diff --git a/sound/pci/oxygen/cs4245.h b/sound/pci/oxygen/cs4245.h new file mode 100644 index 0000000..5e0197e --- /dev/null +++ b/sound/pci/oxygen/cs4245.h @@ -0,0 +1,107 @@ +#define CS4245_CHIP_ID 0x01 +#define CS4245_POWER_CTRL 0x02 +#define CS4245_DAC_CTRL_1 0x03 +#define CS4245_ADC_CTRL 0x04 +#define CS4245_MCLK_FREQ 0x05 +#define CS4245_SIGNAL_SEL 0x06 +#define CS4245_PGA_B_CTRL 0x07 +#define CS4245_PGA_A_CTRL 0x08 +#define CS4245_ANALOG_IN 0x09 +#define CS4245_DAC_A_CTRL 0x0a +#define CS4245_DAC_B_CTRL 0x0b +#define CS4245_DAC_CTRL_2 0x0c +#define CS4245_INT_STATUS 0x0d +#define CS4245_INT_MASK 0x0e +#define CS4245_INT_MODE_MSB 0x0f +#define CS4245_INT_MODE_LSB 0x10 + +/* Chip ID */ +#define CS4245_CHIP_PART_MASK 0xf0 +#define CS4245_CHIP_REV_MASK 0x0f + +/* Power Control */ +#define CS4245_FREEZE 0x80 +#define CS4245_PDN_MIC 0x08 +#define CS4245_PDN_ADC 0x04 +#define CS4245_PDN_DAC 0x02 +#define CS4245_PDN 0x01 + +/* DAC Control */ +#define CS4245_DAC_FM_MASK 0xc0 +#define CS4245_DAC_FM_SINGLE 0x00 +#define CS4245_DAC_FM_DOUBLE 0x40 +#define CS4245_DAC_FM_QUAD 0x80 +#define CS4245_DAC_DIF_MASK 0x30 +#define CS4245_DAC_DIF_LJUST 0x00 +#define CS4245_DAC_DIF_I2S 0x10 +#define CS4245_DAC_DIF_RJUST_16 0x20 +#define CS4245_DAC_DIF_RJUST_24 0x30 +#define CS4245_RESERVED_1 0x08 +#define CS4245_MUTE_DAC 0x04 +#define CS4245_DEEMPH 0x02 +#define CS4245_DAC_MASTER 0x01 + +/* ADC Control */ +#define CS4245_ADC_FM_MASK 0xc0 +#define CS4245_ADC_FM_SINGLE 0x00 +#define CS4245_ADC_FM_DOUBLE 0x40 +#define CS4245_ADC_FM_QUAD 0x80 +#define CS4245_ADC_DIF_MASK 0x10 +#define CS4245_ADC_DIF_LJUST 0x00 +#define CS4245_ADC_DIF_I2S 0x10 +#define CS4245_MUTE_ADC 0x04 +#define CS4245_HPF_FREEZE 0x02 +#define CS4245_ADC_MASTER 0x01 + +/* MCLK Frequency */ +#define CS4245_MCLK1_MASK 0x70 +#define CS4245_MCLK1_SHIFT 4 +#define CS4245_MCLK2_MASK 0x07 +#define CS4245_MCLK2_SHIFT 0 +#define CS4245_MCLK_1 0 +#define CS4245_MCLK_1_5 1 +#define CS4245_MCLK_2 2 +#define CS4245_MCLK_3 3 +#define CS4245_MCLK_4 4 + +/* Signal Selection */ +#define CS4245_A_OUT_SEL_MASK 0x60 +#define CS4245_A_OUT_SEL_HIZ 0x00 +#define CS4245_A_OUT_SEL_DAC 0x20 +#define CS4245_A_OUT_SEL_PGA 0x40 +#define CS4245_LOOP 0x02 +#define CS4245_ASYNCH 0x01 + +/* Channel B/A PGA Control */ +#define CS4245_PGA_GAIN_MASK 0x3f + +/* ADC Input Control */ +#define CS4245_PGA_SOFT 0x10 +#define CS4245_PGA_ZERO 0x08 +#define CS4245_SEL_MASK 0x07 +#define CS4245_SEL_MIC 0x00 +#define CS4245_SEL_INPUT_1 0x01 +#define CS4245_SEL_INPUT_2 0x02 +#define CS4245_SEL_INPUT_3 0x03 +#define CS4245_SEL_INPUT_4 0x04 +#define CS4245_SEL_INPUT_5 0x05 +#define CS4245_SEL_INPUT_6 0x06 + +/* DAC Channel A/B Volume Control */ +#define CS4245_VOL_MASK 0xff + +/* DAC Control 2 */ +#define CS4245_DAC_SOFT 0x80 +#define CS4245_DAC_ZERO 0x40 +#define CS4245_INVERT_DAC 0x20 +#define CS4245_INT_ACTIVE_HIGH 0x01 + +/* Interrupt Status/Mask/Mode */ +#define CS4245_ADC_CLK_ERR 0x08 +#define CS4245_DAC_CLK_ERR 0x04 +#define CS4245_ADC_OVFL 0x02 +#define CS4245_ADC_UNDRFL 0x01 + + +#define CS4245_SPI_ADDRESS (0x9e << 16) +#define CS4245_SPI_WRITE (0 << 16) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 304f1a5..b59aeef 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -53,13 +53,16 @@ #include #include #include "oxygen.h" +#include "xonar_dg.h" #include "ak4396.h" #include "wm8785.h" MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 driver"); MODULE_LICENSE("GPL v2"); -MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}"); +MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8786}" + ",{C-Media,CMI8787}" + ",{C-Media,CMI8788}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; @@ -79,6 +82,7 @@ enum { MODEL_CLARO_HALO, MODEL_FANTASIA, MODEL_2CH_OUTPUT, + MODEL_XONAR_DG, }; static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { @@ -92,6 +96,8 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, + /* Asus Xonar DG */ + { OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG }, /* PCI 2.0 HD Audio */ { OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT }, /* Kuroutoshikou CMI8787-HG2PCI */ @@ -655,6 +661,9 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.dac_channels_pcm = 2; chip->model.dac_channels_mixer = 2; break; + case MODEL_XONAR_DG: + chip->model = model_xonar_dg; + break; } if (id->driver_data == MODEL_MERIDIAN || id->driver_data == MODEL_CLARO_HALO) { diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 242c1ca..d327c36 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -97,6 +97,16 @@ static int dac_mute_put(struct snd_kcontrol *ctl, return changed; } +static unsigned int upmix_item_count(struct oxygen *chip) +{ + if (chip->model.dac_channels_pcm < 8) + return 2; + else if (chip->model.update_center_lfe_mix) + return 5; + else + return 3; +} + static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { static const char *const names[5] = { @@ -107,7 +117,7 @@ static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) "Front+Surround+Center/LFE+Back", }; struct oxygen *chip = ctl->private_data; - unsigned int count = chip->model.update_center_lfe_mix ? 5 : 3; + unsigned int count = upmix_item_count(chip); info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; info->count = 1; @@ -188,7 +198,7 @@ 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 = chip->model.update_center_lfe_mix ? 5 : 3; + unsigned int count = upmix_item_count(chip); int changed; if (value->value.enumerated.item[0] >= count) diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c new file mode 100644 index 0000000..7ed3284 --- /dev/null +++ b/sound/pci/oxygen/xonar_dg.c @@ -0,0 +1,593 @@ +/* + * card driver for the Xonar DG + * + * 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, see . + */ + +/* + * Xonar DG + * -------- + * + * CMI8788: + * + * SPI 0 -> CS4245 + * + * GPIO 3 <- ? + * GPIO 4 <- headphone detect + * GPIO 5 -> route input jack to line-in (0) or mic-in (1) + * GPIO 6 -> route input jack to line-in (0) or mic-in (1) + * GPIO 7 -> enable rear headphone amp + * GPIO 8 -> enable output to speakers + * + * CS4245: + * + * input 1 <- aux + * input 2 <- front mic + * input 4 <- line/mic + * aux out -> front panel headphones + */ + +#include +#include +#include +#include +#include +#include +#include "oxygen.h" +#include "xonar_dg.h" +#include "cs4245.h" + +#define GPIO_MAGIC 0x0008 +#define GPIO_HP_DETECT 0x0010 +#define GPIO_INPUT_ROUTE 0x0060 +#define GPIO_HP_REAR 0x0080 +#define GPIO_OUTPUT_ENABLE 0x0100 + +struct dg { + unsigned int output_sel; + s8 input_vol[4][2]; + unsigned int input_sel; + u8 hp_vol_att; + u8 cs4245_regs[0x11]; +}; + +static void cs4245_write(struct oxygen *chip, unsigned int reg, u8 value) +{ + struct dg *data = chip->model_data; + + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_3 | + OXYGEN_SPI_CLOCK_1280 | + (0 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI, + CS4245_SPI_ADDRESS | + CS4245_SPI_WRITE | + (value << 8) | reg); + data->cs4245_regs[reg] = value; +} + +static void cs4245_write_cached(struct oxygen *chip, unsigned int reg, u8 value) +{ + struct dg *data = chip->model_data; + + if (value != data->cs4245_regs[reg]) + cs4245_write(chip, reg, value); +} + +static void cs4245_registers_init(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + cs4245_write(chip, CS4245_POWER_CTRL, CS4245_PDN); + cs4245_write(chip, CS4245_DAC_CTRL_1, + data->cs4245_regs[CS4245_DAC_CTRL_1]); + cs4245_write(chip, CS4245_ADC_CTRL, + data->cs4245_regs[CS4245_ADC_CTRL]); + cs4245_write(chip, CS4245_SIGNAL_SEL, + data->cs4245_regs[CS4245_SIGNAL_SEL]); + cs4245_write(chip, CS4245_PGA_B_CTRL, + data->cs4245_regs[CS4245_PGA_B_CTRL]); + cs4245_write(chip, CS4245_PGA_A_CTRL, + data->cs4245_regs[CS4245_PGA_A_CTRL]); + cs4245_write(chip, CS4245_ANALOG_IN, + data->cs4245_regs[CS4245_ANALOG_IN]); + cs4245_write(chip, CS4245_DAC_A_CTRL, + data->cs4245_regs[CS4245_DAC_A_CTRL]); + cs4245_write(chip, CS4245_DAC_B_CTRL, + data->cs4245_regs[CS4245_DAC_B_CTRL]); + cs4245_write(chip, CS4245_DAC_CTRL_2, + CS4245_DAC_SOFT | CS4245_DAC_ZERO | CS4245_INVERT_DAC); + cs4245_write(chip, CS4245_INT_MASK, 0); + cs4245_write(chip, CS4245_POWER_CTRL, 0); +} + +static void