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

[PATCH 0/3] ACPI/ALSA/soundwire: add acpi_get_local_u64_address() helper
by Pierre-Louis Bossart 11 Jun '24

11 Jun '24

The acpi_get_local_address() helper assumes a 32-bit ADR is used. If we want to use this helper for SoundWire/SDCA ASoC codecs, we need an extension where the native 64-bits are used. This patchset suggests a new helper, acpi_get_local_address() may be renamed if desired in a folow-up patch. The path of least resistance would be to merge this patchset in the ASoC tree, since I have additional changes for ASoC/SDCA (SoundWire Device Class) that depend on the new helper. Pierre-Louis Bossart (3): ACPI: utils: introduce acpi_get_local_u64_address() soundwire: slave: simplify code with acpi_get_local_u64_address() ALSA: hda: intel-sdw-acpi: use acpi_get_local_u64_address() drivers/acpi/utils.c | 22 ++++++++++++++++------ drivers/soundwire/slave.c | 13 ++++--------- include/linux/acpi.h | 1 + sound/hda/intel-sdw-acpi.c | 6 +++--- 4 files changed, 24 insertions(+), 18 deletions(-) -- 2.43.0

6 11

[RESEND v5 0/7] ASoC: codecs: wcd937x: add wcd937x audio codec support
by Mohammad Rafi Shaik 11 Jun '24

11 Jun '24

This patchset adds support for Qualcomm WCD9370/WCD9375 codec. Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 3 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 6 DMICs and MBHC. For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on QCM6490 IDP device. Changes since v4: - Removed volatile/read-only registers from defaults list - Added wcd939x_volatile_register() with only volatile registers - Added a wcd939x_readable_register() with read-only and read-write registers, so cache does it's job - Fixed Spurious events for mixer controls and validated with mixer selftest tool - Used TLV instead of enum for ear_pa_gain mixer control - Used enum constraints instead of OneOf in dt-binding patch - Added vdd-px supply property as non optional in dt-binding patch - Reworked and done driver cleanup Changes since v3: - Fixed dt binding check errors. - Added constraints on values in v4-0001 binding patch as suggested by Krzysztof - Change the patch sequence soundwire driver first then codec driver - Added missing .remove soundwire driver function - Reworked and done driver cleanup Changes since v2: - Used common qcom,wcd93xx-common.yaml. removed duplicate properties. - Merged bindings patches "v2-0001" and "v2-0003" in single patch for easy review. - Fixed dt binding check errors. - Added missing "qcom,wcd9375-codec" in v3-0001 dt binding patch. - Added constraints on values in v3-0001 binding patch as suggested by Krzysztof - Fix the typo mistake in v2 cover letter Changes since v1: - Split the patch per driver for easier review as suggested by Krzysztof - Used devm_gpiod_get api to get reset gpio as suggested by Krzysztof Prasad Kumpatla (7): ASoC: dt-bindings: document wcd937x Audio Codec ASoC: codecs: wcd937x-sdw: add SoundWire driver ASoC: codecs: wcd937x: add wcd937x codec driver ASoC: codecs: wcd937x: add basic controls ASoC: codecs: wcd937x: add playback dapm widgets ASoC: codecs: wcd937x: add capture dapm widgets ASoC: codecs: wcd937x: add audio routing and Kconfig .../bindings/sound/qcom,wcd937x-sdw.yaml | 91 + .../bindings/sound/qcom,wcd937x.yaml | 82 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd937x-sdw.c | 1139 +++++++ sound/soc/codecs/wcd937x.c | 3029 +++++++++++++++++ sound/soc/codecs/wcd937x.h | 653 ++++ 7 files changed, 5021 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml create mode 100644 sound/soc/codecs/wcd937x-sdw.c create mode 100644 sound/soc/codecs/wcd937x.c create mode 100644 sound/soc/codecs/wcd937x.h base-commit: 3689b0ef08b70e4e03b82ebd37730a03a672853a -- 2.25.1

2 9

[PATCH v22 00/38] Introduce QC USB SND audio offloading support
by Wesley Cheng 11 Jun '24

11 Jun '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and 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 v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - 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 (36): 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: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 603 ++++++ 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 | 417 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 55 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 684 ++++++ 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 | 1915 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6327 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 v2 0/4] ASoC: qcom: display port changes
by srinivas.kandagatla＠linaro.org 08 Jun '24

08 Jun '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset adds support for. 1. parse Display Port module tokens from ASoC topology 2. add support to DP/HDMI Jack events. 3. fixes a typo in function name in sm8250 Verified these patches on X13s along with changes to tplg in https://git.codelinaro.org/linaro/qcomlt/audioreach-topology/-/tree/topic/x… and ucm changes from https://github.com/Srinivas-Kandagatla/alsa-ucm-conf/tree/topic/x13s-dp Thanks, Srini Changes since v1: - Fixed unused variable warning. - fixed warning 'break;' to avoid fall-through Srinivas Kandagatla (4): ASoC: qcom: q6dsp: parse Display port tokens ASoC: qcom: common: add Display port Jack function ASoC: qcom: sc8280xp: add Display port Jack ASoC: qcom: sm8250: fix a typo in function name sound/soc/qcom/common.c | 29 +++++++++++++++++++++++++++++ sound/soc/qcom/common.h | 3 +++ sound/soc/qcom/qdsp6/topology.c | 26 ++++++++++++++++++++++++++ sound/soc/qcom/sc8280xp.c | 14 ++++++++++++++ sound/soc/qcom/sm8250.c | 4 ++-- 5 files changed, 74 insertions(+), 2 deletions(-) -- 2.25.1

