June 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/4] ASoC: qcom: fixes for Click/Pop Noise
by Srinivas Kandagatla 27 Oct '23

27 Oct '23

Click/Pop Noise was a long pending issue with WSA Codecs which are prone to accumlate DC when ports are active but without any data streams. There are multiple places in the current setup, where this could happen in both startup as well as shutdown path. This patches help fix those issues by making sure the PA is Muted/Unmuted inline with the stream start/stop events. Other than this there is a patch to fix DSP graph saturation issue as we never unload playback graph before loading same graph. Srinivas Kandagatla (4): ASoC: qcom: q6apm-lpass-dai: close graphs before opening a new one ASoC: qcom: sdw: do not restart soundwire ports for every prepare ASoC: codecs: wsa883x: mute/unmute PA in correct sequence ASoC: codecs: wsa881x: mute/unmute PA in correct sequence sound/soc/codecs/wsa881x.c | 30 ++++++++++++++++++++- sound/soc/codecs/wsa883x.c | 36 ++++++++++++++++++++----- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 3 +++ sound/soc/qcom/sdw.c | 7 ++--- 4 files changed, 63 insertions(+), 13 deletions(-) -- 2.21.0

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[alsa-devel] echo-cancellation plugin advice?
by Max Klein 27 Oct '23

27 Oct '23

> Well, asym wouldn't be the answer. Basically you'll need to open two > slave PCMs in one plugin. So, it's no simple filter-type plugin. > It's doable, but not easy as it sounds, I guess. Takashi, can you please further explain why asym is not the solution and what you see as a problem? I am also trying to figure out if it would be possible to create such a plugin for ALSA? I am still searching for some good starting points on how to write an ALSA plugin... Jean-Marc Valin (http://lists.linuxaudio.org/pipermail/linux-audio-dev/2005-June/012454.html) already released his work as part of speex but as far I know there is no plugin that has been made out of it? Thanks for any hint! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

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[PATCH] ALSA: dummy: Fix &dpcm->lock deadlock issues
by YE Chengfeng 18 Sep '23

18 Sep '23

The timer dummy_systimer_callback is executed under softirq context, thus other process context code requiring the same lock should disable interrupt. Otherwise there would be potential deadlock issues when the code executing under process context (i.e., dummy_systimer_pointer, dummy_systimer_start, dummy_systimer_stop) is preempted by the timer while holding the lock. Deadlock scenario: dummy_systimer_pointer -> spin_lock(&dpcm->lock); <timer interrupt> -> dummy_systimer_callback -> spin_lock_irqsave(&dpcm->lock, flags); Fix the potential deadlock by using spin_lock_irqsave. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> --- sound/drivers/dummy.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 9c17b49a2ae1..04fb4f17e05c 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -268,19 +268,23 @@ static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) static int dummy_systimer_start(struct snd_pcm_substream *substream) { struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; - spin_lock(&dpcm->lock); + unsigned long flags; + + spin_lock_irqsave(&dpcm->lock, flags); dpcm->base_time = jiffies; dummy_systimer_rearm(dpcm); - spin_unlock(&dpcm->lock); + spin_unlock_irqrestore(&dpcm->lock, flags); return 0; } static int dummy_systimer_stop(struct snd_pcm_substream *substream) { struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; - spin_lock(&dpcm->lock); + unsigned long flags; + + spin_lock_irqsave(&dpcm->lock, flags); del_timer(&dpcm->timer); - spin_unlock(&dpcm->lock); + spin_unlock_irqrestore(&dpcm->lock, flags); return 0; } @@ -320,11 +324,12 @@ dummy_systimer_pointer(struct snd_pcm_substream *substream) { struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; snd_pcm_uframes_t pos; + unsigned long flags; - spin_lock(&dpcm->lock); + spin_lock_irqsave(&dpcm->lock, flags); dummy_systimer_update(dpcm); pos = dpcm->frac_pos / HZ; - spin_unlock(&dpcm->lock); + spin_unlock_irqrestore(&dpcm->lock, flags); return pos; } -- 2.17.1

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[PATCH v3 00/28] Introduce QC USB SND audio offloading support
by Wesley Cheng 29 Aug '23

