June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] asoc: Update supported rate and format for dummy dai
by Rohit kumar 29 Jun '20

29 Jun '20

Add support for 384KHz sample rate and S24_3LE bitwidth for dummy dai. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/soc-utils.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 922eac9..364b248 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -86,12 +86,13 @@ static const struct snd_soc_component_driver dummy_codec = { .non_legacy_dai_naming = 1, }; -#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_RATES SNDRV_PCM_RATE_8000_384000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_U16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ SNDRV_PCM_FMTBIT_U24_LE | \ SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE | \ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH -next] ASoC: Documentation: fix reference to renamed source file
by Randy Dunlap 29 Jun '20

29 Jun '20

From: Randy Dunlap <rdunlap(a)infradead.org> sound/soc/soc-io.c was merged into sound/soc/soc-component.c, so fixup the Documentation to use the updated file name. Error: Cannot open file ../sound/soc/soc-io.c WARNING: kernel-doc '../scripts/kernel-doc -rst -enable-lineno ../sound/soc/soc-io.c' failed with return code 1 Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> --- Documentation/sound/kernel-api/alsa-driver-api.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20200626.orig/Documentation/sound/kernel-api/alsa-driver-api.rst +++ linux-next-20200626/Documentation/sound/kernel-api/alsa-driver-api.rst @@ -99,7 +99,7 @@ ASoC Core API .. kernel-doc:: include/sound/soc.h .. kernel-doc:: sound/soc/soc-core.c .. kernel-doc:: sound/soc/soc-devres.c -.. kernel-doc:: sound/soc/soc-io.c +.. kernel-doc:: sound/soc/soc-component.c .. kernel-doc:: sound/soc/soc-pcm.c .. kernel-doc:: sound/soc/soc-ops.c .. kernel-doc:: sound/soc/soc-compress.c

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[PATCH] ASoC: qcom: Drop HAS_DMA dependency to fix link failure
by Geert Uytterhoeven 29 Jun '20

29 Jun '20

When building on allyesconfig kernel for a NO_DMA=y platform (e.g. Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n, leading to a link failure: sound/soc/qcom/common.o: In function `qcom_snd_parse_of': common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port' While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845 don't, so the following warning is seen: WARNING: unmet direct dependencies detected for SND_SOC_QDSP6 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n] Selected by [y]: - SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] - SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y] Until recently, this warning was harmless (from a compile-testing point-of-view), but the new user of q6afe_is_rx_port() turned this into a hard failure. As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is safe to increase compile testing coverage. Hence fix the link failure by dropping the HAS_DMA dependency of SND_SOC_QDSP6. Fixes: a2120089251f1fe2 ("ASoC: qcom: common: set correct directions for dailinks") Fixes: 6b1687bf76ef84cb ("ASoC: qcom: add sdm845 sound card support") Fixes: a6f933f63f2ffdb2 ("ASoC: qcom: apq8096: Add db820c machine driver") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d87b5..92f51d0e9fe2bf02 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -72,7 +72,7 @@ config SND_SOC_QDSP6_ASM_DAI config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" - depends on QCOM_APR && HAS_DMA + depends on QCOM_APR select SND_SOC_QDSP6_COMMON select SND_SOC_QDSP6_CORE select SND_SOC_QDSP6_AFE -- 2.17.1

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[PATCH v2 0/3] ASoC: rt1015: fix compilation errors and warnings
by Tzung-Bi Shih 29 Jun '20

29 Jun '20

The series fixes compilation errors and warnings discovered in the thread https://patchwork.kernel.org/patch/11622319/. The original patch: (1) adds acpi.h for fixing implicit declaration of function 'ACPI_PTR' (2) sorts header inclusions in alphabetical The compilation errors and warnings are likely introduced by (2). The 1st and 2nd patches fix the newly discovered errors and warnings. The 3rd patch is the same as the original patch. Tzung-Bi Shih (3): ASoC: core: move definition of enum snd_soc_bias_level ASoC: dapm: declare missing structure prototypes ASoC: rt1015: add missing header inclusion include/sound/soc-dapm.h | 20 ++++++++++++++++++++ include/sound/soc.h | 18 ------------------ sound/soc/codecs/rt1015.c | 17 +++++++++-------- 3 files changed, 29 insertions(+), 26 deletions(-) -- 2.27.0.212.ge8ba1cc988-goog

