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

[PATCH] ASoC: dwc: fix typo in comment
by Myunguk Kim 13 Sep '23

13 Sep '23

This fixes typo in comment Signed-off-by: Myunguk Kim <mwkim(a)gaonchips.com> --- sound/soc/dwc/dwc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 71ff894035a4..45e55e550924 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -138,7 +138,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) irq_valid = true; } - /* Error Handling: TX */ + /* Error Handling: RX */ if (isr[i] & ISR_RXFO) { dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; -- 2.34.1

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[PATCH] ASoC: imx-rpmsg: Set ignore_pmdown_time for dai_link
by Chancel Liu 13 Sep '23

13 Sep '23

i.MX rpmsg sound cards work on codec slave mode. MCLK will be disabled by CPU DAI driver in hw_free(). Some codec requires MCLK present at power up/down sequence. So need to set ignore_pmdown_time to power down codec immediately before MCLK is turned off. Take WM8962 as an example, if MCLK is disabled before DAPM power down playback stream, FIFO error will arise in WM8962 which will have bad impact on playback next. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/imx-rpmsg.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 3c7b95db2eac..d2ecd5e821b1 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -89,6 +89,13 @@ static int imx_rpmsg_probe(struct platform_device *pdev) SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + /* i.MX rpmsg sound cards work on codec slave mode. MCLK will be + * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK + * present at power up/down sequence. So need to set ignore_pmdown_time + * to power down codec immediately before MCLK is turned off. + */ + data->dai.ignore_pmdown_time = 1; + /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); if (ret) { -- 2.25.1

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time to remove COPYING.GPL from alsa-plugins?
by GitHub issues - opened 13 Sep '23

13 Sep '23

alsa-project/alsa-plugins issue #54 was opened from michael-mw-wang: It seems the only thing in alsa-plugins that could have a GPL license is rate/rate_samplerate.c, which has this at the top of the file: * This plugin code is supposed to be used and distributed primarily * under GPL v2 or later, in order to follow the license of libsamplerate. * However, if you already own a commercial license to use libsamplerate * for dynamic linking, this plugin code can be used and distributed also * under LGPL v2.1 or later. In 2016, libsamplerate dropped its commercial license requirement and was released with the 2-Clause BSD license. (See http://libsndfile.github.io/libsamplerate//license.html) So I was wondering if the license for rate_samplerate.c can be updated to remove the GPLv2, which means the COPYING.GPL file can also be removed from the alsa-plugins package? Thank you! Issue URL : https://github.com/alsa-project/alsa-plugins/issues/54 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH v5 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 13 Sep '23

13 Sep '23

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 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 Changelog -------------------------------------------- Changes in v5: - Removed some unnescessary 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 userpsace 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. 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. Mathias Nyman (3): xhci: add support to allocate several interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (29): usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index 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 dt-bindings: usb: xhci: Add 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: qcom: Add USB QMI definitions 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 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: soc: qcom: q6usb: Add headphone jack for offload connection 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 | 53 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 13 + .../devicetree/bindings/usb/usb-xhci.yaml | 4 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 4 + drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-hub.c | 29 +- drivers/usb/host/xhci-mem.c | 102 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 48 +- drivers/usb/host/xhci-sideband.c | 195 ++ drivers/usb/host/xhci.c | 111 +- drivers/usb/host/xhci.h | 105 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 86 + include/linux/usb/xhci-sideband.h | 57 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 51 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 55 + sound/soc/qcom/qdsp6/q6afe.c | 206 +- 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 | 452 ++++ sound/soc/soc-usb.c | 200 ++ sound/usb/Kconfig | 15 + sound/usb/Makefile | 2 +- sound/usb/card.c | 108 + sound/usb/card.h | 23 + sound/usb/endpoint.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 86 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1837 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 45 files changed, 4863 insertions(+), 192 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.yaml create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-intr.h 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/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] ASoC: cs35l56: Disable low-power hibernation mode
by Richard Fitzgerald 13 Sep '23

13 Sep '23

Do not allow the CS35L56 to be put into its lowest power "hibernation" mode. This only affects I2C because "hibernation" is already disabled on SPI and SoundWire. Recent firmwares need a different wake-up sequence. Until that sequence has been specified, the chip "hibernation" mode must be disabled otherwise it can intermittently fail to wake. THIS WILL NOT APPLY CLEANLY TO 6.5 AND EARLIER: We will send a separate backport patch to stable. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-i2c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 9f4f2f4f23f5..d10e0e2380e8 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -27,7 +27,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client) return -ENOMEM; cs35l56->base.dev = dev; - cs35l56->base.can_hibernate = true; i2c_set_clientdata(client, cs35l56); cs35l56->base.regmap = devm_regmap_init_i2c(client, regmap_config); -- 2.30.2

2 1

[PATCH 0/6] ASoC: rt5640: Fix various IRQ handling issues
by Hans de Goede 13 Sep '23

