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

How to make ALSA relocatable in the filesystem?
by GitHub issues - opened 07 Mar '20

07 Mar '20

alsa-project/alsa-lib issue #34 was opened from probonopd: We want to bundle a private copy of ALSA with our application, so that the ALSA from the Linux distribution is not used at all. Yet it seems ALSA is relying on some absolute paths compiled in with no clear way to pass in a different location for ALSA at runtime. How to do this? Reference: https://github.com/mumble-voip/mumble/issues/3959#issuecomment-596095335 Issue URL : https://github.com/alsa-project/alsa-lib/issues/34 Repository URL: https://github.com/alsa-project/alsa-lib

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[GIT PULL] ASoC fixes for v5.6
by Mark Brown 07 Mar '20

07 Mar '20

The following changes since commit 96781fd941b39e1f78098009344ebcd7af861c67: ASoC: sun8i-codec: Fix setting DAI data format (2020-02-17 21:58:41 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.6-rc4 for you to fetch changes up to 3fb83cbee1de58fcd5d22f1db89460bb7c08b6e8: ASoC: wm8741: Fix typo in Kconfig prompt (2020-03-05 12:53:51 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.6 More fixes that have arrived since the merge window, spread out all over. There's a few things like the operation callback addition for rt1015 and the meson reset addition which add small new bits of functionality to fix non-working systems, they're all very small and for parts of newly added functionality. ---------------------------------------------------------------- Amadeusz Sławiński (1): ASoC: Intel: Skylake: Fix available clock counter incrementation Axel Lin (1): ASoC: wm8741: Fix typo in Kconfig prompt Charles Keepax (1): ASoC: dapm: Correct DAPM handling of active widgets during shutdown Dan Carpenter (1): ASoC: SOF: Fix snd_sof_ipc_stream_posn() Dan Murphy (2): ASoC: tas2562: Return invalid for when bitwidth is invalid ASoC: tas2562: Fix sample rate error message Dragos Tarcatu (2): ASoC: topology: Fix memleak in soc_tplg_link_elems_load() ASoC: topology: Fix memleak in soc_tplg_manifest_load() Jack Yu (2): ASoC: rt1015: add operation callback function for rt1015_dai[] ASoC: rt1015: modify pre-divider for sysclk Jerome Brunet (1): ASoC: meson: g12a: add tohdmitx reset Kuninori Morimoto (2): ASoC: soc-pcm/soc-compress: don't use snd_soc_dapm_stream_stop() ASoC: soc-component: tidyup snd_soc_pcm_component_sync_stop() Matthias Reichl (1): ASoC: pcm512x: Fix unbalanced regulator enable call in probe error path Olivier Moysan (1): ASoC: stm32: sai: manage rebind issue Pierre-Louis Bossart (1): ASoC: soc-core: fix for_rtd_codec_dai_rollback() macro Takashi Iwai (3): ASoC: intel: skl: Fix pin debug prints ASoC: intel: skl: Fix possible buffer overflow in debug outputs ASoC: pcm: Fix possible buffer overflow in dpcm state sysfs output include/sound/soc.h | 2 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/pcm512x.c | 8 +++++--- sound/soc/codecs/rt1015.c | 3 ++- sound/soc/codecs/tas2562.c | 5 +++-- sound/soc/intel/skylake/skl-debug.c | 32 +++++++++++++++++--------------- sound/soc/intel/skylake/skl-ssp-clk.c | 4 +++- sound/soc/meson/g12a-tohdmitx.c | 6 ++++++ sound/soc/soc-component.c | 2 +- sound/soc/soc-compress.c | 2 +- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-pcm.c | 18 +++++++++--------- sound/soc/soc-topology.c | 17 ++++++++++------- sound/soc/sof/ipc.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 15 files changed, 71 insertions(+), 52 deletions(-)

2 1

[alsa-devel] [PATCH alsa-ucm-conf] bytcht-es8316: Fix missing including of HeadPhones.conf after ucm2 conversion
by Hans de Goede 07 Mar '20

