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

USB-Audio: Add support for Focusrite 4th Gen devices
by GitHub pull_request - opened 21 Aug '24

21 Aug '24

alsa-project/alsa-ucm-conf pull request #438 was opened from ChrisLane: Extending the existing support for Focusrite Scarlett Solo and 2i2 interfaces with support for the 4th Gen devices. I have a Focusrite Scarlett 2i2 which is what I have tested with and used to get the device ID. For the Solo device, I grabbed the device ID from [linux-hardware.org](https://linux-hardware.org/?id=usb:1235-8218). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/438 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/438.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v24 00/34] Introduce QC USB SND audio offloading support
by Wesley Cheng 21 Aug '24

21 Aug '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... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. 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 v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. 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. 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 (32): 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 usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device 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 ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: qcom: Populate PCM and USB chip information ALSA: usb-audio: qcom: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add 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 .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 435 ++++ 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 | 419 ++++ 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 | 135 ++ 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 | 372 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 331 +++ 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 | 101 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1937 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 5768 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

5 86

No sound from XonarDX
by GitHub issues - opened 21 Aug '24

21 Aug '24

alsa-project/alsa-lib issue #408 was opened from 3bitbrain: No matter what wiki or advice I try, no sound at all from a known good Xonar DX card. Distro : Gentoo, fresh install. Always un-muted everything in the console alsamixer. alsa-lib version 1.2.11 [alsa.txt](https://github.com/user-attachments/files/16697602/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-lib/issues/408 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested()
by Kuninori Morimoto 21 Aug '24

21 Aug '24

soc-pcm.c has snd_soc_dpcm_stream_lock_irqsave_nested() / snd_soc_dpcm_stream_unlock_irqrestore() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased to latest branch (for-6.12) sound/soc/soc-pcm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f79abcf36105..baf8b81e7110 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,12 +49,6 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) - -#define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ - snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) - #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -2144,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); + snd_pcm_stream_lock_irqsave_nested(be_substream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) @@ -2291,7 +2285,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, break; } next: - snd_soc_dpcm_stream_unlock_irqrestore(be, stream, flags); + snd_pcm_stream_unlock_irqrestore(be_substream, flags); if (ret) break; } -- 2.43.0

2 1

[PATCH] ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure
by Vijendar Mukunda 21 Aug '24

21 Aug '24

Remove unused structure member 'rev' from sof_amd_acp_desc structure. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp.h | 1 - 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 - 5 files changed, 5 deletions(-) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 321c40cc6d50..11def07efc0f 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -190,7 +190,6 @@ struct acp_dsp_stream { }; struct sof_amd_acp_desc { - unsigned int rev; const char *name; unsigned int host_bridge_id; u32 pgfsm_base; diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index e90658ba2bd7..b54ed61b79ed 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -28,7 +28,6 @@ #define ACP6x_REG_END 0x125C000 static const struct sof_amd_acp_desc acp63_chip_info = { - .rev = 6, .host_bridge_id = HOST_BRIDGE_ACP63, .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_enb = ACP6X_EXTERNAL_INTR_ENB, diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index a366f904e6f3..c45256bf4fda 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -28,7 +28,6 @@ #define ACP6X_FUTURE_REG_ACLK_0 0x1854 static const struct sof_amd_acp_desc rembrandt_chip_info = { - .rev = 6, .host_bridge_id = HOST_BRIDGE_RMB, .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_stat = ACP6X_EXT_INTR_STAT, diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 2b7c53470ce8..386a0f1e7ee0 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -28,7 +28,6 @@ #define ACP3X_FUTURE_REG_ACLK_0 0x1860 static const struct sof_amd_acp_desc renoir_chip_info = { - .rev = 3, .host_bridge_id = HOST_BRIDGE_CZN, .pgfsm_base = ACP3X_PGFSM_BASE, .ext_intr_stat = ACP3X_EXT_INTR_STAT, diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c index eba580840100..cb845f81795e 100644 --- a/sound/soc/sof/amd/pci-vangogh.c +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -26,7 +26,6 @@ #define ACP5X_FUTURE_REG_ACLK_0 0x1864 static const struct sof_amd_acp_desc vangogh_chip_info = { - .rev = 5, .name = "vangogh", .host_bridge_id = HOST_BRIDGE_VGH, .pgfsm_base = ACP5X_PGFSM_BASE, -- 2.34.1

2 1

[PATCH] ASoC: codecs: wcd934x: make read-only array minCode_param static const
by Colin Ian King 21 Aug '24

21 Aug '24

Don't populate the read-only array minCode_param on the stack at run time, instead make it static const. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/wcd934x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 291d0c80a6fc..910852eb9698 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -2643,8 +2643,8 @@ static void wcd934x_mbhc_get_result_params(struct wcd934x_codec *wcd934x, s16 c1; s32 x1, d1; int32_t denom; - int minCode_param[] = { - 3277, 1639, 820, 410, 205, 103, 52, 26 + static const int minCode_param[] = { + 3277, 1639, 820, 410, 205, 103, 52, 26 }; regmap_update_bits(wcd934x->regmap, WCD934X_ANA_MBHC_ZDET, 0x20, 0x20); -- 2.39.2