13 Sep '23

Hi All, The recent(ish) rt5640 changes to add HDA header jack-detect support and the related suspend/resume handling fixes have introduced several issues with IRQ handling on boards not using the HDA header jack-detect support. This series fixes these issues, see the individual commit messages for details. Regards, Hans Hans de Goede (6): ASoC: rt5640: Revert "Fix sleep in atomic context" ASoC: rt5640: Fix sleep in atomic context ASoC: rt5640: Do not disable/enable IRQ twice on suspend/resume ASoC: rt5640: Enable the IRQ on resume after configuring jack-detect ASoC: rt5640: Fix IRQ not being free-ed for HDA jack detect mode ASoC: rt5640: Only cancel jack-detect work on suspend if active sound/soc/codecs/rt5640.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) -- 2.41.0

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AIF2 ADC Stereo Capture Route is unitialised in alsa-ucm-conf
by GitHub issues - edited 12 Sep '23

12 Sep '23

alsa-project/alsa-ucm-conf issue #351 was edited from broukema: DESCRIPTION: `AIF2 ADC Stereo Capture Route` is not initialised in `alsa-ucm-conf`. This implies that if its setting is significant, any accidental change by the user (using any of the various `alsa` tools) to a wrong setting will not be corrected by any of the three `PinePhone/*.conf` files, leading, in particular, to audio errors during a phone call. This was discovered by testing variations on the Mobian version of the PinePhone alsa-ucm-conf files. CODING ARGUMENT: All parameters that can significantly affect the audio system should be either initialised or overridden in configuration files that the user expects to necessarily work, such as those for phone calls. In `alsa-ucm-conf` commit b68aa52 (one beyond v1.2.10): ```` $ grep -ri "AIF2 ADC Stereo Capture" . $ ```` shows that `AIF2 ADC Stereo Capture Route` is not set for any device at all (PinePhone or other). HARDWARE ARGUMENT: For the PinePhone (PP), Megi recommends in https://xnux.eu/devices/feature/audio-pp.html that `AIF2 ADC Stereo Capture Route` is set to `Mix Mono`. TESTING ARGUMENT: Experimentation with different stored alsa settings (from ```` sudo alsactl store -f - > YYYYMMDD_example_alsa_settings ```` ) showed that settings files whose only substantial difference was in the setting of `AIF2 ADC Stereo Capture Route` had the effect that audio from the PP was heard on the remote phone when the value was `Mix Mono` and was not heard with the value `Stereo`. Testing was done with the equivalent of https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… with the settings file https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… that has the `Mix Mono` value selected. RECOMMENDATIONS: * `AIF2 ADC Stereo Capture Route` should be initialised or set to *something* in one of the .conf files (in the PP case, most likely in `VoiceCall.conf`) to prevent users who experiment and have difficulty remembering which of the 50 PP alsa parameters should be set in which way from accidentally sabotaging their otherwise working audio calls; * Based on Megi's 2020 page and my testing, it seems that the correct value should be `Mix Mono`. COMMENT: While setting all 50 of the alsa settings ```` $ sudo alsactl store -f - |grep "name " |wc 50 258 1788 ```` in the .conf files is most likely unnecessary, there could be others that are necessary too. Ideally, someone who really understands how the PP audio system works (including Megi's two diagrams at https://xnux.eu/devices/feature/audio-pp.html) should be able to make these recommendations. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/351 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Bug: AIF2 ADC Stereo Capture Route is unitialised in alsa-ucm-conf
by GitHub issues - opened 12 Sep '23

12 Sep '23

alsa-project/alsa-ucm-conf issue #351 was opened from broukema: DESCRIPTION: `AIF2 ADC Stereo Capture Route` is not initialised in `alsa-ucm-conf`. This implies that if its setting is significant, any accidental change by the user (using any of the various `alsa` tools) to a wrong setting will not be corrected by any of the three `PinePhone/*.conf` files, leading, in particular, to audio errors during a phone call. This was discovered by testing variations on the Mobian version of the PinePhone alsa-ucm-conf files. CODING ARGUMENT: All parameters that can significantly affect the audio system should be either initialised or overridden in configuration files that the user expects to necessarily work, such as those for phone calls. In `alsa-ucm-conf` commit b68aa52 (one beyond v1.2.10): ```` $ grep -ri "AIF2 ADC Stereo Capture" . $ ```` shows that `AIF2 ADC Stereo Capture Route` is not set for any device at all (PinePhone or other). HARDWARE ARGUMENT: For the PinePhone (PP), Megi recommends in https://xnux.eu/devices/feature/audio-pp.html that `AIF2 ADC Stereo Capture Route` is set to `Mix Mono`. TESTING ARGUMENT: Experimentation with different stored alsa settings (from ```` sudo alsactl store -f - > YYYYMMDD_example_alsa_settings ```` ) showed that settings files whose only substantial difference was in the setting of `AIF2 ADC Stereo Capture Route` had the effect that audio from the PP was heard on the remote phone when the value was `Mix Mono` and was not heard with the value `Stereo`. Testing was done with the equivalent of https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… with the settings file https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… that has the `Mix Mono` value selected. RECOMMENDATIONS: * `AIF2 ADC Stereo Capture Route` should be initialised or set to *something* in one of the .conf files (in the PP case, most likely in `VoiceCall.conf`) to prevent users who experiment and have difficulty remembering which of the 50 PP alsa parameters should be set in which way from accidentally sabotaging their otherwise working audio calls; * Based on Megi's 2020 page and my testing, it seems that the correct value should be `Mix Mono`. COMMENT: While setting all 50 of the alsa settings ```` $ sudo alsactl store -f - |grep "name " |wc 50 258 1788 ```` in the .conf files is most likely unnecessary, there could be others that are necessary too. Ideally, someone who really understands how the PP audio system works (including Megi's two diagrams at https://xnux.eu/devices/feature/audio-pp.html) should be able to make these recommendations. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/351 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Remove optional dependency on GTK 2
by GitHub issues - opened 12 Sep '23

