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

[PATCH] ASoC: rt715-sdca-sdw: Fix wrong complete waiting in rt715_dev_resume()
by Pierre-Louis Bossart 10 May '24

10 May '24

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> enumeration_complete will be completed when a peripheral is attached. And initialization_complete will be completed when a peripheral is initialized. rt715_dev_resume() should wait for initialization_complete instead of enumeration_complete. the issue exists since commit 20d17057f0a8 ("ASoC: rt715-sdca: Add RT715 sdca vendor-specific driver"), but the commit can only apply to commit f892e66fcabc ("ASoC: rt-sdw*: add __func__ to all error logs"). Fixes: f892e66fcabc ("ASoC: rt-sdw*: add __func__ to all error logs") Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt715-sdca-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index d3fb02e8f31a..c8dabb9b16b5 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -234,10 +234,10 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) if (!slave->unattach_request) goto regmap_sync; - time = wait_for_completion_timeout(&slave->enumeration_complete, + time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "%s: Enumeration not complete, timed out\n", __func__); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; -- 2.40.1

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[PATCH] ASoC: audio-graph-card2: call of_node_get() before of_get_next_child()
by Kuninori Morimoto 10 May '24

10 May '24

commit c6f597bc598a8 ("ASoC: audio-graph-card2: remove unneeded of_node_get()") removed of_node_get(), but it need to keep original reference, so of_node_get() itself is needed. Because of_get_next_child() will call of_node_put() inside, if the reference count dropped to zero, then of_node_get() after that will be use afer free. Need to call of_node_get() *before* that. Fixes: c6f597bc598a ("ASoC: audio-graph-card2: remove unneeded of_node_get()") Link: https://lore.kernel.org/r/f930862e-9d30-4ea3-b3e7-b4b4f411f6d1@moroto.mount… Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index c4d52fadebb6..81e84095107e 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1197,7 +1197,7 @@ static int graph_count_c2c(struct simple_util_priv *priv, { struct device_node *ports = of_get_parent(lnk); struct device_node *port0 = lnk; - struct device_node *port1 = of_get_next_child(ports, lnk); + struct device_node *port1 = of_get_next_child(ports, of_node_get(lnk)); struct device_node *ep0 = port_to_endpoint(port0); struct device_node *ep1 = port_to_endpoint(port1); struct device_node *codec0 = of_graph_get_remote_port(ep0); -- 2.25.1

3 2

[PATCH v21 00/39] Introduce QC USB SND audio offloading support
by Wesley Cheng 10 May '24

10 May '24

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 Requesting to see if we can get some Acked-By tags, and merge on usb-next. Changelog -------------------------------------------- Changs in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some 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 (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (37): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index 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: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: qcom: Populate SoC components string ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 611 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 36 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 418 ++++ drivers/usb/host/xhci.c | 43 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 68 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 188 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- 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 | 428 ++++ sound/soc/qcom/sm8250.c | 26 +- sound/soc/qcom/usb_offload_utils.c | 77 + sound/soc/qcom/usb_offload_utils.h | 36 + sound/soc/soc-usb.c | 678 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1914 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6371 insertions(+), 50 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h 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 0/9] ALSA/ASoC: Intel: clarify Copyright information
by Pierre-Louis Bossart 09 May '24

09 May '24

3 12

[PATCH 0/2] ASoC: Intel: avs: Switch to acpi-nhlt
by Cezary Rojewski 09 May '24

09 May '24

The change is based on rafael/acpi-nhlt [1] immutable branch which Rafael kindly prepared for me. Without the topmost changes to ACPI/NHLT, the patches present will fail to compile. Recent changes for the ACPI tree [2] refactored interfaces of the NHLT table. Currently we have two implementations - one found in acpi subsystem (unused) and one in sound/hda/. As NHLT is part of ACPI, idea is to make the former useful and then switch all users of existing sound/hda/intel-nhlt.c to this new interface over time and remove the duplicate afterward. Two patches present here migrate the avs-driver and the intel-dsp-config module. [1]: https://git.kernel.org/pub/scm/linux/kernel/git/rafael/linux-pm.git/?h=acpi… [2]: https://lore.kernel.org/linux-acpi/20240319083018.3159716-1-cezary.rojewski… Amadeusz Sławiński (2): ALSA: hda: intel-dsp-config: Switch to ACPI NHLT ASoC: Intel: avs: Switch to ACPI NHLT sound/hda/Kconfig | 1 + sound/hda/intel-dsp-config.c | 16 ++++++++------ sound/soc/intel/Kconfig | 1 + sound/soc/intel/avs/avs.h | 1 - sound/soc/intel/avs/board_selection.c | 7 +++--- sound/soc/intel/avs/core.c | 10 ++++----- sound/soc/intel/avs/path.c | 32 +++++++++++++-------------- 7 files changed, 34 insertions(+), 34 deletions(-) base-commit: a640acab545b21ed1f347376f34d34e461ea92ba -- 2.25.1

