October 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: fsl: fix dependencies for wm8962
by Arnd Bergmann 01 Oct '14

01 Oct '14

The wm8962 driver uses the input subsystem, but is selected by SND_SOC_FSL_ASOC_CARD, which can be built with CONFIG_INPUT disabled, resulting in this link error: ERROR: "input_event" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! ERROR: "input_register_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! ERROR: "devm_input_allocate_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! This adds an explicit Kconfig dependency to prevent this configuration from being used. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 708b4351f08 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6164e78b466a..99e9386f7956 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -282,7 +282,7 @@ config SND_SOC_IMX_MC13783 config SND_SOC_FSL_ASOC_CARD tristate "Generic ASoC Sound Card with ASRC support" - depends on OF && I2C + depends on OF && I2C && INPUT select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_ESAI

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Re: [alsa-devel] ASoC: ssm4567: Add driver for Analog Devices SSM4567 amplifier
by Dan Carpenter 01 Oct '14

01 Oct '14

Hello Anatol Pomozov, The patch 1ee44ce03011: "ASoC: ssm4567: Add driver for Analog Devices SSM4567 amplifier" from Sep 26, 2014, leads to the following static checker warning: sound/soc/codecs/ssm4567.c:176 ssm4567_hw_params() warn: we tested 'rate <= 96000' before and it was 'false' sound/soc/codecs/ssm4567.c 160 static int ssm4567_hw_params(struct snd_pcm_substream *substream, 161 struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) 162 { 163 struct snd_soc_codec *codec = dai->codec; 164 struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec); 165 unsigned int rate = params_rate(params); 166 unsigned int dacfs; 167 168 if (rate >= 8000 && rate <= 12000) 169 dacfs = SSM4567_DAC_FS_8000_12000; 170 else if (rate >= 16000 && rate <= 24000) 171 dacfs = SSM4567_DAC_FS_16000_24000; 172 else if (rate >= 32000 && rate <= 48000) 173 dacfs = SSM4567_DAC_FS_32000_48000; 174 else if (rate >= 64000 && rate <= 96000) 175 dacfs = SSM4567_DAC_FS_64000_96000; 176 else if (rate >= 64000 && rate <= 96000) 177 dacfs = SSM4567_DAC_FS_64000_96000; These two lines are a cut and paste error. The intent is not clear. 178 else if (rate >= 128000 && rate <= 192000) 179 dacfs = SSM4567_DAC_FS_128000_192000; 180 else 181 return -EINVAL; 182 183 return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL, 184 SSM4567_DAC_FS_MASK, dacfs); 185 } regards, dan carpenter

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[alsa-devel] [PATCH] ASoC: fsl: Do not force codecs selection by SND_SOC_FSL_ASOC_CARD
by Fabio Estevam 01 Oct '14

01 Oct '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The wm8962 driver uses the input subsystem, but it is selected by SND_SOC_FSL_ASOC_CARD, which can be built with CONFIG_INPUT disabled, resulting in this link error: ERROR: "input_event" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! ERROR: "input_register_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! ERROR: "devm_input_allocate_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! Do not force the selection of the codecs by SND_SOC_FSL_ASOC_CARD to avoid such problem. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6164e78..081e406 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -288,9 +288,6 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_ESAI select SND_SOC_FSL_SAI select SND_SOC_FSL_SSI - select SND_SOC_CS42XX8_I2C - select SND_SOC_SGTL5000 - select SND_SOC_WM8962 help ALSA SoC Audio support with ASRC feature for Freescale SoCs that have ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: rt5640: Add function for enabling DMIC from ACPI probed machine
by Jarkko Nikula 01 Oct '14

01 Oct '14

There is no code enabling DMIC clock in systems that don't provide platform data for rt5640 after commit 71d97a794301 ("ASoC: rt5640: Use the platform data for DMIC settings"). I think it's worth to keep this static DMIC clock and alternative data pin setting during probe time. For making possible to use DMIC from ACPI probed machine (prior ACPI 5.1 with _DSD) this patch moves DMIC configuration to new exported rt5640_dmic_enable() that machine drivers can call. Please note, this patch moves DMIC configuration from i2c probe to codec probe in case platform data for rt5640 is set. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> --- for-next. I don't think there is regression caused by 71d97a794301 for the byt-rt5640 machines. Theoretically yes for internal development platform (which is don't care) but also byt-rt5640 got DMI quirks support for machine specific routes only recently (i.e. machine without quirk was never working completely). --- sound/soc/codecs/rt5640.c | 49 +++++++++++++++++++++++++++++------------------ sound/soc/codecs/rt5640.h | 3 +++ 2 files changed, 33 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index f1ec6e6bd08a..c3f2decd643c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1906,6 +1906,32 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, return 0; } +int rt5640_dmic_enable(struct snd_soc_codec *codec, + bool dmic1_data_pin, bool dmic2_data_pin) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); + + if (dmic1_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); + } + + if (dmic2_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5640_dmic_enable); + static int rt5640_probe(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); @@ -1945,6 +1971,10 @@ static int rt5640_probe(struct snd_soc_codec *codec) return -ENODEV; } + if (rt5640->pdata.dmic_en) + rt5640_dmic_enable(codec, rt5640->pdata.dmic1_data_pin, + rt5640->pdata.dmic2_data_pin); + return 0; } @@ -2195,25 +2225,6 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); - if (rt5640->pdata.dmic_en) { - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); - - if (rt5640->pdata.dmic1_data_pin) { - regmap_update_bits(rt5640->regmap, RT5640_DMIC, - RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); - } - - if (rt5640->pdata.dmic2_data_pin) { - regmap_update_bits(rt5640->regmap, RT5640_DMIC, - RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); - } - } - rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 58ebe96b86da..3deb8babeabb 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2097,4 +2097,7 @@ struct rt5640_priv { bool hp_mute; }; +int rt5640_dmic_enable(struct snd_soc_codec *codec, + bool dmic1_data_pin, bool dmic2_data_pin); + #endif -- 2.1.0