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[PATCH v4 0/3] ALSA: compress: Document stream states and fix gapless SM
by Vinod Koul 29 Jun '20

29 Jun '20

Srini found issue with gapless implementation which prompted to look deeper into SM for compressed stream. So documenting SM was first step, so first two patches add that. Last patch fixes the issue by keeping track on partial_drain and then moving state to 'running' in snd_compr_drain_notify() for partial_drain case on success. While at it, noticed snd_compr_drain_notify() is lockless state change, so fixed that as well. I have tested this on Dragon board RB3, compressed audio works out of the box on that platform and Srini will send driver and fcplay patches for gapless soon. Changes in v4: - Add review tag by Charles - fix the lock deadlock pointed by Charles Changes in v3: - Add pause->stop->free transition - Add setup->free transition - remove running->free as that goes thru stop Changes in v2: - Added tested tag by Srini - Update compress SM with Free state and compr_stop() transitions Vinod Koul (3): ALSA: compress: document the compress audio state machine ALSA: compress: document the compress gapless audio state machine ALSA: compress: fix partial_drain completion state .../sound/designs/compress-offload.rst | 83 +++++++++++++++++++ include/sound/compress_driver.h | 10 ++- sound/core/compress_offload.c | 4 + 3 files changed, 96 insertions(+), 1 deletion(-) -- 2.26.2

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[PATCH v3 0/3] ALSA: compress: Document stream states and fix gapless SM
by Vinod Koul 29 Jun '20

29 Jun '20

Srini found issue with gapless implementation which prompted to look deeper into SM for compressed stream. So documenting SM was first step, so first two patches add that. Last patch fixes the issue by keeping track on partial_drain and then moving state to 'running' in snd_compr_drain_notify() for partial_drain case on success. While at it, noticed snd_compr_drain_notify() is lockless state change, so fixed that as well. I have tested this on Dragon board RB3, compressed audio works out of the box on that platform and Srini will send driver and fcplay patches for gapless soon. Changes in v3: - Add pause->stop->free transition - Add setup->free transition - remove running->free as that goes thru stop Changes in v2: - Added tested tag by Srini - Update compress SM with Free state and compr_stop() transitions Vinod Koul (3): ALSA: compress: document the compress audio state machine ALSA: compress: document the compress gapless audio state machine ALSA: compress: fix partial_drain completion state .../sound/designs/compress-offload.rst | 83 +++++++++++++++++++ include/sound/compress_driver.h | 12 ++- sound/core/compress_offload.c | 4 + 3 files changed, 98 insertions(+), 1 deletion(-) -- 2.26.2

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Re: [PATCH v2 08/21] ASoC: codecs: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 29 Jun '20

29 Jun '20

Hi Peter, Mark Thank you for your review > > static const struct snd_soc_dai_ops tfa9879_dai_ops = { > > .hw_params = tfa9879_hw_params, > > - .digital_mute = tfa9879_digital_mute, > > + .mute_stream = tfa9879_mute, > > .set_fmt = tfa9879_set_fmt, > > + .no_capture_mute = 1, > > }; (snip) > - .digital_mute = tfa9879_digital_mute, > + .mute_stream = tfa9879_mute_stream, Yes, indeed. I can fix it by additional patch if v2 was accepted, or, in v3 if needed. Is it OK for you, and Mark ? Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH 10/21] ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 29 Jun '20

