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

Applied "ASoC: sof: sof-audio: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: sof: sof-audio: use for_each_pcm_streams() macro 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 525c4107da8c0a86aa3548dc6e1d0014749e95f7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:47 +0900 Subject: [PATCH] ASoC: sof: sof-audio: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87tv3paa8i.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/sof-audio.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 75f2ef2bd94b..fc4ed2a8a914 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -23,7 +23,7 @@ bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev) int dir; list_for_each_entry(spcm, &sdev->pcm_list, list) { - for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + for_each_pcm_streams(dir) { substream = spcm->stream[dir].substream; if (!substream || !substream->runtime) continue; @@ -71,7 +71,7 @@ int sof_set_hw_params_upon_resume(struct device *dev) * have been suspended. */ list_for_each_entry(spcm, &sdev->pcm_list, list) { - for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + for_each_pcm_streams(dir) { /* * do not reset hw_params upon resume for streams that * were kept running during suspend @@ -319,16 +319,11 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp, int dir; list_for_each_entry(spcm, &sdev->pcm_list, list) { - dir = SNDRV_PCM_STREAM_PLAYBACK; - if (spcm->stream[dir].comp_id == comp_id) { - *direction = dir; - return spcm; - } - - dir = SNDRV_PCM_STREAM_CAPTURE; - if (spcm->stream[dir].comp_id == comp_id) { - *direction = dir; - return spcm; + for_each_pcm_streams(dir) { + if (spcm->stream[dir].comp_id == comp_id) { + *direction = dir; + return spcm; + } } } -- 2.20.1

1 0

Applied "ALSA: usx2y: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ALSA: usx2y: use for_each_pcm_streams() macro 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 ffd11d1e7ad4602d5d1c2b4517ad316f7587d4d9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:51 +0900 Subject: [PATCH] ALSA: usx2y: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87sgj9aa8e.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/usb/usx2y/usbusx2yaudio.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 772f6f3ccbb1..37d290fe9d43 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -906,11 +906,12 @@ static const struct snd_pcm_ops snd_usX2Y_pcm_ops = */ static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream) { - kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]); - usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL; + int stream; - kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]); - usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL; + for_each_pcm_streams(stream) { + kfree(usX2Y_substream[stream]); + usX2Y_substream[stream] = NULL; + } } static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm) -- 2.20.1

1 0

[PATCH v2 0/2] INtroduce the TLV320ADCx140 codec family
by Dan Murphy 18 Feb '20

18 Feb '20

Hello Introducing the Texas Instruments TLV320ADCx140 quad channel audio ADC. This device supports 4 analog inputs, 8 digital inputs or a combination of analog and digital microphones. TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Dan Murphy (2): dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family .../bindings/sound/tlv320adcx140.yaml | 83 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adcx140.c | 851 ++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 130 +++ 5 files changed, 1075 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h -- 2.25.0

2 6

[alsa-devel] [PATCH RFC] ASoC: core: ensure component names are unique
by Jerome Brunet 18 Feb '20

18 Feb '20

Make sure each ASoC component is registered with a unique name. The component is derived from the device name. If a device registers more than one component, the component names will be the same. This usually brings up a warning about the debugfs directory creation of the component since directory already exists. In such case, start numbering the component of the device so the names don't collide anymore. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- * Here is an example of /sys/kernel/debug/asoc/components before the change: hdmi-audio-codec.3.auto c1105400.audio-controller c1105400.audio-controller c1105400.audio-controller c8832000.audio-controller analog-amplifier snd-soc-dummy snd-soc-dummy * and after: hdmi-audio-codec.3.auto c1105400.audio-controller-2 c1105400.audio-controller-1 c1105400.audio-controller c8832000.audio-controller analog-amplifier snd-soc-dummy-1 snd-soc-dummy I wanted to keep the name as they were for the devices registering only one component but we could also start the numbering from the first component instead of the second. sound/soc/soc-core.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 03b87427faa7..6a58a8f6e3c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2446,6 +2446,33 @@ static int snd_soc_register_dais(struct snd_soc_component *component, return ret; } +static char *snd_soc_component_unique_name(struct device *dev, + struct snd_soc_component *component) +{ + struct snd_soc_component *pos; + int count = 0; + char *name, *unique; + + name = fmt_single_name(dev, &component->id); + if (!name) + return name; + + /* Count the number of components registred by the device */ + for_each_component(pos) { + if (dev == pos->dev) + count++; + } + + /* Keep naming as it is for the 1st component */ + if (!count) + return name; + + unique = devm_kasprintf(dev, GFP_KERNEL, "%s-%d", name, count); + devm_kfree(dev, name); + + return unique; +} + static int snd_soc_component_initialize(struct snd_soc_component *component, const struct snd_soc_component_driver *driver, struct device *dev) { @@ -2454,7 +2481,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->card_list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = snd_soc_component_unique_name(dev, component); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.24.1