29 Aug '23

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. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. 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 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 userpsace 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 platorm 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... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Mathias Nyman (1): xhci: Add support to allocate several interrupters Wesley Cheng (27): usb: xhci: Add XHCI APIs to support USB offloading usb: host: xhci-mem: Cleanup pending secondary event ring events ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: Add USB backend ASoC driver for Q6 sound: usb: card: Introduce USB SND platform op callbacks sound: usb: Export USB SND APIs for modules dt-bindings: usb: dwc3: Add snps,num-hc-interrupters definition usb: dwc3: Add DT parameter to specify maximum number of interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present sound: usb: pcm: Export fixed rate check USB SND API sound: usb: Introduce QC USB SND offloading support sound: usb: card: Check for support for requested audio format sound: soc: soc-usb: Add PCM format check API for USB backend sound: soc: qcom: qusb6: Ensure PCM format is supported by USB audio device sound: usb: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend bindings ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: q6afe: Split USB AFE dev_token param into separate API sound: Pass USB SND card and PCM information to SOC USB sound: soc: qdsp6: Add SND kcontrol to select offload device sound: soc: qdsp6: Add SND kcontrol for fetching offload status sound: usb: qc_audio_offload: Use card and PCM index from QMI request sound: usb: card: Allow for rediscovery of connected USB SND devices sound: soc: soc-usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,q6usb-dais.yaml | 49 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 13 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-mem.c | 171 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 2 +- drivers/usb/host/xhci.c | 181 +- drivers/usb/host/xhci.h | 2 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 94 + include/sound/pcm_params.h | 4 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 39 + sound/core/oss/pcm_oss.c | 58 - sound/core/pcm_lib.c | 65 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 50 + sound/soc/qcom/qdsp6/q6afe.c | 208 +- sound/soc/qcom/qdsp6/q6afe.h | 47 +- 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 | 445 ++++ sound/soc/soc-usb.c | 197 ++ sound/usb/Kconfig | 14 + sound/usb/Makefile | 2 +- sound/usb/card.c | 85 + sound/usb/card.h | 31 + sound/usb/endpoint.c | 2 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 29 +- sound/usb/pcm.h | 12 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1790 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 42 files changed, 4646 insertions(+), 100 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.yaml create mode 100644 include/linux/usb/xhci-intr.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/soc-usb.c 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

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[PATCH] ALSA: ac97: Fix possible NULL dereference in snd_ac97_mixer
by Su Hui 23 Aug '23

23 Aug '23

smatch error: sound/pci/ac97/ac97_codec.c:2354 snd_ac97_mixer() error: we previously assumed 'rac97' could be null (see line 2072) remove redundant assignment, return error if rac97 is NULL. Fixes: da3cec35dd3c ("ALSA: Kill snd_assert() in sound/pci/*") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/pci/ac97/ac97_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 9afc5906d662..80a65b8ad7b9 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2069,8 +2069,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, .dev_disconnect = snd_ac97_dev_disconnect, }; - if (rac97) - *rac97 = NULL; + if (!rac97) + return -EINVAL; if (snd_BUG_ON(!bus || !template)) return -EINVAL; if (snd_BUG_ON(template->num >= 4)) -- 2.30.2

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[PATCH 0/4] ASoC: amd: acp: Add sound support for a line of HUAWEI laptops
by Marian Postevca 21 Aug '23