2 1

[PATCH v1] ASoC: tas2781: Remove unnecessary line feed for tasdevice_codec_remove
by Baojun Xu 21 Aug '24

21 Aug '24

Remove unnecessary line feed for tasdevice_codec_remove. Add comma at the end the last member of the array. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v1: --- sound/soc/codecs/tas2781-i2c.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..eb8732b1bb99 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -582,13 +582,13 @@ static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), SND_SOC_DAPM_OUTPUT("OUT"), - SND_SOC_DAPM_INPUT("DMIC") + SND_SOC_DAPM_INPUT("DMIC"), }; static const struct snd_soc_dapm_route tasdevice_audio_map[] = { {"SPK", NULL, "ASI"}, {"OUT", NULL, "SPK"}, - {"ASI OUT", NULL, "DMIC"} + {"ASI OUT", NULL, "DMIC"}, }; static int tasdevice_startup(struct snd_pcm_substream *substream, @@ -730,8 +730,7 @@ static void tasdevice_deinit(void *context) tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; } -static void tasdevice_codec_remove( - struct snd_soc_component *codec) +static void tasdevice_codec_remove(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); -- 2.43.0

2 2

[PATCH] ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested()
by Kuninori Morimoto 21 Aug '24

21 Aug '24

soc-pcm.c has snd_soc_dpcm_stream_lock_irqsave_nested() / snd_soc_dpcm_stream_unlock_irqrestore() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 58381801e41d4..4571785736f76 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,12 +49,6 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) - -#define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ - snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) - #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -2144,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); + snd_pcm_stream_lock_irqsave_nested(be_substream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) @@ -2291,7 +2285,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, break; } next: - snd_soc_dpcm_stream_unlock_irqrestore(be, stream, flags); + snd_pcm_stream_unlock_irqrestore(be_substream, flags); if (ret) break; } -- 2.43.0

2 2

Re: [PATCH] ALSA: hda: cs35l56: Don't use the device index as a calibration index
by Takashi Iwai 21 Aug '24

21 Aug '24

On Wed, 21 Aug 2024 14:47:11 +0200, Simon Trimmer wrote: > > The HDA driver cannot assume that the order that the devices are > specified in the cirrus,dev-index matches the order of calibration > entries. > > Only a calibration entry with a matching silicon id will be used. > > Fixes: cfa43aaa7948 ("ALSA: hda: cs35l56: Apply amp calibration from EFI data") > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Thanks, applied now. Takashi

1 0

Re: [RESEND] ASoC: mediatek: mt8188-mt6359: Modify key
by Mark Brown 21 Aug '24

21 Aug '24

On Wed, Aug 21, 2024 at 03:40:51PM +0800, Zhang Yi wrote: > In order to get the correct keys when using the ES8326.We will associate > SND_JACK_BTN_1 to KEY_VOLUMEUP and SND_JACK_BTN_2 to KEY_VOLUMEDOWN > when the ES8326 flag is recognized. Please please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review.

1 0

[PATCH v2 1/1] ASoC: dt-bindings: Convert tpa6130a2.txt to yaml
by Frank Li 20 Aug '24

20 Aug '24

Convert binding doc tpa6130a2.txt to yaml format. Additional change: - add ref to dai-common.yaml - add i2c node in example Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a20000/amp@60: failed to match any schema with compatible: ['ti,tpa6130a2'] Reviewed-by: Rob Herring (Arm) <robh(a)kernel.org> Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- Change from v1 to v2 - fix MAINTAINERS --- .../bindings/sound/ti,tpa6130a2.yaml | 55 +++++++++++++++++++ .../devicetree/bindings/sound/tpa6130a2.txt | 27 --------- MAINTAINERS | 2 +- 3 files changed, 56 insertions(+), 28 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml new file mode 100644 index 0000000000000..a42bf9bde6940 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tpa6130a2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tpa6130a2 Codec module + +maintainers: + - Sebastian Reichel <sre(a)kernel.org> + +description: + Stereo, analog input headphone amplifier + +properties: + compatible: + enum: + - ti,tpa6130a2 + - ti,tpa6140a2 + + reg: + maxItems: 1 + + Vdd-supply: + description: power supply regulator + + power-gpio: + description: gpio pin to power the device + +required: + - compatible + - reg + - Vdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt deleted file mode 100644 index 6dfa740e4b2d8..0000000000000 --- a/Documentation/devicetree/bindings/sound/tpa6130a2.txt +++ /dev/null @@ -1,27 +0,0 @@ -Texas Instruments - tpa6130a2 Codec module - -The tpa6130a2 serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tpa6130a2" - TPA6130A2 - "ti,tpa6140a2" - TPA6140A2 - - -- reg - <int> - I2C slave address - -- Vdd-supply - <phandle> - power supply regulator - -Optional properties: - -- power-gpio - gpio pin to power the device - -Example: - -tpa6130a2: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - reg = <0x60>; - Vdd-supply = <&vmmc2>; - power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; -}; diff --git a/MAINTAINERS b/MAINTAINERS index 118d9cf64e9cb..4ced49519930b 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22713,12 +22713,12 @@ F: Documentation/devicetree/bindings/sound/tas2552.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h -- 2.34.1

