September 2024 - Alsa-devel - mailman.alsa-project.org

Handling values
by GitHub issues - opened 19 Sep '24

19 Sep '24

alsa-project/alsa-ucm-conf issue #442 was opened from thenameisluk: i have similar config to https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/Qualcomm/sc7… for [homestar](https://wiki.postmarketos.org/wiki/Lenovo_IdeaPad_Duet_5_(google-… ``` # Use case configuration for ALC5682+MAX98357A on SC7180 SectionVerb { Value { TQ "HiFi" } } SectionDevice."Speaker" { Comment "Speaker" Value { PlaybackChannels 4 PlaybackPCM "hw:${CardId},1" PlaybackPriority 100 PlaybackMixerElem "Speaker" PlaybackVolume "Speaker Playback Volume" } } SectionDevice."Mic" { Comment "Internal Microphone" EnableSequence [ cset "name='IF1 01 ADC Swap Mux' 1" cset "name='Stereo1 ADC L2 Mux' 1" cset "name='Stereo1 ADC R2 Mux' 1" cset "name='Stereo1 ADC MIXL ADC2 Switch' 1" cset "name='Stereo1 ADC MIXR ADC2 Switch' 1" ] DisableSequence [ cset "name='Stereo1 ADC L2 Mux' 0" cset "name='Stereo1 ADC R2 Mux' 0" cset "name='Stereo1 ADC MIXL ADC2 Switch' 0" cset "name='Stereo1 ADC MIXR ADC2 Switch' 0" ] Value { CapturePCM "hw:${CardId},0" CapturePriority 100 CaptureMixerElem "STO1 ADC" CaptureVolume "STO1 ADC Capture Volume" } } SectionDevice."HDMI" { Comment "HDMI Jack" ConflictingDevice [ "Speaker" ] EnableSequence [ ] DisableSequence [ ] Value { PlaybackPriority 200 PlaybackPCM "hw:${CardId},2" JackControl "HDMI Jack" } } ``` the speaker and mic section is irrelevant here hdmi is causing issues without ``` ConflictingDevice [ "Speaker" ] ``` pipewire doesn't want to use the ucm at all unless the hdmi is plugged in with it pipewire will only configure hdmi if hdmi is plugged in (when you restart it or at boot) it's all on the same card and the interface never goes away, only switches between ``` # amixer -c 0 contents numid=9,iface=CARD,name='HDMI Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off ... ``` without hdmi and ``` # amixer -c 0 contents numid=9,iface=CARD,name='HDMI Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on ``` with hdmi ```sudo alsactl monitor```also shows ``` node hw:0, #9 (0,0,0,HDMI Jack,0) VALUE ``` when plugging in and unplugging knowing all this i am unable to figure out how to handle it with ucm2 and would appreciate some help getting it working without that ConflictingDevice Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/442 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: cs35l45: Corrects cs35l45_get_clk_freq_id function data type
by Ricardo Rivera-Matos 19 Sep '24

19 Sep '24

Changes cs35l45_get_clk_freq_id() function data type from unsigned int to int. This function is returns a positive index value if successful or a negative error code if unsuccessful. Functionally there should be no difference as long as the unsigned int return is interpreted as an int, however it should be corrected for readability. Signed-off-by: Ricardo Rivera-Matos <rriveram(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45-tables.c | 2 +- sound/soc/codecs/cs35l45.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c index e1cebb9e4dc6..405dab137b3b 100644 --- a/sound/soc/codecs/cs35l45-tables.c +++ b/sound/soc/codecs/cs35l45-tables.c @@ -315,7 +315,7 @@ static const struct { { 0x3B, 24576000 }, }; -unsigned int cs35l45_get_clk_freq_id(unsigned int freq) +int cs35l45_get_clk_freq_id(unsigned int freq) { int i; diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index e2ebcf58d7e0..7a790d2acac7 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -507,7 +507,7 @@ extern const struct dev_pm_ops cs35l45_pm_ops; extern const struct regmap_config cs35l45_i2c_regmap; extern const struct regmap_config cs35l45_spi_regmap; int cs35l45_apply_patch(struct cs35l45_private *cs35l45); -unsigned int cs35l45_get_clk_freq_id(unsigned int freq); +int cs35l45_get_clk_freq_id(unsigned int freq); int cs35l45_probe(struct cs35l45_private *cs35l45); void cs35l45_remove(struct cs35l45_private *cs35l45); -- 2.43.0