29 Jun '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dai_digital_mute() is internally using both mute_stream() (1) or digital_mute() (2), but the difference between these 2 are only handling direction. We can merge digital_mute() into mute_stream int snd_soc_dai_digital_mute(xxx, int direction) { ... else if (dai->driver->ops->mute_stream) (1) return dai->driver->ops->mute_stream(xxx, direction); else if (direction == SNDRV_PCM_STREAM_PLAYBACK && dai->driver->ops->digital_mute) (2) return dai->driver->ops->digital_mute(xxx); ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This was missed on v2 patch-set. sound/soc/codecs/tas2552.c | 5 +++-- sound/soc/codecs/tas2562.c | 5 +++-- sound/soc/codecs/tas2770.c | 5 +++-- sound/soc/codecs/tas571x.c | 5 +++-- sound/soc/codecs/tas5720.c | 5 +++-- sound/soc/codecs/tas6424.c | 5 +++-- 6 files changed, 18 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 529c0fb93f9b..3a153526e47f 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -465,7 +465,7 @@ static int tas2552_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tas2552_mute(struct snd_soc_dai *dai, int mute) +static int tas2552_mute(struct snd_soc_dai *dai, int mute, int direction) { u8 cfg1_reg = 0; struct snd_soc_component *component = dai->component; @@ -519,7 +519,8 @@ static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .set_sysclk = tas2552_set_dai_sysclk, .set_fmt = tas2552_set_dai_fmt, .set_tdm_slot = tas2552_set_dai_tdm_slot, - .digital_mute = tas2552_mute, + .mute_stream = tas2552_mute, + .no_capture_mute = 1, }; /* Formats supported by TAS2552 driver. */ diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 7fae88655a0f..8721b0ecc403 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -334,7 +334,7 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int tas2562_mute(struct snd_soc_dai *dai, int mute) +static int tas2562_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; @@ -552,7 +552,8 @@ static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = { .hw_params = tas2562_hw_params, .set_fmt = tas2562_set_dai_fmt, .set_tdm_slot = tas2562_set_dai_tdm_slot, - .digital_mute = tas2562_mute, + .mute_stream = tas2562_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tas2562_dai[] = { diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 54c8135fe43c..4538b2d0216f 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -189,7 +189,7 @@ static const struct snd_soc_dapm_route tas2770_audio_map[] = { {"VSENSE", "Switch", "VMON"}, }; -static int tas2770_mute(struct snd_soc_dai *dai, int mute) +static int tas2770_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; int ret; @@ -530,10 +530,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, } static struct snd_soc_dai_ops tas2770_dai_ops = { - .digital_mute = tas2770_mute, + .mute_stream = tas2770_mute, .hw_params = tas2770_hw_params, .set_fmt = tas2770_set_fmt, .set_tdm_slot = tas2770_set_dai_tdm_slot, + .no_capture_mute = 1, }; #define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 5b7f9fcf6cbf..835a723ce5bc 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -301,7 +301,7 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream, TAS571X_SDI_FMT_MASK, val); } -static int tas571x_mute(struct snd_soc_dai *dai, int mute) +static int tas571x_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; u8 sysctl2; @@ -354,7 +354,8 @@ static int tas571x_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops tas571x_dai_ops = { .set_fmt = tas571x_set_dai_fmt, .hw_params = tas571x_hw_params, - .digital_mute = tas571x_mute, + .mute_stream = tas571x_mute, + .no_capture_mute = 1, }; diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index e159f839d928..139ac5e683bf 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -199,7 +199,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, return ret; } -static int tas5720_mute(struct snd_soc_dai *dai, int mute) +static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; int ret; @@ -604,7 +604,8 @@ static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = { .hw_params = tas5720_hw_params, .set_fmt = tas5720_set_dai_fmt, .set_tdm_slot = tas5720_set_dai_tdm_slot, - .digital_mute = tas5720_mute, + .mute_stream = tas5720_mute, + .no_capture_mute = 1, }; /* diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index aaba39295079..6198138e693a 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -252,7 +252,7 @@ static int tas6424_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tas6424_mute(struct snd_soc_dai *dai, int mute) +static int tas6424_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; struct tas6424_data *tas6424 = snd_soc_component_get_drvdata(component); @@ -382,7 +382,8 @@ static const struct snd_soc_dai_ops tas6424_speaker_dai_ops = { .hw_params = tas6424_hw_params, .set_fmt = tas6424_set_dai_fmt, .set_tdm_slot = tas6424_set_dai_tdm_slot, - .digital_mute = tas6424_mute, + .mute_stream = tas6424_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tas6424_dai[] = { -- 2.25.1

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[PATCH v2 0/7] ASoC: QCOM: Add support for SC7180 lpass variant
by Ajit Pandey 29 Jun '20