2 1

[PATCH] ASoC: amd: acp: replace desc->rev check with acp pci revision id
by Vijendar Mukunda 20 Aug '24

20 Aug '24

Replace acp descriptor structure member 'rev' check with acp pci revision id. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 0f6115c8b005..e4d46fdda88b 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -236,7 +236,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, unsigned int image_length) { struct snd_sof_dev *sdev = adata->dev; - const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int tx_count, fw_qualifier, val; int ret; @@ -265,8 +264,9 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* psp_send_cmd only required for vangogh platform */ + if (adata->pci_rev == ACP_VANGOGH_PCI_ID && + !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { /* Modify IRAM and DRAM size */ ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); if (ret) @@ -285,8 +285,8 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for renoir platform (rev - 3) */ - if (desc->rev == 3) { + /* psp_send_cmd only required for renoir platform*/ + if (adata->pci_rev == ACP_RN_PCI_ID) { ret = psp_send_cmd(adata, MBOX_ACP_SHA_DMA_COMMAND); if (ret) return ret; @@ -405,7 +405,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_sw0_i2s_err_reason, 0); /* ACP_SW1_I2S_ERROR_REASON is newly added register from rmb platform onwards */ - if (desc->rev >= 6) + if (adata->pci_rev >= ACP_RMB_PCI_ID) snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SW1_I2S_ERROR_REASON, 0); snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_error_stat, 0); irq_flag = 1; @@ -431,6 +431,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) static int acp_power_on(struct snd_sof_dev *sdev) { const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); + struct acp_dev_data *adata = sdev->pdata->hw_pdata; unsigned int base = desc->pgfsm_base; unsigned int val; unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; @@ -441,13 +442,14 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - switch (desc->rev) { - case 3: - case 5: + switch (adata->pci_rev) { + case ACP_RN_PCI_ID: + case ACP_VANGOGH_PCI_ID: acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; break; - case 6: + case ACP_RMB_PCI_ID: + case ACP63_PCI_ID: acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; break; -- 2.34.1

2 1

[RFC PATCH 00/16] soundwire/ASoC: speed-up downloads with BTP/BRA protocol
by Pierre-Louis Bossart 20 Aug '24

20 Aug '24

This RFC patchset suggests a new API for ASoC codec drivers to use for firmware/table downloads. Problem statement: All existing transfers initiated by codec drivers rely on SoundWire read/write commands, which can only support ONE byte per frame. With the typical 48kHz frame rate, this means 384 kbits/s. In addition, the command/control is typically handled with a FIFO and interrupts which adds more software overhead. To give a practical reference, sending 32Kb takes 2.5s on Intel platforms, which means about 105kbit/s only. Additional non-audio activity is likely to adversely impact interrupt scheduling and further decrease the transfer speeds. New SDCA-based codecs have a need to download tables and DSP firmware which are typically between 20 and 256 Kb. The slow bus operation has a direct impact on boot/resume times, and clearly having to wait more than 300ms is a showstopper in terms of latency requirements and user-experience. Suggested solution: The MIPI specification and most of the new codecs support the Bulk Transfer Protocol (BTP) and specifically the Bulk Register Access (BRA) configuration. This mode reclaims the 'audio' data space of the SoundWire frame to send firmware/coefficients over the DataPort 0 (DP0). The API suggested is rather simple, with the following sequence expected: open(): reserve resources and prepare hardware send_async(): trigger DMAs and perform SoundWire bank switch wait(): wait for DMA completion and disable DMAs close(): release resources Benefits: Even after accounting for the protocol overhead, the data can be sent 8x or 16x faster on the link than with the regular commands. With the use of DMAs, the software overhead becomes limited to the initialization. Measured results show that transferring the same 32Kb takes about 100ms, a 25x improvement on the baseline write() commands with an actual bitrate of 2.6 Mbits/s. These results are a measure of bus/hardware performance mainly, and will typically not be too modified by the CPU activity and scheduling. The performance for reads is similar, with a 25x speedup measured. Limitations: Setting up the transfers over DP0 takes time, and the reliance on DMAs on the host side brings alignment restrictions. The BTP/BRA protocol is really only relevant for "large" transfers done during boot/resume *before* audio transfers take place. Mixing BTP/BRA and audio is a nightmare, this patchset suggests a mutual-exclusion between two usages. Scope: This patchset only exposes the API and a debugfs interface to initiate commands, validate results and measure performance. The actual use of the API is left as an exercise for codec driver developers. This patchset depends on a number of pre-requisite patches and will not build on top of any for-next branches. The main intent of this RFC is to gather comments on the usage, API, benefits and restrictions. The code and functionality was tested on an Intel LunarLake RVP platform connected to a Realtek RT711-SDCA device. Acknowledgements: Thanks to Zeek Tsai at Realtek for providing test sequences that helped reconcile the data formatted by the host driver with the expected results on the code side. Pierre-Louis Bossart (16): Documentation: driver: add SoundWire BRA description soundwire: cadence: add BTP support for DP0 soundwire: stream: extend sdw_alloc_stream() to take 'type' parameter soundwire: extend sdw_stream_type to BPT soundwire: stream: special-case the bus compute_params() routine soundwire: stream: reuse existing code for BPT stream soundwire: bus: add API for BPT protocol soundwire: bus: add bpt_stream pointer soundwire: crc8: add constant table soundwire: cadence: add BTP/BRA helpers to format data soundwire: intel_auxdevice: add indirection for BPT open/close/send_async/wait ASoC: SOF: Intel: hda-sdw-bpt: add helpers for SoundWire BPT DMA soundwire: intel: add BPT context definition soundwire: intel_ace2x: add BPT open/close/send_async/wait soundwire: debugfs: add interface for BPT/BRA transfers ASoC: rt711-sdca: add DP0 support Documentation/driver-api/soundwire/bra.rst | 478 +++++++++++++ Documentation/driver-api/soundwire/index.rst | 1 + Documentation/driver-api/soundwire/stream.rst | 2 +- .../driver-api/soundwire/summary.rst | 5 +- drivers/soundwire/Kconfig | 1 + drivers/soundwire/Makefile | 4 +- drivers/soundwire/amd_manager.c | 2 +- drivers/soundwire/bus.c | 77 +++ drivers/soundwire/bus.h | 18 + drivers/soundwire/cadence_master.c | 646 +++++++++++++++++- drivers/soundwire/cadence_master.h | 30 + drivers/soundwire/crc8.c | 277 ++++++++ drivers/soundwire/crc8.h | 11 + drivers/soundwire/debugfs.c | 122 +++- .../soundwire/generic_bandwidth_allocation.c | 84 ++- drivers/soundwire/intel.h | 12 + drivers/soundwire/intel_ace2x.c | 377 ++++++++++ drivers/soundwire/intel_auxdevice.c | 55 ++ drivers/soundwire/qcom.c | 2 +- drivers/soundwire/stream.c | 137 +++- include/linux/soundwire/sdw.h | 91 ++- include/linux/soundwire/sdw_intel.h | 16 + include/sound/hda-sdw-bpt.h | 76 +++ sound/soc/codecs/rt711-sdca-sdw.c | 8 + sound/soc/qcom/sdw.c | 2 +- sound/soc/sof/intel/Kconfig | 8 +- sound/soc/sof/intel/Makefile | 4 + sound/soc/sof/intel/hda-sdw-bpt.c | 328 +++++++++ 28 files changed, 2810 insertions(+), 64 deletions(-) create mode 100644 Documentation/driver-api/soundwire/bra.rst create mode 100644 drivers/soundwire/crc8.c create mode 100644 drivers/soundwire/crc8.h create mode 100644 include/sound/hda-sdw-bpt.h create mode 100644 sound/soc/sof/intel/hda-sdw-bpt.c -- 2.39.2

