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

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...

4 7

[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

4 119

[PATCH 0/1] Add support for primary mi2s on SM8250
by Jens Reidel 11 Sep '24

11 Sep '24

This patch adds support for the primary mi2s interface on devices using SM8250 audio drivers. Tested on SM7150 (xiaomi-davinci). SM7150 sound is close to SM8250 and we intend to use it as a fallback in the future. To: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux(a)mainlining.org Jens Reidel (1): ASoC: qcom: sm8250: enable primary mi2s sound/soc/qcom/sm8250.c | 8 ++++++++ 1 file changed, 8 insertions(+) -- 2.46.0

4 5

[Patch v5 00/12] Add audio support for LPC32XX CPUs
by Piotr Wojtaszczyk 11 Sep '24

11 Sep '24

This pach set is to bring back audio to machines with a LPC32XX CPU. The legacy LPC32XX SoC used to have audio spport in linux 2.6.27. The support was dropped due to lack of interest from mainaeners. Piotr Wojtaszczyk (12): dt-bindings: dma: pl08x: Add dma-cells description dt-bindings: dma: Add lpc32xx DMA mux binding ASoC: dt-bindings: lpc32xx: Add lpc32xx i2s DT binding ARM: dts: lpc32xx: Use simple-mfd for clock control block ARM: dts: lpc32xx: Add missing dma properties ARM: dts: lpc32xx: Add missing i2s properties clk: lpc32xx: initialize regmap using parent syscon dmaengine: Add dma router for pl08x in LPC32XX SoC ARM: lpc32xx: Remove pl08x platform data in favor for device tree mtd: rawnand: lpx32xx: Request DMA channels using DT entries ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs i2x: pnx: Fix potential deadlock warning from del_timer_sync() call in isr .../devicetree/bindings/dma/arm-pl08x.yaml | 7 + .../bindings/dma/nxp,lpc3220-dmamux.yaml | 49 +++ .../bindings/sound/nxp,lpc3220-i2s.yaml | 73 ++++ MAINTAINERS | 20 + arch/arm/boot/dts/nxp/lpc/lpc32xx.dtsi | 53 ++- arch/arm/mach-lpc32xx/phy3250.c | 54 --- drivers/clk/Kconfig | 1 + drivers/clk/nxp/clk-lpc32xx.c | 26 +- drivers/dma/Kconfig | 9 + drivers/dma/Makefile | 1 + drivers/dma/lpc32xx-dmamux.c | 195 +++++++++ drivers/i2c/busses/i2c-pnx.c | 48 +-- drivers/mtd/nand/raw/lpc32xx_mlc.c | 26 +- drivers/mtd/nand/raw/lpc32xx_slc.c | 26 +- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/lpc3xxx-i2s.c | 375 ++++++++++++++++++ sound/soc/fsl/lpc3xxx-i2s.h | 79 ++++ sound/soc/fsl/lpc3xxx-pcm.c | 72 ++++ 19 files changed, 993 insertions(+), 130 deletions(-) create mode 100644 Documentation/devicetree/bindings/dma/nxp,lpc3220-dmamux.yaml create mode 100644 Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml create mode 100644 drivers/dma/lpc32xx-dmamux.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.h create mode 100644 sound/soc/fsl/lpc3xxx-pcm.c -- 2.25.1

7 19

[PATCH] soundwire: stream: fix programming slave ports for non-continous port maps
by Krzysztof Kozlowski 09 Sep '24

09 Sep '24

Two bitmasks in 'struct sdw_slave_prop' - 'source_ports' and 'sink_ports' - define which ports to program in sdw_program_slave_port_params(). The masks are used to get the appropriate data port properties ('struct sdw_get_slave_dpn_prop') from an array. Bitmasks can be non-continuous or can start from index different than 0, thus when looking for matching port property for given port, we must iterate over mask bits, not from 0 up to number of ports. This fixes allocation and programming slave ports, when a source or sink masks start from further index. Fixes: f8101c74aa54 ("soundwire: Add Master and Slave port programming") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 7aa4900dcf31..f275143d7b18 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - u8 num_ports; + unsigned long mask; int i; if (direction == SDW_DATA_DIR_TX) { - num_ports = hweight32(slave->prop.source_ports); + mask = slave->prop.source_ports; dpn_prop = slave->prop.src_dpn_prop; } else { - num_ports = hweight32(slave->prop.sink_ports); + mask = slave->prop.sink_ports; dpn_prop = slave->prop.sink_dpn_prop; } - for (i = 0; i < num_ports; i++) { + for_each_set_bit(i, &mask, 32) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