5 11

[RESEND PATCH] ASoC: tas2562: Return invalid for when bitwidth is invalid
by Dan Murphy 18 Feb '20

18 Feb '20

If the bitwidth passed in to the set_bitwidth function is not supported then return an error. Fixes: 29b74236bd57 ("ASoC: tas2562: Introduce the TAS2562 amplifier") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 729acd874c48..5b4803a76719 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -215,7 +215,8 @@ static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) break; default: - dev_info(tas2562->dev, "Not supported params format\n"); + dev_info(tas2562->dev, "Unsupported bitwidth format\n"); + return -EINVAL; } ret = snd_soc_component_update_bits(tas2562->component, -- 2.25.0

1 0

[PATCH] ASoC: tas2562: Return invalid for when bitwidth is invalid
by Dan Murphy 18 Feb '20

18 Feb '20

If the bitwidth passed in to the set_bitwidth function is not supported then return an error. 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z=SaGP7KbI}*Z(Za@V3a;5Q)k6DEa?D<jYH5(wCq^dnU3P`QEmCHdz~;Zzea&OZ`8h zfaadmm-|Ta|JfK8{*&`l>dA4uAG#zW_2qt3(;R?EJ4}-OmHLv>d#hDH``%Mws>phz z0jVeXVPpuC^{e9ju2kr!MEj+@Dl>x8`#p&W@ocHom+vpKK$^2sIrpRt(OO|pzQ53Y tM#ClBFXJc8Z`pQav#H`*GfQVTtx47^F<M^Z>CdN)DWp>-$^Pc+|2IHPD@*_Y literal 0 HcmV?d00001 diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 729acd874c48..5b4803a76719 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -215,7 +215,8 @@ static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) break; default: - dev_info(tas2562->dev, "Not supported params format\n"); + dev_info(tas2562->dev, "Unsupported bitwidth format\n"); + return -EINVAL; } ret = snd_soc_component_update_bits(tas2562->component, -- 2.25.0

2 2

[asoc:for-5.7 21/76] wcd9335.c:undefined reference to `slim_stream_prepare'
by kbuild test robot 18 Feb '20