3 8

Re: [syzbot] [kernel?] INFO: rcu detected stall in sys_wait4 (4)
by syzbot 09 May '24

09 May '24

syzbot has bisected this issue to: commit 51ea51b18904cd1a0fb244ce41dfd903c2ada628 Author: Shuming Fan <shumingf(a)realtek.com> Date: Fri Dec 23 05:58:46 2022 +0000 ASoC: rt711-sdca: add jack detection mode for JD2 100K bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=123e6f88980000 start commit: fe46a7dd189e Merge tag 'sound-6.9-rc1' of git://git.kernel.. git tree: upstream final oops: https://syzkaller.appspot.com/x/report.txt?x=113e6f88980000 console output: https://syzkaller.appspot.com/x/log.txt?x=163e6f88980000 kernel config: https://syzkaller.appspot.com/x/.config?x=fe78468a74fdc3b7 dashboard link: https://syzkaller.appspot.com/bug?extid=6969434de600a6ba9f07 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1091a5f6180000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17a22c13180000 Reported-by: syzbot+6969434de600a6ba9f07(a)syzkaller.appspotmail.com Fixes: 51ea51b18904 ("ASoC: rt711-sdca: add jack detection mode for JD2 100K") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

1 0

[PATCH v2 0/7] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 09 May '24

09 May '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Changes in v2: - Added vendor prefix to threshold properties - Added separate headset mic bias regulator - Changed some cases of dev_err + return with return dev_err_probe - Added an extra patch to replace some previous dev_err + return cases with dev_err_probe - Moved tab3 DTS wm1811 codec config changes to separate commit --- Artur Weber (7): ASoC: dt-bindings: samsung,midas-audio: Add headset mic bias supply ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Add headset mic bias supply support ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use dev_err_probe where appropriate ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection ARM: dts: samsung: exynos4212-tab3: Fix up wm1811 codec config .../bindings/sound/samsung,midas-audio.yaml | 33 ++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 31 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 343 +++++++++++++++++++-- 4 files changed, 379 insertions(+), 30 deletions(-) --- base-commit: e67572cd2204894179d89bd7b984072f19313b03 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

3 9

[PATCH] ASoC: audio-graph-card2: remove unneeded of_node_get()
by Kuninori Morimoto 09 May '24

09 May '24

"lnk" is used as "port0", and it will be used to get "port1" by of_get_next_child(ports, lnk). It will call of_node_put() inside. This function is calling of_node_get() to make up for it, but it doesn't call paired of_node_put(port0) when it quit function. This of_node_get() itself is not needed, let's remove it. Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card2.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index b1c675c6b6db..825125d8985b 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1021,8 +1021,6 @@ static int graph_count_c2c(struct asoc_simple_priv *priv, struct device_node *codec0 = of_graph_get_remote_port(ep0); struct device_node *codec1 = of_graph_get_remote_port(ep1); - of_node_get(lnk); - /* * codec2codec { * ports { -- 2.25.1

