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

ALSA 1.2.8 release
by Jaroslav Kysela 12 Nov '24

12 Nov '24

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 23 Oct '24

23 Oct '24

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

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[PATCH] ALSA: hda/realtek: tas2781: Fix ROG ALLY X audio
by Jonathan LoBue 11 Oct '24

11 Oct '24

This patch enables the TI TAS2781 amplifier SoC for the ASUS ROG ALLY X. This is a design change from the original ASUS ROG ALLY, creating the need for this patch. All other Realtek Codec settings seem to be re-used from the original ROG ALLY design (on the ROG ALLY X). This patch maintains the previous settings for the Realtek codec portion, but enables the I2C binding for the TI TAS2781 amplifier (instead of the Cirrus CS35L41 amp used on the original ASUS ROG ALLY). One other requirement must be met for audio to work on the ASUS ROG ALLY X. A proper firmware file in the correct location with a proper symlink. We had reached out to TI engineers and confirmed that the firmware found in the Windows' driver package has a GPL license. Bazzite Github is hosting this firmware file for now until proper linux-firmware upstreaming can occur. https://github.com/ublue-os/bazzite This firmware file should be placed in /usr/lib/firmware/ti/tas2781/TAS2XXX1EB3.bin with a symlink to it from /usr/lib/firmware/TAS2XXX1EB3.bin Co-developed by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Signed-off-by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Co-developed by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Antheas Kapenekakis <lkml(a)antheas.dev> Tested-by: Richard Alvarez <alvarez.richard(a)gmail.com> Tested-by: Miles Montierth <cyber_dopamine(a)intheblackmedia.com> Signed-off-by: Jonathan LoBue <jlobue10(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 480e82df7a4c..9bc39c2ee6b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7410,6 +7410,7 @@ enum { ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_X, ALC294_FIXUP_ASUS_ALLY_PINS, ALC294_FIXUP_ASUS_ALLY_VERBS, ALC294_FIXUP_ASUS_ALLY_SPEAKER, @@ -8877,6 +8878,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS }, + [ALC294_FIXUP_ASUS_ALLY_X] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, [ALC294_FIXUP_ASUS_ALLY_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10313,6 +10320,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X), SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.46.0

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Add stacked and parallel memories support in spi-nor
by Mahapatra, Amit Kumar 10 Oct '24

10 Oct '24

Hello Everyone, As https://lore.kernel.org/all/20231125092137.2948-8-amit.kumar-mahapatra@amd.… thread discussing the addition of stacked and parallel mode support in SPI-NOR has become too long and difficult to follow. Therefore, I'm starting a new email thread to focus on the DT binding changes and the overall implementation architecture for adding stacked and parallel (multi-CS) support in the Kernel. Please share your input on this approach. Background: The AMD QSPI controller supports two advanced connection modes(Stacked and Dual Parallel) which allow the controller to treat two different flashes as one storage. Stacked: Flashes share the same SPI bus, but different CS line, controller driver asserts the CS of the flash to which it needs to communicate. Stacked mode is a software abstraction rather than a controller feature or capability. At any given time, the controller communicates with one of the two connected flash devices, as determined by the requested address and data length. If an operation starts on one flash and ends on the other, the core needs to split it into two separate operations and adjust the data length accordingly. Dual Parallel: Both the flashes have their separate SPI bus, CS of both the flashes will be asserted/de-asserted at the same time. In this mode data will be split across both the flashes by enabling the STRIPE setting in the controller. Parallel mode is a controller feature where if the STRIPE bit is set then the controller internally handles the data split during data write to the flashes and while reading data from the flash the controller internally merges data from both the flashes before writing to the controller FIFO. If STRIPE is not enabled, then same data will be sent to both the devices. In parallel mode both the flashes should be identical. For more information on the modes please feel free to go through the controller flash interface below https://docs.amd.com/r/en-US/am011-versal-acap-trm/QSPI-Flash-Device-Interf… Mirochip QSPI controller also supports "Dual Parallel 8-bit IO mode", but they call it "Twin Quad Mode". https://ww1.microchip.com/downloads/aemDocuments/documents/MPU32/ProductDoc… Initially in [1] [2] [3] Miquel had tried to extend MTD-CONCAT driver to support Stacked mode, but the bindings were not accepted. [1] https://lore.kernel.org/all/20191113171505.26128-4-miquel.raynal@bootlin.co… [2] https://lore.kernel.org/all/20191127105522.31445-5-miquel.raynal@bootlin.co… [3]https://lore.kernel.org/all/20211112152411.818321-1-miquel.raynal@bootlin… So, the MTD-CONCAT approach was dropped and the DT bindings that got accepted [4] [5] [6] describe the two flash devices as being one. [4] https://github.com/torvalds/linux/commit/f89504300e94524d5d5846ff8b728592ac… [5] https://github.com/torvalds/linux/commit/eba5368503b4291db7819512600fa014ea… [6] https://github.com/torvalds/linux/commit/e2edd1b64f1c79e8abda365149ed62a2a9… SPI core changes to support the above bindings were added through the following commit: https://github.com/torvalds/linux/commit/4d8ff6b0991d5e86b17b235fc46ec62e91… While adding the support in SPI-NOR we received some inputs/suggestions from Tudor, based on that I am planning to add a new layer between the SPI-NOR and MTD layers to support stacked and parallel configurations. This new layer will be part of the spi-nor and located in mtd/spi-nor/. This layer would perform the following tasks: - During probing, store information from all the connected flashes, whether in stacked or parallel mode, and present it as a single device to the MTD layer. - Register callbacks and manage MTD device registration within the new layer instead of in spi-nor/core.c - There will very minimal change in spinor/core.c as the stacked & parallel handling will be managed by the new layer on top of SPI-NOR. - In stacked mode, the new layer will select the appropriate spi-nor flash based on the address provided by the MTD layer during flash operations. - Manage flash crossover operations in stacked mode. - Handle odd byte count requests from the MTD layer during flash operations in parallel mode. For implementing this the current DT binding[1] [2] [3] need to be updated as follows. stacked-memories DT changes: - Flash size information can be retrieved directly from the flash, so it has been removed from the DT binding. - Each stacked flash will have its own flash node. This approach allows flashes of different makes and sizes to be stacked together, as each flash will be probed individually. - Each of the flash node will have its own "reg" property that will contain its physical CS. - The stacked-memories DT bindings will contain the phandles of the flash nodes connected in stacked mode. - The first flash node will contain the mtd partition that would have the cross over memory staring at a memory location in the first flash and ending at some memory location of the 2nd flash - The new layer will update the mtd->size and other mtd_info parameters after both the flashes are probed and will call mtd_device_register with the combined information. spi@0 { flash@0 { compatible = "jedec,spi-nor" reg = <0x00>; stacked-memories = <&flash@0 &flash@1>; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-0"; reg = <0x0 0xf00000>; }; } flash@1 { compatible = "jedec,spi-nor" reg = <0x01>; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-1"; reg = <0x0 0x800000>; }; } } parallel-memories DT changes: - Flash size information can be retrieved directly from the flash, so it has been removed from the DT binding. - Each flash connected in parallel mode should be identical and will have one flash node for both the flash devices. - The "reg" prop will contain the physical CS number for both the connected flashes. - The new layer will double the mtd-> size and register it with the mtd layer. spi@0 { flash@0 { compatible = "jedec,spi-nor" reg = <0x00 0x01>; parallel-memories; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-0"; reg = <0x0 0xf00000>; } }; } Regards, Amit