18 Feb '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.7 head: 3cd23f021e2e5f3350125abcb39f12430df87d06 commit: ea00d95200d02ece71f5814d41b14f2eb16d598b [21/76] ASoC: Use imply for SND_SOC_ALL_CODECS config: x86_64-randconfig-s2-20200217 (attached as .config) compiler: gcc-7 (Debian 7.5.0-5) 7.5.0 reproduce: git checkout ea00d95200d02ece71f5814d41b14f2eb16d598b # save the attached .config to linux build tree make ARCH=x86_64 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): ld: sound/soc/codecs/ab8500-codec.o: in function `ab8500_codec_read_reg': ab8500-codec.c:(.text+0x479): undefined reference to `abx500_get_register_interruptible' ld: sound/soc/codecs/ab8500-codec.o: in function `ab8500_codec_probe': ab8500-codec.c:(.text+0xdf1): undefined reference to `abx500_get_register_interruptible' ld: ab8500-codec.c:(.text+0xe18): undefined reference to `abx500_set_register_interruptible' ld: sound/soc/codecs/ab8500-codec.o: in function `ab8500_codec_write_reg': ab8500-codec.c:(.text+0x4b1): undefined reference to `abx500_set_register_interruptible' ld: sound/soc/codecs/mc13783.o: in function `mc13783_probe': mc13783.c:(.text+0x24c): undefined reference to `mc13xxx_reg_write' ld: mc13783.c:(.text+0x25e): undefined reference to `mc13xxx_reg_write' ld: mc13783.c:(.text+0x270): undefined reference to `mc13xxx_reg_write' ld: mc13783.c:(.text+0x282): undefined reference to `mc13xxx_reg_write' ld: mc13783.c:(.text+0x294): undefined reference to `mc13xxx_reg_write' ld: sound/soc/codecs/mc13783.o:mc13783.c:(.text+0x2a6): more undefined references to `mc13xxx_reg_write' follow ld: sound/soc/codecs/mc13783.o: in function `mc13783_probe': mc13783.c:(.text+0x2c5): undefined reference to `mc13xxx_reg_rmw' ld: mc13783.c:(.text+0x2e4): undefined reference to `mc13xxx_reg_rmw' ld: sound/soc/codecs/mc13783.o: in function `mc13783_remove': mc13783.c:(.text+0x212): undefined reference to `mc13xxx_reg_rmw' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_trigger': >> wcd9335.c:(.text+0x323): undefined reference to `slim_stream_prepare' >> ld: wcd9335.c:(.text+0x32f): undefined reference to `slim_stream_enable' >> ld: wcd9335.c:(.text+0x345): undefined reference to `slim_stream_unprepare' >> ld: wcd9335.c:(.text+0x351): undefined reference to `slim_stream_disable' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_status': >> wcd9335.c:(.text+0x1970): undefined reference to `of_slim_get_device' >> ld: wcd9335.c:(.text+0x1995): undefined reference to `slim_get_logical_addr' >> ld: wcd9335.c:(.text+0x19b2): undefined reference to `__regmap_init_slimbus' ld: wcd9335.c:(.text+0x19f3): undefined reference to `__regmap_init_slimbus' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_hw_params': >> wcd9335.c:(.text+0x348f): undefined reference to `slim_stream_allocate' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_driver_init': >> wcd9335.c:(.init.text+0x3c): undefined reference to `__slim_driver_register' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_driver_exit': >> wcd9335.c:(.exit.text+0x8): undefined reference to `slim_driver_unregister' ld: sound/soc/codecs/wcd934x.o: in function `wcd934x_trigger': >> wcd934x.c:(.text+0x30d): undefined reference to `slim_stream_prepare' >> ld: wcd934x.c:(.text+0x319): undefined reference to `slim_stream_enable' >> ld: wcd934x.c:(.text+0x32f): undefined reference to `slim_stream_unprepare' >> ld: wcd934x.c:(.text+0x33b): undefined reference to `slim_stream_disable' ld: sound/soc/codecs/wcd934x.o: in function `wcd934x_hw_params': >> wcd934x.c:(.text+0x2620): undefined reference to `slim_stream_allocate' ld: sound/soc/codecs/wcd934x.o: in function `wcd934x_codec_probe': >> wcd934x.c:(.text+0x27ee): undefined reference to `of_slim_get_device' >> ld: wcd934x.c:(.text+0x2818): undefined reference to `slim_get_logical_addr' >> ld: wcd934x.c:(.text+0x2836): undefined reference to `__regmap_init_slimbus' --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

3 2

[alsa-devel] [PATCH v4 00/18] ASoC: soc-pcm cleanup step3
by Kuninori Morimoto 18 Feb '20