12 Sep '23

alsa-project/alsa-tools issue #18 was opened from ChaseKnowlden: GTK 2 is not maintained since December 2020. Issue URL : https://github.com/alsa-project/alsa-tools/issues/18 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH] ASoC: rsnd: remove unneeded of_node_put()
by Kuninori Morimoto 12 Sep '23

12 Sep '23

The loop is not using "node", of_node_put(node) is not needed. Cc: Julia Lawall <julia.lawall(a)inria.fr> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e29c2fee95219..7552fa0a2578d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1302,7 +1302,6 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph) i++; if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); - of_node_put(node); break; } } -- 2.25.1

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[PATCH 0/2] SM6115 TX Macro
by Konrad Dybcio 12 Sep '23

12 Sep '23

Like most Qualcomm SoCs, SM6115 has a TX Macro. Only some minor changes were required. Signed-off-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> --- Konrad Dybcio (2): ASoC: dt-bindings: qcom,lpass-tx-macro: Add SM6115 ASoC: codecs: lpass-tx-macro: Add SM6115 support .../bindings/sound/qcom,lpass-tx-macro.yaml | 18 ++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 2 ++ sound/soc/codecs/lpass-tx-macro.c | 22 +++++++++++++++------- 3 files changed, 35 insertions(+), 7 deletions(-) --- base-commit: 6269320850097903b30be8f07a5c61d9f7592393 change-id: 20230825-topic-6115tx-b523a0209639 Best regards, -- Konrad Dybcio <konrad.dybcio(a)linaro.org>

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[tiwai-sound:test/format-truncation-fixes 20/23] sound/pci/cmipci.c:3106:35: warning: '%d' directive output may be truncated writing between 1 and 11 bytes into a region of size 8
by kernel test robot 12 Sep '23