4 60

[PATCH] ASoC: dt-bindings: samsung,odroid: drop stale clocks
by Krzysztof Kozlowski 20 Aug '24

20 Aug '24

Clocks property was present only to allow usage of assigned-clocks in the sound card node, however in upstream DTS the assigned-clocks were moved in commit 4afb06afd768 ("ARM: dts: exynos: move assigned-clock* properties to i2s0 node in Odroid XU4") to respective I2S nodes. Linux drivers never parsed "clocks" so it can be safely dropped. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/sound/samsung,odroid.yaml | 5 ----- 1 file changed, 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index b77284e3e26a..c3dea852cc8d 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -27,11 +27,6 @@ properties: - const: samsung,odroid-xu4-audio deprecated: true - assigned-clock-parents: true - assigned-clock-rates: true - assigned-clocks: true - clocks: true - cpu: type: object additionalProperties: false -- 2.43.0

3 2

[PATCH alsa-lib] pcm: implement snd_pcm_hw_params_get_sync() and obsolete snd_pcm_info_get_sync()
by Jaroslav Kysela 20 Aug '24

20 Aug '24

Use the new clock source mechanism to get information about similar PCM clock sources for PCM streams. Link: https://lore.kernel.org/linux-sound/20240625172836.589380-1-perex@perex.cz/ Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/pcm.h | 4 ++++ include/sound/uapi/asound.h | 3 ++- src/pcm/pcm.c | 30 ++++++++++++++++++++++++++---- 3 files changed, 32 insertions(+), 5 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 102ff812..b5c67889 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -508,6 +508,9 @@ typedef union _snd_pcm_sync_id { unsigned int id32[4]; } snd_pcm_sync_id_t; +/** synchronization ID size (see snd_pcm_hw_params_get_sync) */ +#define SND_PCM_HW_PARAMS_SYNC_SIZE 16 + /** Infinite wait for snd_pcm_wait() */ #define SND_PCM_WAIT_INFINITE (-1) /** Wait for next i/o in snd_pcm_wait() */ @@ -747,6 +750,7 @@ int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_den); int snd_pcm_hw_params_get_sbits(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_fifo_size(const snd_pcm_hw_params_t *params); +const unsigned char * snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t *params); #if 0 typedef struct _snd_pcm_hw_strategy snd_pcm_hw_strategy_t; diff --git a/include/sound/uapi/asound.h b/include/sound/uapi/asound.h index f3b2b94d..012fd890 100644 --- a/include/sound/uapi/asound.h +++ b/include/sound/uapi/asound.h @@ -426,7 +426,8 @@ struct snd_pcm_hw_params { unsigned int rate_num; /* R: rate numerator */ unsigned int rate_den; /* R: rate denominator */ snd_pcm_uframes_t fifo_size; /* R: chip FIFO size in frames */ - unsigned char reserved[64]; /* reserved for future */ + unsigned char sync[16]; /* R: synchronization ID (perfect sync - one clock source) */ + unsigned char reserved[48]; /* reserved for future */ }; enum { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 62e76e7e..f8581191 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -494,8 +494,8 @@ software parameter. There are two functions allowing link multiple streams together. In the case, the linking means that all operations are synchronized. Because the drivers cannot guarantee the synchronization (sample resolution) on hardware -lacking this feature, the #snd_pcm_info_get_sync() function -returns synchronization ID - #snd_pcm_sync_id_t, which is equal +lacking this feature, the #snd_pcm_hw_params_get_sync() function +returns 16-byte synchronization ID, which is equal for hardware synchronized streams. When the #snd_pcm_link() function is called, all operations managing the stream state for these two streams are joined. The opposite function is #snd_pcm_unlink(). @@ -3948,6 +3948,25 @@ int snd_pcm_hw_params_get_fifo_size(const snd_pcm_hw_params_t *params) return params->fifo_size; } +/** + * \brief Get hardware synchronization ID from a PCM info container + * \param params Configuration space + * \return 16-byte synchronization ID (use #SND_PCM_HW_PARAMS_SYNC_SIZE) + * + * This synchronization ID determines the similar clocks for the + * PCM stream between multiple devices (including different cards). + * "All zeros" means "not set". The contents of the ID can be used + * only for a comparison with the contents of another ID returned + * from this function. Applications should not do a comparison with + * hard-coded values, because the implementation generating such + * synchronization IDs may be changed in future. + */ +const unsigned char *snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t *params) +{ + assert(params); + return params->sync; +} + /** * \brief Fill params with a full configuration space for a PCM * \param pcm PCM handle @@ -7332,7 +7351,7 @@ unsigned int snd_pcm_info_get_subdevices_avail(const snd_pcm_info_t *obj) } /** - * \brief Get hardware synchronization ID from a PCM info container + * \brief (DEPRECATED) Get hardware synchronization ID from a PCM info container * \param obj PCM info container * \return hardware synchronization ID */ @@ -7340,9 +7359,12 @@ snd_pcm_sync_id_t snd_pcm_info_get_sync(const snd_pcm_info_t *obj) { snd_pcm_sync_id_t res; assert(obj); - memcpy(&res, &obj->sync, sizeof(res)); + bzero(&res, sizeof(res)); return res; } +#ifndef DOC_HIDDEN +link_warning(snd_pcm_info_get_sync, "Warning: snd_pcm_info_get_sync is deprecated, consider to use snd_pcm_hw_params_get_sync"); +#endif /** * \brief Set wanted device inside a PCM info container (see #snd_ctl_pcm_info) -- 2.45.2

