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

Re: [PATCH] ALSA: Add logitech Audio profile quirk
by Joshua Pius 06 Sep '24

06 Sep '24

1 3

Suspend/resume Issue on pcm_dmix.c in alsa-lib
by Chancel Liu 06 Sep '24

06 Sep '24

Hi Takashi Iwai, Jaroslav Kysela We found an issue on dmix in alsa-lib when do suspend and resume. It can be easily reproduced by following steps: 1. Run two dmix clients in parallel. (Only one client doesn't has such issue) ~# aplay xxx1.wav & ~# aplay xxx2.wav & Here I attach the asound.conf we're using. ~# cat /etc/asound.conf defaults.pcm.rate_converter "linear" pcm.dmix_44100{ type dmix ipc_key 5678293 ipc_key_add_uid yes slave{ pcm "hw:0,0" period_time 40000 format S16_LE rate 44100 } } pcm.asymed{ type asym playback.pcm "dmix_44100" capture.pcm "dsnoop_44100" } pcm.!default{ type plug route_policy "average" slave.pcm "asymed" } 2. Let linux enter into suspend and then resume(Repeat this step if not reproduced) 3. After resume, aplay will get stuck in snd_pcm_wait(). The GDB shows: (gdb) bt #0 0x0000fffff7da9264 in __GI___poll (fds=fds@entry=0xfffffffff480, nfds=nfds@entry=1, timeout=timeout@entry=240) at /usr/src/debug/glibc/2.39+git/sysdeps/unix/sysv/linux/poll.c:41 #1 0x0000fffff7edf468 in poll (__timeout=240, __nfds=1, __fds=0xfffffffff480) #2 snd1_pcm_wait_nocheck (pcm=pcm@entry=0xaaaaaaad2cb0, timeout=240, timeout@entry=-10001) at pcm.c:2993 #3 0x0000fffff7ee54a0 in snd1_pcm_write_areas (pcm=pcm@entry=0xaaaaaaad2cb0, areas=areas@entry=0xfffffffff560, offset=<optimized out>, offset@entry=0, size=<optimized out>, size@entry=1768, func=func@entry=0xfffff7ef5190 <snd_pcm_plugin_write_areas>) at pcm.c:7699 #4 0x0000fffff7ef5020 in snd_pcm_plugin_writei (pcm=0xaaaaaaad2cb0, buffer=<optimized out>, size=1768) at pcm_plugin.c:354 It seems that sometimes after suspend and resume there's no available space for data written into buffer. Then aplay keeps stuck in snd_pcm_wait(). I checked the hw_ptr of dmix and found that hw_ptr is always 0 after resume. I don't have a solution now so I turn to you for help. The version of alsa-lib is v1.2.11. Could you please help check it? Regards, Chancel Liu

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[PATCH v26 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 04 Sep '24

04 Sep '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 userspace 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 platform 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 v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections 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. - Addressed 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 unnecessary 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 userspsace 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 (31): 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 ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for 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: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 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: 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: 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 .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 429 ++++ 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 | 34 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 45 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 71 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 135 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- 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 | 194 +- 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 | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 338 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 110 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5820 insertions(+), 56 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/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[PATCH] ASoC: tlv320aic32x4: Add multi endpoint support
by Marek Vasut 04 Sep '24