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[PATCH -next 0/5] sound/soc: fix some macro definitions and usages
by Hongbo Li 01 Oct '24

01 Oct '24

Many variable in macro are not used as we used macro_check script to detect and mamually check, let us address these issues. Hongbo Li (5): sound/soc: fix macro definition on TWL4030_OUTPUT_PGA sound/soc remove unused substream in macro soc_component_mark_pop sound/soc remove unused substream in macro soc_dai_mark_pop sound/soc remove unused substream in macro soc_link_mark_pop sound/soc: fix macro definition on STM_SAI_HAS_EXT_SYNC sound/soc/codecs/twl4030.c | 12 ++++++------ sound/soc/soc-component.c | 14 +++++++------- sound/soc/soc-dai.c | 10 +++++----- sound/soc/soc-link.c | 10 +++++----- sound/soc/stm/stm32_sai_sub.c | 2 +- 5 files changed, 24 insertions(+), 24 deletions(-) -- 2.34.1

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[PATCH v2 0/2] {ASoC,media}: constify snd_soc_component_driver when used as read-only
by Javier Carrasco 01 Oct '24

01 Oct '24

Declare `snd_soc_component_driver` as const to move their declarations to read-only sections for the drivers that do not modify the struct after its declaration. Apart from a single case under media/, the affected drivers are members of the ASoC subsystem. To: Tim Harvey <tharvey(a)gateworks.com> To: Mauro Carvalho Chehab <mchehab(a)kernel.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> To: Ray Jui <rjui(a)broadcom.com> To: Scott Branden <sbranden(a)broadcom.com> To: Broadcom internal kernel review list <bcm-kernel-feedback-list(a)broadcom.com> To: Shenghao Ding <shenghao-ding(a)ti.com> To: Kevin Lu <kevin-lu(a)ti.com> To: Baojun Xu <baojun.xu(a)ti.com> To: Olivier Moysan <olivier.moysan(a)foss.st.com> To: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> To: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> To: Alexandre Torgue <alexandre.torgue(a)foss.st.com> To: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> To: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: linux-media(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> Changes in v2: - drop cs43130 and sti-sas, as they modifiy the struct in the probe function. - Link to v1: https://lore.kernel.org/r/20240725-const_snd_soc_component_driver-v1-0-3d7e… --- Javier Carrasco (2): media: i2c: tda1997x: constify snd_soc_component_driver struct ASoC: constify snd_soc_component_driver struct drivers/media/i2c/tda1997x.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/pcm186x.c | 4 ++-- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 12 files changed, 13 insertions(+), 13 deletions(-) --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240725-const_snd_soc_component_driver-b9629a95b948 Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

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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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[RESEND PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 14 Sep '24

14 Sep '24

Hi Mark, resending this as it failed to apply in my last submission. Added Neil Armstrong to Cc: as hopefully he will be able to better review this. Thanks, Geraldo Nascimento --- Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10), for example, if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,28 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending on the HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

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[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

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