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

[PATCH 1/2] ASoC: dt-bindings: Added schema for "nuvoton,nau8325"
by Seven Lee 02 Feb '24

02 Feb '24

Added a DT schema for describing nau8325 audio amplifiers. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../bindings/sound/nuvoton,nau8325.yaml | 82 +++++++++++++++++++ 1 file changed, 82 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml new file mode 100644 index 000000000000..9105985357aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8325.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8325 audio Amplifier + +maintainers: + - Seven Lee <WTLI(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8325 + + reg: + maxItems: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF impedance selection. + enum: + - 0 # Open + - 1 # 25kOhm + - 2 # 125kOhm + - 3 # 2.5kOhm + default: 2 + + nuvoton,dac-vref: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + DAC Reference Voltage Setting. + enum: + - 0 # VDDA + - 1 # VDDA*1.5/1.8V + - 2 # VDDA*1.6/1.8V + - 3 # VDDA*1.7/1.8V + default: 2 + + nuvoton,alc-enable: + description: + Enable digital automatic level control (ALC) function. + type: boolean + + nuvoton,clock-detection-disable: + description: + When clock detection is enabled, it will detect whether MCLK + and FS are within the range. MCLK range is from 2.048MHz to 24.576MHz. + FS range is from 8kHz to 96kHz. + type: boolean + + nuvoton,clock-det-data: + description: + Request clock detection to require 2048 non-zero samples before enabling + the audio paths. If set then non-zero samples is required, otherwise it + doesn't matter. + type: boolean + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@21 { + compatible = "nuvoton,nau8325"; + reg = <0x21>; + nuvoton,vref-impedance = <2>; + nuvoton,dac-vref = <2>; + nuvoton,alc-enable; + nuvoton,clock-det-data; + }; + }; -- 2.25.1

4 11

[PATCH 00/18] ALSA: Various fixes for Cirrus Logic CS35L56 support
by Richard Fitzgerald 01 Feb '24

01 Feb '24

This chain of patches fixes various things that were undocumented, unknown or uncertain when the original driver code was written. And also a few things that were just bugs. The HDA patches have dependencies on the ASoC patches, except for the final patch that removes a bogus test stub function. Richard Fitzgerald (18): ASoC: wm_adsp: Fix firmware file search order ASoC: wm_adsp: Don't overwrite fwf_name with the default ASoC: cs35l56: cs35l56_component_remove() must clear cs35l56->component ASoC: cs35l56: cs35l56_component_remove() must clean up wm_adsp ASoC: cs35l56: Don't add the same register patch multiple times ASoC: cs35l56: Remove buggy checks from cs35l56_is_fw_reload_needed() ASoC: cs35l56: Fix to ensure ASP1 registers match cache ASoC: cs35l56: Fix default SDW TX mixer registers ALSA: hda: cs35l56: Initialize all ASP1 registers ASoC: cs35l56: Fix for initializing ASP1 mixer registers ASoC: cs35l56: Fix misuse of wm_adsp 'part' string for silicon revision ASoC: cs35l56: Firmware file must match the version of preloaded firmware ASoC: cs35l56: Load tunings for the correct speaker models ASoC: cs35l56: Allow more time for firmware to boot ALSA: hda: cs35l56: Fix order of searching for firmware files ALSA: hda: cs35l56: Fix filename string field layout ALSA: hda: cs35l56: Firmware file must match the version of preloaded firmware ALSA: hda: cs35l56: Remove unused test stub function include/sound/cs35l56.h | 7 +- sound/pci/hda/cs35l56_hda.c | 138 ++++++++------ sound/soc/codecs/cs35l56-shared.c | 140 ++++++++++++-- sound/soc/codecs/cs35l56.c | 307 +++++++++++++++++++++++++----- sound/soc/codecs/cs35l56.h | 2 + sound/soc/codecs/wm_adsp.c | 73 +++---- 6 files changed, 498 insertions(+), 169 deletions(-) -- 2.39.2