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Re: [alsa-devel] [PATCH] ASoC: remove use of gpiochip_remove() retval
by Mark Brown 01 Oct '14

01 Oct '14

On Wed, Oct 01, 2014 at 05:17:44PM +0530, Pramod Gurav wrote: > On Wednesday 01 October 2014 05:08 PM, Mark Brown wrote: > > Applied, I'm assuming that the API change is going to be introduced in a > > future kernel version since obviously it'll break the build otherwise. > Mark please revert the changes as these are already present in > linux-next master. I realized it late. Sorry for trouble. Done.

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Re: [alsa-devel] [PATCH] ASoC: remove use of gpiochip_remove() retval
by Mark Brown 01 Oct '14

01 Oct '14

On Wed, Oct 01, 2014 at 11:17:46AM +0530, Pramod Gurav wrote: > Get rid of using return value from gpiochip_remove() as it returns > void with a new change in kernel. Applied, I'm assuming that the API change is going to be introduced in a future kernel version since obviously it'll break the build otherwise.

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[alsa-devel] [BUG] Bad decibel data on C-Media USB sound card
by Allen M. 01 Oct '14

01 Oct '14

Raymond Yau wrote: >It affect your measurent if you are using the mic on your usb audio if the >automatic gain control is switch on Automatic gain control is now switched off. >Background noise affect the result if you are not using line in and line >out with a loopback cable I am using loopback, noise should not be an issue. >How many steps need to reduce amplitude of the recorded sine wave by >half when you play a sine wav through your usb audio and record it back >using audacity? Using recorded sine wav provided by 'speaker-test -t sin' amplitude of recorded sine wave reduced by half at step 77 output of 'amixer -c 2 set Headphone 77': Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 151 Mono: Front Left: Playback 77 [51%] [-13.94dB] [on] Front Right: Playback 77 [51%] [-13.94dB] [on] >You many need to change the name of the volume control to headphone >playback volume in alsamixertest Done. alsamixertest output: http://ix.io/ezd

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Re: [alsa-devel] [PATCH] ASoC:kirkwood: Don't raise an error when no DAI format
by Russell King - ARM Linux 01 Oct '14

01 Oct '14

On Sun, Sep 28, 2014 at 04:19:27PM +0200, Jean-Francois Moine wrote: > The two DAIs of the kirkwood controller have a unique PCM format. > > The simple-card sets the audio hardware definitions of all CPU DAIs. > The PCM format is defined only when it is present in the DT. > > This patch prevents the controller to raise an error when > the DT audio card definition by the simple card contains the PCM > format of one CPU DAI only. I think this is a silly idea - why should every driver have additional code to detect when it's called to do thing. Why doesn't the simple card code always pass the required format? Looking at other drivers, no one else does this; they all appear to require the proper format to be specified. What some drivers do (eg, omap-mcbsp.c) is to block set_fmt when the DAI is already in use - setting a flag "configured" in hw_params, and clearing it in shutdown. Maybe following this will solve your problem. In any case, random drivers doing stuff differently without reason is really not a good idea. -- FTTC broadband for 0.8mile line: currently at 9.5Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [GIT PULL] Another fix for v3.17
by Mark Brown 01 Oct '14

01 Oct '14

The following changes since commit 7d1311b93e58ed55f3a31cc8f94c4b8fe988a2b9: Linux 3.17-rc1 (2014-08-16 10:40:26 -0600) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.17-rc7 for you to fetch changes up to fe2a08b3bf1a6e35c00e18843bc19aa1778432c3: ASoC: ssm2602: do not hardcode type to SSM2602 (2014-09-30 13:27:19 +0100) ---------------------------------------------------------------- ASoC: Fix SSM2602 device identification Another fix for v3.17, fixing device identification after the split out of the I2C and SPI code. ---------------------------------------------------------------- Stefan Kristiansson (1): ASoC: ssm2602: do not hardcode type to SSM2602 sound/soc/codecs/ssm2602.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

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