2 1

Motu 828 (28 x 32) support
by GitHub issues - opened 08 May '24

08 May '24

alsa-project/alsa-ucm-conf issue #415 was opened from beatboxa: The newest [Motu 828 (28x32)](https://motu.com/en-us/products/828/) does not appear to have a UCM yet. It is a class compliant USB device; so as a result, it generally "works" at a very basic level; but default channel mappings and profiles do not. The system appears to default to only 2 channels in and 2 channels out. So, for example, surround playback seems to mix into either the left or right channel. In qpwgraph, it shows 30 inputs (I believe this includes an additional 2-channel loopback at the end) and 32 outputs, all marked as "AUX#," where # starts with 0. I was able to identify which AUX# corresponds to which input or output on the device, through the use of qpwgraph and the device's built-in digital VU meter. I have tried to create a UCM or even an asoundrc; but I have no idea what I'm doing. The documentation and examples are not intuitive for me. Could I collaborate with someone in order to get this audio interface supported? I'd be happy to run any commands and tests you advise on. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/415 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: x1e80100: add recording via DMIC01
by GitHub pull_request - opened 08 May '24

08 May '24

alsa-project/alsa-ucm-conf pull request #414 was opened from krzk: Add support for Digital Microphones DMIC01 on Qualcomm X1E80100 CRD board. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/414 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/414.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda: cs35l56: Fix lifetime of cs_dsp instance
by Richard Fitzgerald 08 May '24

08 May '24

The cs_dsp instance is initialized in the driver probe() so it should be freed in the driver remove(). Also fix a missing call to cs_dsp_remove() in the error path of cs35l56_hda_common_probe(). The call to cs_dsp_remove() was being done in the component unbind callback cs35l56_hda_unbind(). This meant that if the driver was unbound and then re-bound it would be using an uninitialized cs_dsp instance. It is best to initialize the cs_dsp instance in probe() so that it can return an error if it fails. The component binding API doesn't have any error handling so there's no way to handle a failure if cs_dsp was initialized in the bind. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") --- sound/pci/hda/cs35l56_hda.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 558c1f38fe97..11b0570ff56d 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -732,8 +732,6 @@ static void cs35l56_hda_unbind(struct device *dev, struct device *master, void * if (cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); - cs_dsp_remove(&cs35l56->cs_dsp); - if (comps[cs35l56->index].dev == dev) memset(&comps[cs35l56->index], 0, sizeof(*comps)); @@ -1035,7 +1033,7 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) ARRAY_SIZE(cs35l56_hda_dai_config)); ret = cs35l56_force_sync_asp1_registers_from_cache(&cs35l56->base); if (ret) - goto err; + goto dsp_err; /* * By default only enable one ASP1TXn, where n=amplifier index, @@ -1061,6 +1059,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) pm_err: pm_runtime_disable(cs35l56->base.dev); +dsp_err: + cs_dsp_remove(&cs35l56->cs_dsp); err: gpiod_set_value_cansleep(cs35l56->base.reset_gpio, 0); @@ -1078,6 +1078,8 @@ void cs35l56_hda_remove(struct device *dev) component_del(cs35l56->base.dev, &cs35l56_hda_comp_ops); + cs_dsp_remove(&cs35l56->cs_dsp); + kfree(cs35l56->system_name); pm_runtime_put_noidle(cs35l56->base.dev); -- 2.39.2

2 1

[PATCH] ALSA: hda: hda_component: Initialize shared data during bind callback
by Richard Fitzgerald 08 May '24

