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

[alsa-devel] [PATCH] ASoC: sgtl5000: Allow the codec to work in slave mode
by Fabio Estevam 02 Oct '14

02 Oct '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> When configuring sgtl5000 to work in slave mode the following error is seen: root@freescale /$ aplay dtmf_mono_16b_48000.wav [ 11.406460] random: nonblocking pool is initialized [ 11.600461] sgtl5000 1-000a: PLL not supported in slave mode [ 11.609052] sgtl5000 1-000a: ASoC: can't set sgtl5000 hw params: -22 ALSA lib pcm_direct.c:980:(snd1_pcm_direct_initialize_slave) unable to install hw params ALSA lib pcm_dmix.c:1030:(snd_pcm_dmix_open) unable to initialize slave aplay: main:660: audio open error: Invalid argument Do not error out when sgtl5000_set_clock() reaches the default path for slave mode. While at it, fix a typo in the comment. Tested on a imx53-qsb configured as ssi in master and sgtl5000 as slave. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index a604a22..b641ed1 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -619,14 +619,10 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) SGTL5000_MCLK_FREQ_SHIFT; break; default: - /* if mclk not satisify the divider, use pll */ + /* if mclk not satisfy the divider, use pll */ if (sgtl5000->master) { clk_ctl |= SGTL5000_MCLK_FREQ_PLL << SGTL5000_MCLK_FREQ_SHIFT; - } else { - dev_err(codec->dev, - "PLL not supported in slave mode\n"); - return -EINVAL; } } -- 1.9.1

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[alsa-devel] [PATCH v6 0/2] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 02 Oct '14

02 Oct '14

The NXP TDA998x HDMI transmitter may transmit audio to the HDMI link from 2 different sources, I2S and S/PDIF. This patch set adds an interface between the HDMI transmitter and the HDMI CODEC. The interface is used by the TDA998x driver to update the audio constraints from the display characteristics (EDID) and by the audio subsystem to connect the chosen audio source to the HDMI link. v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (2): ASoC:codecs: Add a transmitter interface to the HDMI CODEC drm/i2c:tda998x: Use the HDMI audio CODEC .../devicetree/bindings/drm/i2c/tda998x.txt | 18 ++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 299 +++++++++++++++++++-- include/sound/hdmi.h | 20 ++ sound/soc/codecs/hdmi.c | 174 +++++++++++- 5 files changed, 488 insertions(+), 24 deletions(-) create mode 100644 include/sound/hdmi.h -- 2.1.1

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[alsa-devel] [PATCH] ASoC: ssm4567: Remove duplicated else-if branch
by Anatol Pomozov 02 Oct '14

02 Oct '14

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/ssm4567.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 1dadacb..4b5c17f 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -173,8 +173,6 @@ static int ssm4567_hw_params(struct snd_pcm_substream *substream, dacfs = SSM4567_DAC_FS_32000_48000; else if (rate >= 64000 && rate <= 96000) dacfs = SSM4567_DAC_FS_64000_96000; - else if (rate >= 64000 && rate <= 96000) - dacfs = SSM4567_DAC_FS_64000_96000; else if (rate >= 128000 && rate <= 192000) dacfs = SSM4567_DAC_FS_128000_192000; else -- 2.1.0.rc2.206.gedb03e5

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[alsa-devel] [PATCH 0/2] ASoC: davinci-mcasp: Improved suspend/resume support
by Peter Ujfalusi 02 Oct '14

02 Oct '14

Hi, With this series the McASP will continue to work after deep sleep during audio activity. Without these changes the ongoing audio will not resume and it will time out, but streams started after resume were worked correctly. Regards, Peter --- Peter Ujfalusi (2): ASoC: davinci-mcasp: Convert the context save/restore to use array ASoC: davinvi-mcasp: Proper suspend/resume support while audio is active sound/soc/davinci/davinci-mcasp.c | 79 ++++++++++++++++++++++++++++----------- 1 file changed, 58 insertions(+), 21 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH] ASoC: rt286: Fix compilation warning for !PM
by Thierry Reding 02 Oct '14

02 Oct '14

From: Thierry Reding <treding(a)nvidia.com> The rt286_index_sync() function is only called in the resume path. If PM is disabled it becomes unused and shouldn't be built either. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/codecs/rt286.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index b86b426f159d..4aa555cbcca8 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -269,6 +269,7 @@ static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) return 0; } +#ifdef CONFIG_PM static void rt286_index_sync(struct snd_soc_codec *codec) { struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); @@ -279,6 +280,7 @@ static void rt286_index_sync(struct snd_soc_codec *codec) rt286->index_cache[i].def); } } +#endif static int rt286_support_power_controls[] = { RT286_DAC_OUT1, -- 2.1.0