21 Aug '23

This series adds support for a line of HUAWEI laptops with AMD CPUs that connect using the ACP3x module to a ES8336 codec. The codec driver must be extended to support the S32 LE format and the MCLK div by 2 option. MCLK div by 2 is needed for one specific SKU which uses a 48Mhz MCLK which seems to be too high of a frequency for the codec and must be divided by 2. The acp legacy driver must also be extended by using callbacks so that the more complicated handling for this specific codec can be moved outside the more generic ACP code. The last patch tries to avoid anoying pop sounds when the speaker/headphones are enabled/disabled by delaying the handling of the GPIOs and using a mutex to avoid race conditions between the speaker power event callback and the trigger callback. Marian Postevca (4): ASoC: es8316: Enable support for S32 LE format and MCLK div by 2 ASoC: amd: acp: Add support for splitting the codec specific code from the ACP driver ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec ASoC: amd: acp: Improve support for speaker power events sound/soc/amd/acp-config.c | 70 ++ sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-legacy-mach.c | 105 ++- sound/soc/amd/acp/acp-mach-common.c | 8 + sound/soc/amd/acp/acp-mach.h | 67 ++ sound/soc/amd/acp/acp-renoir.c | 4 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 615 ++++++++++++++++++ sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h | 12 + sound/soc/codecs/es8316.c | 21 +- sound/soc/codecs/es8316.h | 3 + 10 files changed, 886 insertions(+), 21 deletions(-) create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h -- 2.39.1

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Fwd: [Bug 217440] New: ALC236 audio disappears from HP 15z-fc000 on warm boot
by Joseph C. Sible 16 Aug '23

16 Aug '23

Hello, Forwarding my regression bug report submission per the instructions in "Reporting regressions". #regzbot introduced: 5aec989130 https://bugzilla.kernel.org/show_bug.cgi?id=217440 Regards, Joseph C. Sible ---------- Forwarded message --------- From: <bugzilla-daemon(a)kernel.org> Date: Sat, May 13, 2023 at 3:46 PM Subject: [Bug 217440] New: ALC236 audio disappears from HP 15z-fc000 on warm boot To: <josephcsible(a)gmail.com> https://bugzilla.kernel.org/show_bug.cgi?id=217440 Bug ID: 217440 Summary: ALC236 audio disappears from HP 15z-fc000 on warm boot Product: Drivers Version: 2.5 Kernel Version: 6.3.2 Hardware: All OS: Linux Status: NEW Severity: normal Priority: P3 Component: Sound(ALSA) Assignee: perex(a)perex.cz Reporter: josephcsible(a)gmail.com Regression: Yes Bisected 5aec98913095ed3b4424ed6c5fdeb6964e9734da commit-id: On an HP 15z-fc000 laptop that uses Realtek ALC236 for sound, once the system is warm booted ("reboot" from within Linux), sound stops working completely and the device doesn't even show up anymore. Once this happens, the only way to get it to come back is to cold boot ("poweroff" from within Linux, wait for it to turn completely off, and turn it back on with the physical power button). In particular, once you boot from a "bad" kernel, no number of warm boots with "good" kernels will ever bring sound back. I bisected this to commit 5aec989, which first appeared in 5.15-rc6. In addition, I confirmed by doing a custom build of 6.3.2 with reverts of both f30741c (just to avoid a conflict) and 5aec989, which works correctly for me. -- You may reply to this email to add a comment. You are receiving this mail because: You reported the bug.

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Audio output heavily distorted on Xen PV dom0 if not recording (Intel ADL)
by Marek Marczykowski-Górecki 01 Aug '23