18 Feb '20

Hi Mark These are v4 of soc-pcm cleanup step3. Main change from v3 is that - tidyup Subject on [10/18] patch - added Takashi's Reviewed-by / Acked-by on [10/18], [18/18] Kuninori Morimoto (18): 1) ASoC: soc-pcm: move dai_get_widget() 2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() 3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() 4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() 5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() 6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() 7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code 8) ASoC: soc-pcm: use goto and remove multi return 9) ASoC: soc-pcm: merge playback/cature_active into stream_active 10) ASoC: soc.h: add for_each_pcm_streams() 11) ASoC: soc-core: use for_each_pcm_streams() macro 12) ASoC: soc-pcm: use for_each_pcm_streams() macro 13) ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro 14) ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro 15) ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro 16) ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro 17) ASoC: sof: sof-audio: use for_each_pcm_streams() macro 18) ALSA: usx2y: use for_each_pcm_streams() macro include/sound/pcm.h | 5 + include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 - sound/soc/codecs/cs4271.c | 4 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/soc-core.c | 48 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 +- sound/soc/soc-pcm.c | 327 +++++++++----------------- sound/soc/sof/sof-audio.c | 19 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 12 files changed, 162 insertions(+), 284 deletions(-) -- 2.17.1

2 19

[alsa-devel] [PATCH v4 00/11] ASoC: SOF: Data probing
by Cezary Rojewski 18 Feb '20

18 Feb '20

This set of patches achieves few goals in order to enable data probing feature for audio DSP. First, provide new and alter existing interfaces (page allocation, runtime flow adjustments) to make them compress friendly. For HDA part, work has been done to account for compress streams when servicing IRQs, setting up BDLs and assigning DMAs. Finally, the end goal which are the probe APIs and usage itself. Probes can be treated as endpoints which allow for data extraction from or injection to target module - a great ally when debugging problematic audio issues such as distortions, glitches or gaps. Compress streams are a weapon of choice here to provide a lightweight implementation. While all available IPCs have been defined, current implementation covers extraction only, with injection scheduled for a later date. Initial review and development of probes can be found under: https://github.com/thesofproject/linux/pull/1276 with the hda-compress-enable set of patches being separated and reviewed on: https://github.com/thesofproject/linux/pull/1571 Tested on CML-U with rt5682 i2s board. Changes in v4: - No changes to ALSA core and hda patches - 'probe' source added in patch "ASoC: SOF: Implement Probe IPC API" now only compiled if SND_SOC_SOF_DEBUG_PROBES is enabled. Said kconfig initial declaration has been moved to that patch too - 'extractor_stream_tag' field now decorated with _DEBUG_PROBES to account for the change above - Debugfs probe handlers now check if extractor stream is running before continuing. To achieve this, 'extractor_stream_tag' is both initialized and set to SOF_PROBE_INVALID_NODE_ID on sof_probe_compr_free(). Suggestion provided by Pierre - Reverted SND_SOC_SOF_HDA_LINK requirement-removal change for HDA probes. Code requires SND_HDA_EXT_CORE in order to function on Intel platforms - Relocated probe-compress related code from /sof/pcm.c directly to /sof/compress.c as requested by Daniel Changes in v3: - No changes to ALSA core and hda patches - Addressed sparse and all doc related warnings as reported by Pierre - Moved _DEBUG_PROBES config outside of _DEVELOPER_SUPPORT block - Renamed 'extractor' field to 'extractor_stream_tag' as requested - Relocated 'extractor_stream_tag' declaration from patch 07 to 08 Changes in v2: - No changes to ALSA core and hda patches - Removed "ASoC: Intel: sof_rt5682: Add compress probe DAI links" from the patchset list as requested by Pierre - updated copyright header for newly added files (dates) - probes over HDA no longer require SND_SOC_SOF_HDA_LINK enabled - renamed debugfs probe functions as requested by Pierre - probe IPC API has been updated to align with newest SOF FW & probe debug app (struct_size macro has been enlisted to make the size calculations transparent). This targets only "ASoC: SOF: Implement Probe IPC API" patch Cezary Rojewski (11): ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: SOF: Intel: Account for compress streams when servicing IRQs ASoC: SOF: Implement Probe IPC API ASoC: SOF: Generic probe compress operations ASoC: SOF: Intel: Probe compress operations ASoC: SOF: Provide probe debugfs support ASoC: SOF: Intel: Add Probe compress CPU DAIs include/sound/compress_driver.h | 40 +++- include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 2 + include/sound/sof/header.h | 11 ++ sound/core/compress_offload.c | 42 +++++ sound/hda/ext/hdac_ext_stream.c | 49 ++++- sound/hda/hdac_controller.c | 4 +- sound/hda/hdac_stream.c | 52 ++++-- sound/soc/sof/Kconfig | 9 + sound/soc/sof/Makefile | 1 + sound/soc/sof/compress.c | 146 +++++++++++++++ sound/soc/sof/compress.h | 31 ++++ sound/soc/sof/core.c | 6 + sound/soc/sof/debug.c | 226 +++++++++++++++++++++++ sound/soc/sof/intel/Kconfig | 10 + sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 + sound/soc/sof/intel/cnl.c | 9 + sound/soc/sof/intel/hda-compress.c | 132 +++++++++++++ sound/soc/sof/intel/hda-dai.c | 28 +++ sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/intel/hda-stream.c | 26 ++- sound/soc/sof/intel/hda.h | 30 +++ sound/soc/sof/ops.h | 43 +++++ sound/soc/sof/pcm.c | 7 + sound/soc/sof/probe.c | 286 +++++++++++++++++++++++++++++ sound/soc/sof/probe.h | 85 +++++++++ sound/soc/sof/sof-priv.h | 25 +++ 28 files changed, 1277 insertions(+), 39 deletions(-) create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/intel/hda-compress.c create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h -- 2.17.1