04 Sep '24

Support multiple endpoints on TLV320AIC32x4 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: Custom STM32MP157C board uses TLV320AIC32x4 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (TLV320AIC32x4) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: tlv320aic32x4_port: port { tlv320aic32x4_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; tlv320aic32x4_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Baojun Xu <baojun.xu(a)ti.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: Christoph Niedermaier <cniedermaier(a)dh-electronics.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kevin Lu <kevin-lu(a)ti.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Shenghao Ding <shenghao-ding(a)ti.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: kernel(a)dh-electronics.com Cc: linux-sound(a)vger.kernel.org --- sound/soc/codecs/tlv320aic32x4.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 5c0c81da06dba..54ea4bc58c276 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1073,6 +1073,13 @@ static int aic32x4_component_probe(struct snd_soc_component *component) return 0; } +static int aic32x4_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .probe = aic32x4_component_probe, .set_bias_level = aic32x4_set_bias_level, @@ -1082,6 +1089,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), .dapm_routes = aic32x4_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, @@ -1203,6 +1211,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = .num_dapm_widgets = ARRAY_SIZE(aic32x4_tas2505_dapm_widgets), .dapm_routes = aic32x4_tas2505_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_tas2505_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.45.2

2 1

[PATCH v2 0/3] ASoC: dt-bindings: amlogic-sound-cards: document clocks property
by Neil Armstrong 02 Sep '24

02 Sep '24

Following an off-list discution with Jerome about fixing the following DTBs check errors: sound: Unevaluated properties are not allowed ('assigned-clock-parents', 'assigned-clock-rates', 'assigned-clocks' were unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml# sound: Unevaluated properties are not allowed ('assigned-clock-parents', 'assigned-clock-rates', 'assigned-clocks' were unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,gx-sound-card.yaml# sound: 'anyOf' conditional failed, one must be fixed: 'clocks' is a required property '#clock-cells' is a required property from schema $id: http://devicetree.org/schemas/clock/clock.yaml# It has been agreed documenting the clock in the sound card is a better solution than moving them to a random clock controller or consumer node which is not related to the actual meaning of those root frequencies. The patchset adds the clocks proprty to the bindings and finally adds the properties to the DT files. Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Changes in v2: - drop clock-names in bindings & commit messages - fix new boards - drop RFS - Link to v1: https://lore.kernel.org/r/20240614-topic-amlogic-upstream-bindings-fixes-au… --- Neil Armstrong (3): ASoC: dt-bindings: amlogic,axg-sound-card: document clocks property ASoC: dt-bindings: amlogic,gx-sound-card: document clocks property arm64: dts: amlogic: add clock and clock-names to sound cards .../devicetree/bindings/sound/amlogic,axg-sound-card.yaml | 7 +++++++ Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 7 +++++++ arch/arm64/boot/dts/amlogic/meson-axg-s400.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-fbx8am.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-radxa-zero.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-sei510.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-u200.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-x96-max.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-bananapi-cm4-cm4io.dts | 4 ++++ .../arm64/boot/dts/amlogic/meson-g12b-bananapi-cm4-mnt-reform2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-bananapi.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-dreambox.dtsi | 6 ++++++ arch/arm64/boot/dts/amlogic/meson-g12b-gsking-x.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-gtking-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-gtking.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-go-ultra.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-n2.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-n2l.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-radxa-zero2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-ugoos-am6.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gx-libretech-pc.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gx-p23x-q20x.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-kii-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-nanopi-k2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-nexbox-a95x.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-odroidc2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-p200.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-p201.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-vega-s95.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-wetek-hub.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-wetek-play2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s805x-libretech-ac.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s805x-p241.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-khadas-vim.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-libretech-cc-v2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-libretech-cc.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-p212.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-vero4k.dts | 5 +++++ arch/arm64/boot/dts/amlogic/meson-gxm-khadas-vim2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxm-nexbox-a1.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxm-rbox-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-khadas-vim3.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-libretech-cottonwood.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-a95xf3-air-gbit.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-a95xf3-air.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-bananapi-m2-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-bananapi-m5.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-h96-max.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-odroid.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-sei610.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-x96-air-gbit.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-x96-air.dts | 4 ++++ 52 files changed, 217 insertions(+) --- base-commit: cb2361c78068ec47c3a6222d71f664b67665550e change-id: 20240614-topic-amlogic-upstream-bindings-fixes-audio-snd-card-2c642ccd2c6e Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