12 Sep '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/format-truncation-fixes head: 5eb64dfc2dc9c05e50f2f122d77b28cede293b6b commit: 1c87c035c832a55414b42288df76bb10a5b11b34 [20/23] ALSA: cmipci: Fix -Wformat-truncation warning config: alpha-allyesconfig (https://download.01.org/0day-ci/archive/20230913/202309130112.VUtCZ4wu-lkp@…) compiler: alpha-linux-gcc (GCC) 13.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20230913/202309130112.VUtCZ4wu-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202309130112.VUtCZ4wu-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/pci/cmipci.c: In function 'snd_cmipci_create': >> sound/pci/cmipci.c:3106:35: warning: '%d' directive output may be truncated writing between 1 and 11 bytes into a region of size 8 [-Wformat-truncation=] 3106 | " (model %d)", cm->chip_version); | ^~ sound/pci/cmipci.c:3106:26: note: directive argument in the range [-2147483648, 67] 3106 | " (model %d)", cm->chip_version); | ^~~~~~~~~~~~~ sound/pci/cmipci.c:3105:17: note: 'snprintf' output between 11 and 21 bytes into a destination of size 16 3105 | snprintf(modelstr, sizeof(modelstr), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 3106 | " (model %d)", cm->chip_version); | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +3106 sound/pci/cmipci.c 3041 3042 /* initialize codec registers */ 3043 snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_RESET); 3044 snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_RESET); 3045 snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0); /* disable ints */ 3046 snd_cmipci_ch_reset(cm, CM_CH_PLAY); 3047 snd_cmipci_ch_reset(cm, CM_CH_CAPT); 3048 snd_cmipci_write(cm, CM_REG_FUNCTRL0, 0); /* disable channels */ 3049 snd_cmipci_write(cm, CM_REG_FUNCTRL1, 0); 3050 3051 snd_cmipci_write(cm, CM_REG_CHFORMAT, 0); 3052 snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_ENDBDAC|CM_N4SPK3D); 3053 #if CM_CH_PLAY == 1 3054 snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); 3055 #else 3056 snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); 3057 #endif 3058 if (cm->chip_version) { 3059 snd_cmipci_write_b(cm, CM_REG_EXT_MISC, 0x20); /* magic */ 3060 snd_cmipci_write_b(cm, CM_REG_EXT_MISC + 1, 0x09); /* more magic */ 3061 } 3062 /* Set Bus Master Request */ 3063 snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_BREQ); 3064 3065 /* Assume TX and compatible chip set (Autodetection required for VX chip sets) */ 3066 switch (pci->device) { 3067 case PCI_DEVICE_ID_CMEDIA_CM8738: 3068 case PCI_DEVICE_ID_CMEDIA_CM8738B: 3069 if (!pci_dev_present(intel_82437vx)) 3070 snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_TXVX); 3071 break; 3072 default: 3073 break; 3074 } 3075 3076 if (cm->chip_version < 68) { 3077 val = pci->device < 0x110 ? 8338 : 8738; 3078 } else { 3079 switch (snd_cmipci_read_b(cm, CM_REG_INT_HLDCLR + 3) & 0x03) { 3080 case 0: 3081 val = 8769; 3082 break; 3083 case 2: 3084 val = 8762; 3085 break; 3086 default: 3087 switch ((pci->subsystem_vendor << 16) | 3088 pci->subsystem_device) { 3089 case 0x13f69761: 3090 case 0x584d3741: 3091 case 0x584d3751: 3092 case 0x584d3761: 3093 case 0x584d3771: 3094 case 0x72848384: 3095 val = 8770; 3096 break; 3097 default: 3098 val = 8768; 3099 break; 3100 } 3101 } 3102 } 3103 sprintf(card->shortname, "C-Media CMI%d", val); 3104 if (cm->chip_version < 68) 3105 snprintf(modelstr, sizeof(modelstr), > 3106 " (model %d)", cm->chip_version); 3107 else 3108 modelstr[0] = '\0'; 3109 scnprintf(card->longname, sizeof(card->longname), 3110 "%s%s at %#lx, irq %i", 3111 card->shortname, modelstr, cm->iobase, cm->irq); 3112 3113 if (cm->chip_version >= 39) { 3114 val = snd_cmipci_read_b(cm, CM_REG_MPU_PCI + 1); 3115 if (val != 0x00 && val != 0xff) { 3116 if (mpu_port[dev]) 3117 iomidi = cm->iobase + CM_REG_MPU_PCI; 3118 integrated_midi = 1; 3119 } 3120 } 3121 if (!integrated_midi) { 3122 val = 0; 3123 iomidi = mpu_port[dev]; 3124 switch (iomidi) { 3125 case 0x320: val = CM_VMPU_320; break; 3126 case 0x310: val = CM_VMPU_310; break; 3127 case 0x300: val = CM_VMPU_300; break; 3128 case 0x330: val = CM_VMPU_330; break; 3129 default: 3130 iomidi = 0; break; 3131 } 3132 if (iomidi > 0) { 3133 snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val); 3134 /* enable UART */ 3135 snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_UART_EN); 3136 if (inb(iomidi + 1) == 0xff) { 3137 dev_err(cm->card->dev, 3138 "cannot enable MPU-401 port at %#lx\n", 3139 iomidi); 3140 snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, 3141 CM_UART_EN); 3142 iomidi = 0; 3143 } 3144 } 3145 } 3146 3147 if (cm->chip_version < 68) { 3148 err = snd_cmipci_create_fm(cm, fm_port[dev]); 3149 if (err < 0) 3150 return err; 3151 } 3152 3153 /* reset mixer */ 3154 snd_cmipci_mixer_write(cm, 0, 0); 3155 3156 snd_cmipci_proc_init(cm); 3157 3158 /* create pcm devices */ 3159 pcm_index = pcm_spdif_index = 0; 3160 err = snd_cmipci_pcm_new(cm, pcm_index); 3161 if (err < 0) 3162 return err; 3163 pcm_index++; 3164 err = snd_cmipci_pcm2_new(cm, pcm_index); 3165 if (err < 0) 3166 return err; 3167 pcm_index++; 3168 if (cm->can_ac3_hw || cm->can_ac3_sw) { 3169 pcm_spdif_index = pcm_index; 3170 err = snd_cmipci_pcm_spdif_new(cm, pcm_index); 3171 if (err < 0) 3172 return err; 3173 } 3174 3175 /* create mixer interface & switches */ 3176 err = snd_cmipci_mixer_new(cm, pcm_spdif_index); 3177 if (err < 0) 3178 return err; 3179 3180 if (iomidi > 0) { 3181 err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, 3182 iomidi, 3183 (integrated_midi ? 3184 MPU401_INFO_INTEGRATED : 0) | 3185 MPU401_INFO_IRQ_HOOK, 3186 -1, &cm->rmidi); 3187 if (err < 0) 3188 dev_err(cm->card->dev, 3189 "no UART401 device at 0x%lx\n", iomidi); 3190 } 3191 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

topology: pre-processor: Add support for enum controls
by GitHub pull_request - opened 12 Sep '23