07 Mar '20

The conversion to ucm2 format missed adding an include for: codecs/es8316/HeadPhones.conf Leading to no sound on the headphones output, this commit adds the missing include fixing this. Cc: youling 257 <youling257(a)gmail.com> Reported-by: youling 257 <youling257(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/bytcht-es8316/HiFi-Components.conf | 2 ++ ucm2/bytcht-es8316/HiFi-LongName.conf | 2 ++ 2 files changed, 4 insertions(+) diff --git a/ucm2/bytcht-es8316/HiFi-Components.conf b/ucm2/bytcht-es8316/HiFi-Components.conf index 314d355..c40bd49 100644 --- a/ucm2/bytcht-es8316/HiFi-Components.conf +++ b/ucm2/bytcht-es8316/HiFi-Components.conf @@ -20,6 +20,8 @@ If.mono { } } +<codecs/es8316/HeadPhones.conf> + If.in1 { Condition { Type String diff --git a/ucm2/bytcht-es8316/HiFi-LongName.conf b/ucm2/bytcht-es8316/HiFi-LongName.conf index ea7d1c3..03cf17b 100644 --- a/ucm2/bytcht-es8316/HiFi-LongName.conf +++ b/ucm2/bytcht-es8316/HiFi-LongName.conf @@ -20,6 +20,8 @@ If.mono { } } +<codecs/es8316/HeadPhones.conf> + If.in1 { Condition { Type String -- 2.24.1

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[PATCH v4 0/8] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 07 Mar '20

07 Mar '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for change DT binding asrc-width to asrc-format Shengjiu Wang (8): ASoC: dt-bindings: fsl_asrc: Change asrc-width to asrc-format ARM: dts: imx6qdl: Change asrc-width to asrc-format ASoC: fsl-asoc-card: Change asrc-width to asrc-format ASoC: fsl_asrc: Change asrc_width to asrc_format ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI and platform drivers changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 +- .../devicetree/bindings/sound/fsl,easrc.yaml | 96 + arch/arm/boot/dts/imx6qdl.dtsi | 2 +- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 9 +- sound/soc/fsl/fsl_asrc.c | 282 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 105 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2111 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 12 files changed, 3169 insertions(+), 232 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

5 14

[PATCH] ASoC: wcd9335: fix address map representation
by Srinivas Kandagatla 06 Mar '20