3 6

[tiwai-sound:for-next 11/11] sound/core/pcm_memory.c:196:37: error: incompatible pointer types passing 'size_t *' (aka 'unsigned int *') to parameter of type 'unsigned long *'
by kernel test robot 31 Aug '24

31 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 61bc4deff033181992408f973b48fca08757d3ff commit: 61bc4deff033181992408f973b48fca08757d3ff [11/11] ALSA: pcm: replace simple_strtoul to kstrtoul config: hexagon-randconfig-002-20240901 (https://download.01.org/0day-ci/archive/20240901/202409010509.nOL6xTD7-lkp@…) compiler: clang version 20.0.0git (https://github.com/llvm/llvm-project 46fe36a4295f05d5d3731762e31fc4e6e99863e9) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240901/202409010509.nOL6xTD7-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409010509.nOL6xTD7-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:548:31: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 548 | val = __raw_readb(PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:561:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 561 | val = __le16_to_cpu((__le16 __force)__raw_readw(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:37:51: note: expanded from macro '__le16_to_cpu' 37 | #define __le16_to_cpu(x) ((__force __u16)(__le16)(x)) | ^ In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:574:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 574 | val = __le32_to_cpu((__le32 __force)__raw_readl(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:35:51: note: expanded from macro '__le32_to_cpu' 35 | #define __le32_to_cpu(x) ((__force __u32)(__le32)(x)) | ^ In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:585:33: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 585 | __raw_writeb(value, PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:595:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 595 | __raw_writew((u16 __force)cpu_to_le16(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:605:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 605 | __raw_writel((u32 __force)cpu_to_le32(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ In file included from sound/core/pcm_memory.c:14: In file included from include/sound/pcm.h:15: In file included from include/linux/mm.h:2228: include/linux/vmstat.h:517:36: warning: arithmetic between different enumeration types ('enum node_stat_item' and 'enum lru_list') [-Wenum-enum-conversion] 517 | return node_stat_name(NR_LRU_BASE + lru) + 3; // skip "nr_" | ~~~~~~~~~~~ ^ ~~~ >> sound/core/pcm_memory.c:196:37: error: incompatible pointer types passing 'size_t *' (aka 'unsigned int *') to parameter of type 'unsigned long *' [-Werror,-Wincompatible-pointer-types] 196 | buffer->error = kstrtoul(str, 10, &size); | ^~~~~ include/linux/kstrtox.h:30:90: note: passing argument to parameter 'res' here 30 | static inline int __must_check kstrtoul(const char *s, unsigned int base, unsigned long *res) | ^ 7 warnings and 1 error generated. vim +196 sound/core/pcm_memory.c 174 175 /* 176 * write callback for prealloc proc file 177 * 178 * accepts the preallocation size in kB. 179 */ 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 181 struct snd_info_buffer *buffer) 182 { 183 struct snd_pcm_substream *substream = entry->private_data; 184 struct snd_card *card = substream->pcm->card; 185 char line[64], str[64]; 186 size_t size; 187 struct snd_dma_buffer new_dmab; 188 189 guard(mutex)(&substream->pcm->open_mutex); 190 if (substream->runtime) { 191 buffer->error = -EBUSY; 192 return; 193 } 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { 195 snd_info_get_str(str, line, sizeof(str)); > 196 buffer->error = kstrtoul(str, 10, &size); 197 if (buffer->error != 0) 198 return; 199 size *= 1024; 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { 201 buffer->error = -EINVAL; 202 return; 203 } 204 if (substream->dma_buffer.bytes == size) 205 return; 206 memset(&new_dmab, 0, sizeof(new_dmab)); 207 new_dmab.dev = substream->dma_buffer.dev; 208 if (size > 0) { 209 if (do_alloc_pages(card, 210 substream->dma_buffer.dev.type, 211 substream->dma_buffer.dev.dev, 212 substream->stream, 213 size, &new_dmab) < 0) { 214 buffer->error = -ENOMEM; 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", 216 substream->pcm->card->number, substream->pcm->device, 217 substream->stream ? 'c' : 'p', substream->number, 218 substream->pcm->name, size); 219 return; 220 } 221 substream->buffer_bytes_max = size; 222 } else { 223 substream->buffer_bytes_max = UINT_MAX; 224 } 225 if (substream->dma_buffer.area) 226 do_free_pages(card, &substream->dma_buffer); 227 substream->dma_buffer = new_dmab; 228 } else { 229 buffer->error = -EINVAL; 230 } 231 } 232 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 11/11] sound/core/pcm_memory.c:196:51: error: passing argument 3 of 'kstrtoul' from incompatible pointer type
by kernel test robot 31 Aug '24