cs4245_init(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + data->cs4245_regs[CS4245_DAC_CTRL_1] = + CS4245_DAC_FM_SINGLE | CS4245_DAC_DIF_LJUST; + data->cs4245_regs[CS4245_ADC_CTRL] = + CS4245_ADC_FM_SINGLE | CS4245_ADC_DIF_LJUST; + data->cs4245_regs[CS4245_SIGNAL_SEL] = + CS4245_A_OUT_SEL_HIZ | CS4245_ASYNCH; + data->cs4245_regs[CS4245_PGA_B_CTRL] = 0; + data->cs4245_regs[CS4245_PGA_A_CTRL] = 0; + data->cs4245_regs[CS4245_ANALOG_IN] = + CS4245_PGA_SOFT | CS4245_PGA_ZERO | CS4245_SEL_INPUT_4; + data->cs4245_regs[CS4245_DAC_A_CTRL] = 0; + data->cs4245_regs[CS4245_DAC_B_CTRL] = 0; + cs4245_registers_init(chip); + snd_component_add(chip->card, "CS4245"); +} + +static void dg_output_enable(struct oxygen *chip) +{ + msleep(2500); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); +} + +static void dg_init(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + data->output_sel = 0; + data->input_sel = 3; + data->hp_vol_att = 2 * 16; + + cs4245_init(chip); + + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_MAGIC | GPIO_HP_DETECT); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_INPUT_ROUTE | GPIO_HP_REAR | GPIO_OUTPUT_ENABLE); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_INPUT_ROUTE | GPIO_HP_REAR); + dg_output_enable(chip); +} + +static void dg_cleanup(struct oxygen *chip) +{ + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); +} + +static void dg_suspend(struct oxygen *chip) +{ + dg_cleanup(chip); +} + +static void dg_resume(struct oxygen *chip) +{ + cs4245_registers_init(chip); + dg_output_enable(chip); +} + +static void set_cs4245_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct dg *data = chip->model_data; + u8 value; + + value = data->cs4245_regs[CS4245_DAC_CTRL_1] & ~CS4245_DAC_FM_MASK; + if (params_rate(params) <= 50000) + value |= CS4245_DAC_FM_SINGLE; + else if (params_rate(params) <= 100000) + value |= CS4245_DAC_FM_DOUBLE; + else + value |= CS4245_DAC_FM_QUAD; + cs4245_write_cached(chip, CS4245_DAC_CTRL_1, value); +} + +static void set_cs4245_adc_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct dg *data = chip->model_data; + u8 value; + + value = data->cs4245_regs[CS4245_ADC_CTRL] & ~CS4245_ADC_FM_MASK; + if (params_rate(params) <= 50000) + value |= CS4245_ADC_FM_SINGLE; + else if (params_rate(params) <= 100000) + value |= CS4245_ADC_FM_DOUBLE; + else + value |= CS4245_ADC_FM_QUAD; + cs4245_write_cached(chip, CS4245_ADC_CTRL, value); +} + +static int output_switch_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "Speakers", "Headphones", "FP Headphones" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int output_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.enumerated.item[0] = data->output_sel; + mutex_unlock(&chip->mutex); + return 0; +} + +static int output_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + u8 reg; + int changed; + + if (value->value.enumerated.item[0] > 2) + return -EINVAL; + + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != data->output_sel; + if (changed) { + data->output_sel = value->value.enumerated.item[0]; + + reg = data->cs4245_regs[CS4245_SIGNAL_SEL] & + ~CS4245_A_OUT_SEL_MASK; + reg |= data->output_sel == 2 ? + CS4245_A_OUT_SEL_DAC : CS4245_A_OUT_SEL_HIZ; + cs4245_write_cached(chip, CS4245_SIGNAL_SEL, reg); + + cs4245_write_cached(chip, CS4245_DAC_A_CTRL, + data->output_sel ? data->hp_vol_att : 0); + cs4245_write_cached(chip, CS4245_DAC_B_CTRL, + data->output_sel ? data->hp_vol_att : 0); + + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + data->output_sel == 1 ? GPIO_HP_REAR : 0, + GPIO_HP_REAR); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int hp_volume_offset_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "< 64 ohms", "64-150 ohms", "150-300 ohms" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int hp_volume_offset_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + if (data->hp_vol_att > 2 * 7) + value->value.enumerated.item[0] = 0; + else if (data->hp_vol_att > 0) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + mutex_unlock(&chip->mutex); + return 0; +} + +static int hp_volume_offset_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + static const s8 atts[3] = { 2 * 16, 2 * 7, 0 }; + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + s8 att; + int changed; + + if (value->value.enumerated.item[0] > 2) + return -EINVAL; + att = atts[value->value.enumerated.item[0]]; + mutex_lock(&chip->mutex); + changed = att != data->hp_vol_att; + if (changed) { + data->hp_vol_att = att; + if (data->output_sel) { + cs4245_write_cached(chip, CS4245_DAC_A_CTRL, att); + cs4245_write_cached(chip, CS4245_DAC_B_CTRL, att); + } + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int input_vol_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 = 2 * -12; + info->value.integer.max = 2 * 12; + return 0; +} + +static int input_vol_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + unsigned int idx = ctl->private_value; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = data->input_vol[idx][0]; + value->value.integer.value[1] = data->input_vol[idx][1]; + mutex_unlock(&chip->mutex); + return 0; +} + +static int input_vol_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + unsigned int idx = ctl->private_value; + int changed = 0; + + if (value->value.integer.value[0] < 2 * -12 || + value->value.integer.value[0] > 2 * 12 || + value->value.integer.value[1] < 2 * -12 || + value->value.integer.value[1] > 2 * 12) + return -EINVAL; + mutex_lock(&chip->mutex); + changed = data->input_vol[idx][0] != value->value.integer.value[0] || + data->input_vol[idx][1] != value->value.integer.value[1]; + if (changed) { + data->input_vol[idx][0] = value->value.integer.value[0]; + data->input_vol[idx][1] = value->value.integer.value[1]; + if (idx == data->input_sel) { + cs4245_write_cached(chip, CS4245_PGA_A_CTRL, + data->input_vol[idx][0]); + cs4245_write_cached(chip, CS4245_PGA_B_CTRL, + data->input_vol[idx][1]); + } + } + mutex_unlock(&chip->mutex); + return changed; +} + +static DECLARE_TLV_DB_SCALE(cs4245_pga_db_scale, -1200, 50, 0); + +static int input_sel_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[4] = { + "Mic", "Aux", "Front Mic", "Line" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 4; + info->value.enumerated.item &= 3; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int input_sel_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.enumerated.item[0] = data->input_sel; + mutex_unlock(&chip->mutex); + return 0; +} + +static int input_sel_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + static const u8 sel_values[4] = { + CS4245_SEL_MIC, + CS4245_SEL_INPUT_1, + CS4245_SEL_INPUT_2, + CS4245_SEL_INPUT_4 + }; + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + int changed; + + if (value->value.enumerated.item[0] > 3) + return -EINVAL; + + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != data->input_sel; + if (changed) { + data->input_sel = value->value.enumerated.item[0]; + + cs4245_write(chip, CS4245_ANALOG_IN, + (data->cs4245_regs[CS4245_ANALOG_IN] & + ~CS4245_SEL_MASK) | + sel_values[data->input_sel]); + + cs4245_write_cached(chip, CS4245_PGA_A_CTRL, + data->input_vol[data->input_sel][0]); + cs4245_write_cached(chip, CS4245_PGA_B_CTRL, + data->input_vol[data->input_sel][1]); + + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + data->input_sel ? 0 : GPIO_INPUT_ROUTE, + GPIO_INPUT_ROUTE); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { "Active", "Frozen" }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + info->value.enumerated.item &= 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + value->value.enumerated.item[0] = + !!