6 15

How to handle complex Codec2Codec
by Kuninori Morimoto 09 Sep '24

09 Sep '24

Hi ALSA ML ALSA SoC is supporting Codec2Codec connection, and Audio Graph Card2 is supporting it. I think its image is... +------+ +--- |Codec1| <-- | +------+ | +------+ +--> |Codec2| --> +------+ This is OK, but how to handle below ? +----------------------+ |CPU | +------+ | +--[rate convert]<--| <-- |Codec1| <-- | | | +------+ | | | +------+ | +------------------>| --> |Codec2| --> +----------------------+ +------+ I think there are 2 issues. (1) How to handle [CPU] part on Codec2Codec ? (2) How to handle [rate convert] on Codec2Codec ? I guess (1) can be handled if [CPU] was worked as platform, but do you have any other idea/solution for it ? I think (2) [rate convert] needs DPCM connection, but how to use it with Codec2Codec ? Or, can we handle Codec2Codec via DPCM somehow ?? FE BE +------+ +------+ --> |Codec1| -> |Codec2| -> +------+ +------+ If we can do this, both (1) and (2) can be solved, like below ? In such case, [Codec1] need to work as [capture] when playback. But I'm not sure how to implement it... FE BE +------+ +-----------------+ +-----+ +------+ --> |Codec1|-|CPU[rate convert]| -> |dummy|-|Codec2| -> +------+ +-----------------+ +-----+ +------+ Thank you for your help !! Best regards --- Kuninori Morimoto

2 8

Re: [PATCH] ALSA: Add logitech Audio profile quirk
by Joshua Pius 06 Sep '24

06 Sep '24

1 3

Suspend/resume Issue on pcm_dmix.c in alsa-lib
by Chancel Liu 06 Sep '24

06 Sep '24

Hi Takashi Iwai, Jaroslav Kysela We found an issue on dmix in alsa-lib when do suspend and resume. It can be easily reproduced by following steps: 1. Run two dmix clients in parallel. (Only one client doesn't has such issue) ~# aplay xxx1.wav & ~# aplay xxx2.wav & Here I attach the asound.conf we're using. ~# cat /etc/asound.conf defaults.pcm.rate_converter "linear" pcm.dmix_44100{ type dmix ipc_key 5678293 ipc_key_add_uid yes slave{ pcm "hw:0,0" period_time 40000 format S16_LE rate 44100 } } pcm.asymed{ type asym playback.pcm "dmix_44100" capture.pcm "dsnoop_44100" } pcm.!default{ type plug route_policy "average" slave.pcm "asymed" } 2. Let linux enter into suspend and then resume(Repeat this step if not reproduced) 3. After resume, aplay will get stuck in snd_pcm_wait(). The GDB shows: (gdb) bt #0 0x0000fffff7da9264 in __GI___poll (fds=fds@entry=0xfffffffff480, nfds=nfds@entry=1, timeout=timeout@entry=240) at /usr/src/debug/glibc/2.39+git/sysdeps/unix/sysv/linux/poll.c:41 #1 0x0000fffff7edf468 in poll (__timeout=240, __nfds=1, __fds=0xfffffffff480) #2 snd1_pcm_wait_nocheck (pcm=pcm@entry=0xaaaaaaad2cb0, timeout=240, timeout@entry=-10001) at pcm.c:2993 #3 0x0000fffff7ee54a0 in snd1_pcm_write_areas (pcm=pcm@entry=0xaaaaaaad2cb0, areas=areas@entry=0xfffffffff560, offset=<optimized out>, offset@entry=0, size=<optimized out>, size@entry=1768, func=func@entry=0xfffff7ef5190 <snd_pcm_plugin_write_areas>) at pcm.c:7699 #4 0x0000fffff7ef5020 in snd_pcm_plugin_writei (pcm=0xaaaaaaad2cb0, buffer=<optimized out>, size=1768) at pcm_plugin.c:354 It seems that sometimes after suspend and resume there's no available space for data written into buffer. Then aplay keeps stuck in snd_pcm_wait(). I checked the hw_ptr of dmix and found that hw_ptr is always 0 after resume. I don't have a solution now so I turn to you for help. The version of alsa-lib is v1.2.11. Could you please help check it? Regards, Chancel Liu

3 12

[PATCH v26 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 04 Sep '24

04 Sep '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (31): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 429 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 34 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 45 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 71 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 135 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 338 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 110 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5820 insertions(+), 56 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 64

[PATCH] ASoC: tlv320aic32x4: Add multi endpoint support
by Marek Vasut 04 Sep '24

04 Sep '24

Support multiple endpoints on TLV320AIC32x4 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: Custom STM32MP157C board uses TLV320AIC32x4 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (TLV320AIC32x4) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: tlv320aic32x4_port: port { tlv320aic32x4_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; tlv320aic32x4_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Baojun Xu <baojun.xu(a)ti.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: Christoph Niedermaier <cniedermaier(a)dh-electronics.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kevin Lu <kevin-lu(a)ti.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Shenghao Ding <shenghao-ding(a)ti.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: kernel(a)dh-electronics.com Cc: linux-sound(a)vger.kernel.org --- sound/soc/codecs/tlv320aic32x4.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 5c0c81da06dba..54ea4bc58c276 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1073,6 +1073,13 @@ static int aic32x4_component_probe(struct snd_soc_component *component) return 0; } +static int aic32x4_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .probe = aic32x4_component_probe, .set_bias_level = aic32x4_set_bias_level, @@ -1082,6 +1089,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), .dapm_routes = aic32x4_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, @@ -1203,6 +1211,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = .num_dapm_widgets = ARRAY_SIZE(aic32x4_tas2505_dapm_widgets), .dapm_routes = aic32x4_tas2505_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_tas2505_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.45.2

2 1