February 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: dmic: Add optional wakeup delay
by Matthias Kaehlcke 16 Feb '18

16 Feb '18

On some systems a delay is needed after switching on the clocks, to allow the output to stabilize and avoid a popping noise at the beginning of the recording. Add the optional device tree property 'wakeup-delay-ms' and apply the specified delay after enabling the mic. A blocking delay can't be applied in dmic_daiops_trigger() since the function is called in atomic context. Instead use a DAPM event handler to set the enable GPIO and apply the delay in the handler. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v2: - use DAPM event handler instead of _prepare() and get rid of _trigger() - skip error check for optional 'wakeup-delay-ms' property - updated commit message .../devicetree/bindings/sound/dmic.txt | 2 + sound/soc/codecs/dmic.c | 63 ++++++++++--------- 2 files changed, 37 insertions(+), 28 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt index f7bf65611453..e957b4136716 100644 --- a/Documentation/devicetree/bindings/sound/dmic.txt +++ b/Documentation/devicetree/bindings/sound/dmic.txt @@ -8,6 +8,7 @@ Required properties: Optional properties: - dmicen-gpios: GPIO specifier for dmic to control start and stop - num-channels: Number of microphones on this DAI + - wakeup-delay-ms: Delay (in ms) after enabling the DMIC Example node: @@ -15,4 +16,5 @@ Example node: compatible = "dmic-codec"; dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; num-channels = <1>; + wakeup-delay-ms <50>; }; diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index cf83c423394d..b5f60ac22ace 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,7 @@ * */ +#include <linux/delay.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <linux/platform_device.h> @@ -29,34 +30,33 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -static int dmic_daiops_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); +struct dmic { + struct gpio_desc *gpio_en; + int wakeup_delay; +}; - if (!dmic_en) - return 0; +static int dmic_aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct dmic *dmic = snd_soc_codec_get_drvdata(codec); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(dmic_en, 1); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 1); + + if (dmic->wakeup_delay) + msleep(dmic->wakeup_delay); break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(dmic_en, 0); + case SND_SOC_DAPM_POST_PMD: + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 0); break; } return 0; } -static const struct snd_soc_dai_ops dmic_dai_ops = { - .trigger = dmic_daiops_trigger, -}; - static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -68,26 +68,33 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &dmic_dai_ops, }; static int dmic_codec_probe(struct snd_soc_codec *codec) { - struct gpio_desc *dmic_en; + struct dmic *dmic; + + dmic = devm_kzalloc(codec->dev, sizeof(*dmic), GFP_KERNEL); + if (!dmic) + return -ENOMEM; + + dmic->gpio_en = devm_gpiod_get_optional(codec->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic->gpio_en)) + return PTR_ERR(dmic->gpio_en); - dmic_en = devm_gpiod_get_optional(codec->dev, - "dmicen", GPIOD_OUT_LOW); - if (IS_ERR(dmic_en)) - return PTR_ERR(dmic_en); + device_property_read_u32(codec->dev, "wakeup-delay-ms", + &dmic->wakeup_delay); - snd_soc_codec_set_drvdata(codec, dmic_en); + snd_soc_codec_set_drvdata(codec, dmic); return 0; } static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { - SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("DMIC AIF", "Capture", 0, + SND_SOC_NOPM, 0, 0, dmic_aif_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_INPUT("DMic"), }; -- 2.16.1.291.g4437f3f132-goog

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[alsa-devel] [asoc:topic/ak4458 1/2] sound/soc/codecs/ak4458.c:577:28: sparse: symbol 'ak4458_regmap' was not declared. Should it be static?
by kbuild test robot 16 Feb '18

16 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/ak4458 head: 17e2304f0ad31d3f9bba4285b7024bf3c3626600 commit: 08660086eff9e9e69246a65ee48aba5c3901bf5e [1/2] ASoC: ak4458: Add support for AK4458 DAC driver reproduce: # apt-get install sparse git checkout 08660086eff9e9e69246a65ee48aba5c3901bf5e make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/ak4458.c:577:28: sparse: symbol 'ak4458_regmap' was not declared. Should it be >> sound/soc/codecs/ak4458.c:587:25: sparse: symbol 'ak4458_pm' was not declared. Should it be Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] MAINTAINERS: Add myself as sgtl5000 maintainer
by Fabio Estevam 16 Feb '18