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[PATCH v1] ALSA: hda/tas2781: Add new quirk for Lenovo Y990 Laptop
by Baojun Xu 19 Sep '24

19 Sep '24

Add new vendor_id and subsystem_id in quirk for Lenovo Y990 Laptop. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 480e82df7a4c..4f69fd30fdce 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10612,6 +10612,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.43.0

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[RFC PATCH v3 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 19 Sep '24

19 Sep '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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[PATCH] ASoC: amd: acp: don't set card long_name
by Vijendar Mukunda 18 Sep '24

18 Sep '24

UCM can load a board-specific file based on the card long_name. Remove the constant "AMD Soundwire SOF" long_name so that the ASoC core can set the long_name based on DMI information. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 6c50c8276538..306854fb08e3 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -400,9 +400,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) return ret; } -/* SoC card */ -static const char sdw_card_long_name[] = "AMD Soundwire SOF"; - static int mc_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev); @@ -463,8 +460,6 @@ static int mc_probe(struct platform_device *pdev) if (!card->components) return -ENOMEM; - card->long_name = sdw_card_long_name; - /* Register the card */ ret = devm_snd_soc_register_card(card->dev, card); if (ret) { -- 2.34.1

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[PATCH 0/8] ASoC: amd: acp: refactor acp version differentiation logic
by Vijendar Mukunda 18 Sep '24

18 Sep '24

Currently different logics being used in the code for acp version differentiation. This patch series refactors the code to use acp pci revision id for handling acp version specific code. Vijendar Mukunda (8): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver sound/soc/amd/acp/acp-i2s.c | 38 +++++++++++++-------------- sound/soc/amd/acp/acp-legacy-common.c | 24 ++++++++--------- sound/soc/amd/acp/acp-legacy-mach.c | 8 +++--- sound/soc/amd/acp/acp-mach-common.c | 38 +++++++++++++++++++-------- sound/soc/amd/acp/acp-mach.h | 12 +++------ sound/soc/amd/acp/acp-pci.c | 7 +---- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 14 +++++----- sound/soc/amd/acp/acp-rembrandt.c | 4 +-- sound/soc/amd/acp/acp-renoir.c | 4 +-- sound/soc/amd/acp/acp-sof-mach.c | 6 ++--- sound/soc/amd/acp/acp63.c | 4 +-- sound/soc/amd/acp/acp70.c | 14 ++++------ sound/soc/amd/acp/acp_common.h | 19 ++++++++++++++ sound/soc/amd/acp/amd.h | 9 ++----- 15 files changed, 108 insertions(+), 95 deletions(-) create mode 100644 sound/soc/amd/acp/acp_common.h -- 2.34.1

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Re: [PATCH] [RFC] ASoC: Conditional PCM support
by Pierre-Louis Bossart 17 Sep '24