06 Mar '20

slimbus addresses are 16 bit wide, masking page numbers to wcd register at offset of 12 will limit the number for pages. So it becomes impossible to write to page 0x10 registers. Remove masking 0x800 (slimbus address range) from register address and making use of window parameters in regmap config should fix it and also will represent the registers exactly inline with Datasheet. Remove this unnessary masking and make the registers be inline with datasheet. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd9335.c | 18 +++++++++--------- sound/soc/codecs/wcd9335.h | 7 ++++--- 2 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f11ffa28683b..700cc1212770 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4926,11 +4926,11 @@ static const struct regmap_range_cfg wcd9335_ranges[] = { .name = "WCD9335", .range_min = 0x0, .range_max = WCD9335_MAX_REGISTER, - .selector_reg = WCD9335_REG(0x0, 0), + .selector_reg = WCD9335_SEL_REGISTER, .selector_mask = 0xff, .selector_shift = 0, - .window_start = 0x0, - .window_len = 0x1000, + .window_start = 0x800, + .window_len = 0x100, }, }; @@ -4968,12 +4968,12 @@ static const struct regmap_range_cfg wcd9335_ifc_ranges[] = { { .name = "WCD9335-IFC-DEV", .range_min = 0x0, - .range_max = WCD9335_REG(0, 0x7ff), - .selector_reg = WCD9335_REG(0, 0x0), - .selector_mask = 0xff, + .range_max = WCD9335_MAX_REGISTER, + .selector_reg = WCD9335_SEL_REGISTER, + .selector_mask = 0xfff, .selector_shift = 0, - .window_start = 0x0, - .window_len = 0x1000, + .window_start = 0x800, + .window_len = 0x400, }, }; @@ -4981,7 +4981,7 @@ static struct regmap_config wcd9335_ifc_regmap_config = { .reg_bits = 16, .val_bits = 8, .can_multi_write = true, - .max_register = WCD9335_REG(0, 0x7FF), + .max_register = WCD9335_MAX_REGISTER, .ranges = wcd9335_ifc_ranges, .num_ranges = ARRAY_SIZE(wcd9335_ifc_ranges), }; diff --git a/sound/soc/codecs/wcd9335.h b/sound/soc/codecs/wcd9335.h index 4d9be2496c30..72060824c743 100644 --- a/sound/soc/codecs/wcd9335.h +++ b/sound/soc/codecs/wcd9335.h @@ -8,9 +8,9 @@ * in slimbus mode the reg base starts from 0x800 * in i2s/i2c mode the reg base is 0x0 */ -#define WCD9335_REG(pg, r) ((pg << 12) | (r) | 0x800) +#define WCD9335_REG(pg, r) ((pg << 8) | (r)) #define WCD9335_REG_OFFSET(r) (r & 0xFF) -#define WCD9335_PAGE_OFFSET(r) ((r >> 12) & 0xFF) +#define WCD9335_PAGE_OFFSET(r) ((r >> 8) & 0xFF) /* Page-0 Registers */ #define WCD9335_PAGE0_PAGE_REGISTER WCD9335_REG(0x00, 0x000) @@ -600,7 +600,8 @@ #define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE BIT(0) #define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_DISABLE 0 #define WCD9335_CDC_TOP_TOP_CFG1 WCD9335_REG(0x0d, 0x082) -#define WCD9335_MAX_REGISTER WCD9335_REG(0x80, 0x0FF) +#define WCD9335_MAX_REGISTER 0xffff +#define WCD9335_SEL_REGISTER 0x800 /* SLIMBUS Slave Registers */ #define WCD9335_SLIM_PGD_PORT_INT_EN0 WCD9335_REG(0, 0x30) -- 2.21.0

2 1

[PATCH 0/2] ASoC: wcd934x: minor fixes
by Srinivas Kandagatla 06 Mar '20

06 Mar '20

This patchset has 1 fix and a header cleanup found during recent testing. Srinivas Kandagatla (2): ASoC: wcd934x: fix High Accuracy Buck enable ASoC: wcd934x: remove unused headers sound/soc/codecs/wcd934x.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) -- 2.21.0

2 4

Applied "ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates" to the asoc tree
by Mark Brown 06 Mar '20

06 Mar '20

The patch ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4769bfb9dada678b31a2ec275372624dbfeed9d1 Mon Sep 17 00:00:00 2001 From: Samuel Holland <samuel(a)sholland.org> Date: Wed, 4 Mar 2020 23:11:41 -0600 Subject: [PATCH] ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates It can be useful to derive min/max rates of a snd_pcm_hardware without having a snd_pcm_runtime, such as before constructing an ASoC DAI link. Create a new helper that takes a pointer to a snd_pcm_hardware directly, and refactor the original function as a wrapper around it, to avoid needing to update any call sites. Signed-off-by: Samuel Holland <samuel(a)sholland.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200305051143.60691-2-samuel@sholland.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/pcm.h | 9 ++++++++- sound/core/pcm_misc.c | 18 +++++++++--------- 2 files changed, 17 insertions(+), 10 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2628246b76fa..f7a95b711100 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1127,7 +1127,14 @@ snd_pcm_kernel_readv(struct snd_pcm_substream *substream, return __snd_pcm_lib_xfer(substream, bufs, false, frames, true); } -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw); + +static inline int +snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) +{ + return snd_pcm_hw_limit_rates(&runtime->hw); +} + unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a, diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index a6a541511534..5dd2e5335900 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -474,32 +474,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int EXPORT_SYMBOL(snd_pcm_format_set_silence); /** - * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields - * @runtime: the runtime instance + * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields + * @hw: the pcm hw instance * * Determines the rate_min and rate_max fields from the rates bits of - * the given runtime->hw. + * the given hw. * * Return: Zero if successful. */ -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw) { int i; for (i = 0; i < (int)snd_pcm_known_rates.count; i++) { - if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_min = snd_pcm_known_rates.list[i]; + if (hw->rates & (1 << i)) { + hw->rate_min = snd_pcm_known_rates.list[i]; break; } } for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) { - if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_max = snd_pcm_known_rates.list[i]; + if (hw->rates & (1 << i)) { + hw->rate_max = snd_pcm_known_rates.list[i]; break; } } return 0; } -EXPORT_SYMBOL(snd_pcm_limit_hw_rates); +EXPORT_SYMBOL(snd_pcm_hw_limit_rates); /** * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit -- 2.20.1