16 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> I would like helping maintaining and reviewing/testing sgtl5000 related patches. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 9a7f76e..c488c8a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -9922,6 +9922,13 @@ F: Documentation/ABI/stable/sysfs-bus-nvmem F: include/linux/nvmem-consumer.h F: include/linux/nvmem-provider.h +NXP SGTL5000 DRIVER +M: Fabio Estevam <fabio.estevam(a)nxp.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/sgtl5000.txt +F: sound/soc/codecs/sgtl5000* + NXP TDA998X DRM DRIVER M: Russell King <linux(a)armlinux.org.uk> S: Supported -- 2.7.4

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[alsa-devel] [PATCH v3] ASoC: sgtl5000: Fix suspend/resume
by Fabio Estevam 16 Feb '18

16 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 8419caa72702 ("ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF") causes the sgtl5000 to fail after a suspend/resume sequence: # aplay /media/a2002011001-e02.wav Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2051: write error: Input/output error The problem is caused by the fact that the aforementioned commit dropped the cache handling, so re-introduce the register map resync to fix the problem. Cc: <stable(a)vger.kernel.org> Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v2: - Rebased against Linus' tree sound/soc/codecs/sgtl5000.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e1ab553..7ff5cb7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -871,15 +871,26 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct sgtl5000_priv *sgtl = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: + regcache_cache_only(sgtl->regmap, false); + ret = regcache_sync(sgtl->regmap); + if (ret) { + regcache_cache_only(sgtl->regmap, true); + return ret; + } + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, SGTL5000_REFTOP_POWERUP); break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl->regmap, true); snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, 0); break; -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] Add support for AK5558 ADC
by Daniel Baluta 16 Feb '18

16 Feb '18

We support normal mode, TDM mode and pm. Changes since v2: [addressed comments from Andy, Fabio and Rob] * sort include files * use probe_new * reword the binding document * use adc@10 instead of ak5558@10 * remove file name at the beginning of codec source code. * make i2c_probe and i2c_remove parameters naming consistent. Changes since v1: [addressed comments from Andy and Fabio] * fix GPIO polarity from active high to active low for correct documentation * fix license header by using SPDX identifier * remove debug prints at the beginning of functions. * only support auto clock switching (manual switching was dead code anyway) (in the future we could add a DT property to choose between manual and auto) * Use gpiod API * use GENMASK * introduce power_off/power_on One open question is the resume sequence which appears to need power_off/power_on. Just power_on alone isn't enough. With just power_on after resume aplay plays a song for 1 seconds and then the sound stops. Datasheet says, page 55" (1) The PDN pin should be held to “L” for more than 150 ns after AVDD and TVDD are powered up. Daniel Baluta (2): ASoC: codecs: Add support for AK5558 ADC driver ASoC: ak5558: Add bindings for AK5558 ADC Documentation/devicetree/bindings/sound/ak5558.txt | 22 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5558.c | 618 +++++++++++++++++++++ sound/soc/codecs/ak5558.h | 52 ++ 5 files changed, 700 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5558.txt create mode 100644 sound/soc/codecs/ak5558.c create mode 100644 sound/soc/codecs/ak5558.h -- 2.7.4

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[alsa-devel] [PATCH 0/2] ASoC: Intel: kabylake: Adds da7219+max98537 machine driver
by Naveen Manohar 16 Feb '18

16 Feb '18

Add kabylake machine driver & corresponding driver entry in machine table for maxim speaker codec - max98357a, configured to ssp0 and dialog headset codec - da7219, configured to ssp1. Naveen Manohar (2): ASoC: Intel: Add Kabylake-y Dialog Maxim machine driver ASoC: Intel: Add Kabylake Dialog+Maxim machine driver entry sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 613 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 13 + 4 files changed, 642 insertions(+) create mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c -- 1.9.1

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[alsa-devel] [PATCH v3 0/2] Add support for AK4458 DAC
by Cosmin-Gabriel Samoila 16 Feb '18