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[PATCH] ALSA: usb-audio: Add quirk for PioneerDJ DDJ-1000
by rp＠nrp.de 01 Feb '24

01 Feb '24

Problems: - PioneerDJ DDJ-1000 audio channels are not recognized. - Jog dial displays are showing permanently "NO AUDIO DRIVER" The attached patch fixes this. The jog displays and all output and input channels are working now. Signed-off-by: Ralf Pachali <rp(a)nrp.de> --- sound/usb/quirks-table.h +++ sound/usb/quirks-table.h @@ -3942,6 +3942,64 @@ } }, +{ + /* + * PIONEER DJ DDJ-1000 + * PCM is 6 channels out, 12 channels in @ 44.1 fixed + * The Feedback for the output is the input + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0020), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2100/MS2106 based AV capture cards * --- sound/usb/quirks.c +++ sound/usb/quirks.c @@ -1390,6 +1390,16 @@ return 0; } +static int snd_usb_pioneer_ddj1000_boot_quirk(struct usb_device *dev) +{ + /* disable "no audio driver" message on jog wheel displays */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0x8003, NULL, 0); + + return 0; +} + /* * Setup quirks */ @@ -1630,6 +1640,8 @@ switch (id) { case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ return snd_usb_motu_m_series_boot_quirk(dev); + case USB_ID(0x2b73, 0x0020): /* Pioneer DDJ-1000 */ + return snd_usb_pioneer_ddj1000_boot_quirk(dev); } return 0;

2 1

[PATCH v12 00/41] Introduce QC USB SND audio offloading support
by Wesley Cheng 01 Feb '24

01 Feb '24

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (3): xhci: add support to allocate several interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (38): usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information dt-bindings: usb: xhci: Add num-hc-interrupters definition dt-bindings: usb: dwc3: Limit num-hc-interrupters definition usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Add Q6USB backend ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Add SND kcontrol for fetching USB offload status ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: mixer: Add USB offloading mixer control ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add ASoC: qcom: Populate SoC components string ASoC: doc: Add documentation for SOC USB .../devicetree/bindings/sound/qcom,q6usb.yaml | 55 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 4 + .../devicetree/bindings/usb/usb-xhci.yaml | 6 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 611 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-mem.c | 113 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 48 +- drivers/usb/host/xhci-sideband.c | 374 ++++ drivers/usb/host/xhci.c | 115 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 67 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 90 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/common.c | 41 + sound/soc/qcom/common.h | 4 +- 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 | 401 ++++ sound/soc/qcom/sm8250.c | 14 +- sound/soc/soc-usb.c | 538 +++++ sound/usb/Kconfig | 19 + sound/usb/Makefile | 3 +- sound/usb/card.c | 123 ++ sound/usb/card.h | 24 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer.c | 5 + sound/usb/mixer_usb_offload.c | 81 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 94 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1909 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 52 files changed, 6163 insertions(+), 82 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb.yaml 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/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

2 49

[PATCH v7 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 31 Jan '24

31 Jan '24

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. This series is mostly just a resend keeping pace with the kernel under it, except for a minor fixup in the ASoC stuff. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 98 + drivers/mfd/cs42l43-sdw.c | 239 ++ drivers/mfd/cs42l43.c | 1188 +++++++++ drivers/mfd/cs42l43.h | 28 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 32 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 284 ++ include/linux/mfd/cs42l43-regs.h | 1184 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 946 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7615 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

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[v3 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 31 Jan '24

31 Jan '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 32 +++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 113 insertions(+), 10 deletions(-) -- 2.25.1

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ALSA-LIB: xrun recovery in iec958 plugin for 8ch HDMI?
by Pavel Hofman 31 Jan '24