2 15

Questions about event implementation in ALSA Sequencer
by Takashi Sakamoto 18 Feb '20

18 Feb '20

Hi, I'm working for alsa-gobject[1] so that it supports API for ALSA Sequencer. At present I've mostly done it just with direct dispatch support[2] (thus transmission via queue is for my later work). Then I have some questions about the design of event in ALSA Sequencer. I'd like to ask for some advices (mainly Iwai-san, perhaps). 1. use case of SNDRV_SEQ_EVENT_LENGTH_VARUSR in 'struct snd_seq_event' In my understanding, the flag is used for a case that sender transmits the value of pointer itself and its length to the receiver in the shape of 'struct snd_seq_ev_ext'. I assume that two use cases. If the sender and receiver are in the same process, the event is a request for the receiver to operate data in the same VMA. If the sender and receiver are in different processes, the event is a request for pointer-based calculation between the peer. If the above understanding is correct, it's hard to represent this type of event for g-i interface because g-i is the object-based framework. Any raw pointer without explicit type is hard to be exposed to g-i applications as long as I know, and it's going to be unsupported, perhaps. 2. event data type corresponding to event type Some event types are expected to specific data type; e.g. SNDRV_SEQ_EVENT_NOTE is for 'struct snd_seq_ev_note' and SNDRV_SEQ_EVENT_CONTROLLER is for 'struct snd_seq_ev_ctrl'. However there're some types for any data type; e.g. SNDRV_SEQ_EVENT_ECHO or SNDRV_SEQ_EVENT_USR0. For the kind of types, the event structure has no information about which data type should be used and user applications voluntarily decide the data type. Therefore the sender and receiver should share a kind of protocol in advance. This means that userspace applications require API to select data type independent of event type itself. 3. quote event and specific event types. Two event types are reserved for 'struct snd_seq_ev_quote'; i.e. SNDRV_SEQ_EVENT_KERNEL_ERROR and SNDRV_SEQ_EVENT_KERNEL_QUOTE(obsoleted?). However, the quote structure is exposed to userspace itself. Furthermore, as of v5.5 kernel, there's no in-kernel code to check the quote data from/to user client. Is it better to produce API so that userspace application can transfer the quote event? [1] https://github.com/alsa-project/alsa-gobject [2] https://github.com/takaswie/alsa-gobject/tree/topic/seq Regards Takashi Sakamoto

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