31 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 61bc4deff033181992408f973b48fca08757d3ff commit: 61bc4deff033181992408f973b48fca08757d3ff [11/11] ALSA: pcm: replace simple_strtoul to kstrtoul config: sh-allmodconfig (https://download.01.org/0day-ci/archive/20240901/202409010425.YPS7cWeJ-lkp@…) compiler: sh4-linux-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240901/202409010425.YPS7cWeJ-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409010425.YPS7cWeJ-lkp@intel.com/ All errors (new ones prefixed by >>): sound/core/pcm_memory.c: In function 'snd_pcm_lib_preallocate_proc_write': >> sound/core/pcm_memory.c:196:51: error: passing argument 3 of 'kstrtoul' from incompatible pointer type [-Wincompatible-pointer-types] 196 | buffer->error = kstrtoul(str, 10, &size); | ^~~~~ | | | size_t * {aka unsigned int *} In file included from include/linux/kernel.h:25, from arch/sh/include/asm/fixmap.h:13, from arch/sh/include/asm/pgtable.h:22, from include/linux/pgtable.h:6, from arch/sh/include/asm/io.h:21, from include/linux/io.h:14, from sound/core/pcm_memory.c:7: include/linux/kstrtox.h:30:90: note: expected 'long unsigned int *' but argument is of type 'size_t *' {aka 'unsigned int *'} 30 | static inline int __must_check kstrtoul(const char *s, unsigned int base, unsigned long *res) | ~~~~~~~~~~~~~~~^~~ vim +/kstrtoul +196 sound/core/pcm_memory.c 174 175 /* 176 * write callback for prealloc proc file 177 * 178 * accepts the preallocation size in kB. 179 */ 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 181 struct snd_info_buffer *buffer) 182 { 183 struct snd_pcm_substream *substream = entry->private_data; 184 struct snd_card *card = substream->pcm->card; 185 char line[64], str[64]; 186 size_t size; 187 struct snd_dma_buffer new_dmab; 188 189 guard(mutex)(&substream->pcm->open_mutex); 190 if (substream->runtime) { 191 buffer->error = -EBUSY; 192 return; 193 } 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { 195 snd_info_get_str(str, line, sizeof(str)); > 196 buffer->error = kstrtoul(str, 10, &size); 197 if (buffer->error != 0) 198 return; 199 size *= 1024; 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { 201 buffer->error = -EINVAL; 202 return; 203 } 204 if (substream->dma_buffer.bytes == size) 205 return; 206 memset(&new_dmab, 0, sizeof(new_dmab)); 207 new_dmab.dev = substream->dma_buffer.dev; 208 if (size > 0) { 209 if (do_alloc_pages(card, 210 substream->dma_buffer.dev.type, 211 substream->dma_buffer.dev.dev, 212 substream->stream, 213 size, &new_dmab) < 0) { 214 buffer->error = -ENOMEM; 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", 216 substream->pcm->card->number, substream->pcm->device, 217 substream->stream ? 'c' : 'p', substream->number, 218 substream->pcm->name, size); 219 return; 220 } 221 substream->buffer_bytes_max = size; 222 } else { 223 substream->buffer_bytes_max = UINT_MAX; 224 } 225 if (substream->dma_buffer.area) 226 do_free_pages(card, &substream->dma_buffer); 227 substream->dma_buffer = new_dmab; 228 } else { 229 buffer->error = -EINVAL; 230 } 231 } 232 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Fix TLV dB parser in case of used container
by GitHub pull_request - edited 31 Aug '24