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Applied "ASoC: pcm: Export parameter intersection logic" to the asoc tree
by Mark Brown 06 Mar '20

06 Mar '20

The patch ASoC: pcm: Export parameter intersection logic has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5854a46486ad5b8d73766735fb0d77f05956b22c Mon Sep 17 00:00:00 2001 From: Samuel Holland <samuel(a)sholland.org> Date: Wed, 4 Mar 2020 23:11:42 -0600 Subject: [PATCH] ASoC: pcm: Export parameter intersection logic The logic to calculate the subset of stream parameters supported by all DAIs associated with a PCM stream is nontrivial. Export a helper function so it can be used to set up simple codec2codec DAI links. Signed-off-by: Samuel Holland <samuel(a)sholland.org> Link: https://lore.kernel.org/r/20200305051143.60691-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 56 ++++++++++++++++++++++++++++++--------------- 2 files changed, 41 insertions(+), 18 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 81e5d17be935..9543d9246ca4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -471,6 +471,9 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd); void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream); void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream); +int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hardware *hw, int stream); + int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e3a2c4f7757b..de4226357e2b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -587,11 +587,18 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) soc_pcm_set_msb(substream, cpu_bits); } -static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) +/** + * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream + * @rtd: ASoC PCM runtime + * @hw: PCM hardware parameters (output) + * @stream: Direction of the PCM stream + * + * Calculates the subset of stream parameters supported by all DAIs + * associated with the PCM stream. + */ +int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hardware *hw, int stream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_pcm_hardware *hw = &runtime->hw; - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; struct snd_soc_pcm_stream *codec_stream; @@ -602,7 +609,6 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX; unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX; u64 formats = ULLONG_MAX; - int stream = substream->stream; int i; /* first calculate min/max only for CPUs in the DAI link */ @@ -613,12 +619,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * Otherwise, since the rate, channel, and format values will * zero in that case, we would have no usable settings left, * causing the resulting setup to fail. - * At least one CPU should match, otherwise we should have - * bailed out on a higher level, since there would be no - * CPU to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(cpu_dai, - substream->stream)) + if (!snd_soc_dai_stream_valid(cpu_dai, stream)) continue; cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream); @@ -640,9 +642,6 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * Otherwise, since the rate, channel, and format values will * zero in that case, we would have no usable settings left, * causing the resulting setup to fail. - * At least one CODEC should match, otherwise we should have - * bailed out on a higher level, since there would be no - * CODEC to support the transfer direction in that case. */ if (!snd_soc_dai_stream_valid(codec_dai, stream)) continue; @@ -657,6 +656,10 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates); } + /* Verify both a valid CPU DAI and a valid CODEC DAI were found */ + if (!chan_min || !cpu_chan_min) + return -EINVAL; + /* * chan min/max cannot be enforced if there are multiple CODEC DAIs * connected to CPU DAI(s), use CPU DAI's directly and let @@ -670,18 +673,35 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* finally find a intersection between CODECs and CPUs */ hw->channels_min = max(chan_min, cpu_chan_min); hw->channels_max = min(chan_max, cpu_chan_max); - if (hw->formats) - hw->formats &= formats; - else - hw->formats = formats; + hw->formats = formats; hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates); - snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_limit_rates(hw); hw->rate_min = max(hw->rate_min, cpu_rate_min); hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max); hw->rate_max = min_not_zero(hw->rate_max, rate_max); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_runtime_calc_hw); + +static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) +{ + struct snd_pcm_hardware *hw = &substream->runtime->hw; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + u64 formats = hw->formats; + + /* + * At least one CPU and one CODEC should match. Otherwise, we should + * have bailed out on a higher level, since there would be no CPU or + * CODEC to support the transfer direction in that case. + */ + snd_soc_runtime_calc_hw(rtd, hw, substream->stream); + + if (formats) + hw->formats &= formats; } static int soc_pcm_components_open(struct snd_pcm_substream *substream) -- 2.20.1