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[alsa-devel] [PATCH] ASoC: tas2552: Fix compilation warning for !PM_RUNTIME
by Thierry Reding 02 Oct '14

02 Oct '14

From: Thierry Reding <treding(a)nvidia.com> The tas2552_sw_shutdown() function is only used by runtime suspend support, so only build it when necessary. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/codecs/tas2552.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 1ed57a7e57b6..f039dc825971 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -115,6 +115,7 @@ static const struct snd_soc_dapm_route tas2552_audio_map[] = { {"ClassD", NULL, "PLL"}, }; +#ifdef CONFIG_PM_RUNTIME static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) { u8 cfg1_reg; @@ -127,6 +128,7 @@ static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, TAS2552_SWS_MASK, cfg1_reg); } +#endif static int tas2552_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, -- 2.1.0

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[alsa-devel] [PATCH 1/2] ASoC: Intel: byt-rt5640: Add quirk for Dell Venue 8 Pro tablet
by Jarkko Nikula 02 Oct '14

02 Oct '14

It was found with help of Jan-Michael Brummer <jan.brummer(a)tabos.org> that Dell Venue 8 Pro tablet has a digital microphone connected to DMIC2 interface of the RT564x. This patch adds a DAPM route to DMIC2 and a quirk using it for that tablet. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Reported-by: Jan-Michael Brummer <jan.brummer(a)tabos.org> --- Hi Jan. Could you give a second test to this patch? I noticed I had a few lines missing from the version I sent you yesterday. If this still works please reply with your Tested-by:. Mark: I don't have this hw but Jan helped me to debug the right configuration and sent the needed machine information for the DMI quirk. --- sound/soc/intel/byt-rt5640.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 8392c160d9e2..a9619b4201f9 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -49,6 +49,10 @@ static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { {"DMIC1", NULL, "Internal Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"DMIC2", NULL, "Internal Mic"}, +}; + static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, @@ -56,6 +60,7 @@ static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { enum { BYT_RT5640_DMIC1_MAP, + BYT_RT5640_DMIC2_MAP, BYT_RT5640_IN1_MAP, }; @@ -111,6 +116,15 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP, }, + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "DellInc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"), + }, + .driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP | + BYT_RT5640_DMIC_EN), + }, {} }; @@ -138,6 +152,10 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5640_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); break; + case BYT_RT5640_DMIC2_MAP: + custom_map = byt_rt5640_intmic_dmic2_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map); + break; default: custom_map = byt_rt5640_intmic_dmic1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); -- 2.1.0

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[alsa-devel] Differences between tlv320aic3x and tlv320aic31xx
by Alexandre Belloni 01 Oct '14

01 Oct '14

Hi, I'm trying to add support for the tlv320aic3104a present on a custom board using an am335x. I see that Benoit seemed to have been successful when using the tlv320aic3x driver with minor modifications. http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058966.ht… On the other hand, in the past, I've used a tlv320aic3100 with the tlv320aic31xx driver. My plan for now is to use the tlv320aic3x driver with snd-soc-evm as this is the one it is using (it is using .codec_dai_name = "tlv320aic3x-hifi"). I had no luck until now but I still have to review my pinmuxing. But before I investigate to much, can you confirm what codecs are supported by which driver? I'm pretty sure that would be a nice addition to the documentation too. Thanks, -- Alexandre Belloni, Free Electrons Embedded Linux, Kernel and Android engineering http://free-electrons.com

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: Minor coding style modify
by Oder Chiou 01 Oct '14

01 Oct '14

Minor coding style modify Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a7762d0..dfbe74a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1838,7 +1838,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, RT5645_I2S_DL_MASK, val_len); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; - case RT5645_AIF2: + case RT5645_AIF2: mask_clk = RT5645_I2S_BCLK_MS2_MASK | RT5645_I2S_PD2_MASK; val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; @@ -1983,7 +1983,7 @@ static int rt5645_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, snd_soc_update_bits(codec, RT5645_GLB_CLK, RT5645_PLL1_SRC_MASK, RT5645_PLL1_SRC_BCLK1); break; - case RT5645_AIF2: + case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_GLB_CLK, RT5645_PLL1_SRC_MASK, RT5645_PLL1_SRC_BCLK2); break; -- 1.8.1.1.439.g50a6b54

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[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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