17 Sep '24

On 8/21/24 12:18, Cezary Rojewski wrote: > Conditional PCM (condpcm) helps facilitate modern audio usecases such as > Echo Cancellations and Noise Reduction. These are not invoked by the > means of userspace application opening an endpoint (FrontEnd) but are a > "side effect" of selected PCMs running simultaneously e.g.: if both > Speaker (source) and Microphone Array (sink) are running, reference > data from the Speaker and take it into account when processing capture > for better voice command detection ratio. The point about dependencies between capture/playback usages is certainly valid, and we've faced it multiple times for SOF - and even before in the mobile phone days. I am not convinced however that the graph management suggested here solves the well-known DPCM routing problems? See notes in no specific order below. I am not following what the 'source' and 'sink' concepts refer to in this context. It looks like you are referring to regular PCM devices, i.e. Front Ends in soc-pcm parlance but examples and code make references to Back Ends. There are also complicated cases where the amplifiers can provide an echo reference for AEC and I/V sensing for speaker protection. You would want to capture both even if there's no capture happening at the userspace level. This is a well-know DPCM routing issue where we have to rely on a Front-End being opened and some tags in UCM to deal with loose coupling. It would help if you added precisions on your assumptions of where the processing takes place. In some cases Echo cancellation is handled in userspace, others in SOC firmware and others externally in a codec. The notion of source/sink is also problematic when the same BE provides two sources of information that will be split, again same problem with amplifier feedback being used for two separate functions. What happens if you have multiple sinks for one source? Same for the cases where the mic input is split multiple ways with different processing added on different PCM capture devices, e.g. for WebRTC there's an ask for a raw input, an AEC-processed input and AEC+NS-processed input. That's typically implemented with two splitters, the echo reference would be used by an intermediate node inside a firmware graph, not at the DAI/BE or host/FE levels, and such intermediate nodes are usually not handled by soc-pcm. We really need more than the notion of FE and BE, a two-layer solution is no longer sufficient. The other thing that looks weird is the dependency on both sink and source sharing a common state. For a noise reduction there are cases where you'd want the mic input to be stored in a history buffer so that the noise parameters can be estimated as soon as the actual capture starts. > Which PCMs are needed for given conditional PCM to be spawned is > determinated by the driver when registering the condpcm. Presumably such links should be described by a topology file? It would be odd for a driver to have to guess when to connect processing elements. > The functionality was initially proposed for the avs-driver [1] and, > depending on feedback and review may either go back into avs -or- become > a ASoC-core feature. Implementation present here is an example of how > such functionality could look and work on the ASoC side. Compared to > what was provided initially, the patch carries simplified version of the > feature: no priority/overriding for already running conditional PCMs. > Whatever is spawned is treated as a non-conflicting entity. > > Assumptions and design decisions: > > - existence and outcome of condpcm operations is entirely optional and > shall not impact the runtime flow of PCMs that spawned given condpcm, > e.g.: fail in cpcm->hw_params() shall not impact fe->hw_params() or > be->hw_params() negatively. Think of it as of debugfs. Useful? Yes. > Required for system to operate? No. that's debatable, if the AEC setup isn't successful then is the functionality implemented correctly? My take is no, don't fail silently if the AEC doesn't work. If this functionality is listed as a product requirement then it cannot be treated as a debugfs optional thing. Exhibit A for this is the countless cases where validation reported a problem with a path remaining active or conversely not being setup, or a voice quality issue. Those are not optional... > - a condpcm is a runtime entity that's audio format independent - since > certain FE/BEs are its dependencies already, that's no need to do > format ruling twice. Driver may still do custom checks thanks to > ->match() operation. > > - a condpcm allows for additional processing of data that flows from > data-source - a substream instance acting as data provider - > to sink - a substream acting as data consumer. At the same time, > regardless of substream->stream, given substream may act as data > source for one condpcm and data sink for another, simultaneously. > > - while condpcm's behaviour mimics standard PCM one, there is no > ->open() and ->close() - FE/BEs are treated as operational starting > with successful ->hw_params(), when hw_ruling is done and hardware is > configured. > > - cpcm->prepare() gets called only when both data source and sink are > prepared > - cpcm->trigger(START) gets called only when both data source and sink > are running > - cpcm->trigger(STOP) gets called when either data source or sink is > stopped > > Simplified state machine: > > | > register_condpcm() | > v > +--+-------------+ > | DISCONNECTED |<-+ > +--+-------------+ | > | | > condpcm_hw_params() | | > v | > +--+-------------+ | > | SETUP | | condpcm_hw_free() > +--+-------------+ | > | | > condpcm_prepare() | | > v | > +--+----+--------+ | > | PREPARED |--+ > +--+----------+--+ > | ^ > condpcm_start() | | condpcm_stop() > v | > +--+----------+--+ > | RUNNING | > +----------------+ > > What's missing? > I've not covered the locking part yet. While some operations are covered > by default thanks to snd_soc_dpcm_mutex(), it is insufficient. > If feature goes back to the avs-driver, then I'm set due to path->mutex. > > The locking is one of the reasons I'm leaning towards leaving the > condpcm within the avs-driver. For soc_condpcm_find_match() to be > precise and do no harm, a lock must prepend the list_for_each_entry(). > Entries (substreams) of that list may be part of number of different > components and the search may negatively impact runtime flow of > substreams that do not care about condpcms at all. > > Has this been tested? > > Unit-like only. Typical case below with avs_condpcm_ops representing > bunch of stubs with printfs. > > static struct snd_soc_condpcm_pred pred1 = { > .card_name = "ssp0-loopback", > .link_name = "SSP0-Codec", /* BE */ > .direction = SNDRV_PCM_STREAM_PLAYBACK, > }; > > static struct snd_soc_condpcm_pred pred2 = { > .card_name = "hdaudioB0D2", > .link_name = "HDMI1", /* FE */ > .direction = SNDRV_PCM_STREAM_PLAYBACK, > }; It's not intuitive to follow what HDMI and SSP might have to do with each other, nor why one is a BE and one is an FE? If I follow the code below, the SSP loopback is a source feeds into an HDMI sink, and SSP is a BE and HDMI an FE? Confusing example... > static void avs_condpcms_register(struct avs_dev *adev) > { > (...) > snd_soc_register_condpcm(&pred1, &pred2, &avs_condpcm_ops, adev); > } > > [1]: https://lore.kernel.org/alsa-devel/20211208111301.1817725-21-cezary.rojewsk… > > Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> > --- > include/sound/pcm.h | 1 + > include/sound/soc.h | 65 ++++++++ > sound/core/pcm.c | 1 + > sound/soc/Makefile | 2 +- > sound/soc/soc-condpcm.c | 348 ++++++++++++++++++++++++++++++++++++++++ > sound/soc/soc-condpcm.h | 17 ++ > sound/soc/soc-core.c | 2 + > sound/soc/soc-pcm.c | 11 ++ > 8 files changed, 446 insertions(+), 1 deletion(-) > create mode 100644 sound/soc/soc-condpcm.c > create mode 100644 sound/soc/soc-condpcm.h > > diff --git a/include/sound/pcm.h b/include/sound/pcm.h > index ac8f3aef9205..7e635b3103a2 100644 > --- a/include/sound/pcm.h > +++ b/include/sound/pcm.h > @@ -482,6 +482,7 @@ struct snd_pcm_substream { > struct list_head link_list; /* linked list member */ > struct snd_pcm_group self_group; /* fake group for non linked substream (with substream lock inside) */ > struct snd_pcm_group *group; /* pointer to current group */ > + struct list_head cpcm_candidate_node; It wouldn't hurt to describe what 'candidate' might mean here? > /* -- assigned files -- */ > int ref_count; > atomic_t mmap_count; > diff --git a/include/sound/soc.h b/include/sound/soc.h > index e844f6afc5b5..32a6b5092192 100644 > --- a/include/sound/soc.h > +++ b/include/sound/soc.h > @@ -426,6 +426,7 @@ struct snd_jack; > struct snd_soc_card; > struct snd_soc_pcm_stream; > struct snd_soc_ops; > +struct snd_soc_condpcm; > struct snd_soc_pcm_runtime; > struct snd_soc_dai; > struct snd_soc_dai_driver; > @@ -1154,6 +1155,64 @@ struct snd_soc_card { > #define for_each_card_widgets_safe(card, w, _w) \ > list_for_each_entry_safe(w, _w, &card->widgets, list) > > +/* Conditional PCM operations called by soc-pcm.c. */ > +struct snd_soc_condpcm_ops { > + int (*match)(struct snd_soc_condpcm *, struct snd_pcm_substream *, > + struct snd_pcm_substream *); > + int (*hw_params)(struct snd_soc_condpcm *, struct snd_pcm_hw_params *); > + int (*hw_free)(struct snd_soc_condpcm *); > + int (*prepare)(struct snd_soc_condpcm *, struct snd_pcm_substream *); > + int (*trigger)(struct snd_soc_condpcm *, struct snd_pcm_substream *, int); > +}; > + > +/** > + * struct snd_soc_condpcm_pred - Predicate, describes source or sink (substream) > + * dependency for given conditional PCM. > + * > + * @card_name: Name of card owning substream to find. > + * @link_name: Name of DAI LINK owning substream to find. > + * @direction: Whether its SNDRV_PCM_STREAM_PLAYBACK or CAPTURE. > + */ > +struct snd_soc_condpcm_pred { > + const char *card_name; Please tell me the runtimes and links are in the same card... If not, there's all kinds of power management and probe/remove issues... > + const char *link_name; dai link name? > + int direction; > +}; > + > +/** > + * struct snd_soc_condpcm - Conditional PCM descriptor. > + * > + * @ops: custom PCM operations. > + * @preds: predicates for identifying source and sink for given conditional PCM. predicate is a verb and a noun, not clear what you are trying to document. > + * > + * @source: substreaming acting as a data source, assigned at runtime. > + * @sink: substreaming acting as a data sink, assigned at runtime. > + * @state: current runtime state. presumably this state is already defined that the state of sink/source? > + * > + * @source_node: list navigation for rtd->source_list. > + * @sink_node: list navigation for rtd->sink_list. > + * @node: list navigation for condpcm_list (soc-condpcm.c). > + */ > +struct snd_soc_condpcm { > + const struct snd_soc_condpcm_ops *ops; > + struct snd_soc_condpcm_pred preds[2]; > + > + struct snd_pcm_substream *source; > + struct snd_pcm_substream *sink; > + snd_pcm_state_t