12 Sep '23

alsa-project/alsa-utils pull request #236 was opened from ranj063: Add support for adding enum controls in the topology pre-processor. Request URL : https://github.com/alsa-project/alsa-utils/pull/236 Patch URL : https://github.com/alsa-project/alsa-utils/pull/236.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] ALSA: hda/tas2781: Support ACPI_ID both TXNW2781 and TIAS2781
by Shenghao Ding 12 Sep '23

12 Sep '23

Support ACPI_ID both TXNW2781 and TIAS2781, TXNW2781 is the only one legal ACPI ID, TIAS2781 is the widely-used ACPI ID named by our customers, so far it is not registered. We have discussed this with them, they requested TIAS2781 must be supported for the laptops already released to market, their new laptops will switch to TXNW2781. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v1: - Add TXNW2781 into tas2781_acpi_hda_match and move it to the top - Redefine tas2781_generic_fixup, remove hid param - TIAS2781 has been used by our customers, see following dstd.dsl. We have discussed this with them, they requested TIAS2781 must be supported for the laptops already released to market, their new laptops will switch to TXNW2781 Name (_HID, "TIAS2781") // _HID: Hardware ID Name (_UID, Zero) // _UID: Unique ID Method (_SUB, 0, NotSerialized) // _SUB: Subsystem ID { If ((SPID == Zero)) { Return ("17AA3886") } If ((SPID == One)) { Return ("17AA3884") } } - Add TXNW2781 support in comp_match_tas2781_dev_name --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++++++--------------- sound/pci/hda/tas2781_hda_i2c.c | 33 ++++++++++++++++++------------ 2 files changed, 40 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b7e78bfcff..6dae58a8ef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6770,24 +6770,35 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +/* TIAS2781 is the unofficial ACPI id, but widely used in current devices. + * TXNW2781 is the official ACPI id, and will be used in the new devices. + * Check TIAS2781 or TXNW2781 + */ static int comp_match_tas2781_dev_name(struct device *dev, void *data) { - struct scodec_dev_name *p = data; + const char c[][10] = { "TXNW2781", "TIAS2781" }; const char *d = dev_name(dev); - int n = strlen(p->bus); + const char *bus = data; + int n = strlen(bus), i; char tmp[32]; /* check the bus name */ - if (strncmp(d, p->bus, n)) + if (strncmp(d, bus, n)) return 0; /* skip the bus number */ if (isdigit(d[n])) n++; - /* the rest must be exact matching */ - snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); - return !strcmp(d + n, tmp); + for (i = 0; i < ARRAY_SIZE(c); i++) { + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", c[i]); + + if (!strcmp(d + n, tmp)) + return 1; + } + + return 0; } static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, @@ -6824,24 +6835,17 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } static void tas2781_generic_fixup(struct hda_codec *cdc, int action, - const char *bus, const char *hid) + const char *bus) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct scodec_dev_name *rec; int ret; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); - if (!rec) - return; - rec->bus = bus; - rec->hid = hid; - rec->index = 0; spec->comps[0].codec = cdc; component_match_add(dev, &spec->match, - comp_match_tas2781_dev_name, rec); + comp_match_tas2781_dev_name, (void *)bus); ret = component_master_add_with_match(dev, &comp_master_ops, spec->match); if (ret) @@ -6888,7 +6892,7 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { - tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); + tas2781_generic_fixup(cdc, action, "i2c"); } /* for alc295_fixup_hp_top_speakers */ diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index fb80280293..8493952305 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -65,6 +65,16 @@ enum calib_data { CALIB_MAX }; +/* TIAS2781 is the unofficial ACPI id, but widely used in current devices. + * TXNW2781 is the official ACPI id, and will be used in the new devices. + */ +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TIAS2781", 0 }, + {"TXNW2781", 1 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) { struct tasdevice_priv *tas_priv = data; @@ -644,20 +654,23 @@ static void tas2781_hda_remove(struct device *dev) static int tas2781_hda_i2c_probe(struct i2c_client *clt) { struct tasdevice_priv *tas_priv; - const char *device_name; - int ret; + int ret, i; - if (strstr(dev_name(&clt->dev), "TIAS2781")) - device_name = "TIAS2781"; - else - return -ENODEV; + /* Check TIAS2781 or TXNW2781 */ + for (i = 0; i < ARRAY_SIZE(tas2781_acpi_hda_match); i++) + if (strstr(dev_name(&clt->dev), tas2781_acpi_hda_match[i].id)) + break; + + if (i == ARRAY_SIZE(tas2781_acpi_hda_match)) + return dev_err_probe(tas_priv->dev, -ENODEV, + "Device not available\n"); tas_priv = tasdevice_kzalloc(clt); if (!tas_priv) return -ENOMEM; tas_priv->irq_info.irq = clt->irq; - ret = tas2781_read_acpi(tas_priv, device_name); + ret = tas2781_read_acpi(tas_priv, tas2781_acpi_hda_match[i].id); if (ret) return dev_err_probe(tas_priv->dev, ret, "Platform not supported\n"); @@ -822,12 +835,6 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { {} }; -static const struct acpi_device_id tas2781_acpi_hda_match[] = { - {"TIAS2781", 0 }, - {} -}; -MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); - static struct i2c_driver tas2781_hda_i2c_driver = { .driver = { .name = "tas2781-hda", -- 2.34.1