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[PATCH v4 0/3] simple-audio-card codec2codec support
by Samuel Holland 06 Mar '20

06 Mar '20

We are currently using simple-audio-card on the Allwinner A64 SoC. The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the CPU, the modem, and Bluetooth. Adding support for the secondary AIFs requires adding codec2codec DAI links. Since the modem and bt-sco codec DAI drivers only have one set of possible PCM parameters (namely, 8kHz mono S16LE), there's no real need for a machine driver to specify the DAI link configuration. The parameters for these "simple" DAI links can be chosen automatically. This series adds codec2codec DAI link support to simple-audio-card. Codec to codec links are automatically detected when all DAIs in the link belong to codec components. I tried to reuse as much code as possible, so the first two patches refactor a couple of helper functions to be more generic. The last patch adds the new feature and its documentation. Changes in v4: - Rebased on top of asoc/for-next, several changes to patch 2 - Removed unused variable from patch 3 Changes in v3: - Update use of for_each_rtd_components for v5.6 Changes in v2: - Drop patch 1 as it was merged - Automatically detect codec2codec links instead of using a DT property Samuel Holland (3): ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec2codec DAI links Documentation/sound/soc/codec-to-codec.rst | 9 +++- include/sound/pcm.h | 9 +++- include/sound/soc.h | 3 ++ sound/core/pcm_misc.c | 18 +++---- sound/soc/generic/simple-card-utils.c | 48 ++++++++++++++++++ sound/soc/soc-pcm.c | 59 ++++++++++++++-------- 6 files changed, 114 insertions(+), 32 deletions(-) -- 2.24.1

3 5

Applied "ASoC: soc-pcm: move dpcm_fe_dai_close()" to the asoc tree
by Mark Brown 06 Mar '20

06 Mar '20

The patch ASoC: soc-pcm: move dpcm_fe_dai_close() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 30fca26f8e2277ccd14fe3277a330b4f21cadca7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 6 Mar 2020 10:09:44 +0900 Subject: [PATCH] ASoC: soc-pcm: move dpcm_fe_dai_close() move dpcm_fe_dai_close() next to dpcm_fe_dai_open(). This is prepare for dpcm_fe_dai_open() cleanup Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/87pndqp9uv.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e3a2c4f7757b..3686dda097e2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2978,6 +2978,26 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) return ret; } +static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = fe_substream->stream, ret; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + ret = dpcm_fe_dai_shutdown(fe_substream); + + /* mark FE's links ready to prune */ + for_each_dpcm_be(fe, stream, dpcm) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + + fe->dpcm[stream].runtime = NULL; + mutex_unlock(&fe->card->mutex); + return ret; +} + static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; @@ -3017,26 +3037,6 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) return ret; } -static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) -{ - struct snd_soc_pcm_runtime *fe = fe_substream->private_data; - struct snd_soc_dpcm *dpcm; - int stream = fe_substream->stream, ret; - - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - ret = dpcm_fe_dai_shutdown(fe_substream); - - /* mark FE's links ready to prune */ - for_each_dpcm_be(fe, stream, dpcm) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - - fe->dpcm[stream].runtime = NULL; - mutex_unlock(&fe->card->mutex); - return ret; -} - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { -- 2.20.1

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