5 24

[PATCHv4 9/9] ASoC: dt-bindings: fsl-asoc-card: add compatible for generic codec
by Elinor Montmasson 06 Jun '24

06 Jun '24

Add documentation about new dts bindings following new support for compatible "fsl,imx-audio-generic". Some CPU DAI don't require a real audio codec. The new compatible "fsl,imx-audio-generic" allows using the driver with codec drivers SPDIF DIT and SPDIF DIR as dummy codecs. It also allows using not pre-configured audio codecs which don't require specific control through a codec driver. The new dts properties give the possibility to set some parameters about the CPU DAI usually set through the codec configuration. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- .../bindings/sound/fsl-asoc-card.yaml | 96 ++++++++++++++++++- 1 file changed, 92 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..332d8bf96e06 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -23,6 +23,16 @@ description: and PCM DAI formats. However, it'll be also possible to support those non AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long as the driver has been properly upgraded. + To use CPU DAIs that do not require a codec such as an S/PDIF controller, + or to use a DAI to output or capture raw I2S/TDM data, you can + use the compatible "fsl,imx-audio-generic". + +definitions: + imx-audio-generic-dependency: + properties: + compatible: + contains: + const: fsl,imx-audio-generic maintainers: - Shengjiu Wang <shengjiu.wang(a)nxp.com> @@ -81,6 +91,7 @@ properties: - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 + - fsl,imx-audio-generic model: $ref: /schemas/types.yaml#/definitions/string @@ -93,8 +104,14 @@ properties: need to add ASRC support via DPCM. audio-codec: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of an audio codec + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandle of an audio codec. + If using the "fsl,imx-audio-generic" compatible, give instead a pair of + phandles with the spdif_transmitter first (driver SPDIF DIT) and the + spdif_receiver second (driver SPDIF DIR). + items: + maxItems: 1 audio-cpu: $ref: /schemas/types.yaml#/definitions/phandle @@ -150,8 +167,8 @@ properties: description: dai-link uses bit clock inversion. mclk-id: - $ref: /schemas/types.yaml#/definitions/uint32 - description: main clock id, specific for each card configuration. + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Main clock id for each codec, specific for each card configuration. mux-int-port: $ref: /schemas/types.yaml#/definitions/uint32 @@ -167,10 +184,68 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller + # Properties relevant only with "fsl,imx-audio-generic" compatible + dai-tdm-slot-width: + description: See tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-num: + description: See tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + clocks: + description: | + The CPU DAI system clock, used to retrieve + the CPU DAI system clock frequency with the generic codec. + maxItems: 1 + + clock-names: + items: + - const: cpu_sysclk + + cpu-system-clock-direction-out: + description: | + Specifies cpu system clock direction as 'out' on initialization. + If not set, direction is 'in'. + $ref: /schemas/types.yaml#/definitions/flag + +dependencies: + dai-tdm-slot-width: + $ref: "#/definitions/imx-audio-generic-dependency" + dai-tdm-slot-num: + $ref: "#/definitions/imx-audio-generic-dependency" + clocks: + $ref: "#/definitions/imx-audio-generic-dependency" + cpu-system-clock-direction-out: + $ref: "#/definitions/imx-audio-generic-dependency" + required: - compatible - model +allOf: + - if: + $ref: "#/definitions/imx-audio-generic-dependency" + then: + properties: + audio-codec: + items: + - description: SPDIF DIT phandle + - description: SPDIF DIR phandle + mclk-id: + maxItems: 1 + items: + minItems: 1 + maxItems: 2 + else: + properties: + audio-codec: + maxItems: 1 + mclk-id: + maxItems: 1 + items: + maxItems: 1 + unevaluatedProperties: false examples: @@ -195,3 +270,16 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; }; + + - | + #include <dt-bindings/clock/imx8mn-clock.h> + sound-spdif-asrc { + compatible = "fsl,imx-audio-generic"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + clocks = <&clk IMX8MN_CLK_SAI5_ROOT>; + clock-names = "cpu_sysclk"; + cpu-system-clock-direction-out; + }; -- 2.34.1

3 10

[PATCH v4 0/2] ASoC: fsl_xcvr: Support i.MX95 platform
by Shengjiu Wang 05 Jun '24

05 Jun '24

On i.MX95 wakeup domain, there is one instance of Audio XCVR supporting SPDIF mode with a connection to the Audio XCVR physical interface. changes in v4: - refine the constarint for 'clocks' according to Rob's comments changes in v3: - refine the constraint for 'resets', 'clocks', 'interrupts' in document. changes in v2: - Merge patch 1&2, 3&4 from v1 together. - Add more comments in commit message - Add constaint for clocks used on i.mx95 - Add 'select SND_SOC_FSL_UTILS' for compiling issue. Shengjiu Wang (2): ASoC: dt-bindings: fsl,xcvr: Add compatible string for i.MX95 ASoC: fsl_xcvr: Add support for i.MX95 platform .../devicetree/bindings/sound/fsl,xcvr.yaml | 32 ++++- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_xcvr.c | 128 ++++++++++++------ sound/soc/fsl/fsl_xcvr.h | 91 +++++++++++++ 4 files changed, 208 insertions(+), 44 deletions(-) -- 2.34.1