(data->cs4245_regs[CS4245_ADC_CTRL] & CS4245_HPF_FREEZE); + return 0; +} + +static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + u8 reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->cs4245_regs[CS4245_ADC_CTRL] & ~CS4245_HPF_FREEZE; + if (value->value.enumerated.item[0]) + reg |= CS4245_HPF_FREEZE; + changed = reg != data->cs4245_regs[CS4245_ADC_CTRL]; + if (changed) + cs4245_write(chip, CS4245_ADC_CTRL, reg); + mutex_unlock(&chip->mutex); + return changed; +} + +#define INPUT_VOLUME(xname, index) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = input_vol_info, \ + .get = input_vol_get, \ + .put = input_vol_put, \ + .tlv = { .p = cs4245_pga_db_scale }, \ + .private_value = index, \ +} +static const struct snd_kcontrol_new dg_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Output Playback Enum", + .info = output_switch_info, + .get = output_switch_get, + .put = output_switch_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphones Impedance Playback Enum", + .info = hp_volume_offset_info, + .get = hp_volume_offset_get, + .put = hp_volume_offset_put, + }, + INPUT_VOLUME("Mic Capture Volume", 0), + INPUT_VOLUME("Aux Capture Volume", 1), + INPUT_VOLUME("Front Mic Capture Volume", 2), + INPUT_VOLUME("Line Capture Volume", 3), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = input_sel_info, + .get = input_sel_get, + .put = input_sel_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC High-pass Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, + }, +}; + +static int dg_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "Master Playback ", 16)) + return 1; + return 0; +} + +static int dg_mixer_init(struct oxygen *chip) +{ + unsigned int i; + int err; + + for (i = 0; i < ARRAY_SIZE(dg_controls); ++i) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&dg_controls[i], chip)); + if (err < 0) + return err; + } + return 0; +} + +static void dump_cs4245_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct dg *data = chip->model_data; + unsigned int i; + + snd_iprintf(buffer, "\nCS4245:"); + for (i = 1; i <= 0x10; ++i) + snd_iprintf(buffer, " %02x", data->cs4245_regs[i]); + snd_iprintf(buffer, "\n"); +} + +struct oxygen_model model_xonar_dg = { + .shortname = "Xonar DG", + .longname = "C-Media Oxygen HD Audio", + .chip = "CMI8786", + .init = dg_init, + .control_filter = dg_control_filter, + .mixer_init = dg_mixer_init, + .cleanup = dg_cleanup, + .suspend = dg_suspend, + .resume = dg_resume, + .set_dac_params = set_cs4245_dac_params, + .set_adc_params = set_cs4245_adc_params, + .dump_registers = dump_cs4245_registers, + .model_data_size = sizeof(struct dg), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1, + .dac_channels_pcm = 6, + .dac_channels_mixer = 0, + .function_flags = OXYGEN_FUNCTION_SPI, + .dac_mclks = OXYGEN_MCLKS(256, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; diff --git a/sound/pci/oxygen/xonar_dg.h b/sound/pci/oxygen/xonar_dg.h new file mode 100644 index 0000000..5688d78 --- /dev/null +++ b/sound/pci/oxygen/xonar_dg.h @@ -0,0 +1,8 @@ +#ifndef XONAR_DG_H_INCLUDED +#define XONAR_DG_H_INCLUDED + +#include "oxygen.h" + +extern struct oxygen_model model_xonar_dg; + +#endif -- cgit v0.10.2 From b532d6b8d3aa163e1dc143bc729e9ee92f75baf5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:23:57 +0100 Subject: ALSA: virtuoso: add Xonar HDAV1.3 Slim support Add experimental support for the Asus Xonar HDAV1.3 Slim sound card. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 805ce91..3c1eddd 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -2004,9 +2004,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-virtuoso ------------------- - Module for sound cards based on the Asus AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, DS, HDAV1.3 (Deluxe), Essence ST - (Deluxe) and Essence STX. + Module for sound cards based on the Asus AV66/AV100/AV200 chips, + i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), Essence STX, + HDAV1.3 (Deluxe), and HDAV1.3 Slim. This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index f0bcf68..9823d59 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -819,8 +819,8 @@ config SND_VIRTUOSO Say Y here to include support for sound cards based on the Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), and Essence STX. - Support for the HDAV1.3 (Deluxe) is experimental; for the - HDAV1.3 Slim and Xense, missing. + Support for the HDAV1.3 (Deluxe) and HDAV1.3 Slim is experimental; + for the Xense, missing. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/oxygen/xonar_hdmi.c b/sound/pci/oxygen/xonar_hdmi.c index b12db1f..136dac6 100644 --- a/sound/pci/oxygen/xonar_hdmi.c +++ b/sound/pci/oxygen/xonar_hdmi.c @@ -1,5 +1,5 @@ /* - * helper functions for HDMI models (Xonar HDAV1.3) + * helper functions for HDMI models (Xonar HDAV1.3/HDAV1.3 Slim) * * Copyright (c) Clemens Ladisch * diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 003c480..bf4fd4b 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -1152,9 +1152,6 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; - case 0x835e: - snd_printk(KERN_ERR "the HDAV1.3 Slim is not supported\n"); - return -ENODEV; default: return -EINVAL; } diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 4f96570..ad48356 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -1,5 +1,5 @@ /* - * card driver for models with WM8776/WM8766 DACs (Xonar DS) + * card driver for models with WM8776/WM8766 DACs (Xonar DS/HDAV1.3 Slim) * * Copyright (c) Clemens Ladisch * @@ -77,6 +77,13 @@ #define GPIO_DS_OUTPUT_FRONTLR 0x0080 #define GPIO_DS_OUTPUT_ENABLE 0x0100 +#define GPIO_SLIM_HDMI_DISABLE 0x0001 +#define GPIO_SLIM_OUTPUT_ENABLE 0x0002 +#define GPIO_SLIM_FIRMWARE_CLK 0x0040 +#define GPIO_SLIM_FIRMWARE_DATA 0x0080 + +#define I2C_DEVICE_WM8776 0x34 /* 001101, 0, /W=0 */ + #define LC_CONTROL_LIMITER 0x40000000 #define LC_CONTROL_ALC 0x20000000 @@ -88,19 +95,37 @@ struct xonar_wm87x6 { struct snd_kcontrol *mic_adcmux_control; struct snd_kcontrol *lc_controls[13]; struct snd_jack *hp_jack; + struct xonar_hdmi hdmi; }; -static void wm8776_write(struct oxygen *chip, - unsigned int reg, unsigned int value) +static void wm8776_write_spi(struct oxygen *chip, + unsigned int reg, unsigned int value) { - struct xonar_wm87x6 *data = chip->model_data; - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (1 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); +} + +static void wm8776_write_i2c(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_WM8776, + (reg << 1) | (value >> 8), value); +} + +static void wm8776_write(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + if ((chip->model.function_flags & OXYGEN_FUNCTION_2WIRE_SPI_MASK) == + OXYGEN_FUNCTION_SPI) + wm8776_write_spi(chip, reg, value); + else + wm8776_write_i2c(chip, reg, value); if (reg < ARRAY_SIZE(data->wm8776_regs)) { if (reg >= WM8776_HPLVOL && reg <= WM8776_DACMASTER) value &= ~WM8776_UPDATE; @@ -267,17 +292,50 @@ static void xonar_ds_init(struct oxygen *chip) snd_component_add(chip->card, "WM8766"); } +static void xonar_hdav_slim_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->generic.anti_pop_delay = 300; + data->generic.output_enable_bit = GPIO_SLIM_OUTPUT_ENABLE; + + wm8776_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_SLIM_HDMI_DISABLE | + GPIO_SLIM_FIRMWARE_CLK | + GPIO_SLIM_FIRMWARE_DATA); + + xonar_hdmi_init(chip, &data->hdmi); + xonar_enable_output(chip); + + snd_component_add(chip->card, "WM8776"); +} + static void xonar_ds_cleanup(struct oxygen *chip) { xonar_disable_output(chip); wm8776_write(chip, WM8776_RESET, 0); } +static void xonar_hdav_slim_cleanup(struct oxygen *chip) +{ + xonar_hdmi_cleanup(chip); + xonar_disable_output(chip); + wm8776_write(chip, WM8776_RESET, 0); + msleep(2); +} + static void xonar_ds_suspend(struct oxygen *chip) { xonar_ds_cleanup(chip); } +static void xonar_hdav_slim_suspend(struct oxygen *chip) +{ + xonar_hdav_slim_cleanup(chip); +} + static void xonar_ds_resume(struct oxygen *chip) { wm8776_registers_init(chip); @@ -286,6 +344,15 @@ static void xonar_ds_resume(struct oxygen *chip) xonar_ds_handle_hp_jack(chip); } +static void xonar_hdav_slim_resume(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + wm8776_registers_init(chip); + xonar_hdmi_resume(chip, &data->hdmi); + xonar_enable_output(chip); +} + static void wm8776_adc_hardware_filter(unsigned int channel, struct snd_pcm_hardware *hardware) { @@ -300,6 +367,13 @@ static void wm8776_adc_hardware_filter(unsigned int channel, } } +static void xonar_hdav_slim_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware) +{ + wm8776_adc_hardware_filter(channel, hardware); + xonar_hdmi_pcm_hardware_filter(channel, hardware); +} + static void