1 0

[PATCH v2 00/14] ASoC: Intel: Remove skylake driver
by Cezary Rojewski 20 Aug '24

20 Aug '24

A spiritual successor to haswell/baytrail removal series [1]. The avs-driver found in sound/soc/intel/avs is a direct replacement to the existing skylake-driver. It covers all features supported by it and more and aligns with the recommended flows and requirements based on Windows driver equivalent. The skylake-driver related UAPI has been removed with "ASoC: Drop soc-topology ABI v4 support" [2]. For the official kernel tree the deprecation begun with v6.0. Most skylake-drivers users moved to avs- or SOF-driver when AudioDSP capabilities are available on the platform or to snd-hda-intel (sound/pci/hda) when such capabilities are not. For the supported trees the deprecation begun with v5.4 with v5.15 being the first where the skylake-driver is disabled entirely. All machine board drivers that consume this DSP driver have their replacements present within sound/soc/intel/avs/boards/ directory. Changes in v2: - added two new patches (leading the series) - the first moves all device entries previously under the skylake-driver dependency under the avs-driver - the second patch drops struct skl_machine_pdata usage to fix compilation errors as pointed out by IKP - no further changes, retained Acked-by tags for these [1]: https://lore.kernel.org/alsa-devel/20201006064907.16277-1-cezary.rojewski@i… [2]: https://lore.kernel.org/alsa-devel/20240403091629.647267-1-cezary.rojewski@… Cezary Rojewski (14): ALSA: hda: Move SST device entries to AVS ASoC: Intel: Drop skl_machine_pdata usage ASoC: Intel: Remove bxt_rt298 board driver ASoC: Intel: Remove bxt_da7219_max98357a board driver ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver ASoC: Intel: Remove kbl_rt5663_max98927 board driver ASoC: Intel: Remove kbl_rt5660 board driver ASoC: Intel: Remove kbl_da7219_max98927 board driver ASoC: Intel: Remove kbl_da7219_max98357a board driver ASoC: Intel: Remove skl_rt286 board driver ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver ASoC: Intel: Remove skl_nau88l25_max98357a board driver ASoC: Intel: Remove skylake driver ASoC: Intel: avs: Enable by default for all SST configurations sound/hda/intel-dsp-config.c | 111 +- sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/avs/core.c | 8 +- sound/soc/intel/boards/Kconfig | 152 +- sound/soc/intel/boards/Makefile | 20 - sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ---- sound/soc/intel/boards/bxt_rt298.c | 670 --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ---- sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------ sound/soc/intel/boards/kbl_rt5660.c | 567 --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ---- .../soc/intel/boards/skl_nau88l25_max98357a.c | 704 ---- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ---- sound/soc/intel/boards/skl_rt286.c | 568 --- .../intel/common/soc-acpi-intel-cnl-match.c | 6 - .../intel/common/soc-acpi-intel-ehl-match.c | 1 - .../intel/common/soc-acpi-intel-hda-match.c | 6 - .../intel/common/soc-acpi-intel-icl-match.c | 6 - .../intel/common/soc-acpi-intel-kbl-match.c | 11 - .../intel/common/soc-acpi-intel-skl-match.c | 5 - sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 --- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 --- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 ------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 ------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 -- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 -- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ----- sound/soc/intel/skylake/skl-sst-ipc.h | 169 - sound/soc/intel/skylake/skl-sst-utils.c | 425 -- sound/soc/intel/skylake/skl-sst.c | 599 --- sound/soc/intel/skylake/skl-topology.c | 3605 ----------------- sound/soc/intel/skylake/skl-topology.h | 524 --- sound/soc/intel/skylake/skl.c | 1177 ------ sound/soc/intel/skylake/skl.h | 207 - 46 files changed, 59 insertions(+), 22871 deletions(-) delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/bxt_rt298.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_rt286.c delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h -- 2.25.1