3 4

[PATCH] dt-bindings: sound: tlv320adc3xxx: Fix incorrect GPIO description
by Ricard Wanderlof 05 Jun '24

05 Jun '24

Fix the description for the ti,dmdin-gpio1 and ti,dmclk-gpio2 properties to correctly describe that when configured as general purpose outputs (ADC3XXX_GPIO_GPO), the pins are available via the GPIO framework. Signed-off-by: Ricard Wanderlof <ricard.wanderlof(a)axis.com> --- Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml index ede14ca2c07a..8ac741f4cd56 100644 --- a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -58,8 +58,8 @@ properties: description: | Configuration for DMDIN/GPIO1 pin. - When ADC3XXX_GPIO_GPO is configured, this causes corresponding the - ALSA control "GPIOx Output" to appear, as a switch control. + When ADC3XXX_GPIO_GPO is selected, the pin may be controlled via the + GPIO framework, as pin number 0 on the device. ti,dmclk-gpio2: $ref: /schemas/types.yaml#/definitions/uint32 @@ -76,8 +76,8 @@ properties: description: | Configuration for DMCLK/GPIO2 pin. - When ADC3XXX_GPIO_GPO is configured, this causes corresponding the - ALSA control "GPIOx Output" to appear, as a switch control. + When ADC3XXX_GPIO_GPO is selected, the pin may be controlled via the + GPIO framework, as pin number 1 on the device. Note that there is currently no support for reading the GPIO pins as inputs. --- base-commit: 47d09270d7776e46858a161f94b735640b2fb056 change-id: 20240528-tlv320adc3xxx-dt-gpio-fix-a3384e435148 Best regards, -- -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 3

[PATCH 00/11] ASoC: simple-card: sync support
by Kuninori Morimoto 05 Jun '24

05 Jun '24

Hi Mark We have simle-card / audio-graph / audio-graph2, basically these supports same feature but is using different DT style. Because we are using 3 drivers, some feature was added to one driver, but other drivers doesn't have it. This patch set try to sync it on these 3 drivers. Kuninori Morimoto (11): ASoC: simple-card-utils: remove both playback/capture_only check ASoC: audio-graph-card2: add ep_to_port() / port_to_ports() ASoC: audio-graph-card2: remove ports node name check ASoC: audio-graph-card2: expand dai_link property part ASoC: audio-graph-card2: merge graph_parse_mclk_fs() into graph_link_init() ASoC: audio-graph-card: add ep_to_port() / port_to_ports() ASoC: audio-graph-card: remove ports node name check ASoC: audio-graph-card: enable playback/capture_only property ASoC: audio-graph-card: merge graph_parse_mclk_fs() into graph_link_init() ASoC: simple-audio-card: enable playback/capture_only property ASoC: simple-audio-card: merge simple_parse_mclk_fs() into simple_link_init() include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 95 ++++++++------ sound/soc/generic/audio-graph-card2.c | 171 +++++++++++++++----------- sound/soc/generic/simple-card-utils.c | 23 ++-- sound/soc/generic/simple-card.c | 56 +++++---- 5 files changed, 198 insertions(+), 149 deletions(-) -- 2.43.0

2 12

[PATCH v2 0/2] ASoC: fsl_xcvr: Support i.MX95 platform
by Shengjiu Wang 05 Jun '24

05 Jun '24

On i.MX95 wakeup domain, there is one instance of Audio XCVR supporting SPDIF mode with a connection to the Audio XCVR physical interface. changes in v2: - Merge patch 1&2, 3&4 from v1 together. - Add more comments in commit message - Add constaint for clocks used on i.mx95 - Add 'select SND_SOC_FSL_UTILS' for compiling issue. Shengjiu Wang (2): ASoC: dt-bindings: fsl,xcvr: Add compatible string for i.MX95 ASoC: fsl_xcvr: Add support for i.MX95 platform .../devicetree/bindings/sound/fsl,xcvr.yaml | 55 ++++++-- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_xcvr.c | 128 ++++++++++++------ sound/soc/fsl/fsl_xcvr.h | 91 +++++++++++++ 4 files changed, 222 insertions(+), 53 deletions(-) -- 2.34.1

4 5

[PATCH 0/2] tlv320adc3xxx: Allow MICBIAS pins to be used as GPOs
by Ricard Wanderlof 04 Jun '24

04 Jun '24

In some cases, depending on system design, the MICBIAS pins on the chip are not needed as such, but a couple of extra GPIO pins would be useful. This patch allows the MICBIAS pins to be configured in the device tree as general purpose output pins, controlled via the GPIO framework. Owing to their originally intended purpose there are some limitations: when the MICBIAS pins are deactivated, they will float, so will likely need a pulldown in many applications. When activated, they will assume the voltage specified by the micbias1-vg and micbias2-vg properties, respectively, meaning that the resulting output voltage will be 2.0 V, 2.5 V or AVDD . Signed-off-by: Ricard Wanderlof <ricard.wanderlof(a)axis.com> --- Ricard Wanderlof (2): ASoC: dt-bindings: tlv320adc3xxx: Add MICBIAS-as-GPO properties tlv320adc3xxx: Add support for using MICBIAS pins as GPO .../bindings/sound/ti,tlv320adc3xxx.yaml | 20 ++++ sound/soc/codecs/tlv320adc3xxx.c | 105 ++++++++++++++++----- 2 files changed, 104 insertions(+), 21 deletions(-) --- base-commit: 47d09270d7776e46858a161f94b735640b2fb056 change-id: 20240528-tlv320adc3xxx-micbias-gpo-b8b4d1c846d3 Best regards, -- -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 3

