From e48d73c697b77b798a82e86c937fc41e597a1471 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:32 +0200 Subject: ASoC: ak4671: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 998fa0c..686cacb 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -611,20 +611,7 @@ static struct snd_soc_dai_driver ak4671_dai = { .ops = &ak4671_dai_ops, }; -static int ak4671_probe(struct snd_soc_codec *codec) -{ - return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -static int ak4671_remove(struct snd_soc_codec *codec) -{ - ak4671_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { - .probe = ak4671_probe, - .remove = ak4671_remove, .set_bias_level = ak4671_set_bias_level, .controls = ak4671_snd_controls, .num_controls = ARRAY_SIZE(ak4671_snd_controls), -- cgit v0.10.2 From eb58960e9ebf15cde8ca1248e15ecb0c8f3a28bd Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Fri, 24 Oct 2014 21:25:59 +0200 Subject: ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex. The CMR divider register is shared by playback and capture. The SSC driver therefore tries to enforce rules so that the needed register content do not conflict during simultaneous playback/capture. However, the implementation also prevents changing the register content in half-duplex scenarios, which is needed when using the OSS API. Thus, only lock the divider if there is a stream in the other direction. Fixes the below program to not fail with the atmel ssc dai in master mode. int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22025; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin Acked-by: Bo Shen Signed-off-by: Mark Brown diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f39..b1cc2a4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, * transmit and receive, so if a value has already * been set, it must match this value. */ - if (ssc_p->cmr_div == 0) + if (ssc_p->dir_mask != + (SSC_DIR_MASK_PLAYBACK | SSC_DIR_MASK_CAPTURE)) + ssc_p->cmr_div = div; + else if (ssc_p->cmr_div == 0) ssc_p->cmr_div = div; else if (div != ssc_p->cmr_div) -- cgit v0.10.2 From ef326f4bb2675e9309ba318b19442d9823e58ee2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 12 Nov 2014 14:55:26 +0000 Subject: ASoC: arizona: Add support for 768kHz DMIC operation The new IPs supports a new lower frequency 768kHz DMIC operation add support for this into the OSR control. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0c05e7a..786464f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -648,7 +648,7 @@ SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum, EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); static const char * const arizona_in_dmic_osr_text[] = { - "1.536MHz", "3.072MHz", "6.144MHz", + "1.536MHz", "3.072MHz", "6.144MHz", "768kHz", }; const struct soc_enum arizona_in_dmic_osr[] = { -- cgit v0.10.2 From e9c7f34a7eba13e1a53212246c607d13574f9eff Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 12 Nov 2014 16:12:46 +0000 Subject: ASoC: arizona: Add DSP_B and LEFT_J mode support Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 786464f..19887bf 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -61,6 +61,11 @@ #define ARIZONA_FLL_MIN_OUTDIV 2 #define ARIZONA_FLL_MAX_OUTDIV 7 +#define ARIZONA_FMT_DSP_MODE_A 0 +#define ARIZONA_FMT_DSP_MODE_B 1 +#define ARIZONA_FMT_I2S_MODE 2 +#define ARIZONA_FMT_LEFT_JUSTIFIED_MODE 3 + #define arizona_fll_err(_fll, fmt, ...) \ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_warn(_fll, fmt, ...) \ @@ -946,10 +951,26 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - mode = 0; + mode = ARIZONA_FMT_DSP_MODE_A; + break; + case SND_SOC_DAIFMT_DSP_B: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "DSP_B not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_DSP_MODE_B; break; case SND_SOC_DAIFMT_I2S: - mode = 2; + mode = ARIZONA_FMT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "LEFT_J not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_LEFT_JUSTIFIED_MODE; break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", @@ -1298,7 +1319,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { + val &= ARIZONA_AIF1_FMT_MASK; + if ((channels & 1) && (val == ARIZONA_FMT_I2S_MODE)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); bclk_target /= channels; bclk_target *= channels + 1; -- cgit v0.10.2 From 2849bde56aac38645c5ed2af3971358b89a929f6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:47:50 +0100 Subject: ASoC: alc5623: Cleanup bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Also remove the manual sequencing back to SND_SOC_BIAS_ON in resume as this is already handled by the ASoC core. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 9d0755a..bdf8c5a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -866,7 +866,6 @@ static int alc5623_suspend(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(alc5623->regmap, true); return 0; @@ -887,15 +886,6 @@ static int alc5623_resume(struct snd_soc_codec *codec) return ret; } - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* charge alc5623 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - alc5623_set_bias_level(codec, codec->dapm.bias_level); - } - return 0; } @@ -906,9 +896,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) alc5623_reset(codec); - /* power on device */ - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (alc5623->add_ctrl) { snd_soc_write(codec, ALC5623_ADD_CTRL_REG, alc5623->add_ctrl); @@ -964,19 +951,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5623_remove(struct snd_soc_codec *codec) -{ - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .probe = alc5623_probe, - .remove = alc5623_remove, .