16 Feb '18

We support Normal Mode, TDM Mode and PM. Changes since V2: - remove unused code left from V1 - rewrite set_dai_fmt to only remember the format since - the format is rewritten in hw_params - use defined constants for mode in set_tdm_slot function - use GENMASK - add commit message for bindings patch - modify bindings description text since we only support I2C for now - rename ak4458_codec_probe to ak4458_probe - write whole implementation of regmap init and gpiod get in i2c_probe - added ak4458_remove where we power_off the chip Changes since V1: - support only i2c - i2c probe / remove/suspend/ resume moved into shared code - change enum array to separate variables - aggregate Volume control for Left and Right Channel - remove mux from DAPM - remove unused functions - remove unused controls - remove unused macros - use snd_soc_update_bits instead of soc_read + soc_write - use GPIOD - use SPDX identifier - use dac@10 instgitead of ak4458@10 - use explicit power_on/off functions - add __maybe_unused attribute - use probe_new for i2c_probe Cosmin-Gabriel Samoila (2): ASoC: codecs: Add support for AK4458 DAC driver ASoC: ak4458: Add bindings for AK4458 DAC Documentation/devicetree/bindings/sound/ak4458.txt | 23 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4458.c | 659 +++++++++++++++++++++ sound/soc/codecs/ak4458.h | 86 +++ 5 files changed, 776 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4458.txt create mode 100644 sound/soc/codecs/ak4458.c create mode 100644 sound/soc/codecs/ak4458.h -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix suspend/resume
by Fabio Estevam 16 Feb '18

16 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 8419caa72702 ("ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF") causes the sgtl5000 to fail after a suspend/resume sequence: # aplay /media/a2002011001-e02.wav Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2051: write error: Input/output error The problem is caused by the fact that the aforementioned commit dropped the cache handling, so re-introduce the register map resync to fix the problem. Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Re-introduce cache handling (Mark). Mark, If this gets applied into next, then I can generate the fixes for the 4.9 and 4.14 stable versions later. Thanks sound/soc/codecs/sgtl5000.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c26ac22..7514c77 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -871,15 +871,26 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, static int sgtl5000_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { + struct sgtl5000_priv *sgtl = snd_soc_component_get_drvdata(component); + int ret; + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: + regcache_cache_only(sgtl->regmap, false); + ret = regcache_sync(sgtl->regmap); + if (ret) { + regcache_cache_only(sgtl->regmap, true); + return ret; + } + snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, SGTL5000_REFTOP_POWERUP); break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl->regmap, true); snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, 0); break; -- 2.7.4

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[alsa-devel] WM8985 Machine Driver Write Error
by Mohammad, Jamal M 16 Feb '18

16 Feb '18

Hi Guys, We have a custom i.MX6 Board which has WM8985 audio codec. I developed machine level driver ( imx-wm8985.c ) looking at imx-wm8962 code. Driver loads fine without any error. "aplay -L" list the sound card. I2C interface works fine but it is write only peripheral. But when I am trying to play a Wav file using aplay, it fails with "aplay: pcm_write:1939: write error: Input/output error". Reading Online found that this happens when routing is incorrect. I am unable to proceed after this..Please help me to fix this issue. Here is my device tree. sound { compatible = "fsl,imx6q-sabresd-wm8985", "fsl,imx-audio-wm8985"; model = "wm8985-audio"; cpu-dai = <&ssi2>; audio-codec = <&codec>; audio-routing = "Ext Spk", "SPKL", "Ext Spk", "SPKR"; mux-int-port = <2>; mux-ext-port = <3>; }; Please provide me clue/hint to proceed and solve the issue. I can provide you more details if you need. Thanks for your help. Mohammad Jamal Mohiuddin Software Engineer, Retail Hosp/HW NCR Corporation India Pvt Ltd. MohammadJamal.Mohiuddin(a)ncr.com | www.ncr.com<http://www.ncr.com>

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[alsa-devel] sgtl5000 fails after suspend/resume
by Fabio Estevam 15 Feb '18

15 Feb '18

Hi, I noticed that sgtl5000 does not work after a suspend/resume cycle: # aplay /media/a2002011001-e02.wav Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2051: write error: Input/output error Tested several 4.16-rc1 (and also previous kernels like 4.4) and all of them failed. I think it worked in some point in time. Tested on a imx53qsb and imx6 wandboard. Does anyone have any suggestions? Thanks, Fabio Estevam

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