4 3

[PATCH] ASoC: amd: yc: Fix non-functional mic on Lenovo 82QF and 82UG
by August Wikerfors 12 Sep '23

12 Sep '23

Like the Lenovo 82TL and 82V2, the Lenovo 82QF (Yoga 7 14ARB7) and 82UG (Legion S7 16ARHA7) both need a quirk entry for the internal microphone to function. Commit c008323fe361 ("ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ") restricted the quirk that previously matched "82" to "82V2", breaking microphone functionality on these devices. Fix this by adding specific quirks for these models, as was done for the Lenovo 82TL. Fixes: c008323fe361 ("ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ") Closes: https://github.com/tomsom/yoga-linux/issues/51 Link: https://bugzilla.kernel.org/show_bug.cgi?id=208555#c780 Cc: stable(a)vger.kernel.org Signed-off-by: August Wikerfors <git(a)augustwikerfors.se> --- sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 59aa2e9d3a79..94e9eb8e73f2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -213,6 +213,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82QF"), + } + }, { .driver_data = &acp6x_card, .matches = { @@ -220,6 +227,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "82TL"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82UG"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.42.0

2 1

[PATCH] ASoC: amd: yc: Fix a non-functional mic on Lenovo 82TL
by Mario Limonciello 12 Sep '23

12 Sep '23

Lenovo 82TL has DMIC connected like 82V2 does. Also match 82TL. Reported-by: wildjim(a)kiwinet.org Closes: https://bugzilla.kernel.org/show_bug.cgi?id=217063 Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index b304b3562c82..7c3b32d6f2fb 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -213,6 +213,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82TL"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.34.1

2 1

[PATCH v4 00/11] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 12 Sep '23

12 Sep '23

Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. Compared to previous version, I picked up Pierre's changes to add a probe_no_wq, added remove_no_wq myself and then fixed the SOF changes to use it. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org Maarten Lankhorst (9): ALSA: hda/intel: Fix error handling in azx_probe() ALSA: hda/i915: Allow override of gpu binding. ALSA: hda/i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda/i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ALSA: hda/intel: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Move binding to display driver outside of deferred probe ALSA: hda/i915: Remove extra argument from snd_hdac_i915_init Pierre-Louis Bossart (2): ASoC: SOF: core: add 'no_wq' probe and remove callbacks ASoC: SOF: Intel: hda: start splitting the probe sound/hda/hdac_i915.c | 24 ++++++----- sound/pci/hda/hda_intel.c | 60 ++++++++++++++-------------- sound/soc/intel/avs/core.c | 13 ++++-- sound/soc/intel/skylake/skl.c | 31 +++++--------- sound/soc/sof/core.c | 11 ++++- sound/soc/sof/intel/hda-common-ops.c | 2 + sound/soc/sof/intel/hda.c | 31 ++++++++------ sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/ops.h | 16 ++++++++ sound/soc/sof/sof-priv.h | 2 + 10 files changed, 112 insertions(+), 80 deletions(-) -- 2.39.2

6 27

[PATCH] ALSA: core: Use dev_name of card_dev as debugfs directory name
by Peter Ujfalusi 12 Sep '23

12 Sep '23

There is no need to use temporary string for the debugfs directory name as we can use the device name of the card. This change will also fixes the following compiler warning/error (W=1): sound/core/init.c: In function ‘snd_card_init’: sound/core/init.c:367:28: error: ‘%d’ directive writing between 1 and 10 bytes into a region of size 4 [-Werror=format-overflow=] 367 | sprintf(name, "card%d", idx); | ^~ sound/core/init.c:367:23: note: directive argument in the range [0, 2147483646] 367 | sprintf(name, "card%d", idx); | ^~~~~~~~ sound/core/init.c:367:9: note: ‘sprintf’ output between 6 and 15 bytes into a destination of size 8 367 | sprintf(name, "card%d", idx); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors The idx is guarantied to be less than SNDRV_CARDS (max 256 or 8) by the code in snd_card_init(), however the compiler does not see that. The warnings got brought to light by a recent patch upstream: commit 6d4ab2e97dcf ("extrawarn: enable format and stringop overflow warnings in W=1") Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/core/init.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index d61bde1225f2..22c0d217b860 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -278,9 +278,6 @@ static int snd_card_init(struct snd_card *card, struct device *parent, size_t extra_size) { int err; -#ifdef CONFIG_SND_DEBUG - char name[8]; -#endif if (extra_size > 0) card->private_data = (char *)card + sizeof(struct snd_card); @@ -364,8 +361,8 @@ static int snd_card_init(struct snd_card *card, struct device *parent, } #ifdef CONFIG_SND_DEBUG - sprintf(name, "card%d", idx); - card->debugfs_root = debugfs_create_dir(name, sound_debugfs_root); + card->debugfs_root = debugfs_create_dir(dev_name(&card->card_dev), + sound_debugfs_root); #endif return 0; -- 2.42.0