[PATCH] MAINTAINERS: copy linux-arm-msm for sound/qcom changes
by Dmitry Baryshkov 04 Jun '24

04 Jun '24

Not having linux-arm-msm@ in cc for audio-related changes for Qualcomm platforms means that interested parties can easily miss the patches. Add corresponding L: entry so that linux-arm-msm ML gets CC'ed for audio patches too. Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 523d84b2d613..b9bdd1f08005 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18227,6 +18227,7 @@ QCOM AUDIO (ASoC) DRIVERS M: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> M: Banajit Goswami <bgoswami(a)quicinc.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +L: linux-arm-msm(a)vger.kernel.org S: Supported F: Documentation/devicetree/bindings/soc/qcom/qcom,apr* F: Documentation/devicetree/bindings/sound/qcom,* --- base-commit: 9d99040b1bc8dbf385a8aa535e9efcdf94466e19 change-id: 20240531-asoc-qcom-cc-lamsm-be80e19f8cd8 Best regards, -- Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org>

3 2

[PATCH v6] ASoC: tas2781: Fix wrong loading calibrated data sequence
by Shenghao Ding 04 Jun '24

04 Jun '24

3 4

[PATCH 0/7] ALSA: hda: Improvements to hda_component
by Simon Trimmer 01 Jun '24

01 Jun '24

This series of patches makes sure that the existing consumers of the infrastructure unbind their interface before they begin destroying the driver, moves duplicated members from the instanced component structures into a new parent structure and introduces locking so that consumers of the interface do not use stale data. Simon Trimmer (7): ALSA: hda: cs35l56: Component should be unbound before deconstruction ALSA: hda: cs35l41: Component should be unbound before deconstruction ALSA: hda/tas2781: Component should be unbound before deconstruction ALSA: hda: hda_component: Introduce component parent structure ALSA: hda: hda_component: Change codecs to use component parent structure ALSA: hda: hda_component: Move codec field into the parent ALSA: hda: hda_component: Protect shared data with a mutex sound/pci/hda/cs35l41_hda.c | 47 +++++++++++--------- sound/pci/hda/cs35l56_hda.c | 29 +++++++------ sound/pci/hda/hda_component.c | 76 ++++++++++++++++++++------------- sound/pci/hda/hda_component.h | 48 ++++++++++++++------- sound/pci/hda/patch_realtek.c | 17 ++++---- sound/pci/hda/tas2781_hda_i2c.c | 37 ++++++++-------- 6 files changed, 148 insertions(+), 106 deletions(-) -- 2.34.1

2 9

[PATCH v2 0/6] ASoC: Drop or replace of_gpio.h
by Andy Shevchenko 31 May '24

31 May '24

Replace or drop the legacy header that is subject to remove. Not all of them were compile-tested, the series might have hidden compilation errors. In v2: - added tags (Kuninori, Charles) - ripped out TAS2781 (it's a mess from GPIO handling perspective) Andy Shevchenko (6): ASoC: codecs: Remove unused of_gpio.h ASoC: fsl: Remove unused of_gpio.h ASoC: rockchip: Remove unused of_gpio.h ASoC: codecs: Replace of_gpio.h by proper one ASoC: generic: Replace of_gpio.h by proper one ASoC: samsung: Replace of_gpio.h by proper one sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4458.c | 1 - sound/soc/codecs/aw88395/aw88395.c | 2 +- sound/soc/codecs/aw88399.c | 1 - sound/soc/codecs/cs53l30.c | 1 - sound/soc/codecs/max98390.c | 1 - sound/soc/codecs/pcm3168a.c | 1 - sound/soc/codecs/rk817_codec.c | 1 - sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tlv320adc3xxx.c | 1 - sound/soc/codecs/tlv320adcx140.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/ts3a227e.c | 1 - sound/soc/codecs/wsa883x.c | 1 - sound/soc/fsl/imx-es8328.c | 1 - sound/soc/fsl/imx-rpmsg.c | 2 -- sound/soc/generic/audio-graph-card2-custom-sample.c | 3 ++- sound/soc/rockchip/rockchip_i2s.c | 1 - sound/soc/rockchip/rockchip_spdif.c | 1 - sound/soc/samsung/aries_wm8994.c | 2 +- 23 files changed, 4 insertions(+), 24 deletions(-) -- 2.43.0.rc1.1336.g36b5255a03ac

2 7

[PATCHv4 7/9] ASoC: fsl-asoc-card: add DT clock "cpu_sysclk" with generic codec
by Elinor Montmasson 31 May '24

31 May '24

Add an optional DT clock "cpu_sysclk" to get the CPU DAI system-clock frequency when using the generic codec. It is set for both Tx and Rx. The way the frequency value is used is up to the CPU DAI driver implementation. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 9aca8ad15372..c7fc9c16f761 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -754,6 +754,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) snd_soc_of_parse_tdm_slot(np, NULL, NULL, &priv->cpu_priv.slot_num, &priv->cpu_priv.slot_width); + struct clk *cpu_sysclk = clk_get(&pdev->dev, "cpu_sysclk"); + if (!IS_ERR(cpu_sysclk)) { + priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(cpu_sysclk); + priv->cpu_priv.sysclk_freq[RX] = priv->cpu_priv.sysclk_freq[TX]; + clk_put(cpu_sysclk); + } } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; -- 2.34.1