31 Jan '24

Hi, Users of RPi's hdmi:vc4-hdmi report random channel swapping after an xrun. Tested (among other clients) on speaker-test, details https://forums.raspberrypi.com/viewtopic.php?p=2187582#p2187540 Experienced RPi developer has diagnosed the driver - HDMI device and DMA are correctly reset at xrun recovery (i.e. snd_pcm_stop(XRUN) in the driver -> snd_pcm_prepare in the client). Therefore an option is that alsa supplies misaligned 8ch samples after the xrun recovery. Users checked the older HDMI driver bcm2835_audio (bcm2835-pcm.c) which accepts standard S16/24_LE formats and does not require the iec958 plugin and xruns did not cause any channel swaps. Hence aiming at the iec958 plugin. It seems to keep state (e.g. iec->counter) which perhaps may not be reset at snd_pcm_prepare. Thanks a lot, Pavel.

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ASoC: rockchip_i2s_tdm calibration clocking problem
by Pavel Hofman 31 Jan '24

31 Jan '24

Hi, I am hitting a clock issue with rockchip_i2s_tdm.c + simple-audio-card (+ RK3308). At boot the mclk clk_i2s0_8ch_tx is (somehow) initialized to some (unimportant?) value 50176000 Hz. Note that this frequency is not multiple of either 48kHz or 44.1kHz. Method asoc_simple_parse_clk() reads this value and sets it to simple_dai->sysclk. Subsequently at asoc_simple_dai_init this "random" initial value is stored to i2s_tdm->mclk_tx_freq: 17.839330] rockchip_i2s_tdm_set_sysclk+0x50/0xbc [snd_soc_rockchip_i2s_tdm] [ 17.839367] snd_soc_dai_set_sysclk+0x38/0xb8 [snd_soc_core] [ 17.839596] asoc_simple_init_dai+0x94/0xc0 [snd_soc_simple_card_utils] [ 17.839640] asoc_simple_dai_init+0x130/0x230 [snd_soc_simple_card_utils] [ 17.839672] snd_soc_link_init+0x28/0x90 [snd_soc_core] [ 17.839843] snd_soc_bind_card+0x60c/0xbb4 [snd_soc_core] When starting playback, called by rockchip_i2s_tdm_hw_params(), rockchip_i2s_tdm_calibrate_mclk() correctly switches parent of mclk_parent to correct root pll clock mclk_root0/1 for the given samplerate and correctly configures mclk_parent frequency. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… But right after that, the next line of rockchip_i2s_tdm_hw_params() calls rockchip_i2s_tdm_set_mclk() https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… This method calls clk_set_rate(i2s_tdm->mclk_tx, i2s_tdm->mclk_tx_freq), which resets the clock and its parental chain to the original incorrect value stored in i2s_tdm->mclk_tx_freq from the dai initialization. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… As a result, no matter what sample rate is being played, the i2s mclk clock always ends up configured incorrectly. DTS I2S sets all clocks, therefore the clk calibration in rockchip_i2s_tdm.c should be (and is) used: i2s_8ch_0: i2s@ff300000 { compatible = "rockchip,rk3308-i2s-tdm"; reg = <0x0 0xff300000 0x0 0x1000>; interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; clocks = <&cru SCLK_I2S0_8CH_TX>, <&cru SCLK_I2S0_8CH_RX>, <&cru HCLK_I2S0_8CH>, <&cru SCLK_I2S0_8CH_TX_SRC>, <&cru SCLK_I2S0_8CH_RX_SRC>, <&cru PLL_VPLL0>, <&cru PLL_VPLL1>; clock-names = "mclk_tx", "mclk_rx", "hclk", "mclk_tx_src", "mclk_rx_src", "mclk_root0", "mclk_root1"; ......... It seems to me that the calibration code should also rewrite the initially incorrect i2s_tdm->mclk_tx_freq and i2s_tdm->mclk_rx_freq with correct values corresponding to the momentary hw_params rate, or maybe rockchip_i2s_tdm_set_mclk() should not be called if rockchip_i2s_tdm_calibrate_mclk() is called a line above (i.e. putting the call to rockchip_i2s_tdm_set_mclk() into "else" branch). Thank you very much for help. With regards, Pavel.