08 May '24

Move the initialization of the shared struct hda_component array into hda_component_manager_bind(). The purpose of the manager bind() callback is to allow it to perform initialization before binding in the component drivers. This is the correct place to initialize the shared data. The original implementation initialized the shared data in hda_component_manager_init(). This is only done once during probe() of the manager driver. So if the component binding was unbound and then rebound, the shared data would not be re-initialized. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: fd895a74dc1d ("ALSA: hda: realtek: Move hda_component implementation to module") --- sound/pci/hda/hda_component.c | 16 +++++++++++++++- sound/pci/hda/hda_component.h | 7 ++----- sound/pci/hda/patch_realtek.c | 2 +- 3 files changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_component.c b/sound/pci/hda/hda_component.c index cd299d7d84ba..d02589014a3f 100644 --- a/sound/pci/hda/hda_component.c +++ b/sound/pci/hda/hda_component.c @@ -123,6 +123,21 @@ static int hda_comp_match_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +int hda_component_manager_bind(struct hda_codec *cdc, + struct hda_component *comps, int count) +{ + int i; + + /* Init shared data */ + for (i = 0; i < count; ++i) { + memset(&comps[i], 0, sizeof(comps[i])); + comps[i].codec = cdc; + } + + return component_bind_all(hda_codec_dev(cdc), comps); +} +EXPORT_SYMBOL_NS_GPL(hda_component_manager_bind, SND_HDA_SCODEC_COMPONENT); + int hda_component_manager_init(struct hda_codec *cdc, struct hda_component *comps, int count, const char *bus, const char *hid, @@ -143,7 +158,6 @@ int hda_component_manager_init(struct hda_codec *cdc, sm->hid = hid; sm->match_str = match_str; sm->index = i; - comps[i].codec = cdc; component_match_add(dev, &match, hda_comp_match_dev_name, sm); } diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index c80a66691b5d..c70b3de68ab2 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -75,11 +75,8 @@ int hda_component_manager_init(struct hda_codec *cdc, void hda_component_manager_free(struct hda_codec *cdc, const struct component_master_ops *ops); -static inline int hda_component_manager_bind(struct hda_codec *cdc, - struct hda_component *comps) -{ - return component_bind_all(hda_codec_dev(cdc), comps); -} +int hda_component_manager_bind(struct hda_codec *cdc, + struct hda_component *comps, int count); static inline void hda_component_manager_unbind(struct hda_codec *cdc, struct hda_component *comps) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c157158f1cb..8c39bbdc6c5e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6785,7 +6785,7 @@ static int comp_bind(struct device *dev) struct alc_spec *spec = cdc->spec; int ret; - ret = hda_component_manager_bind(cdc, spec->comps); + ret = hda_component_manager_bind(cdc, spec->comps, ARRAY_SIZE(spec->comps)); if (ret) return ret; -- 2.39.2

2 1

[PATCH] ALSA: hda/cs_dsp_ctl: Use private_free for control cleanup
by Richard Fitzgerald 08 May '24

08 May '24

Use the control private_free callback to free the associated data block. This ensures that the memory won't leak, whatever way the control gets destroyed. The original implementation didn't actually remove the ALSA controls in hda_cs_dsp_control_remove(). It only freed the internal tracking structure. This meant it was possible to remove/unload the amp driver while leaving its ALSA controls still present in the soundcard. Obviously attempting to access them could cause segfaults or at least dereferencing stale pointers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 3233b978af23 ("ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls") --- This replaces my earlier "keep it simple" attempt at cleanup. I decided that implementing private_free is the only sensible way to fix the cleanup. --- sound/pci/hda/hda_cs_dsp_ctl.c | 47 ++++++++++++++++++++++------------ 1 file changed, 31 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 463ca06036bf..9db45d7c17e5 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <sound/soc.h> +#include <linux/cleanup.h> #include <linux/firmware/cirrus/cs_dsp.h> #include <linux/firmware/cirrus/wmfw.h> #include "hda_cs_dsp_ctl.h" @@ -97,11 +98,23 @@ static unsigned int wmfw_convert_flags(unsigned int in) return out; } -static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char *name) +static void hda_cs_dsp_free_kcontrol(struct snd_kcontrol *kctl) { + struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; + + /* NULL priv to prevent a double-free in hda_cs_dsp_control_remove() */ + cs_ctl->priv = NULL; + kfree(ctl); +} + +static void hda_cs_dsp_add_kcontrol(struct cs_dsp_coeff_ctl *cs_ctl, + const struct hda_cs_dsp_ctl_info *info, + const char *name) +{ struct snd_kcontrol_new kcontrol = {0}; struct snd_kcontrol *kctl; + struct hda_cs_dsp_coeff_ctl *ctl __free(kfree) = NULL; int ret = 0; if (cs_ctl->len > ADSP_MAX_STD_CTRL_SIZE) { @@ -110,6 +123,13 @@ static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char return; } + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) + return; + + ctl->cs_ctl = cs_ctl; + ctl->card = info->card; + kcontrol.name = name; kcontrol.info = hda_cs_dsp_coeff_info; kcontrol.iface = SNDRV_CTL_ELEM_IFACE_MIXER; @@ -117,20 +137,22 @@ static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char kcontrol.get = hda_cs_dsp_coeff_get; kcontrol.put = hda_cs_dsp_coeff_put; - /* Save ctl inside private_data, ctl is owned by cs_dsp, - * and will be freed when cs_dsp removes the control */ kctl = snd_ctl_new1(&kcontrol, (void *)ctl); if (!kctl) return; - ret = snd_ctl_add(ctl->card, kctl); + kctl->private_free = hda_cs_dsp_free_kcontrol; + ctl->kctl = kctl; + + /* snd_ctl_add() calls our private_free on error, which will kfree(ctl) */ + cs_ctl->priv = no_free_ptr(ctl); + ret = snd_ctl_add(info->card, kctl); if (ret) { dev_err(cs_ctl->dsp->dev, "Failed to add KControl %s = %d\n", kcontrol.name, ret); return; } dev_dbg(cs_ctl->dsp->dev, "Added KControl: %s\n", kcontrol.name); - ctl->kctl = kctl; } static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, @@ -138,7 +160,6 @@ static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, { struct cs_dsp *cs_dsp = cs_ctl->dsp; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - struct hda_cs_dsp_coeff_ctl *ctl; const char *region_name; int ret; @@ -163,15 +184,7 @@ static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, " %.*s", cs_ctl->subname_len - skip, cs_ctl->subname + skip); } - ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); - if (!ctl) - return; - - ctl->cs_ctl = cs_ctl; - ctl->card = info->card; - cs_ctl->priv = ctl; - - hda_cs_dsp_add_kcontrol(ctl, name); + hda_cs_dsp_add_kcontrol(cs_ctl, info, name); } void hda_cs_dsp_add_controls(struct cs_dsp *dsp, const struct hda_cs_dsp_ctl_info *info) @@ -203,7 +216,9 @@ void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) { struct hda_cs_dsp_coeff_ctl *ctl = cs_ctl->priv; - kfree(ctl); + /* ctl and kctl may already have been removed by ALSA private_free */ + if (ctl && ctl->kctl) + snd_ctl_remove(ctl->card, ctl->kctl); } EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS); -- 2.39.2