4 19

[PATCH 00/12] ASoC: Intel: Remove skylake driver
by Cezary Rojewski 20 Aug '24

20 Aug '24

A spiritual successor to haswell/baytrail removal series [1]. The avs-driver found in sound/soc/intel/avs is a direct replacement to the existing skylake-driver. It covers all features supported by it and more and aligns with the recommended flows and requirements based on Windows driver equivalent. The skylake-driver related UAPI has been removed with "ASoC: Drop soc-topology ABI v4 support" [2]. For the official kernel tree the deprecation begun with v6.0. Most skylake-drivers users moved to avs- or SOF-driver when AudioDSP capabilities are available on the platform or to snd-hda-intel (sound/pci/hda) when such capabilities are not. For the supported trees the deprecation begun with v5.4 with v5.15 being the first where the skylake-driver is disabled entirely. All machine board drivers that consume this DSP driver have their replacements present within sound/soc/intel/avs/boards/ directory. [1]: https://lore.kernel.org/alsa-devel/20201006064907.16277-1-cezary.rojewski@i… [2]: https://lore.kernel.org/alsa-devel/20240403091629.647267-1-cezary.rojewski@… Cezary Rojewski (12): ASoC: Intel: Remove bxt_rt298 board driver ASoC: Intel: Remove bxt_da7219_max98357a board driver ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver ASoC: Intel: Remove kbl_rt5663_max98927 board driver ASoC: Intel: Remove kbl_rt5660 board driver ASoC: Intel: Remove kbl_da7219_max98927 board driver ASoC: Intel: Remove kbl_da7219_max98357a board driver ASoC: Intel: Remove skl_rt286 board driver ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver ASoC: Intel: Remove skl_nau88l25_max98357a board driver ASoC: Intel: Remove skylake driver ASoC: Intel: avs: Enable by default for all SST configurations sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/avs/core.c | 8 +- sound/soc/intel/boards/Kconfig | 152 +- sound/soc/intel/boards/Makefile | 20 - sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ---- sound/soc/intel/boards/bxt_rt298.c | 670 --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ---- sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------ sound/soc/intel/boards/kbl_rt5660.c | 567 --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ---- .../soc/intel/boards/skl_nau88l25_max98357a.c | 704 ---- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ---- sound/soc/intel/boards/skl_rt286.c | 568 --- sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 --- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 --- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 ------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 ------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 -- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 -- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ----- sound/soc/intel/skylake/skl-sst-ipc.h | 169 - sound/soc/intel/skylake/skl-sst-utils.c | 425 -- sound/soc/intel/skylake/skl-sst.c | 599 --- sound/soc/intel/skylake/skl-topology.c | 3605 ----------------- sound/soc/intel/skylake/skl-topology.h | 524 --- sound/soc/intel/skylake/skl.c | 1177 ------ sound/soc/intel/skylake/skl.h | 207 - 39 files changed, 9 insertions(+), 22775 deletions(-) delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/bxt_rt298.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_rt286.c delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h -- 2.25.1

3 14

[REGRESSION] ALSA: firewire-lib: heavy digital distortion with Fireface 800
by edmund.raile 20 Aug '24

20 Aug '24

Between the kernel releases 6.9.9 and 6.10.0, something changed that causes heavy digital distortion on playback from at least ALSA and PipeWire. The rhythm of the music is discernable in the output, so it isn't just random noise. It sounds a bit like bitcrushing, but not quite. The generated overtones appear to be dependet on the sample rate. I am sorry I can not be more specific, there are no kernel messages this time. As Mr. Sakamoto recently committed all these changes to firewire he might be able to identify the issue easily, which is why I'd like to ask him to look into it. Kind regards, Edmund Raile. #regzbot introduced: v6.9.9..v6.10