2 7

[PATCH v1 0/6] ASoC: Drop or replace of_gpio.h
by Andy Shevchenko 31 May '24

31 May '24

Replace or drop the legacy header that is subject to remove. Not all of them were compile-tested, the series might have hidden compilation errors. Andy Shevchenko (6): ASoC: codecs: Remove unused of_gpio.h ASoC: fsl: Remove unused of_gpio.h ASoC: rockchip: Remove unused of_gpio.h ASoC: codecs: Replace of_gpio.h by proper one ASoC: generic: Replace of_gpio.h by proper one ASoC: samsung: Replace of_gpio.h by proper one sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4458.c | 1 - sound/soc/codecs/aw88395/aw88395.c | 2 +- sound/soc/codecs/aw88399.c | 1 - sound/soc/codecs/cs53l30.c | 1 - sound/soc/codecs/max98390.c | 1 - sound/soc/codecs/pcm3168a.c | 1 - sound/soc/codecs/rk817_codec.c | 1 - sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tas2781-comlib.c | 1 - sound/soc/codecs/tas2781-fmwlib.c | 1 - sound/soc/codecs/tas2781-i2c.c | 1 - sound/soc/codecs/tlv320adc3xxx.c | 1 - sound/soc/codecs/tlv320adcx140.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/ts3a227e.c | 1 - sound/soc/codecs/wsa883x.c | 1 - sound/soc/fsl/imx-es8328.c | 1 - sound/soc/fsl/imx-rpmsg.c | 2 -- sound/soc/generic/audio-graph-card2-custom-sample.c | 3 ++- sound/soc/rockchip/rockchip_i2s.c | 1 - sound/soc/rockchip/rockchip_spdif.c | 1 - sound/soc/samsung/aries_wm8994.c | 2 +- 26 files changed, 4 insertions(+), 27 deletions(-) -- 2.43.0.rc1.1336.g36b5255a03ac

4 11

[PATCH v4] ASoc: tas2781: Enable RCA-based playback without DSP firmware download
by Shenghao Ding 31 May '24

31 May '24

In only RCA (Reconfigurable Architecture) binary case, no DSP program will be working inside tas2563/tas2781, that is dsp-bypass mode, do not support speaker protection, and audio acoustic algorithms in this mode. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - add a description of what the state machine looks like, and why FW_PENDING/FAIL remain but are not used. - Remove TASDEVICE_DSP_FW_NONE because of unused. - Remove stray change. - Fix broken indentation. v3: - Add description on RCA is Reconfigurable Architecture. - Add the description on enabling - Reword the commit - Remove question mark in the comments. - Add spaces in comments. v2: - Correct comment. - Add Fixes. - Move header file to the first. v1: - Split out the different logical changes into different patches. - rename tasdevice_dsp_fw_state -> tasdevice_fw_state, the fw are not only DSP fw, but also RCA(Reconfigurable data, such as acoustic data and register setting, etc). - Add TASDEVICE_RCA_FW_OK in tasdevice_fw_state to identify the state that only RCA binary file has been download successfully, but DSP fw is not loaded or loading failure. - Add the this strategy into tasdevice_tuning_switch. - If one side of the if/else has a braces both should in tasdevice_tuning_switch. - Identify whehter both RCA and DSP have been loaded or only RCA has been loaded in tasdevice_fw_ready. - Add check fw load status in tasdevice_startup. - remove ret in tasdevice_startup to make the code neater. --- include/sound/tas2781-dsp.h | 11 ++++++++-- sound/soc/codecs/tas2781-fmwlib.c | 18 +++++++++++----- sound/soc/codecs/tas2781-i2c.c | 34 +++++++++++++++++++------------ 3 files changed, 43 insertions(+), 20 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 7fba7ea26a4b..3cda9da14f6d 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -117,10 +117,17 @@ struct tasdevice_fw { struct device *dev; }; -enum tasdevice_dsp_fw_state { - TASDEVICE_DSP_FW_NONE = 0, +enum tasdevice_fw_state { + /* Driver in startup mode, not load any firmware. */ TASDEVICE_DSP_FW_PENDING, + /* DSP firmware in the system, but parsing error. */ TASDEVICE_DSP_FW_FAIL, + /* + * Only RCA (Reconfigurable Architecture) firmware load + * successfully. + */ + TASDEVICE_RCA_FW_OK, + /* Both RCA and DSP firmware load successfully. */ TASDEVICE_DSP_FW_ALL_OK, }; diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 265a8ca25cbb..838d29fead96 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2324,14 +2324,21 @@ void tasdevice_tuning_switch(void *context, int state) struct tasdevice_fw *tas_fmw = tas_priv->fmw; int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + /* + * Only RCA-based Playback can still work with no dsp program running + * inside the chip. + */ + switch (tas_priv->fw_state) { + case TASDEVICE_RCA_FW_OK: + case TASDEVICE_DSP_FW_ALL_OK: + break; + default: return; } if (state == 0) { - if (tas_priv->cur_prog < tas_fmw->nr_programs) { - /*dsp mode or tuning mode*/ + if (tas_fmw && tas_priv->cur_prog < tas_fmw->nr_programs) { + /* dsp mode or tuning mode */ profile_cfg_id = tas_priv->rcabin.profile_cfg_id; tasdevice_select_tuningprm_cfg(tas_priv, tas_priv->cur_prog, tas_priv->cur_conf, @@ -2340,9 +2347,10 @@ void tasdevice_tuning_switch(void *context, int state) tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_POWER_UP); - } else + } else { tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } } EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, SND_SOC_TAS2781_FMWLIB); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9350972dfefe..9c3c89cb36de 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -380,23 +380,32 @@ static void tasdevice_fw_ready(const struct firmware *fmw, mutex_lock(&tas_priv->codec_lock); ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) + if (ret) { + tasdevice_config_info_remove(tas_priv); goto out; + } tasdevice_create_control(tas_priv); tasdevice_dsp_remove(tas_priv); tasdevice_calbin_remove(tas_priv); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", tas_priv->dev_name); ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; goto out; } - tasdevice_dsp_create_ctrls(tas_priv); + + /* + * If no dsp-related kcontrol created, the dsp resource will be freed. + */ + ret = tasdevice_dsp_create_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dsp controls error\n"); + goto out; + } tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; @@ -417,9 +426,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - /*If DSP FW fail, kcontrol won't be created */ - tasdevice_config_info_remove(tas_priv); + if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } mutex_unlock(&tas_priv->codec_lock); @@ -466,14 +474,14 @@ static int tasdevice_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int ret = 0; - if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); - ret = -EINVAL; + switch (tas_priv->fw_state) { + case TASDEVICE_RCA_FW_OK: + case TASDEVICE_DSP_FW_ALL_OK: + return 0; + default: + return -EINVAL; } - - return ret; } static int tasdevice_hw_params(struct snd_pcm_substream *substream, -- 2.34.1