29 Jun '20

We've common ASoC cpu driver for QCOM LPASS soc varaints. This patch chain add support for new variant SC7180 soc by doing the required modification in existing common lpass-cpu driver. Below is a brief summary of patch series: PATCH v2 0001 ... 0004: Update lpass-cpu driver and documentation to make it more generic and support newer soc registers configuration. PATCH v2 0005 ... 0007: Add documentation and platform driver for newer SC7180 SOC variant. Ajit Pandey (7): Documentation: device-tree: sound: Update lpass-cpu driver binding ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-cpu: Make "ahbix-clk" an optional clock. ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers include: dt-bindings: sound: Add sc7180-lpass bindings header device-tree: bindings: sound: lpass-cpu: Add new compatible soc ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 ------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 154 +++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 10 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 100 ++++++-- sound/soc/qcom/lpass-cpu.c | 116 ++++++---- sound/soc/qcom/lpass-lpaif-reg.h | 203 ++++++++++------- sound/soc/qcom/lpass-platform.c | 86 ++++--- sound/soc/qcom/lpass-sc7180.c | 252 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 40 +++- 11 files changed, 790 insertions(+), 257 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- ('The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,\na Linux Foundation Collaborative Project')

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[PATCH alsa-lib 1/5] Fix invalid "Regex" Type in various Condition blocks
by Hans de Goede 28 Jun '20

28 Jun '20

Regex is not a valid Condition type, using it leads to errors like these: ALSA lib ucm_cond.c:300:(if_eval) unknown If.Condition.Type ALSA lib main.c:983:(snd_use_case_mgr_open) error: failed to import cht-bsw-rt5672 use case configuration -22 alsaucm: error failed to open sound card cht-bsw-rt5672: Invalid argument Replacing it with "RegexMatch" fixes this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/cht-bsw-rt5672/HiFi.conf | 4 ++-- ucm2/chtnau8824/HiFi.conf | 2 +- ucm2/chtrt5645/HiFi.conf | 6 +++--- ucm2/sof-soundwire/sof-soundwire.conf | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/ucm2/cht-bsw-rt5672/HiFi.conf b/ucm2/cht-bsw-rt5672/HiFi.conf index 394e45d..dc8ddd5 100644 --- a/ucm2/cht-bsw-rt5672/HiFi.conf +++ b/ucm2/cht-bsw-rt5672/HiFi.conf @@ -9,7 +9,7 @@ Define.HeadsetMic "yes" If.cfg-dmic1 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(SoMeThInK1)" # fixme! } @@ -21,7 +21,7 @@ If.cfg-dmic1 { If.cfg-dmic2 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(LENOVO.*ThinkPad8)" } diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf index 6b7624d..3ccd6b1 100644 --- a/ucm2/chtnau8824/HiFi.conf +++ b/ucm2/chtnau8824/HiFi.conf @@ -2,7 +2,7 @@ Define.Speaker "Speaker" If.cfg-mspk { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(cube-i1_TF|PIPO-W2S)" } diff --git a/ucm2/chtrt5645/HiFi.conf b/ucm2/chtrt5645/HiFi.conf index fb8636d..5d0e3bf 100644 --- a/ucm2/chtrt5645/HiFi.conf +++ b/ucm2/chtrt5645/HiFi.conf @@ -4,7 +4,7 @@ Define.DigitalMic "" If.cfg-dmic1 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(ASUSTeK.*T100HAN|ASUSTeK.*T101HA)" } @@ -16,7 +16,7 @@ If.cfg-dmic1 { If.cfg-dmic2 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(LENOVO.*LenovoMIIX320|MEDION.*Wingman)" } @@ -28,7 +28,7 @@ If.cfg-dmic2 { If.cfg-mspk { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(gpd-win-pocket-rt5645|TECLAST-X80Pro)" } diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index fe43d5f..2ba44be 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -38,7 +38,7 @@ DefineRegex { If.hs_init { Condition { - Type Regex + Type RegexMatch Regex "(rt5682|rt700|rt711)" String "${var:HeadsetCodec1}" } @@ -47,7 +47,7 @@ If.hs_init { If.mic_init { Condition { - Type Regex + Type RegexMatch Regex "(rt715)" String "${var:MicCodec1}" } -- 2.26.2

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