2 1

[PATCH] ALSA: core: Increase the name array size for debugfs directory name
by Peter Ujfalusi 12 Sep '23

12 Sep '23

The idx is guarantied to be less than SNDRV_CARDS (max 256 or 8) by the code in snd_card_init(), however the compiler does not see that. Compiling with W=1 results: sound/core/init.c: In function ‘snd_card_init’: sound/core/init.c:367:28: error: ‘%d’ directive writing between 1 and 10 bytes into a region of size 4 [-Werror=format-overflow=] 367 | sprintf(name, "card%d", idx); | ^~ sound/core/init.c:367:23: note: directive argument in the range [0, 2147483646] 367 | sprintf(name, "card%d", idx); | ^~~~~~~~ sound/core/init.c:367:9: note: ‘sprintf’ output between 6 and 15 bytes into a destination of size 8 367 | sprintf(name, "card%d", idx); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors While the code is correct, we need to silence the compiler somehow. It could be done by limiting the range in sprintf like sprintf(name, "card%d", idx % SNDRV_CARDS); sprintf(name, "card%hhd", idx); etc These are too workaroundish. Increase the name array to 15 instead which looks better and only adds 7 bytes on stack. The warnings got brought to light by a recent patch upstream: commit 6d4ab2e97dcf ("extrawarn: enable format and stringop overflow warnings in W=1") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, The mentioned commit causes other build failures with W=1 at least in sound/usb/mixer_scarlett_gen2.c sound/usb/mixer.c sound/soc/codecs/hdac_hdmi.c sound/hda/intel-sdw-acpi.c Some of them are also false and we need to find a workaround, but I think the scarlett case might be valid. Regards, Peter sound/core/init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/init.c b/sound/core/init.c index d61bde1225f2..d8a13a76d241 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -279,7 +279,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, { int err; #ifdef CONFIG_SND_DEBUG - char name[8]; + char name[15]; #endif if (extra_size > 0) -- 2.42.0

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Update alc287_fixup_bind_dacs()
by Kailang 12 Sep '23

12 Sep '23

Hi Takashi, Two speakers platform will common use same function. So, attach patch was an update. BR, Kailang

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[PATCH] ALSA: seq: Avoid delivery of events for disabled UMP groups
by Takashi Iwai 12 Sep '23

12 Sep '23

ALSA sequencer core still delivers events to the disabled UMP Group, leaving this handling to the device. But it's rather risky and it's easy to imagine that such an unexpected event may screw up the device firmware. This patch avoids the superfluous event deliveries by setting the group_filter of the UMP client as default, and evaluate the group_filter properly at delivery from non-UMP clients. The grouop_filter is updated upon the dynamic UMP Function Block updates, so that it follows the change of the disabled UMP Groups, too. Fixes: d2b706077792 ("ALSA: seq: Add UMP group filter") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_ump_client.c | 22 ++++++++++++++++++++++ sound/core/seq/seq_ump_convert.c | 2 ++ 2 files changed, 24 insertions(+) diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index f26a1812dfa7..a60e3f069a80 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -416,6 +416,25 @@ static void setup_client_midi_version(struct seq_ump_client *client) snd_seq_kernel_client_put(cptr); } +/* set up client's group_filter bitmap */ +static void setup_client_group_filter(struct seq_ump_client *client) +{ + struct snd_seq_client *cptr; + unsigned int filter; + int p; + + cptr = snd_seq_kernel_client_get(client->seq_client); + if (!cptr) + return; + filter = ~(1U << 0); /* always allow groupless messages */ + for (p = 0; p < SNDRV_UMP_MAX_GROUPS; p++) { + if (client->groups[p].active) + filter &= ~(1U << (p + 1)); + } + cptr->group_filter = filter; + snd_seq_kernel_client_put(cptr); +} + /* UMP group change notification */ static void handle_group_notify(struct work_struct *work) { @@ -424,6 +443,7 @@ static void handle_group_notify(struct work_struct *work) update_group_attrs(client); update_port_infos(client); + setup_client_group_filter(client); } /* UMP FB change notification */ @@ -492,6 +512,8 @@ static int snd_seq_ump_probe(struct device *_dev) goto error; } + setup_client_group_filter(client); + err = create_ump_endpoint_port(client); if (err < 0) goto error; diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index 7cc84e137999..b141024830ec 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -1197,6 +1197,8 @@ int snd_seq_deliver_to_ump(struct snd_seq_client *source, struct snd_seq_event *event, int atomic, int hop) { + if (dest->group_filter & (1U << dest_port->ump_group)) + return 0; /* group filtered - skip the event */ if (event->type == SNDRV_SEQ_EVENT_SYSEX) return cvt_sysex_to_ump(dest, dest_port, event, atomic, hop); else if (snd_seq_client_is_midi2(dest)) -- 2.35.3