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[PATCHv4 8/9] ASoC: fsl-asoc-card: add DT property "cpu-system-clock-direction-out"
by Elinor Montmasson 31 May '24

31 May '24

Add new optional DT property "cpu-system-clock-direction-out" to set sysclk direction as "out" for the CPU DAI when using the generic codec. It is set for both Tx and Rx. If not set, the direction is "in". The way the direction value is used is up to the CPU DAI driver implementation. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c7fc9c16f761..f3fc2b29c92f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -760,6 +760,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_freq[RX] = priv->cpu_priv.sysclk_freq[TX]; clk_put(cpu_sysclk); } + priv->cpu_priv.sysclk_dir[TX] = + of_property_read_bool(np, "cpu-system-clock-direction-out") ? + SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; + priv->cpu_priv.sysclk_dir[RX] = priv->cpu_priv.sysclk_dir[TX]; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; -- 2.34.1

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[PATCH] ALSA: hda: cs35l41: Possible null pointer dereference in cs35l41_hda_unbind()
by Simon Trimmer 31 May '24

31 May '24

The cs35l41_hda_unbind() function clears the hda_component entry matching it's index and then dereferences the codec pointer held in the first element of the hda_component array, this is an issue when the device index was 0. Instead use the codec pointer stashed in the cs35l41_hda structure as it will still be valid. Fixes: 7cf5ce66dfda ("ALSA: hda: cs35l41: Add device_link between HDA and cs35l41_hda") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 6c49e5c6cd20..d54d4d60b03e 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1495,7 +1495,7 @@ static void cs35l41_hda_unbind(struct device *dev, struct device *master, void * if (comps[cs35l41->index].dev == dev) { memset(&comps[cs35l41->index], 0, sizeof(*comps)); sleep_flags = lock_system_sleep(); - device_link_remove(&comps->codec->core.dev, cs35l41->dev); + device_link_remove(&cs35l41->codec->core.dev, cs35l41->dev); unlock_system_sleep(sleep_flags); } } -- 2.34.1

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[PATCH] ALSA: hda: cs35l56: Fix lifecycle of codec pointer
by Simon Trimmer 31 May '24

31 May '24

The codec should be cleared when the amp driver is unbound and when resuming it should be tested to prevent loading firmware into the device and ALSA in a partially configured system state. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 11b0570ff56d..0923e2589f5f 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -735,6 +735,8 @@ static void cs35l56_hda_unbind(struct device *dev, struct device *master, void * if (comps[cs35l56->index].dev == dev) memset(&comps[cs35l56->index], 0, sizeof(*comps)); + cs35l56->codec = NULL; + dev_dbg(cs35l56->base.dev, "Unbound\n"); } @@ -840,6 +842,9 @@ static int cs35l56_hda_system_resume(struct device *dev) cs35l56->suspended = false; + if (!cs35l56->codec) + return 0; + ret = cs35l56_is_fw_reload_needed(&cs35l56->base); dev_dbg(cs35l56->base.dev, "fw_reload_needed: %d\n", ret); if (ret > 0) { -- 2.34.1

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[PATCH v6 0/1] Add tas2781 driver for SPI.
by Baojun Xu 31 May '24

31 May '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (1): ALSA: hda/tas2781: Add tas2781 hda driver based on SPI drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 12 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 148 ++ sound/pci/hda/tas2781_hda_spi.c | 1266 ++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2101 +++++++++++++++++ 8 files changed, 3557 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

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[GIT PULL] ASoC fixes for v6.10-rc1
by Mark Brown 30 May '24