2 1

[PATCH v1 1/1] ASoC: atmel: tse850-pcm5142: Remove unused of_gpio.h
by Andy Shevchenko 08 May '24

08 May '24

of_gpio.h is deprecated and subject to remove. The driver doesn't use it, simply remove the unused header. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/atmel/tse850-pcm5142.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 611da23325d3..5d208e0b4b90 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -35,10 +35,9 @@ // of the (filtered) output from the PCM5142 codec. #include <linux/clk.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/regulator/consumer.h> #include <sound/soc.h> -- 2.43.0.rc1.1336.g36b5255a03ac

2 1

[PATCH v2 0/4] ASoC: qcom: x1e80100: Correct channel mapping
by Krzysztof Kozlowski 08 May '24

08 May '24

Hi, First patch is a build dependency. Changes in v2: - Re-work most of the idea according to Srini comments: set channel mapping for backend DAIs, not frontend. - Patch #1: no changes - Patch #2 is entirely replaced - now channel mapping is implemented in q6apm-lpass-dais. - Patch #3: rework to new approach, but most of the code stays. - Patch #4: rework significantly, because only backend DAIs is now affected. - Link to v1: https://lore.kernel.org/r/20240507-asoc-x1e80100-4-channel-mapping-v1-0-b12… Description =========== X1E80100 CRD is the first board, which comes with four speakers, so we still keep fixing and adding missing pieces. The board has speaker arranged as left front+back and then right front+back. Using default channel mapping causes front right speaker to play left back stream. Adjust the channel maps for frontend DAIs to fix stereo and four-channel playback. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: Constify channel mapping array arguments in set_channel_map() ASoC: qcom: q6apm-lpass-dais: Implement proper channel mapping ASoC: qcom: qdsp6: Set channel mapping instead of fixed defaults ASoC: qcom: x1e80100: Correct channel mapping include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 ++++---- sound/soc/codecs/adau7118.c | 6 ++++-- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l41.c | 3 ++- sound/soc/codecs/max98504.c | 6 ++++-- sound/soc/codecs/wcd9335.c | 6 ++++-- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/qcom/qdsp6/audioreach.c | 30 +++++++----------------------- sound/soc/qcom/qdsp6/audioreach.h | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++------ sound/soc/qcom/qdsp6/q6apm-dai.c | 2 ++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 21 +++++++++++++-------- sound/soc/qcom/x1e80100.c | 18 ++++++++++++++++++ sound/soc/soc-dai.c | 4 ++-- 15 files changed, 79 insertions(+), 57 deletions(-) --- base-commit: 2b84edefcad14934796fad37b16512b6a2ca467e change-id: 20240507-asoc-x1e80100-4-channel-mapping-ea5f02b9e678 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