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[PATCH] ALSA: docs: Fix a typo of midi2_ump_probe option for snd-usb-audio
by Takashi Iwai 12 Sep '23

12 Sep '23

A simple typo fix: midi2_probe => midi2_ump_probe. Fixes: febdfa0e9c8a ("ALSA: docs: Update MIDI 2.0 documentation for UMP 1.1 enhancement") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/designs/midi-2.0.rst | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/designs/midi-2.0.rst b/Documentation/sound/designs/midi-2.0.rst index 45987f256b97..086487ca7ab1 100644 --- a/Documentation/sound/designs/midi-2.0.rst +++ b/Documentation/sound/designs/midi-2.0.rst @@ -74,8 +74,8 @@ topology based on those information. When the device is older and doesn't respond to the new UMP inquiries, the driver falls back and builds the topology based on Group Terminal Block (GTB) information from the USB descriptor. Some device might be screwed up by the -unexpected UMP command; in such a case, pass `midi2_probe=0` option to -snd-usb-audio driver for skipping the UMP v1.1 inquiries. +unexpected UMP command; in such a case, pass `midi2_ump_probe=0` +option to snd-usb-audio driver for skipping the UMP v1.1 inquiries. When the MIDI 2.0 device is probed, the kernel creates a rawmidi device for each UMP Endpoint of the device. Its device name is -- 2.35.3

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RE: Re: [PATCH] ASoC: dwc: fix typo in comment
by 김명욱 12 Sep '23

12 Sep '23

ISR_RXFO means \"Status of Data Overrun interrupt for the RX channel\" according to the datasheetSo, the comment should be RX, not TXThanksMyunguk------------Original Message------------Subject : Re: [PATCH] ASoC: dwc: fix typo in commentDate : 2023-09-11 21:46:05From : Bagas Sanjaya <bagasdotme@gmailcom>To : Myunguk Kim <mwkim@gaonchipscom>, alsa-devel@alsa-projectorgCc : Liam Girdwood <lgirdwood@gmailcom>, Mark Brown <broonie@kernelorg>, Jaroslav Kysela <perex@perexcz>, Takashi Iwai <tiwai@susecom>, Maxim Kochetkov <fido_max@inboxru>, Kuninori Morimoto <kuninorimorimotogx@renesascom>, Xingyu Wu <xingyuwu@starfivetechcom>, Uwe Kleine-König <ukleine-koenig@pengutronixde>, linux-kernel@vgerkernelorgOn Mon, Sep 11, 2023 at 03:46:46PM +0900, Myunguk Kim wrote:> This fixes typo in commentOn what? Can you describe the typofix in more detail?>> Signed-off-by: Myunguk Kim <mwkim@gaonchipscom>> ---> sound/soc/dwc/dwc-i2sc | 2 +-> 1 file changed, 1 insertion(+), 1 deletion(-)>> diff --git a/sound/soc/dwc/dwc-i2sc b/sound/soc/dwc/dwc-i2sc> index 71ff894035a445e55e550924 100644> --- a/sound/soc/dwc/dwc-i2sc> +++ b/sound/soc/dwc/dwc-i2sc> @@ -138,7 +138,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)> irq_valid = true;> }>> - /* Error Handling: TX */> + /* Error Handling: RX */> if (isr[i] & ISR_RXFO)> { dev_err_ratelimited(dev->dev, \"RX overrun (ch_id=%d)\\n\", i);> irq_valid = true;Oh, you mean the code following the comment checks for RXFO overrunConfused--An old man doll just what I always wanted! - Clara

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[PATCH] ASoC: fsl: imx-pcm-rpmsg: Add SNDRV_PCM_INFO_BATCH flag
by Shengjiu Wang 12 Sep '23

12 Sep '23

The rpmsg pcm device is a device which should support double buffering. Found this issue with pipewire. When there is no SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will set headroom to be zero, and because rpmsg pcm device don't support residue report, when the latency setting is small, the "delay" always larger than "target" in alsa-pcm.c, that reading next period data is not scheduled on time. With SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will select a smaller period size for device, then the task of reading next period data will be scheduled on time. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-pcm-rpmsg.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index d63782b8bdef..bb736d45c9e0 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -19,6 +19,7 @@ static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | -- 2.34.1

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[PATCH v2 0/2] Fix redundant PLLA update
by Sameer Pujar 12 Sep '23

12 Sep '23

This small series fixes redundant PLLA updates that happen for each DAI link in the audio path. This helps to resolve DMIC clock issue seen on Jetson TX2 platform. Sameer Pujar (2): ASoC: soc-utils: Export snd_soc_dai_is_dummy() symbol ASoC: tegra: Fix redundant PLLA and PLLA_OUT0 updates sound/soc/soc-utils.c | 1 + sound/soc/tegra/tegra_audio_graph_card.c | 30 ++++++++++++++---------- 2 files changed, 18 insertions(+), 13 deletions(-) -- 2.17.1

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