30 May '24

The following changes since commit 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0: Linux 6.10-rc1 (2024-05-26 15:20:12 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.10-rc1 for you to fetch changes up to c85578e730135d47271ca874e4436dd87e87f252: ASoC: SOF: ipc4-topology: Fix nhlt configuration blob (2024-05-30 14:33:14 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.10 Several serieses of fixes that have come in since the merge window, mostly for Intel systems. The biggest thing is some updates from Peter which fix support for a series of Intel laptops which have been found to use 16 bit rather than 32 bit DMIC configuration blobs in their firmware descriptions. We also have a bunch of fixes for module annotations, and some smaller single patch fixes. ---------------------------------------------------------------- Charles Keepax (3): ASoC: cs42l43: Only restrict 44.1kHz for the ASP ASoC: wm_adsp: Add missing MODULE_DESCRIPTION() MAINTAINERS: Remove James Schulman from Cirrus audio maintainers Mark Brown (2): ASoC: SOF: add missing MODULE_DESCRIPTION ASoC: SOF: ipc4-topology: Fix nhlt configuration blob Peter Ujfalusi (6): ASoC: SOF: ipc4-topology: Fix input format query of process modules without base extension ASoC: SOF: ipc4-topology: Add support for NHLT with 16-bit only DMIC blob ASoC: SOF: ipc4-topology: Print out the channel count in sof_ipc4_dbg_audio_format ASoC: SOF: ipc4-topology/pcm: Rename sof_ipc4_copier_is_single_format() ASoC: SOF: ipc4-topology: Improve readability of sof_ipc4_prepare_dai_copier() ASoC: SOF: ipc4-topology: Adjust the params based on DAI formats Pierre-Louis Bossart (5): ASoC: SOF: stream-ipc: remove unnecessary MODULE_LICENSE ASoC: SOF: AMD: group all module related information ASoC: SOF: reorder MODULE_ definitions ASoC: SOF: add missing MODULE_DESCRIPTION() ASoC: Intel: sof-sdw: fix missing SPI_MASTER dependency MAINTAINERS | 1 - sound/soc/codecs/cs42l43.c | 5 +- sound/soc/codecs/wm_adsp.c | 1 + sound/soc/intel/boards/Kconfig | 2 +- sound/soc/sof/amd/acp-common.c | 4 +- sound/soc/sof/amd/acp.c | 2 +- sound/soc/sof/amd/acp63.c | 4 - sound/soc/sof/amd/pci-acp63.c | 1 + sound/soc/sof/amd/pci-rmb.c | 1 + sound/soc/sof/amd/pci-rn.c | 1 + sound/soc/sof/amd/pci-vangogh.c | 1 + sound/soc/sof/amd/rembrandt.c | 4 - sound/soc/sof/amd/renoir.c | 4 - sound/soc/sof/amd/vangogh.c | 4 - sound/soc/sof/core.c | 2 +- sound/soc/sof/imx/imx-common.c | 1 + sound/soc/sof/imx/imx8.c | 3 +- sound/soc/sof/imx/imx8m.c | 3 +- sound/soc/sof/imx/imx8ulp.c | 3 +- sound/soc/sof/intel/atom.c | 1 + sound/soc/sof/intel/bdw.c | 1 + sound/soc/sof/intel/byt.c | 1 + sound/soc/sof/intel/hda-codec.c | 1 + sound/soc/sof/intel/hda-ctrl.c | 1 + sound/soc/sof/intel/hda-mlink.c | 1 + sound/soc/sof/intel/hda.c | 1 + sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-lnl.c | 1 + sound/soc/sof/intel/pci-mtl.c | 1 + sound/soc/sof/intel/pci-skl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/intel/pci-tng.c | 1 + sound/soc/sof/ipc4-pcm.c | 12 +- sound/soc/sof/ipc4-topology.c | 163 +++++++++++++++++-------- sound/soc/sof/ipc4-topology.h | 6 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 3 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 3 +- sound/soc/sof/mediatek/mtk-adsp-common.c | 1 + sound/soc/sof/nocodec.c | 2 +- sound/soc/sof/sof-acpi-dev.c | 1 + sound/soc/sof/sof-client-ipc-flood-test.c | 2 +- sound/soc/sof/sof-client-ipc-kernel-injector.c | 2 +- sound/soc/sof/sof-client-ipc-msg-injector.c | 2 +- sound/soc/sof/sof-client-probes.c | 2 +- sound/soc/sof/sof-of-dev.c | 1 + sound/soc/sof/sof-pci-dev.c | 1 + sound/soc/sof/sof-utils.c | 1 + sound/soc/sof/stream-ipc.c | 2 - sound/soc/sof/xtensa/core.c | 2 +- 51 files changed, 173 insertions(+), 95 deletions(-)

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Re:[PATCH 6.8 215/493] cpufreq: exit() callback is optional
by lizhe 30 May '24