state; > + void *private_data; > + > + /* Condpcms navigation for the pcm runtime. */ > + struct list_head source_node; > + struct list_head sink_node; > + struct list_head node; > +}; > + > +int snd_soc_unregister_condpcm(struct snd_soc_condpcm *cpcm); > +struct snd_soc_condpcm *snd_soc_register_condpcm(struct snd_soc_condpcm_pred *source, > + struct snd_soc_condpcm_pred *sink, > + const struct snd_soc_condpcm_ops *ops, > + void *private_data); > > static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card) > { > @@ -1161,6 +1220,10 @@ static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card) > } > > /* SoC machine DAI configuration, glues a codec and cpu DAI together */ > +/* > + * @source_cpcm_list: List of condpcms this pcm is source for. > + * @sink_cpcm_list: List of condpcms this pcm is sink for. > + */ > struct snd_soc_pcm_runtime { > struct device *dev; > struct snd_soc_card *card; > @@ -1172,6 +1235,8 @@ struct snd_soc_pcm_runtime { > /* Dynamic PCM BE runtime data */ > struct snd_soc_dpcm_runtime dpcm[SNDRV_PCM_STREAM_LAST + 1]; > struct snd_soc_dapm_widget *c2c_widget[SNDRV_PCM_STREAM_LAST + 1]; > + struct list_head cpcm_source_list; > + struct list_head cpcm_sink_list; > > long pmdown_time; > > diff --git a/sound/core/pcm.c b/sound/core/pcm.c > index dc37f3508dc7..12243cecaa11 100644 > --- a/sound/core/pcm.c > +++ b/sound/core/pcm.c > @@ -663,6 +663,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) > substream->pstr = pstr; > substream->number = idx; > substream->stream = stream; > + INIT_LIST_HEAD(&substream->cpcm_candidate_node); > sprintf(substream->name, "subdevice #%i", idx); > substream->buffer_bytes_max = UINT_MAX; > if (prev == NULL) > diff --git a/sound/soc/Makefile b/sound/soc/Makefile > index 775bb38c2ed4..8004de7c500c 100644 > --- a/sound/soc/Makefile > +++ b/sound/soc/Makefile > @@ -1,6 +1,6 @@ > # SPDX-License-Identifier: GPL-2.0 > snd-soc-core-y := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o > -snd-soc-core-y += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o > +snd-soc-core-y += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o soc-condpcm.o > snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o > > ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) > diff --git a/sound/soc/soc-condpcm.c b/sound/soc/soc-condpcm.c > new file mode 100644 > index 000000000000..786b3fafd714 > --- /dev/null > +++ b/sound/soc/soc-condpcm.c > @@ -0,0 +1,348 @@ > +// SPDX-License-Identifier: GPL-2.0-only > +// > +// Conditional-PCM management > +// > +// Copyright(c) 2024 Intel Corporation > +// > +// Author: Cezary Rojewski <cezary.rojewski(a)intel.com> > +// > + > +#include <linux/export.h> > +#include <linux/list.h> > +#include <linux/slab.h> > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include "soc-condpcm.h" > + > +/* > + * condpcm_list - Stores all registered conditional pcms. > + * condpcm_candidate_list - Stores all substreams that passed hw_params() step. > + */ > +static LIST_HEAD(condpcm_list); > +static LIST_HEAD(condpcm_candidate_list); > + > +static bool dpcm_prepare_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state == SND_SOC_DPCM_STATE_PREPARE || > + state == SND_SOC_DPCM_STATE_START || > + state == SND_SOC_DPCM_STATE_PAUSED; > +} > + > +static bool dpcm_hw_params_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state >= SND_SOC_DPCM_STATE_HW_PARAMS && > + state < SND_SOC_DPCM_STATE_HW_FREE; > +} > + > +static bool dpcm_start_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state == SND_SOC_DPCM_STATE_START || > + state == SND_SOC_DPCM_STATE_PAUSED; > +} > + > +static int soc_condpcm_hw_params(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_hw_params *params) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(cpcm->source); > + struct snd_soc_pcm_runtime *rtd2 = snd_soc_substream_to_rtd(cpcm->sink); how are the 'params' defined? I read above " a condpcm is a runtime entity that's audio format independent - since certain FE/BEs are its dependencies already, that's no need to do format ruling twice. " That doesn't tell us how this 'params' is determined. This is important for cases where the speaker output is e.g. 2ch 48kHz and the mic input is 4ch 96kHz. If this condpcm is not managed by any usersapce action, then what is the logic for selecting the settings in 'params'? > + int ret; > + > + ret = cpcm->ops->hw_params(cpcm, params); > + if (ret) > + return ret; > + > + list_add_tail(&cpcm->source_node, &rtd->cpcm_source_list); > + list_add_tail(&cpcm->sink_node, &rtd2->cpcm_sink_list); > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + return 0; > +} There's also the well-known problem that hw_params can be called multiple times. Here this wouldn't work with the same source/sink added multiple times in a list. > + > +static void soc_condpcm_hw_free(struct snd_soc_condpcm *cpcm) > +{ > + cpcm->ops->hw_free(cpcm); > + list_del(&cpcm->source_node); > + list_del(&cpcm->sink_node); > + cpcm->state = SNDRV_PCM_STATE_DISCONNECTED; > +} > + > +static void soc_condpcm_prepare(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream) > +{ > + int ret; > + > + ret = cpcm->ops->prepare(cpcm, substream); > + if (!ret) > + cpcm->state = SNDRV_PCM_STATE_PREPARED; > +} you probably need to look at the xruns and resume cases, where prepare() is used for vastly different purposes. > + > +static void soc_condpcm_start(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int cmd) > +{ > + int ret; > + > + ret = cpcm->ops->trigger(cpcm, substream, cmd); > + if (ret) > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + else > + cpcm->state = SNDRV_PCM_STATE_RUNNING; > +} > + > +static void soc_condpcm_stop(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int cmd) > +{ > + int ret; > + > + ret = cpcm->ops->trigger(cpcm, substream, cmd); > + if (ret || cmd != SNDRV_PCM_TRIGGER_PAUSE_PUSH) > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + else > + cpcm->state = SNDRV_PCM_STATE_PREPARED; > +} > + > +void snd_soc_condpcms_prepare(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Prepare conditional pcms only if source and sink are both preprared. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) { > + if (cpcm->state == SNDRV_PCM_STATE_SETUP && > + dpcm_prepare_done(cpcm->sink)) > + soc_condpcm_prepare(cpcm, substream); > + } > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) { > + if (cpcm->state == SNDRV_PCM_STATE_SETUP && > + dpcm_prepare_done(cpcm->source)) > + soc_condpcm_prepare(cpcm, substream); > + } > +} > + > +static void soc_condpcms_start(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Start conditional pcms only if source and sink are both running. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) { > + if (cpcm->state == SNDRV_PCM_STATE_PREPARED && > + dpcm_start_done(cpcm->sink)) > + soc_condpcm_start(cpcm, substream, cmd); > + } > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) { > + if (cpcm->state == SNDRV_PCM_STATE_PREPARED && > + dpcm_start_done(cpcm->source)) > + soc_condpcm_start(cpcm, substream, cmd); > + } > +} > + > +static void soc_condpcms_stop(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Stop conditional pcms if either source or sink is not running. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) > + if (cpcm->state == SNDRV_PCM_STATE_RUNNING) > + soc_condpcm_stop(cpcm, substream, cmd); > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) > + if (cpcm->state == SNDRV_PCM_STATE_RUNNING) > + soc_condpcm_stop(cpcm, substream, cmd); > +} > + > +void snd_soc_condpcms_trigger(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_RESUME: > + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > + soc_condpcms_start(rtd, substream, cmd); > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > + soc_condpcms_stop(rtd, substream, cmd); > + break; > + } > +} > + > +void snd_soc_condpcms_teardown(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream) > +{ > + struct snd_soc_condpcm *cpcm, *save; > + > + list_del(&substream->cpcm_candidate_node); > + > + /* Free all condpcms this substream spawned. */ > + list_for_each_entry_safe(cpcm, save, &rtd->cpcm_source_list, source_node) > + soc_condpcm_hw_free(cpcm); > + list_for_each_entry_safe(cpcm, save, &rtd->cpcm_sink_list, sink_node) > + soc_condpcm_hw_free(cpcm); > +} > + > +static bool soc_condpcm_test(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int dir) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + struct snd_soc_condpcm_pred *pred = &cpcm->preds[dir]; > + > + return pred->direction == substream->stream && > + !strcmp(pred->card_name, rtd->card->name) && > + !strcmp(pred->link_name, rtd->dai_link->name); > +} > + > +static struct snd_pcm_substream * > +soc_condpcm_find_match(struct snd_soc_condpcm *cpcm, struct snd_pcm_substream *ignore, int dir) > +{ > + struct snd_pcm_substream *substream; > + > + list_for_each_entry(substream, &condpcm_candidate_list, cpcm_candidate_node) { > + if (substream == ignore) > + continue; > + > + /* Only substreams that passed hw_params() are valid candidates. */ > + if (!dpcm_hw_params_done(substream)) > + continue; > + > + if (soc_condpcm_test(cpcm, substream, dir)) > + return substream; > + } > + > + return NULL; > +} > + > +static int soc_condpcm_walk(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, int dir) > +{ > + /* Temporary source/sink cache. */ > + struct snd_pcm_substream *substreams[2]; > + struct snd_soc_condpcm *cpcm; > + int ret; > + > + substreams[dir] = substream; > + > + list_for_each_entry(cpcm, &condpcm_list, node) { > + if (cpcm->state != SNDRV_PCM_STATE_DISCONNECTED) > + continue; > + > + /* Does this cpcm match @substream? */ > + if (!soc_condpcm_test(cpcm, substream, dir)) > + continue; > + > + /* Find pair for the @substream. */ > + substreams[!dir] = soc_condpcm_find_match(cpcm, substream, !dir); > + if (!substreams[!dir]) > + continue; > + > + /* Allow driver to have the final word. */ > + ret = cpcm->ops->match(cpcm, substreams[0], substreams[1]); > + if (ret) > + continue; > + cpcm->source = substreams[0]; > + cpcm->sink = substreams[1]; > + > + ret = soc_condpcm_hw_params(cpcm, params); > + if (ret) { > + cpcm->source = NULL; > + cpcm->sink = NULL; > + return ret; > + } > + } > + > + return 0; > +} > + > +/* Called by soc-pcm.c after each successful hw_params(). */ > +int snd_soc_condpcms_walk_all(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + int ret; > + > + list_add_tail(&substream->cpcm_candidate_node, &condpcm_candidate_list); > + > + /* Spawn all condpcms this substream is the missing source of. */ > + ret = soc_condpcm_walk(rtd, substream, params, SNDRV_PCM_STREAM_CAPTURE); > + if (ret) > + return ret; > + > + /* Spawn all condpcms this substream is the missing sink of. */ > + return soc_condpcm_walk(rtd, substream, params, SNDRV_PCM_STREAM_PLAYBACK); > +} Are loops supported? Is the order between capture and playback intentional? Is the notion of playback/capture even relevant when trying to add loopbacks? Lots of questions...