set_wm87x6_dac_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -316,6 +390,14 @@ static void set_wm8776_adc_params(struct oxygen *chip, wm8776_write_cached(chip, WM8776_MSTRCTRL, reg); } +static void set_hdav_slim_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_wm87x6 *data = chip->model_data; + + xonar_set_hdmi_params(chip, &data->hdmi, params); +} + static void update_wm8776_volume(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; @@ -1007,6 +1089,53 @@ static const struct snd_kcontrol_new ds_controls[] = { .private_value = 0, }, }; +static const struct snd_kcontrol_new hdav_slim_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "HDMI Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = xonar_gpio_bit_switch_get, + .put = xonar_gpio_bit_switch_put, + .private_value = GPIO_SLIM_HDMI_DISABLE | XONAR_GPIO_BIT_INVERT, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .info = wm8776_hp_vol_info, + .get = wm8776_hp_vol_get, + .put = wm8776_hp_vol_put, + .tlv = { .p = wm8776_hp_db_scale }, + }, + WM8776_BIT_SWITCH("Headphone Playback Switch", + WM8776_PWRDOWN, WM8776_HPPD, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Capture Volume", + .info = wm8776_input_vol_info, + .get = wm8776_input_vol_get, + .put = wm8776_input_vol_put, + .tlv = { .p = wm8776_adc_db_scale }, + }, + WM8776_BIT_SWITCH("Mic Capture Switch", + WM8776_ADCMUX, 1 << 0, 0, 0), + WM8776_BIT_SWITCH("Aux Capture Switch", + WM8776_ADCMUX, 1 << 1, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Level Control Capture Enum", + .info = wm8776_level_control_info, + .get = wm8776_level_control_get, + .put = wm8776_level_control_put, + .private_value = 0, + }, +}; static const struct snd_kcontrol_new lc_controls[] = { WM8776_FIELD_CTL_VOLUME("Limiter Threshold", WM8776_ALCCTRL1, 0, 11, 0, 15, 0xf, @@ -1050,6 +1179,26 @@ static const struct snd_kcontrol_new lc_controls[] = { LC_CONTROL_ALC, wm8776_ngth_db_scale), }; +static int add_lc_controls(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + struct snd_kcontrol *ctl; + int err; + + BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); + for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { + ctl = snd_ctl_new1(&lc_controls[i], chip); + if (!ctl) + return -ENOMEM; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + data->lc_controls[i] = ctl; + } + return 0; +} + static int xonar_ds_mixer_init(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; @@ -1071,17 +1220,26 @@ static int xonar_ds_mixer_init(struct oxygen *chip) } if (!data->line_adcmux_control || !data->mic_adcmux_control) return -ENXIO; - BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); - for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { - ctl = snd_ctl_new1(&lc_controls[i], chip); + + return add_lc_controls(chip); +} + +static int xonar_hdav_slim_mixer_init(struct oxygen *chip) +{ + unsigned int i; + struct snd_kcontrol *ctl; + int err; + + for (i = 0; i < ARRAY_SIZE(hdav_slim_controls); ++i) { + ctl = snd_ctl_new1(&hdav_slim_controls[i], chip); if (!ctl) return -ENOMEM; err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; - data->lc_controls[i] = ctl; } - return 0; + + return add_lc_controls(chip); } static void dump_wm8776_registers(struct oxygen *chip, @@ -1145,6 +1303,38 @@ static const struct oxygen_model model_xonar_ds = { .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; +static const struct oxygen_model model_xonar_hdav_slim = { + .shortname = "Xonar HDAV1.3 Slim", + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_hdav_slim_init, + .mixer_init = xonar_hdav_slim_mixer_init, + .cleanup = xonar_hdav_slim_cleanup, + .suspend = xonar_hdav_slim_suspend, + .resume = xonar_hdav_slim_resume, + .pcm_hardware_filter = xonar_hdav_slim_hardware_filter, + .set_dac_params = set_hdav_slim_dac_params, + .set_adc_params = set_wm8776_adc_params, + .update_dac_volume = update_wm8776_volume, + .update_dac_mute = update_wm8776_mute, + .uart_input = xonar_hdmi_uart_input, + .dump_registers = dump_wm8776_registers, + .dac_tlv = wm87x6_dac_db_scale, + .model_data_size = sizeof(struct xonar_wm87x6), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1, + .dac_channels_pcm = 8, + .dac_channels_mixer = 2, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + int __devinit get_xonar_wm87x6_model(struct oxygen *chip, const struct pci_device_id *id) { @@ -1152,6 +1342,9 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, case 0x838e: chip->model = model_xonar_ds; break; + case 0x835e: + chip->model = model_xonar_hdav_slim; + break; default: return -EINVAL; } -- cgit v0.10.2 From 9600732b6caba595f34acf2abd930098ec9a0b2b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:25:44 +0100 Subject: ALSA: core, oxygen, virtuoso: add an enum control info helper Introduce the helper function snd_ctl_enum_info() to fill out the elem_info fields for an enumerated control. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/include/sound/control.h b/include/sound/control.h index 112374d..7715e6f 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -160,12 +160,14 @@ static inline struct snd_ctl_elem_id *snd_ctl_build_ioff(struct snd_ctl_elem_id } /* - * Frequently used control callbacks + * Frequently used control callbacks/helpers */ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, + unsigned int items, const char *const names[]); /* * virtual master control diff --git a/sound/core/control.c b/sound/core/control.c index 45a8180..9ce00ed 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1488,7 +1488,7 @@ int snd_ctl_create(struct snd_card *card) } /* - * Frequently used control callbacks + * Frequently used control callbacks/helpers */ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1513,3 +1513,29 @@ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_boolean_stereo_info); + +/** + * snd_ctl_enum_info - fills the info structure for an enumerated control + * @info: the structure to be filled + * @channels: the number of the control's channels; often one + * @items: the number of control values; also the size of @names + * @names: an array containing the names of all control values + * + * Sets all required fields in @info to their appropriate values. + * If the control's accessibility is not the default (readable and writable), + * the caller has to fill @info->access. + */ +int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, + unsigned int items, const char *const names[]) +{ + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = channels; + info->value.enumerated.items = items; + if (info->value.enumerated.item >= items) + info->value.enumerated.item = items - 1; + strlcpy(info->value.enumerated.name, + names[info->value.enumerated.item], + sizeof(info->value.enumerated.name)); + return 0; +} +EXPORT_SYMBOL(snd_ctl_enum_info); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index b59aeef..45427d8 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -415,13 +415,7 @@ static int rolloff_info(struct snd_kcontrol *ctl, "Sharp Roll-off", "Slow Roll-off" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int rolloff_get(struct snd_kcontrol *ctl, @@ -473,13 +467,7 @@ static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) "None", "High-pass Filter" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index d327c36..821df1c 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -119,13 +119,7 @@ static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) struct oxygen *chip = ctl->private_data; unsigned int count = upmix_item_count(chip); - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - 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; + return snd_ctl_enum_info(info, 1, count, names); } static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) @@ -659,12 +653,7 @@ static int mic_fmic_source_info(struct snd_kcontrol *ctl, { static const char *const names[] = { "Mic Jack", "Front Panel" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - info->value.enumerated.item &= 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int mic_fmic_source_get(struct snd_kcontrol *ctl, diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 55c52c7..9f72d42 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -298,13 +298,7 @@ static int rolloff_info(struct snd_kcontrol *ctl, "Fast Roll-off", "Slow Roll-off" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int rolloff_get(struct snd_kcontrol *ctl, diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 7ed3284..cc610ac1 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -213,13 +213,7 @@ static int output_switch_info(struct snd_kcontrol *ctl, "Speakers", "Headphones", "FP Headphones" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item >= 3) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 3, names); } static int output_switch_get(struct snd_kcontrol *ctl, @@ -276,13 +270,7 @@ static int hp_volume_offset_info(struct snd_kcontrol *ctl, "< 64 ohms", "64-150 ohms", "150-300 ohms" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item >= 3) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 3, names); } static int hp_volume_offset_get(struct snd_kcontrol *ctl, @@ -390,12 +378,7 @@ static int input_sel_info(struct snd_kcontrol *ctl, "Mic", "Aux", "Front Mic", "Line" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 4; - info->value.enumerated.item &= 3; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 4, names); } static int input_sel_get(struct snd_kcontrol *ctl, @@ -453,12 +436,7 @@ static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { static const char *const names[2] = { "Active", "Frozen" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - info->value.enumerated.item &= 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index bf4fd4b..54cad38 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -678,13 +678,7 @@ static int rolloff_info(struct snd_kcontrol *ctl, "Sharp Roll-off", "Slow Roll-off" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int rolloff_get(struct snd_kcontrol *ctl, @@ -748,13 +742,7 @@ static int st_output_switch_info(struct snd_kcontrol *ctl, "Speakers", "Headphones", "FP Headphones" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item >= 3) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 3, names); } static int st_output_switch_get(struct snd_kcontrol *ctl, @@ -809,13 +797,7 @@ static int st_hp_volume_offset_info(struct snd_kcontrol *ctl, "< 64 ohms", "64-300 ohms", "300-600 ohms" }; - 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; + return snd_ctl_enum_info(info, 1, 3, names); } static int st_hp_volume_offset_get(struct snd_kcontrol *ctl, diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index ad48356..42d1ab1 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -577,11 +577,6 @@ static int wm8776_field_enum_info(struct snd_kcontrol *ctl, const char *const *names; max = (ctl->private_value >> 12) & 0xf; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = max + 1; - if (info->value.enumerated.item > max) - info->value.enumerated.item = max; switch ((ctl->private_value >> 24) & 0x1f) { case WM8776_ALCCTRL2: names = hld; @@ -605,8 +600,7 @@ static int wm8776_field_enum_info(struct snd_kcontrol *ctl, default: return -ENXIO; } - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, max + 1, names); } static int wm8776_field_volume_info(struct snd_kcontrol *ctl, @@ -863,13 +857,8 @@ static int wm8776_level_control_info(struct snd_kcontrol *ctl, static const char *const names[3] = { "None", "Peak Limiter", "Automatic Level Control" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item >= 3) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + + return snd_ctl_enum_info(info, 1, 3, names); } static int wm8776_level_control_get(struct snd_kcontrol *ctl, @@ -955,13 +944,7 @@ static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) "None", "High-pass Filter" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) -- cgit v0.10.2 From dd1224aa3ebfdef6cddcfb68ddcd0e682eaf9f84 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:27:26 +0100 Subject: ALSA: bt87x: use enum control info helper Simplify the info callback by using the snd_ctl_enum_info() helper function. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 37e1b5d..2958a05 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -637,15 +637,9 @@ static struct snd_kcontrol_new snd_bt87x_capture_boost = { static int snd_bt87x_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *info) { - static char *texts[3] = {"TV Tuner", "FM", "Mic/Line"}; + static const char *const texts[3] = {"TV Tuner", "FM", "Mic/Line"}; - 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, texts[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 3, texts); } static int snd_bt87x_capture_source_get(struct snd_kcontrol *kcontrol, -- cgit v0.10.2 From 60c4ce4a0c2ffbd37d6604f68fa4408ca7226157 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:28:19 +0100 Subject: ALSA: cmipci: use enum control info helper Simplify info callbacks by using the snd_ctl_enum_info() helper function. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 329968e..b5bb036 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2507,14 +2507,12 @@ static int snd_cmipci_line_in_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct cmipci *cm = snd_kcontrol_chip(kcontrol); - static char *texts[3] = { "Line-In", "Rear Output", "Bass Output" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = cm->chip_version >= 39 ? 3 : 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 const char *const texts[3] = { + "Line-In", "Rear Output", "Bass Output" + }; + + return snd_ctl_enum_info(uinfo, 1, + cm->chip_version >= 39 ? 3 : 2, texts); } static inline unsigned int get_line_in_mode(struct cmipci *cm) @@ -2564,14 +2562,9 @@ static int snd_cmipci_line_in_mode_put(struct snd_kcontrol *kcontrol, static int snd_cmipci_mic_in_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { "Mic-In", "Center/LFE Output" }; - 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 const char *const texts[2] = { "Mic-In", "Center/LFE Output" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_cmipci_mic_in_mode_get(struct snd_kcontrol *kcontrol, -- cgit v0.10.2 From bed6896d0be1de12eb6237a43a4beaaf7dcfb42c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:29:06 +0100 Subject: ALSA: ymfpci: use enum control info helper Simplify the info callback by using the snd_ctl_enum_info() helper function. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 5518371..c94c051 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1389,15 +1389,9 @@ static struct snd_kcontrol_new snd_ymfpci_spdif_stream __devinitdata = static int snd_ymfpci_drec_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *info) { - static char *texts[3] = {"AC'97", "IEC958", "ZV Port"}; - - 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, texts[info->value.enumerated.item]); - return 0; + static const char *const texts[3] = {"AC'97", "IEC958", "ZV Port"}; + + return snd_ctl_enum_info(info, 1, 3, texts); } static int snd_ymfpci_drec_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value) -- cgit v0.10.2 From 2a1803a7291e82823effef0b48b0039324d224c6 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:30:13 +0100 Subject: ALSA: usb-audio: use enum control info helper Simplify info callbacks by using the snd_ctl_enum_info() helper function. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 25bce7e..d7021cb 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1729,13 +1729,7 @@ static int roland_load_info(struct snd_kcontrol *kcontrol, { static const char *const names[] = { "High Load", "Light Load" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item > 1) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int roland_load_get(struct snd_kcontrol *kcontrol, diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f2d74d6..7df89b3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1633,18 +1633,11 @@ static int parse_audio_extension_unit(struct mixer_build *state, int unitid, voi static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *cval = kcontrol->private_data; - char **itemlist = (char **)kcontrol->private_value; + const char **itemlist = (const char **)kcontrol->private_value; if (snd_BUG_ON(!itemlist)) return -EINVAL; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = cval->max; - if (uinfo->value.enumerated.item >= cval->max) - uinfo->value.enumerated.item = cval->max - 1; - strlcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item], - sizeof(uinfo->value.enumerated.name)); - return 0; + return snd_ctl_enum_info(uinfo, 1, cval->max, itemlist); } /* get callback for selector unit */ -- cgit v0.10.2 From 061b869eca6f725b0119f7dff833288a44bf46c0 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:30:54 +0100 Subject: ALSA: usb-audio: add Edirol SD-90 PCM support Add support for the 24-bit audio I/Os of the Edirol SD-90 interface. Reported-any-tested-by: Jim Grusendorf Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/usb/format.c b/sound/usb/format.c index 6914821..5b792d2 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -76,7 +76,10 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format = 1 << UAC_FORMAT_TYPE_I_PCM; } if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) { - if (sample_width > sample_bytes * 8) { + if (chip->usb_id == USB_ID(0x0582, 0x0016) /* Edirol SD-90 */ && + sample_width == 24 && sample_bytes == 2) + sample_bytes = 3; + else if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, sample_width, sample_bytes); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index ad7079d..3599987 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -705,11 +705,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE + .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE + .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, -- cgit v0.10.2 From af39917d5a59d82f50bf9f5fadaa53944e5abf76 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:32:54 +0100 Subject: MAINTAINERS: add myself Add entries for the remaining drivers that I'm maintaining. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/MAINTAINERS b/MAINTAINERS index 1a1c27b..38c2f92 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1419,6 +1419,14 @@ S: Supported F: block/bsg.c F: include/linux/bsg.h +BT87X AUDIO DRIVER +M: Clemens Ladisch +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +T: git git://git.alsa-project.org/alsa-kernel.git +S: Maintained +F: Documentation/sound/alsa/Bt87x.txt +F: sound/pci/bt87x.c + BT8XXGPIO DRIVER M: Michael Buesch W: http://bu3sch.de/btgpio.php @@ -1444,6 +1452,13 @@ S: Maintained F: Documentation/video4linux/bttv/ F: drivers/media/video/bt8xx/bttv* +C-MEDIA CMI8788 DRIVER +M: Clemens Ladisch +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +T: git git://git.alsa-project.org/alsa-kernel.git +S: Maintained +F: sound/pci/oxygen/ + CACHEFILES: FS-CACHE BACKEND FOR CACHING ON MOUNTED FILESYSTEMS M: David Howells L: linux-cachefs@redhat.com @@ -2234,6 +2249,13 @@ W: bluesmoke.sourceforge.net S: Maintained F: drivers/edac/r82600_edac.c +EDIROL UA-101/UA-1000 DRIVER +M: Clemens Ladisch +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +T: git git://git.alsa-project.org/alsa-kernel.git +S: Maintained +F: sound/usb/misc/ua101.c + EEEPC LAPTOP EXTRAS DRIVER M: Corentin Chary L: acpi4asus-user@lists.sourceforge.net @@ -3378,6 +3400,13 @@ L: linux-serial@vger.kernel.org S: Maintained F: drivers/serial/jsm/ +K10TEMP HARDWARE MONITORING DRIVER +M: Clemens Ladisch +L: lm-sensors@lm-sensors.org +S: Maintained +F: Documentation/hwmon/k10temp +F: drivers/hwmon/k10temp.c + K8TEMP HARDWARE MONITORING DRIVER M: Rudolf Marek L: lm-sensors@lm-sensors.org @@ -4394,6 +4423,13 @@ F: drivers/of F: include/linux/of*.h K: of_get_property +OPL4 DRIVER +M: Clemens Ladisch +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +T: git git://git.alsa-project.org/alsa-kernel.git +S: Maintained +F: sound/drivers/opl4/ + OPROFILE M: Robert Richter L: oprofile-list@lists.sf.net @@ -6127,6 +6163,13 @@ S: Maintained W: http://www.one-eyed-alien.net/~mdharm/linux-usb/ F: drivers/usb/storage/ +USB MIDI DRIVER +M: Clemens Ladisch +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +T: git git://git.alsa-project.org/alsa-kernel.git +S: Maintained +F: sound/usb/midi.* + USB OHCI DRIVER M: David Brownell L: linux-usb@vger.kernel.org -- cgit v0.10.2 From 860cffd57acff68e8bc5465f4dd3b7d338fb8e62 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:34:15 +0100 Subject: ALSA: oxygen: add digital input validity check switch Add a mixer control to prevent capturing S/PDIF samples that are not marked as valid (non-audio or corrupted samples). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 821df1c..9bff14d 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -434,30 +434,31 @@ 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) +static int spdif_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; + u32 bit = ctl->private_value; value->value.integer.value[0] = - !!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) - & OXYGEN_SPDIF_LOOPBACK); + !!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) & bit); return 0; } -static int spdif_loopback_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +static int spdif_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; + u32 bit = ctl->private_value; 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; + newreg = oldreg | bit; else - newreg = oldreg & ~OXYGEN_SPDIF_LOOPBACK; + newreg = oldreg & ~bit; changed = newreg != oldreg; if (changed) oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, newreg); @@ -835,8 +836,17 @@ static const struct snd_kcontrol_new spdif_input_controls[] = { .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, + .get = spdif_bit_switch_get, + .put = spdif_bit_switch_put, + .private_value = OXYGEN_SPDIF_LOOPBACK, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("Validity Check ",CAPTURE,SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = spdif_bit_switch_get, + .put = spdif_bit_switch_put, + .private_value = OXYGEN_SPDIF_SPDVALID, }, }; -- cgit v0.10.2 From 64878dfbf755446f025965b742e56e4739a33b37 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:35:38 +0100 Subject: ALSA: oxygen: X-Meridian: add S/PDIF source selection Add a mixer control to select between the on-board and extension board S/PDIF inputs for the X-Meridian (2G). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 45427d8..22cbf3e 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -28,7 +28,13 @@ * * GPIO 0 -> DFS0 of AK5385 * GPIO 1 -> DFS1 of AK5385 - * GPIO 8 -> enable headphone amplifier on HT-Omega models + * + * X-Meridian models: + * GPIO 4 -> enable extension S/PDIF input + * GPIO 6 -> enable on-board S/PDIF input + * + * Claro models: + * GPIO 8 -> enable headphone amplifier * * CM9780: * @@ -124,6 +130,10 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_AK5385_DFS_DOUBLE 0x0001 #define GPIO_AK5385_DFS_QUAD 0x0002 +#define GPIO_MERIDIAN_DIG_MASK 0x0050 +#define GPIO_MERIDIAN_DIG_EXT 0x0010 +#define GPIO_MERIDIAN_DIG_BOARD 0x0040 + #define GPIO_CLARO_HP 0x0100 struct generic_data { @@ -238,6 +248,10 @@ static void generic_init(struct oxygen *chip) static void meridian_init(struct oxygen *chip) { + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_MERIDIAN_DIG_MASK); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + GPIO_MERIDIAN_DIG_BOARD, GPIO_MERIDIAN_DIG_MASK); ak4396_init(chip); ak5385_init(chip); } @@ -506,6 +520,51 @@ static const struct snd_kcontrol_new hpf_control = { .put = hpf_put, }; +static int meridian_dig_source_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { "On-board", "Extension" }; + + return snd_ctl_enum_info(info, 1, 2, names); +} + +static int meridian_dig_source_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + value->value.enumerated.item[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & + GPIO_MERIDIAN_DIG_EXT); + return 0; +} + +static int meridian_dig_source_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 old_reg, new_reg; + int changed; + + mutex_lock(&chip->mutex); + old_reg = oxygen_read16(chip, OXYGEN_GPIO_DATA); + new_reg = old_reg & ~GPIO_MERIDIAN_DIG_MASK; + if (value->value.enumerated.item[0] == 0) + new_reg |= GPIO_MERIDIAN_DIG_BOARD; + else + new_reg |= GPIO_MERIDIAN_DIG_EXT; + changed = new_reg != old_reg; + if (changed) + oxygen_write16(chip, OXYGEN_GPIO_DATA, new_reg); + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new meridian_dig_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Source Capture Enum", + .info = meridian_dig_source_info, + .get = meridian_dig_source_get, + .put = meridian_dig_source_put, +}; + static int generic_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); @@ -524,6 +583,20 @@ static int generic_wm8785_mixer_init(struct oxygen *chip) return 0; } +static int meridian_mixer_init(struct oxygen *chip) +{ + int err; + + err = generic_mixer_init(chip); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&meridian_dig_source_control, chip)); + if (err < 0) + return err; + return 0; +} + static void dump_ak4396_registers(struct oxygen *chip, struct snd_info_buffer *buffer) { @@ -600,7 +673,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, switch (id->driver_data) { case MODEL_MERIDIAN: chip->model.init = meridian_init; - chip->model.mixer_init = generic_mixer_init; + chip->model.mixer_init = meridian_mixer_init; chip->model.resume = meridian_resume; chip->model.set_adc_params = set_ak5385_params; chip->model.dump_registers = dump_ak4396_registers; -- cgit v0.10.2 From a1f80fcfd51c81960bb32601d9aa0acda9d62504 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:36:23 +0100 Subject: ALSA: oxygen: do not show chip revision in card longname Apparently, the revision is 2 on all sold sound cards, so this information is not actually useful. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 2c5fb9e..c2ae63d 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -129,7 +129,6 @@ struct oxygen { u8 pcm_running; u8 dac_routing; u8 spdif_playback_enable; - u8 revision; u8 has_ac97_0; u8 has_ac97_1; u32 spdif_bits; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 77e1f08..70b7398 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -372,12 +372,7 @@ static void oxygen_init(struct oxygen *chip) (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) + if (!(oxygen_read8(chip, OXYGEN_REVISION) & OXYGEN_REVISION_2)) oxygen_set_bits8(chip, OXYGEN_MISC, OXYGEN_MISC_PCI_MEM_W_1_CLOCK); @@ -669,8 +664,8 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, strcpy(card->driver, chip->model.chip); strcpy(card->shortname, chip->model.shortname); - sprintf(card->longname, "%s (rev %u) at %#lx, irq %i", - chip->model.longname, chip->revision, chip->addr, chip->irq); + sprintf(card->longname, "%s at %#lx, irq %i", + chip->model.longname, chip->addr, chip->irq); strcpy(card->mixername, chip->model.chip); snd_component_add(card, chip->model.chip); -- cgit v0.10.2 From a4b1696916abc4a0b6b11fc4cc2cd52421a0c7ac Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Jan 2011 16:37:19 +0100 Subject: ALSA: oxygen: add some card names Instead of the generic Oxygen, use the actual card name, if known. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 22cbf3e..be264d3 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -84,10 +84,13 @@ MODULE_PARM_DESC(enable, "enable card"); enum { MODEL_CMEDIA_REF, MODEL_MERIDIAN, + MODEL_MERIDIAN_2G, MODEL_CLARO, MODEL_CLARO_HALO, MODEL_FANTASIA, + MODEL_SERENADE, MODEL_2CH_OUTPUT, + MODEL_HG2PCI, MODEL_XONAR_DG, }; @@ -107,15 +110,15 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { /* PCI 2.0 HD Audio */ { OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT }, /* Kuroutoshikou CMI8787-HG2PCI */ - { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_2CH_OUTPUT }, + { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_HG2PCI }, /* TempoTec HiFier Fantasia */ { OXYGEN_PCI_SUBID(0x14c3, 0x1710), .driver_data = MODEL_FANTASIA }, /* TempoTec HiFier Serenade */ - { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_2CH_OUTPUT }, + { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_SERENADE }, /* AuzenTech X-Meridian */ { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, /* AuzenTech X-Meridian 2G */ - { OXYGEN_PCI_SUBID(0x5431, 0x017a), .driver_data = MODEL_MERIDIAN }, + { OXYGEN_PCI_SUBID(0x5431, 0x017a), .driver_data = MODEL_MERIDIAN_2G }, /* HT-Omega Claro */ { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO }, /* HT-Omega Claro halo */ @@ -669,9 +672,20 @@ static const struct oxygen_model model_generic = { static int __devinit get_oxygen_model(struct oxygen *chip, const struct pci_device_id *id) { + static const char *const names[] = { + [MODEL_MERIDIAN] = "AuzenTech X-Meridian", + [MODEL_MERIDIAN_2G] = "AuzenTech X-Meridian 2G", + [MODEL_CLARO] = "HT-Omega Claro", + [MODEL_CLARO_HALO] = "HT-Omega Claro halo", + [MODEL_FANTASIA] = "TempoTec HiFier Fantasia", + [MODEL_SERENADE] = "TempoTec HiFier Serenade", + [MODEL_HG2PCI] = "CMI8787-HG2PCI", + }; + chip->model = model_generic; switch (id->driver_data) { case MODEL_MERIDIAN: + case MODEL_MERIDIAN_2G: chip->model.init = meridian_init; chip->model.mixer_init = meridian_mixer_init; chip->model.resume = meridian_resume; @@ -702,7 +716,9 @@ static int __devinit get_oxygen_model(struct oxygen *chip, CAPTURE_1_FROM_SPDIF; break; case MODEL_FANTASIA: + case MODEL_SERENADE: case MODEL_2CH_OUTPUT: + case MODEL_HG2PCI: chip->model.shortname = "C-Media CMI8787"; chip->model.chip = "CMI8787"; if (id->driver_data == MODEL_FANTASIA) @@ -731,6 +747,8 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.misc_flags = OXYGEN_MISC_MIDI; chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; } + if (id->driver_data < ARRAY_SIZE(names) && names[id->driver_data]) + chip->model.shortname = names[id->driver_data]; return 0; } -- cgit v0.10.2 From e92d457514c957f293c4706a06a25833061d9b88 Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Tue, 11 Jan 2011 13:20:30 +1100 Subject: ALSA: include delay.h for msleep in Xonar DG support Signed-off-by: Stephen Rothwell Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index cc610ac1..dd45cf4 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -40,6 +40,7 @@ */ #include +#include #include #include #include -- cgit v0.10.2 From 5fc51524746430179c676297fd48aaab28e1b85f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 11 Jan 2011 10:33:40 +0100 Subject: ALSA: oxygen: fix CD/MIDI for X-Meridian (2G) Enable the X-Meridian's CD input and the X-Meridian 2G's potential MIDI ports. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index be264d3..db39cfc 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -695,6 +695,8 @@ static int __devinit get_oxygen_model(struct oxygen *chip, PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; + if (id->driver_data == MODEL_MERIDIAN) + chip->model.device_config |= AC97_CD_INPUT; break; case MODEL_CLARO: chip->model.init = claro_init; @@ -743,6 +745,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, break; } if (id->driver_data == MODEL_MERIDIAN || + id->driver_data == MODEL_MERIDIAN_2G || id->driver_data == MODEL_CLARO_HALO) { chip->model.misc_flags = OXYGEN_MISC_MIDI; chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; -- cgit v0.10.2 From d1d7093f3f639e693fa763eb51d8d9c349bfd606 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 11 Jan 2011 10:35:29 +0100 Subject: ALSA: oxygen: add S/PDIF source selection for Claro cards Add a mixer control to switch between the optical and coaxial S/PDIF inputs on the HT-Omega Claro and Claro halo cards. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index db39cfc..d7e8ddd 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -34,6 +34,7 @@ * GPIO 6 -> enable on-board S/PDIF input * * Claro models: + * GPIO 6 -> S/PDIF from optical (0) or coaxial (1) input * GPIO 8 -> enable headphone amplifier * * CM9780: @@ -137,6 +138,7 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_MERIDIAN_DIG_EXT 0x0010 #define GPIO_MERIDIAN_DIG_BOARD 0x0040 +#define GPIO_CLARO_DIG_COAX 0x0040 #define GPIO_CLARO_HP 0x0100 struct generic_data { @@ -268,6 +270,8 @@ static void claro_enable_hp(struct oxygen *chip) static void claro_init(struct oxygen *chip) { + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CLARO_DIG_COAX); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_DIG_COAX); ak4396_init(chip); wm8785_init(chip); claro_enable_hp(chip); @@ -275,6 +279,8 @@ static void claro_init(struct oxygen *chip) static void claro_halo_init(struct oxygen *chip) { + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CLARO_DIG_COAX); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_DIG_COAX); ak4396_init(chip); ak5385_init(chip); claro_enable_hp(chip); @@ -523,14 +529,24 @@ static const struct snd_kcontrol_new hpf_control = { .put = hpf_put, }; -static int meridian_dig_source_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +static int meridian_dig_source_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) { static const char *const names[2] = { "On-board", "Extension" }; return snd_ctl_enum_info(info, 1, 2, names); } -static int meridian_dig_source_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +static int claro_dig_source_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { "Optical", "Coaxial" }; + + return snd_ctl_enum_info(info, 1, 2, names); +} + +static int meridian_dig_source_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; @@ -540,7 +556,19 @@ static int meridian_dig_source_get(struct snd_kcontrol *ctl, struct snd_ctl_elem return 0; } -static int meridian_dig_source_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +static int claro_dig_source_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + value->value.enumerated.item[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & + GPIO_CLARO_DIG_COAX); + return 0; +} + +static int meridian_dig_source_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; u16 old_reg, new_reg; @@ -560,6 +588,25 @@ static int meridian_dig_source_put(struct snd_kcontrol *ctl, struct snd_ctl_elem return changed; } +static int claro_dig_source_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 old_reg, new_reg; + int changed; + + mutex_lock(&chip->mutex); + old_reg = oxygen_read16(chip, OXYGEN_GPIO_DATA); + new_reg = old_reg & ~GPIO_CLARO_DIG_COAX; + if (value->value.enumerated.item[0]) + new_reg |= GPIO_CLARO_DIG_COAX; + changed = new_reg != old_reg; + if (changed) + oxygen_write16(chip, OXYGEN_GPIO_DATA, new_reg); + mutex_unlock(&chip->mutex); + return changed; +} + static const struct snd_kcontrol_new meridian_dig_source_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "IEC958 Source Capture Enum", @@ -568,6 +615,14 @@ static const struct snd_kcontrol_new meridian_dig_source_control = { .put = meridian_dig_source_put, }; +static const struct snd_kcontrol_new claro_dig_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Source Capture Enum", + .info = claro_dig_source_info, + .get = claro_dig_source_get, + .put = claro_dig_source_put, +}; + static int generic_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); @@ -600,6 +655,34 @@ static int meridian_mixer_init(struct oxygen *chip) return 0; } +static int claro_mixer_init(struct oxygen *chip) +{ + int err; + + err = generic_wm8785_mixer_init(chip); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&claro_dig_source_control, chip)); + if (err < 0) + return err; + return 0; +} + +static int claro_halo_mixer_init(struct oxygen *chip) +{ + int err; + + err = generic_mixer_init(chip); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&claro_dig_source_control, chip)); + if (err < 0) + return err; + return 0; +} + static void dump_ak4396_registers(struct oxygen *chip, struct snd_info_buffer *buffer) { @@ -700,13 +783,14 @@ static int __devinit get_oxygen_model(struct oxygen *chip, break; case MODEL_CLARO: chip->model.init = claro_init; + chip->model.mixer_init = claro_mixer_init; chip->model.cleanup = claro_cleanup; chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; break; case MODEL_CLARO_HALO: chip->model.init = claro_halo_init; - chip->model.mixer_init = generic_mixer_init; + chip->model.mixer_init = claro_halo_mixer_init; chip->model.cleanup = claro_cleanup; chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; -- cgit v0.10.2 From 921eebdc18c82268eab446592191b39e35d031d6 Mon Sep 17 00:00:00 2001 From: Karsten Wiese Date: Mon, 3 Jan 2011 02:41:58 +0100 Subject: ALSA: snd-usb-us122l: Fix MIDI output The US-122L always reads 9 bytes per urb unless they are set to 0xFD. Signed-off-by: Karsten Wiese Cc: stable@kernel.org Signed-off-by: Takashi Iwai diff --git a/sound/usb/midi.c b/sound/usb/midi.c index d7021cb..db2dc5f 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -850,8 +850,8 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, return; } - memset(urb->transfer_buffer + count, 0xFD, 9 - count); - urb->transfer_buffer_length = count; + memset(urb->transfer_buffer + count, 0xFD, ep->max_transfer - count); + urb->transfer_buffer_length = ep->max_transfer; } static struct usb_protocol_ops snd_usbmidi_122l_ops = { @@ -1295,6 +1295,13 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ ep->max_transfer = 4; break; + /* + * Some devices only work with 9 bytes packet size: + */ + case USB_ID(0x0644, 0x800E): /* Tascam US-122L */ + case USB_ID(0x0644, 0x800F): /* Tascam US-144 */ + ep->max_transfer = 9; + break; } for (i = 0; i < OUTPUT_URBS; ++i) { buffer = usb_alloc_coherent(umidi->dev, -- cgit v0.10.2 From cdce2db74e156fbd9a2dc3c7b246166f8b70955b Mon Sep 17 00:00:00 2001 From: Karsten Wiese Date: Tue, 4 Jan 2011 01:20:37 +0100 Subject: ALSA: snd-usb-us122l: Fix missing NULL checks Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl(). Wake up poll waiters before returning from usb_stream_hwdep_ioctl(). Signed-off-by: Karsten Wiese Cc: stable@kernel.org Signed-off-by: Takashi Iwai diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 6ef68e4..084e6fc 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -273,29 +273,26 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, struct file *file, poll_table *wait) { struct us122l *us122l = hw->private_data; - struct usb_stream *s = us122l->sk.s; unsigned *polled; unsigned int mask; poll_wait(file, &us122l->sk.sleep, wait); - switch (s->state) { - case usb_stream_ready: - if (us122l->first == file) - polled = &s->periods_polled; - else - polled = &us122l->second_periods_polled; - if (*polled != s->periods_done) { - *polled = s->periods_done; - mask = POLLIN | POLLOUT | POLLWRNORM; - break; + mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; + if (mutex_trylock(&us122l->mutex)) { + struct usb_stream *s = us122l->sk.s; + if (s && s->state == usb_stream_ready) { + if (us122l->first == file) + polled = &s->periods_polled; + else + polled = &us122l->second_periods_polled; + if (*polled != s->periods_done) { + *polled = s->periods_done; + mask = POLLIN | POLLOUT | POLLWRNORM; + } else + mask = 0; } - /* Fall through */ - mask = 0; - break; - default: - mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; - break; + mutex_unlock(&us122l->mutex); } return mask; } @@ -381,6 +378,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, { struct usb_stream_config *cfg; struct us122l *us122l = hw->private_data; + struct usb_stream *s; unsigned min_period_frames; int err = 0; bool high_speed; @@ -426,18 +424,18 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, snd_power_wait(hw->card, SNDRV_CTL_POWER_D0); mutex_lock(&us122l->mutex); + s = us122l->sk.s; if (!us122l->master) us122l->master = file; else if (us122l->master != file) { - if (memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg))) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg))) { err = -EIO; goto unlock; } us122l->slave = file; } - if (!us122l->sk.s || - memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg)) || - us122l->sk.s->state == usb_stream_xrun) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg)) || + s->state == usb_stream_xrun) { us122l_stop(us122l); if (!us122l_start(us122l, cfg->sample_rate, cfg->period_frames)) err = -EIO; @@ -448,6 +446,7 @@ unlock: mutex_unlock(&us122l->mutex); free: kfree(cfg); + wake_up_all(&us122l->sk.sleep); return err; } -- cgit v0.10.2 From c386735264da97e6b6d15aa56361e9ef188b26ab Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 12 Jan 2011 08:30:12 +0100 Subject: ALSA: oxygen: fix Xonar DG input Apparently, this card uses the other I2S input. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index dd45cf4..e4de0b8 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -561,7 +561,7 @@ struct oxygen_model model_xonar_dg = { .model_data_size = sizeof(struct dg), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_1, + CAPTURE_0_FROM_I2S_2, .dac_channels_pcm = 6, .dac_channels_mixer = 0, .function_flags = OXYGEN_FUNCTION_SPI, -- cgit v0.10.2