1 0

[PATCH 1/2] dt-bindings: pinctrl: Unify "input-debounce" schema
by Rob Herring 31 Jan '24

31 Jan '24

nuvoton,npcm845-pinctrl defines the common "input-debounce" property as an array rather than an scalar. Update the common definition to expand it to an uint32-array, and update all the users of the property with array constraints. Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml | 3 ++- .../devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml | 1 - .../devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml | 3 ++- Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml | 2 +- 4 files changed, 5 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml index bb61a30321a1..482acda88e73 100644 --- a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml +++ b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml @@ -93,7 +93,8 @@ properties: input-schmitt-disable: true - input-debounce: true + input-debounce: + maxItems: 1 output-low: true diff --git a/Documentation/devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml index 3e8472898800..20cf0102aa63 100644 --- a/Documentation/devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml @@ -152,7 +152,6 @@ patternProperties: description: Debouncing periods in microseconds, one period per interrupt bank found in the controller - $ref: /schemas/types.yaml#/definitions/uint32-array minItems: 1 maxItems: 4 diff --git a/Documentation/devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml index 7b7f840ffc4c..08442c880f07 100644 --- a/Documentation/devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml @@ -103,7 +103,8 @@ patternProperties: items: pattern: "^gpio1?[0-9]{1,2}$" - input-debounce: true + input-debounce: + maxItems: 1 additionalProperties: false diff --git a/Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml b/Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml index be81ed22a036..d0af21a564b4 100644 --- a/Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml +++ b/Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml @@ -97,7 +97,7 @@ properties: description: disable schmitt-trigger mode input-debounce: - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: /schemas/types.yaml#/definitions/uint32-array description: Takes the debounce time in usec as argument or 0 to disable debouncing -- 2.43.0

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[PATCH v3] ASoC: Intel: mtl-match: Add cs42l43_l0 cs35l56_l23 for MTL
by Maciej Strozek 30 Jan '24

30 Jan '24

The layout is configured as: - Link0: CS42L43 Jack and mics (2ch) - Link2: 2x CS35L56 Speaker (amps 3 and 4, right) - Link3: 2x CS35L56 Speaker (amps 1 and 2, left) Corresponding SOF topology: https://github.com/thesofproject/sof/pull/8773 Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v2 -> v3: added layout description and link to required SOF topology v1 -> v2: swapped AMP3 and AMP4 prefixes .../intel/common/soc-acpi-intel-mtl-match.c | 57 +++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index feb12c6c85d1..23eaf47b3f24 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -377,6 +377,37 @@ static const struct snd_soc_acpi_adr_device cs35l56_2_adr[] = { } }; +static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = { + { + .adr = 0x00023201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00023301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + } + +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_l_adr[] = { + { + .adr = 0x00033001fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033101fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP2" + } +}; + static const struct snd_soc_acpi_link_adr rt5682_link2_max98373_link0[] = { /* Expected order: jack -> amp */ { @@ -554,6 +585,26 @@ static const struct snd_soc_acpi_link_adr mtl_cs42l43_cs35l56[] = { {} }; +static const struct snd_soc_acpi_link_adr cs42l43_link0_cs35l56_link2_link3[] = { + /* Expected order: jack -> amp */ + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_r_adr), + .adr_d = cs35l56_2_r_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_l_adr), + .adr_d = cs35l56_3_l_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { /* mockup tests need to be first */ @@ -599,6 +650,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt1318-l12-rt714-l0.tplg" }, + { + .link_mask = BIT(0) | BIT(2) | BIT(3), + .links = cs42l43_link0_cs35l56_link2_link3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-cs42l43-l0-cs35l56-l23.tplg", + }, { .link_mask = GENMASK(2, 0), .links = mtl_cs42l43_cs35l56, -- 2.34.1

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