01 Aug '23

Hi, I have a rather interesting issue on HP EliteBook 1040, with Intel ADL HDA. Sometimes (often) audio output is heavily distorted, but then opening pavucontrol or just launching `parecord /dev/null` fixes the issue instantly. But, when the mixer or recording is closed, then few seconds later issue comes back. You can hear this at https://cloud.marmarek.net/s/46prqZnwrqDkBdD Initially recording was running, then about 10s stopped and you can hear the issue few seconds later. Then I starter recording again, and the issue is gone. This is how it should sound: https://youtube.com/watch?v=4Tr0otuiQuU I do not see any kernel messages when issue happens. The issue happens on Linux 6.1.26, 6.2.10, and 6.3.2. I haven't tested other versions. It happens only when running under Xen (in PV dom0). The very same Linux started directly works fine. Full dmesg of both native and Xen boots is at https://gist.github.com/marmarek/548ee79282446344a172109bcd943930 When running Linux 6.3.2 there, I noticed that just after boot first playback is okay, but if I start and stop recording it breaks. And then after some time of idle it fixes itself again. I have not observed this behavior on 6.2.10 (it was broken initially too), but I could be also doing some things differently. I have few other ADL-based systems (but not HP), and they do not show this issue, only this HP laptop is affected. Example dmesg from another ADL-based system: https://gist.github.com/marmarek/38c5ba3ec58f9300fbc1842ddc9fa4f5 I'm not sure if that's the relevant difference, but comparing dmesg from working and not working systems, I see those on boot for the problematic case: snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 sof-audio-pci-intel-tgl 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver sof-audio-pci-intel-tgl 0000:00:1f.3: enabling device (0000 -> 0002) xen: registering gsi 16 triggering 0 polarity 1 Already setup the GSI :16 sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 sof-audio-pci-intel-tgl 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) sof-audio-pci-intel-tgl 0000:00:1f.3: use msi interrupt mode sof-audio-pci-intel-tgl 0000:00:1f.3: hda codecs found, mask 5 sof-audio-pci-intel-tgl 0000:00:1f.3: using HDA machine driver skl_hda_dsp_generic now sof-audio-pci-intel-tgl 0000:00:1f.3: DMICs detected in NHLT tables: 2 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 sof-audio-pci-intel-tgl 0000:00:1f.3: unknown sof_ext_man header type 3 size 0x30 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 sof-audio-pci-intel-tgl 0000:00:1f.3: Topology: ABI 3:22:1 Kernel ABI 3:23:0 skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: Parent card not yet available, widget card binding deferred cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: Firmware version: 3 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: cirrus/cs35l41-dsp1-spk-prot-103c8974.wmfw: Fri 27 Aug 2021 14:58:19 W. Europe Daylight Time cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: Firmware: 400a4 vendor: 0x2 v0.43.1, 2 algorithms cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: 0: ID cd v29.63.1 XM@94 YM@e cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: 1: ID f20b v0.1.0 XM@176 YM@0 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: spk-prot: E:\Amp Tuning\HP\840\0527\103C8972_220527.bin snd_hda_codec_realtek ehdaudio0D0: bound spi0-CSC3551:00-cs35l41-hda.0 (ops cs35l41_hda_comp_ops [snd_hda_scodec_cs35l41]) cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: Firmware version: 3 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: cirrus/cs35l41-dsp1-spk-prot-103c8974.wmfw: Fri 27 Aug 2021 14:58:19 W. Europe Daylight Time cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: Firmware: 400a4 vendor: 0x2 v0.43.1, 2 algorithms cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: 0: ID cd v29.63.1 XM@94 YM@e cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: 1: ID f20b v0.1.0 XM@176 YM@0 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: spk-prot: E:\Amp Tuning\HP\840\0527\103C8972_220527.bin snd_hda_codec_realtek ehdaudio0D0: bound spi0-CSC3551:00-cs35l41-hda.1 (ops cs35l41_hda_comp_ops [snd_hda_scodec_cs35l41]) Any ideas? -- Best Regards, Marek Marczykowski-Górecki Invisible Things Lab

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[PATCH 00/11] Even more msm bindings fixes
by Konrad Dybcio 25 Jul '23