2 1

[PATCH v1] ASoc: tas2781: fixed the issue that the chip do not shutdown immediatly after aplay stopped
by Shenghao Ding 20 Aug '24

20 Aug '24

Issue reported from customer that the chip do not shutdown after aplay stopped until 6 mins later. Drop tasdevice_dapm_event and implement .stream_mute in the tasdevice_dai_ops. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Drop tasdevice_dapm_event and implement .stream_mute in the tasdevice_dai_ops - Remove unnecessary line feed for tasdevice_codec_remove --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..145c45349399 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -557,30 +557,22 @@ static void tasdevice_fw_ready(const struct firmware *fmw, release_firmware(fmw); } -static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, int stream) { - struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int state = 0; - /* Codec Lock Hold */ - mutex_lock(&tas_priv->codec_lock); - if (event == SND_SOC_DAPM_PRE_PMD) - state = 1; - tasdevice_tuning_switch(tas_priv, state); - /* Codec Lock Release*/ - mutex_unlock(&tas_priv->codec_lock); + /* Codec Lock/UnLock */ + guard(mutex)(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, mute); return 0; } static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, - 0, 0, tasdevice_dapm_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SPK("SPK", NULL), SND_SOC_DAPM_OUTPUT("OUT"), SND_SOC_DAPM_INPUT("DMIC") }; @@ -667,6 +659,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { .startup = tasdevice_startup, .hw_params = tasdevice_hw_params, .set_sysclk = tasdevice_set_dai_sysclk, + .mute_stream = tasdevice_mute, }; static struct snd_soc_dai_driver tasdevice_dai_driver[] = { -- 2.34.1

3 3

[PATCH v7 2/2] ARM: dts: imx6: update spdif sound card node properties
by Elinor Montmasson 20 Aug '24

20 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi | 9 +++++++-- .../boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 9 +++++++-- 12 files changed, 104 insertions(+), 28 deletions(-) diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts index 95b49fc83f7b..299106fbe51c 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts @@ -127,12 +127,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts index a7d5693c5ab7..8d2b608e0b90 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts @@ -111,12 +111,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts index 7c298d9aa21e..5353a0c24420 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts @@ -90,11 +90,16 @@ sound-sgtl5000 { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi index ea40623d12e5..edf55760a5c1 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi @@ -197,11 +197,20 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi index 3a46ade3b6bd..9e97ef5e43f2 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi @@ -121,11 +121,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi index d3a7a6eeb8e0..b01670cdd52c 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi @@ -142,12 +142,21 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + /* Optional S/PDIF in on SODIMM 88 and out on SODIMM 90, 137 or 168 */ sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi index 761566ae3cf5..bd66430c1d78 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi @@ -100,12 +100,17 @@ v_usb1: regulator-v-usb1 { vin-supply = <&v_5v0>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "Integrated SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; gpio-keys { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi index a955c77cd499..d1ad65ab6b72 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi @@ -140,12 +140,17 @@ sound_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi index 0a3deaf92eea..55fb7b904220 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi @@ -143,12 +143,17 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-sabreauto-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; backlight { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi index 38abb6b50f6c..7130b9c3b3aa 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi @@ -26,11 +26,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; reg_1p5v: regulator-1p5v { diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts index b0c27b9b0244..dfbfb8119bf3 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts @@ -97,11 +97,16 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi index 7d4170c27732..277a6e039045 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi @@ -183,12 +183,17 @@ panel_in: endpoint { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx6sx-sdb-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; -- 2.34.1

1 0

[PATCH v7 1/2] arm64: dts: imx8m: update spdif sound card node properties
by Elinor Montmasson 20 Aug '24

20 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mq-evk.dts | 24 +++++++++++++++---- 3 files changed, 43 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi index 930e14fec423..5f8336217bb8 100644 --- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi @@ -180,12 +180,21 @@ cpu { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi index 9e0259ddf4bc..33d73f3dc187 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi @@ -124,12 +124,21 @@ sound-wm8524 { "Line Out Jack", "LINEVOUTR"; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; sound-micfil { diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts index 7507548cdb16..a87d0692c3bb 100644 --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; + }; + + hdmi_arc_in: hdmi-arc-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; }; sound-hdmi-arc { compatible = "fsl,imx-audio-spdif"; model = "imx-hdmi-arc"; - spdif-controller = <&spdif2>; - spdif-in; + audio-cpu = <&spdif2>; + audio-codec = <&hdmi_arc_in>; }; }; -- 2.34.1

1 0

[PATCH] ASoc: tas2781: replace devm_kzalloc and scnprintf with devm_kasprintf
by Shenghao Ding 20 Aug '24