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config alc5623_regmap = { -- cgit v0.10.2 From 5c9dc0898f343473efa3056fff9d5a9fbd577272 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:47:51 +0100 Subject: ASoC: alc5632: Cleanup bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 85942ca..d1fdbc2 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1038,23 +1038,15 @@ static struct snd_soc_dai_driver alc5632_dai = { }; #ifdef CONFIG_PM -static int alc5632_suspend(struct snd_soc_codec *codec) -{ - alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int alc5632_resume(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); regcache_sync(alc5632->regmap); - alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } #else -#define alc5632_suspend NULL #define alc5632_resume NULL #endif @@ -1062,9 +1054,6 @@ static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - /* power on device */ - alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - switch (alc5632->id) { case 0x5c: snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls, @@ -1077,19 +1066,12 @@ static int alc5632_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5632_remove(struct snd_soc_codec *codec) -{ - alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5632 = { .probe = alc5632_probe, - .remove = alc5632_remove, - .suspend = alc5632_suspend, .resume = alc5632_resume, .set_bias_level = alc5632_set_bias_level, + .suspend_bias_off = true, + .controls = alc5632_snd_controls, .num_controls = ARRAY_SIZE(alc5632_snd_controls), .dapm_widgets = alc5632_dapm_widgets, -- cgit v0.10.2 From 1689092da0815997eeded3f4c7b35be3df444df5 Mon Sep 17 00:00:00 2001 From: Nicolas Ferre Date: Mon, 1 Dec 2014 16:57:52 +0100 Subject: ASoC: snd-soc-afeb9260: delete driver as board has just been removed During the removal of all AT91 !DT boards, we removed the AFEB9260. This driver is !DT and was only used by this board, so we delete it as well. Reported-by: Paul Bolle Signed-off-by: Nicolas Ferre Signed-off-by: Mark Brown diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 5baabc8..466a821 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -17,4 +17,3 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o -obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c deleted file mode 100644 index 9579799..0000000 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ /dev/null @@ -1,151 +0,0 @@ -/* - * afeb9260.c -- SoC audio for AFEB9260 - * - * Copyright (C) 2009 Sergey Lapin - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include - -#include -#include -#include - -#include "../codecs/tlv320aic23.h" -#include "atmel-pcm.h" -#include "atmel_ssc_dai.h" - -#define CODEC_CLOCK 12000000 - -static int afeb9260_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int err; - - /* Set the codec system clock for DAC and ADC */ - err = - snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); - - if (err < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return err; - } - - return err; -} - -static struct snd_soc_ops afeb9260_ops = { - .hw_params = afeb9260_hw_params, -}; - -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route afeb9260_audio_map[] = { - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic Jack"}, -}; - - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link afeb9260_dai = { - .name = "TLV320AIC23", - .stream_name = "AIC23", - .cpu_dai_name = "atmel-ssc-dai.0", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &afeb9260_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_machine_afeb9260 = { - .name = "AFEB9260", - .owner = THIS_MODULE, - .dai_link = &afeb9260_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = afeb9260_audio_map, - .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map), -}; - -static struct platform_device *afeb9260_snd_device; - -static int __init afeb9260_soc_init(void) -{ - int err; - struct device *dev; - - if (!(machine_is_afeb9260())) - return -ENODEV; - - - afeb9260_snd_device = platform_device_alloc("soc-audio", -1); - if (!afeb9260_snd_device) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260); - err = platform_device_add(afeb9260_snd_device); - if (err) - goto err1; - - dev = &afeb9260_snd_device->dev; - - return 0; -err1: - platform_device_put(afeb9260_snd_device); - return err; -} - -static void __exit afeb9260_soc_exit(void) -{ - platform_device_unregister(afeb9260_snd_device); -} - -module_init(afeb9260_soc_init); -module_exit(afeb9260_soc_exit); - -MODULE_AUTHOR("Sergey Lapin "); -MODULE_DESCRIPTION("ALSA SoC for AFEB9260"); -MODULE_LICENSE("GPL"); - -- cgit v0.10.2 From de82bf6c055e1f426845e7e140c7d48dd70b6d16 Mon Sep 17 00:00:00 2001 From: Nicolas Ferre Date: Tue, 2 Dec 2014 10:04:35 +0100 Subject: ASoC: Kconfig: remove not used SND_AT91_SOC_AFEB9260 option Now that the driver snd-soc-afeb9260.c is deleted, remove its Kconfig option. Reported-by: Paul Bolle Signed-off-by: Nicolas Ferre Signed-off-by: Mark Brown diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 27e3fc4..fb38783 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -52,12 +52,3 @@ config SND_AT91_SOC_SAM9X5_WM8731 help Say Y if you want to add support for audio SoC on an at91sam9x5 based board that is using WM8731 codec. - -config SND_AT91_SOC_AFEB9260 - tristate "SoC Audio support for AFEB9260 board" - depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC - select SND_ATMEL_SOC_PDC - select SND_ATMEL_SOC_SSC - select SND_SOC_TLV320AIC23_I2C - help - Say Y here to support sound on AFEB9260 board. -- cgit v0.10.2