1 5

[PATCH v1] ALSA: hda/realtek: Add support for ASUS Zenbook 2024 HN7306W
by Stefan Binding 08 May '24

08 May '24

This laptop uses CS35L41 HDA with 2 amps using I2C and Internal Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8502b30d5d5d..8c157158f1cb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10276,6 +10276,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), -- 2.34.1

2 1

[PATCH 0/4] ASoC: Constify static snd_pcm_hardware
by Krzysztof Kozlowski 08 May '24

08 May '24

No dependencies. Static 'struct snd_pcm_hardware' is not modified by few drivers and its copy is passed to the core, so it can be made const for increased code safety. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: qcom: Constify static snd_pcm_hardware ASoC: fsl: Constify static snd_pcm_hardware ASoC: meson: Constify static snd_pcm_hardware ASoC: uniphier: Constify static snd_pcm_hardware sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- sound/soc/meson/aiu-fifo-i2s.c | 2 +- sound/soc/meson/aiu-fifo-spdif.c | 2 +- sound/soc/meson/aiu-fifo.h | 2 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- sound/soc/uniphier/aio-dma.c | 2 +- 8 files changed, 9 insertions(+), 9 deletions(-) --- base-commit: ebbaba914053d8bcadd7d64eb8d2555299509a27 change-id: 20240429-n-asoc-const-snd-pcm-hardware-f0a5ade885ca Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 5

[PATCH AUTOSEL 6.1 01/25] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 094445036c20f..d6ef8e850412b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH AUTOSEL 6.6 01/43] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e609249cc38d5..651408c6f399d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH AUTOSEL 6.8 01/52] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 05f38d1f7d824..b41a1147f1c34 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH 0/4] ASoC: qcom: x1e80100: Correct channel mapping
by Krzysztof Kozlowski 07 May '24

07 May '24

Hi, First patch is a build dependency. Description =========== X1E80100 CRD is the first board, which comes with four speakers, so we still keep fixing and adding missing pieces. The board has speaker arranged as left front+back and then right front+back. Using default channel mapping causes front right speaker to play left back stream. Adjust the channel maps for frontend DAIs to fix stereo and four-channel playback. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: Constify channel mapping array arguments in set_channel_map() ASoC: qcom: q6dsp: Implement proper channel mapping in Audioreach ASoC: qcom: q6dsp: Set channel mapping instead of fixed defaults ASoC: qcom: x1e80100: Correct channel mapping include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 +++---- sound/soc/codecs/adau7118.c | 6 ++++-- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l41.c | 3 ++- sound/soc/codecs/max98504.c | 6 ++++-- sound/soc/codecs/wcd9335.c | 6 ++++-- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/qcom/qdsp6/audioreach.c | 14 +++++-------- sound/soc/qcom/qdsp6/audioreach.h | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++------ sound/soc/qcom/qdsp6/q6apm-dai.c | 12 +++++++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 6 ++++-- sound/soc/qcom/qdsp6/q6apm.c | 28 ++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6apm.h | 8 +++++++ sound/soc/qcom/qdsp6/topology.c | 12 +++++++++++ sound/soc/qcom/x1e80100.c | 37 +++++++++++++++++++++++++++++++-- sound/soc/soc-dai.c | 4 ++-- 18 files changed, 142 insertions(+), 39 deletions(-) --- base-commit: c5e512ffe106f751c61e5a036560f522e58eadcd change-id: 20240507-asoc-x1e80100-4-channel-mapping-ea5f02b9e678 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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[PATCH 0/3] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 07 May '24

07 May '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Artur Weber (3): ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection .../bindings/sound/samsung,midas-audio.yaml | 30 +++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 23 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 265 ++++++++++++++++++++- 4 files changed, 305 insertions(+), 15 deletions(-) --- base-commit: e67572cd2204894179d89bd7b984072f19313b03 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