20 Aug '24

Replace devm_kzalloc and scnprintf with devm_kasprintf. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..1451ba0fa5b4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -346,13 +346,11 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) } /* Create a mixer item for selecting the active profile */ - name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); + name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, "Speaker Profile Id"); if (!name) { ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -441,18 +439,13 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) goto out; } - /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, - SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); - if (!prog_name || !conf_name) { + /* Create mixer items for selecting the active Program and Config */ + prog_name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, + "Speaker Program Id"); + if (!prog_name) { ret = -ENOMEM; goto out; } - - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -460,8 +453,12 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + conf_name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, "Speaker Config Id"); + if (!conf_name) { + ret = -ENOMEM; + goto out; + } dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; -- 2.34.1

3 2

[PATCH] soundwire: qcom: Use devm_clk_get_enabled() helpers
by Huan Yang 20 Aug '24

20 Aug '24

The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: Huan Yang <link(a)vivo.com> --- drivers/soundwire/qcom.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e..2a57ed9464b4 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1527,24 +1527,22 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (IS_ERR(ctrl->audio_cgcr)) { dev_err(dev, "Failed to get cgcr reset ctrl required for SW gating\n"); ret = PTR_ERR(ctrl->audio_cgcr); - goto err_init; + return ret; } } ctrl->irq = of_irq_get(dev->of_node, 0); if (ctrl->irq < 0) { ret = ctrl->irq; - goto err_init; + return ret; } - ctrl->hclk = devm_clk_get(dev, "iface"); + ctrl->hclk = devm_clk_get_enabled(dev, "iface"); if (IS_ERR(ctrl->hclk)) { ret = dev_err_probe(dev, PTR_ERR(ctrl->hclk), "unable to get iface clock\n"); - goto err_init; + return ret; } - clk_prepare_enable(ctrl->hclk); - ctrl->dev = dev; dev_set_drvdata(&pdev->dev, ctrl); mutex_init(&ctrl->port_lock); @@ -1558,7 +1556,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) ret = qcom_swrm_get_port_config(ctrl); if (ret) - goto err_clk; + return ret; params = &ctrl->bus.params; params->max_dr_freq = DEFAULT_CLK_FREQ; @@ -1586,7 +1584,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) "soundwire", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire irq\n"); - goto err_clk; + return ret; } ctrl->wake_irq = of_irq_get(dev->of_node, 1); @@ -1597,7 +1595,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) "swr_wake_irq", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire wake irq\n"); - goto err_init; + return ret; } } @@ -1612,7 +1610,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (ret) { dev_err(dev, "Failed to register Soundwire controller (%d)\n", ret); - goto err_clk; + return ret; } qcom_swrm_init(ctrl); @@ -1642,9 +1640,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) err_master_add: sdw_bus_master_delete(&ctrl->bus); -err_clk: - clk_disable_unprepare(ctrl->hclk); -err_init: return ret; } @@ -1653,7 +1648,6 @@ static void qcom_swrm_remove(struct platform_device *pdev) struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(&pdev->dev); sdw_bus_master_delete(&ctrl->bus); - clk_disable_unprepare(ctrl->hclk); } static int __maybe_unused swrm_runtime_resume(struct device *dev) -- 2.45.2

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[PATCH v3 0/8] MSM8953/MSM8976 ASoC support
by Adam Skladowski 20 Aug '24

20 Aug '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. Changes since v2 =============== 1. Guarded ioremap on quinary mux with checking if resource is not NULL 2. Introduced compatible based reg overriding in wcd-analog driver 3. Separated PM8950/PM8953 codec patches 4. Moved reg/reg-names back into top-level and rewrote as in comments 5. Separated MSM8953/MSM8976 QDSP6 patches 6. Added Lineout patch [1] - https://lore.kernel.org/lkml/20240723083300.35605-1-krzysztof.kozlowski@lin… Changes since v1 ================ 1. Rebased dt-bindings documentation based on Krzysztof's split patch[1] 2. Resolved clang errors by guarding ret verification inside if 3. Switched quin-iomux to devm_ioremap_resource to not fail on msm8916 Signed-off-by: Adam Skladowski <a39.skl(a)gmail.com> --- Adam Skladowski (4): ASoC: dt-bindings: pm8916-wcd-analog-codec: Document pm8950/pm8953 ASoC: msm8916-wcd-analog: add pm8950 codec ASoC: dt-bindings: apq8016-sbc: Add msm8953/msm8976-qdsp6-sndcard ASoC: qcom: apq8016_sbc: Add support for msm8976 SoC Vladimir Lypak (4): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add pm8953 codec ASoC: qcom: apq8016_sbc: Add support for msm8953 SoC ASoC: msm8916-wcd-analog: add lineout output .../bindings/sound/qcom,apq8016-sbc-sndcard.yaml | 27 ++++ .../sound/qcom,pm8916-wcd-analog-codec.yaml | 6 +- sound/soc/codecs/msm8916-wcd-analog.c | 145 ++++++++++++++++++++- sound/soc/qcom/apq8016_sbc.c | 88 ++++++++++++- 4 files changed, 257 insertions(+), 9 deletions(-) --- base-commit: 31c2a5c4bf3673b0bf098f358ea931b9ada1e7de change-id: 20240729-msm8953-msm8976-asoc-65f973a13165 Best regards, -- Adam Skladowski <a39.skl(a)gmail.com>

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[DISCUSSION] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by xeroxslayer＠proton.me 20 Aug '24

20 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

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