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[PATCH 000/113] ALSA: add snd_pcm_is_playback/capture() macro
by Kuninori Morimoto 17 Sep '24

17 Sep '24

Hi ALSA-ML Cc Staging-ML, USB-ML Many drivers are using below code to know the Sound direction. if (direction == SNDRV_PCM_STREAM_PLAYBACK) This patch-set add snd_pcm_is_playback/capture() macro to handle it. Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH v2 0/2] Updates for Atmel SSC DAI
by Andrei Simion 16 Sep '24

16 Sep '24

This patch set includes two updates for the Atmel SSC DAI driver: - Address the limitation with the S24_LE format. - Add stream names for DPCM and future use-cases. Codrin Ciubotariu (2): ASoC: atmel: atmel_ssc_dai: Add stream names ASoC: atmel: atmel_ssc_dai: Drop S24_LE support due to single channel limitation sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) base-commit: 7083504315d64199a329de322fce989e1e10f4f7 -- 2.34.1

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[PATCH 0/2] Updates for Atmel SSC DAI
by Andrei Simion 16 Sep '24

16 Sep '24

This patch set includes two updates for the Atmel SSC DAI driver: - Address the limitation with the S24_LE format. - Add stream names for DPCM and future use-cases. Codrin Ciubotariu (2): ASoC: atmel: atmel_ssc_dai: Add stream names ASoC: atmel: atmel_ssc_dai: Drop S24_LE support due to single channel limitation sound/soc/atmel/atmel_ssc_dai.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) base-commit: 5acd9952f95fb4b7da6d09a3be39195a80845eb6 -- 2.34.1

2 5