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[PATCH v2] ASoC: Intel: avs: boards: Properly name input device
by Amadeusz Sławiński 07 May '24

07 May '24

Machine boards expose input device for use with userspace. Current name in some cases is incorrect, fix it. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- v2: - Rebase on top of correct tree. --- sound/soc/intel/avs/boards/es8336.c | 2 +- sound/soc/intel/avs/boards/nau8825.c | 2 +- sound/soc/intel/avs/boards/rt274.c | 3 ++- sound/soc/intel/avs/boards/rt286.c | 4 ++-- sound/soc/intel/avs/boards/rt298.c | 4 ++-- 5 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index ff3bd15132694..f4919a6abc278 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -113,7 +113,7 @@ static int avs_es8336_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, &data->jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 3164745b15167..eef718c6b8ad9 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -96,7 +96,7 @@ static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) * 4 buttons here map to the google Reference headset. * The use of these buttons can be decided by the user space. */ - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack, pins, num_pins); if (ret) diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index e3aa28780df5a..51ebc790c6b4d 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -102,7 +102,8 @@ static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins, + num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index d24316fc4db9c..67fd362962c40 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -63,8 +63,8 @@ static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 7fd28544f7869..1003e8689eacc 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -74,8 +74,8 @@ static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; -- 2.34.1

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[PATCH 0/4] ASoC: use 'time_left' instead of 'timeout' with wait_for_*() functions
by Wolfram Sang 07 May '24

07 May '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_*() functions causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code obvious and self explaining. This is part of a tree-wide series. The rest of the patches can be found here (some parts may still be WIP): git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/time_left Because these patches are generated, I audit them before sending. This is why I will send series step by step. Build bot is happy with these patches, though. No functional changes intended. Wolfram Sang (4): ASoC: codecs: wm8962: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8993: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8994: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8996: use 'time_left' variable with wait_for_completion_timeout() sound/soc/codecs/wm8962.c | 12 ++++++------ sound/soc/codecs/wm8993.c | 12 ++++++------ sound/soc/codecs/wm8994.c | 8 ++++---- sound/soc/codecs/wm8996.c | 14 +++++++------- 4 files changed, 23 insertions(+), 23 deletions(-) -- 2.43.0

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[PATCH 00/13] ASoC: Use snd_soc_substream_to_rtd() for accessing private_data
by Krzysztof Kozlowski 07 May '24

07 May '24

Hi, Do not open-code snd_soc_substream_to_rtd() when accessing snd_pcm_substream->private_data. This makes code more consistent with rest of ASoC and allows in the future to move the field to any other place or add additional checks in snd_soc_substream_to_rtd(). Best regards, Krzysztof --- Krzysztof Kozlowski (13): ASoC: qcom: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: tegra: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: ti: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: arm: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: amd: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: fsl: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: img: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: kirkwood: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: loongson: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: mediatek: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: meson: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: samsung: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: sunxi: Use snd_soc_substream_to_rtd() for accessing private_data sound/arm/pxa2xx-pcm-lib.c | 4 ++-- sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/imx-card.c | 6 +++--- sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 6 +++--- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/loongson/loongson_card.c | 2 +- sound/soc/loongson/loongson_dma.c | 2 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 14 +++++++------- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 2 +- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 ++++---- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 6 +++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 10 +++++----- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 4 ++-- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 4 ++-- sound/soc/qcom/qdsp6/q6apm-dai.c | 2 +- sound/soc/qcom/sc7180.c | 10 +++++----- sound/soc/qcom/sc7280.c | 12 ++++++------ sound/soc/qcom/sc8280xp.c | 8 ++++---- sound/soc/qcom/sdw.c | 8 ++++---- sound/soc/qcom/sm8250.c | 10 +++++----- sound/soc/qcom/x1e80100.c | 8 ++++---- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/tegra/tegra_asoc_machine.c | 2 +- sound/soc/tegra/tegra_pcm.c | 6 +++--- sound/soc/ti/omap-hdmi.c | 2 +- 35 files changed, 82 insertions(+), 82 deletions(-) --- base-commit: 82415cf72c7e224be7a6496f3a53c0b365c2fe9d change-id: 20240430-asoc-snd-substream-clean-924b717d8f54 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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