30 May '24

Hi, Viresh Kumar Do you take to heart the efforts and contributions of others to the community ? 1. I submitted this patch. You included it in your patch completely. > > static int cpufreq_offline(unsigned int cpu) > >@@ -1731,7 +1734,7 @@ static void cpufreq_remove_dev(struct device *dev, struct subsys_interface *sif) > > } > > > > /* We did light-weight exit earlier, do full tear down now */ > >- if (cpufreq_driver->offline) > >+ if (cpufreq_driver->offline && cpufreq_driver->exit) > > cpufreq_driver->exit(policy); > > > > up_write(&policy->rwsem); > >-- 2. You have seen the three patches I submitted to you and the main line before submitting this patch. > >@@ -1670,10 +1670,13 @@ static void __cpufreq_offline(unsigned int cpu, struct cpufreq_policy *policy) > > */ > > if (cpufreq_driver->offline) { > > cpufreq_driver->offline(policy); > >- } else if (cpufreq_driver->exit) { > >- cpufreq_driver->exit(policy); > >- policy->freq_table = NULL; > >+ return; > > } > >+ > >+ if (cpufreq_driver->exit) > >+ cpufreq_driver->exit(policy); > >+ > >+ policy->freq_table = NULL; > > } As a senior member of the kernel community, you should uphold the fairness and justice of the community. Why do you have to rush in and take over others' patches? Isn't that selfish of you? Your behavior disrespects others' efforts and enthusiasm, and it undermines the collaborative, harmonious, and fair atmosphere of the community. I have tried to communicate with you several times, but you have refused. Is it because you are the maintainer of the community ? Where does your authority come from ? thanks. Lizhe At 2024-05-28 02:53:37, "Greg Kroah-Hartman" <gregkh(a)linuxfoundation.org> wrote: >6.8-stable review patch. If anyone has any objections, please let me know. > >------------------ > >From: Viresh Kumar <viresh.kumar(a)linaro.org> > >[ Upstream commit b8f85833c05730d631576008daaa34096bc7f3ce ] > >The exit() callback is optional and shouldn't be called without checking >a valid pointer first. > >Also, we must clear freq_table pointer even if the exit() callback isn't >present. > >Signed-off-by: Viresh Kumar <viresh.kumar(a)linaro.org> >Fixes: 91a12e91dc39 ("cpufreq: Allow light-weight tear down and bring up of CPUs") >Fixes: f339f3541701 ("cpufreq: Rearrange locking in cpufreq_remove_dev()") >Reported-by: Lizhe <sensor1010(a)163.com> >Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> >Signed-off-by: Sasha Levin <sashal(a)kernel.org> >--- > drivers/cpufreq/cpufreq.c | 11 +++++++---- > 1 file changed, 7 insertions(+), 4 deletions(-) > >diff --git a/drivers/cpufreq/cpufreq.c b/drivers/cpufreq/cpufreq.c >index 3c2c955fbbbd6..86f1bc7754ea6 100644 >--- a/drivers/cpufreq/cpufreq.c >+++ b/drivers/cpufreq/cpufreq.c >@@ -1670,10 +1670,13 @@ static void __cpufreq_offline(unsigned int cpu, struct cpufreq_policy *policy) > */ > if (cpufreq_driver->offline) { > cpufreq_driver->offline(policy); >- } else if (cpufreq_driver->exit) { >- cpufreq_driver->exit(policy); >- policy->freq_table = NULL; >+ return; > } >+ >+ if (cpufreq_driver->exit) >+ cpufreq_driver->exit(policy); >+ >+ policy->freq_table = NULL; > } > > static int cpufreq_offline(unsigned int cpu) >@@ -1731,7 +1734,7 @@ static void cpufreq_remove_dev(struct device *dev, struct subsys_interface *sif) > } > > /* We did light-weight exit earlier, do full tear down now */ >- if (cpufreq_driver->offline) >+ if (cpufreq_driver->offline && cpufreq_driver->exit) > cpufreq_driver->exit(policy); > > up_write(&policy->rwsem); >-- >2.43.0 > >

1 0

[PATCH v2 0/2] Support Tegra I2S client format conversion
by Sameer Pujar 30 May '24

30 May '24

From: Mohan Kumar <mkumard(a)nvidia.com> The AHUB HW modules are interconnected with CIF which are capable of supporting Channel and Sample bit format conversion. Due to this, the I2S Client can have different Channel and Sample bit from the hw_params() and this config is passed from CIF port of I2S DT node which can help to perform this conversion. - First change to split simple_fixup_sample_fmt to support returning sample format value - Second patch to support Tegra I2S client channel and sample format programming based on CIF port from DT node. Changelog ========= v2: - Fix sparse warning related to incorrect type assignment for variable 'sample_format' in hw_params() call. The defect was reported by "kernel test robot <lkp(a)intel.com>" and details can be found below: https://lore.kernel.org/oe-kbuild-all/202405211805.UMAdiH0d-lkp@intel.com/ Mohan Kumar (2): ASoC: simple-card-utils: Split simple_fixup_sample_fmt func ASoC: tegra: I2S client convert formats handling include/sound/simple_card_utils.h | 2 + sound/soc/generic/simple-card-utils.c | 26 +++++++--- sound/soc/tegra/tegra210_i2s.c | 71 +++++++++++++++++++++++++-- sound/soc/tegra/tegra210_i2s.h | 2 + 4 files changed, 91 insertions(+), 10 deletions(-) -- 2.45.1

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[RFC 00/12] ASoC: remove/replace dpcm_xxx flag
by Kuninori Morimoto 30 May '24

30 May '24

Hi each ASoC driver maintainer As I posted in "ASoC: grace time for DPCM cleanup", dpcm_xxx flag will be remove. Then, some driver might got damage. This patch-set try to remove/replace dpcm_xxx flag, but is not tesed. It is just "RFC" patch, but I'm happy if you can reuse it on your driver. Thank you for your help !! Best regards --- Kuninori Morimoto

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