31 Aug '24

alsa-project/alsa-lib pull request #409 was edited from arkq: In case when dB information does not appear as the only TLV type in the stream (it might be wrapped in a container, but the container can not have any other type), the TLV parser fails to get the dB TLV pointer. This commit fixes it by distinguishing between TLV parse error and dB information not being found in a container, so the parser can iterate over all elements in the container. Also, it fixes out-of-bounds read in case of malicious TLV record. EDIT: Other possibility to fix is to return `-ENOENT` in case where dB information was not found and handle that accordingly when iterating over items of a container. Request URL : https://github.com/alsa-project/alsa-lib/pull/409 Patch URL : https://github.com/alsa-project/alsa-lib/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fix TLV dB parser in case of used container
by GitHub pull_request - opened 31 Aug '24

31 Aug '24

alsa-project/alsa-lib pull request #409 was opened from arkq: In case when dB information does not appear as the only TLV type in the stream (it might be wrapped in a container, but the container can not have any other type), the TLV parser fails to get the dB TLV pointer. This commit fixes it by distinguishing between TLV parse error and dB information not being found in a container, so the parser can iterate over all elements in the container. Also, it fixes out-of-bounds read in case of malicious TLV record. Request URL : https://github.com/alsa-project/alsa-lib/pull/409 Patch URL : https://github.com/alsa-project/alsa-lib/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ASoC: codecs: Switch to use dev_err_probe()
by Yang Ruibin 30 Aug '24

30 Aug '24

Using dev_err_probe() instead of dev_err() in probe() simplifies the error path and standardizes the format of the error code. Signed-off-by: Yang Ruibin <11162571(a)vivo.com> --- sound/soc/codecs/ad1980.c | 8 +++----- sound/soc/codecs/adau1701.c | 19 +++++++------------ sound/soc/codecs/ssm2602.c | 6 ++---- 3 files changed, 12 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3c1ae13c1..d2f679d9d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -237,11 +237,9 @@ static int ad1980_soc_probe(struct snd_soc_component *component) u16 ext_status; ac97 = snd_soc_new_ac97_component(component, 0, 0); - if (IS_ERR(ac97)) { - ret = PTR_ERR(ac97); - dev_err(component->dev, "Failed to register AC97 component: %d\n", ret); - return ret; - } + if (IS_ERR(ac97)) + return dev_err_probe(component->dev, PTR_ERR(ac97), + "Failed to register AC97 component\n"); regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); if (IS_ERR(regmap)) { diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8bd6067df..f4c574447 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -677,10 +677,9 @@ static int adau1701_probe(struct snd_soc_component *component) ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(component->dev, "Failed to enable regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(component->dev, ret, + "Failed to enable regulators\n"); /* * Let the pll_clkdiv variable default to something that won't happen @@ -798,17 +797,13 @@ static int adau1701_i2c_probe(struct i2c_client *client) ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(dev, "Failed to get regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(dev, ret, "Failed to get regulators\n"); ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(dev, "Failed to enable regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(dev, ret, "Failed to enable regulators\n"); adau1701->client = client; adau1701->regmap = devm_regmap_init(dev, NULL, client, diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c29324403..153eb55a3 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -605,10 +605,8 @@ static int ssm260x_component_probe(struct snd_soc_component *component) int ret; ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); - if (ret < 0) { - dev_err(component->dev, "Failed to issue reset: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(component->dev, ret, "Failed to issue reset\n"); regmap_register_patch(ssm2602->regmap, ssm2602_patch, ARRAY_SIZE(ssm2602_patch)); -- 2.34.1

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