25 Jul '23

This series contains couple more random dt schema warning fixes, centered around linux-msm. All of the patches (except) the last two are pretty much independent. Patch 1 is not only a bindings fix, but also a functional one. Signed-off-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> --- Konrad Dybcio (11): arm64: dts: qcom: sc7180: Fix DSI0_PHY reg-names arm64: dts: qcom: msm8939: Drop "qcom,idle-state-spc" compatible arm64: dts: qcom: msm8939: Add missing 'cache-unified' to L2 arm64: dts: qcom: apq8039-t2: Drop inexistent property dt-bindings: sound: Convert pm8916-wcd-analog-codec to YAML dt-bindings: mfd: qcom,spmi-pmic: Reference pm8916 wcd analog codec schema dt-bindings: iommu: qcom_iommu: Allow 'tbu' clock dt-bindings: usb: ci-hdrc-usb2: Fix handling pinctrl properties dt-bindings: arm: msm: kpss-acc: Make the optional reg truly optional dt-bindings: usb: dwc3: Allow just 1 interrupt for MSM8996 arm64: dts: qcom: msm8996: Add missing interrupt to the USB2 controller .../devicetree/bindings/arm/msm/qcom,kpss-acc.yaml | 1 + .../devicetree/bindings/iommu/qcom,iommu.yaml | 4 + .../devicetree/bindings/mfd/qcom,spmi-pmic.yaml | 2 +- .../bindings/sound/qcom,msm8916-wcd-analog.txt | 101 ------------- .../sound/qcom,pm8916-wcd-analog-codec.yaml | 160 +++++++++++++++++++++ .../devicetree/bindings/usb/ci-hdrc-usb2.yaml | 27 +--- .../devicetree/bindings/usb/qcom,dwc3.yaml | 2 +- arch/arm64/boot/dts/qcom/apq8039-t2.dts | 1 - arch/arm64/boot/dts/qcom/msm8939.dtsi | 4 +- arch/arm64/boot/dts/qcom/msm8996.dtsi | 3 + arch/arm64/boot/dts/qcom/sc7180.dtsi | 4 +- 11 files changed, 182 insertions(+), 127 deletions(-) --- base-commit: 3eedd211ad93c322fb360b83a3d76a2c6cd622dc change-id: 20230627-topic-more_bindings-325a6f4b93a7 Best regards, -- Konrad Dybcio <konrad.dybcio(a)linaro.org>

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[PATCH v1 0/5] Add PWM-DAC audio support for StarFive JH7110 RISC-V SoC
by Hal Feng 25 Jul '23

25 Jul '23

This patchset adds PWM-DAC audio support for the StarFive JH7110 SoC. The PWM-DAC module does not require a hardware codec, so add a dummy codec driver for it. The fourth patch depends on the patch [1]. The fifth patch depends on the patchset [2], patch [3]. [1] https://lore.kernel.org/all/20230526145402.450-3-walker.chen@starfivetech.c… [2] https://lore.kernel.org/all/20230518101234.143748-1-xingyu.wu@starfivetech.… [3] https://lore.kernel.org/all/20230322094820.24738-5-walker.chen@starfivetech… Hal Feng (5): ASoC: dt-bindings: Add StarFive JH7110 dummy PWM-DAC transmitter ASoC: codecs: Add StarFive JH7110 dummy PWM-DAC transmitter driver ASoC: dt-bindings: Add StarFive JH7110 PWM-DAC controller ASoC: starfive: Add JH7110 PWM-DAC driver riscv: dts: starfive: Add JH7110 PWM-DAC support .../sound/starfive,jh7110-pwmdac-dit.yaml | 38 + .../sound/starfive,jh7110-pwmdac.yaml | 76 ++ MAINTAINERS | 8 + .../jh7110-starfive-visionfive-2.dtsi | 50 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 13 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jh7110_pwmdac_transmitter.c | 74 ++ sound/soc/starfive/Kconfig | 9 + sound/soc/starfive/Makefile | 1 + sound/soc/starfive/jh7110_pwmdac.c | 787 ++++++++++++++++++ 11 files changed, 1062 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac-dit.yaml create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml create mode 100644 sound/soc/codecs/jh7110_pwmdac_transmitter.c create mode 100644 sound/soc/starfive/jh7110_pwmdac.c base-commit: 45a3e24f65e90a047bef86f927ebdc4c710edaa1 prerequisite-patch-id: 8c735dffc6d5388a35a76b16e914a2f9722ad979 prerequisite-patch-id: ffa1f5831e75722c9f41603f009f762c9fd525e2 prerequisite-patch-id: 36e69700dfc0375b950b0e23086ed3b722cb84a4 prerequisite-patch-id: 0b49b996d7a404ea548e1734c12933ec749e92b9 prerequisite-patch-id: 81f7c65712c4901a7a178ddcd98ffc55f3b473ff prerequisite-patch-id: f342fbf594014b072378528bea94c01fb2186e1a prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: d5abfba63fc07ff97b5023911513c260bb7a53e1 prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 prerequisite-patch-id: 2f7aca99e714a4c590a91baa015080ac0902814d prerequisite-patch-id: 32cabbc4e7a97ec14d5c28a477fa483784f86709 prerequisite-patch-id: d449b1957dd77c2537c38585daa75974c94c529a -- 2.38.1

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