September 2023 - Alsa-devel - mailman.alsa-project.org

topology/topology.c: error: 'LC_ALL' undeclared - compiling alsa-utils 1.2.10 with musl
by GitHub issues - opened 19 Sep '23

19 Sep '23

alsa-project/alsa-utils issue #239 was opened from michaelopdenacker: I'm facing the below issue trying to compile alsa-utils with the musl C library with the Yocto Project Poky master branch: x86_64-poky-linux-musl-gcc -m64 -march=core2 -mtune=core2 -msse3 -mfpmath=sse -fstack-protector-strong -O2 -D_FORTIFY_SOURCE=2 -Wformat -Wformat-security -Werror=format-security --sysroot=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/recipe-sysroot -DHAVE_CONFIG_H -I. -I../../alsa-utils-1.2.10/topology -I../include -Wall -I../../alsa-utils-1.2.10/include -DALSA_TOPOLOGY_PLUGIN_DIR=\"/usr/lib/alsa-topology\" -O2 -pipe -g -feliminate-unused-debug-types -fcanon-prefix-map -fmacro-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/alsa-utils-1.2.10=/usr/src/debug/alsa-utils/1.2.10-r0 -fdebug-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/alsa-utils-1.2.10=/usr/src/debug/alsa-utils/1.2.10-r0 -fmacro-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/build=/usr/src/debug/alsa-utils/1.2.10-r0 -fdebug-prefix -map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/build=/usr/src/debug/alsa-utils/1.2.10-r0 -fdebug-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/recipe-sysroot= -fmacro-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/recipe-sysroot= -fdebug-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/recipe-sysroot-native= -c -o pre-process-dai.o ../../alsa-utils-1.2.10/topology/pre-process-dai.c | ../../alsa-utils-1.2.10/topology/topology.c: In function 'main': | ../../alsa-utils-1.2.10/topology/topology.c:442:9: warning: implicit declaration of function 'setlocale' [-Wimplicit-function-declaration] | 442 | setlocale(LC_ALL, ""); | | ^~~~~~~~~ | ../../alsa-utils-1.2.10/topology/topology.c:442:19: error: 'LC_ALL' undeclared (first use in this function) | 442 | setlocale(LC_ALL, ""); | | ^~~~~~ Including <locale.h> in topology.c makes the error go away. I'll send a patch shortly. Michael Issue URL : https://github.com/alsa-project/alsa-utils/issues/239 Repository URL: https://github.com/alsa-project/alsa-utils

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nhlt-dmic-info.c: error: unknown type name 'u_int8_t' - compiling alsa-utils 1.2.10 with musl
by GitHub issues - opened 19 Sep '23

19 Sep '23

alsa-project/alsa-utils issue #238 was opened from michaelopdenacker: I'm facing the below issue trying to compile alsa-utils with the musl C library with the Yocto Project Poky master branch: | x86_64-poky-linux-musl-gcc -m64 -march=core2 -mtune=core2 -msse3 -mfpmath=sse -fstack-protector-strong -O2 -D_FORTIFY_SOURCE=2 -Wformat -Wformat-security -Werror=format-security --sysroot=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/recipe-sysroot -DHAVE_CONFIG_H -I. -I../../alsa-utils-1.2.10/nhlt -I../include -I../../alsa-utils-1.2.10/include -O2 -pipe -g -feliminate-unused-debug-types -fcanon-prefix-map -fmacro-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/alsa-utils-1.2.10=/usr/src/debug/alsa-utils/1.2.10-r0 -fdebug-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/alsa-utils-1.2.10=/usr/src/debug/alsa-utils/1.2.10-r0 -fmacro-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/build=/usr/src/debug/alsa-utils/1.2.10-r0 -fdebug-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-l inux-musl/alsa-utils/1.2.10/build=/usr/src/debug/alsa-utils/1.2.10-r0 -fdebug-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/recipe-sysroot= -fmacro-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/recipe-sysroot= -fdebug-prefix-map=/home/mike/work/yocto/poky/build/tmp/work/core2-64-poky-linux-musl/alsa-utils/1.2.10/recipe-sysroot-native= -c -o nhlt-dmic-info.o ../../alsa-utils-1.2.10/nhlt/nhlt-dmic-info.c | ../../alsa-utils-1.2.10/nhlt/nhlt-dmic-info.c:48:36: error: unknown type name 'u_int8_t'; did you mean 'uint8_t'? | 48 | static const char *microphone_type(u_int8_t type) | | ^~~~~~~~ | | uint8_t. ... Including <sys/types.h> from nhlt/nhlt-dmic-info.c solves this issue. I'll send a patch shortly. Issue URL : https://github.com/alsa-project/alsa-utils/issues/238 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] mfd: cs42l43: add REGMAP_IRQ requirement
by Randy Dunlap 19 Sep '23

19 Sep '23

The cs42l43 core uses devm_regmap_add_irq_chip(), so add REGMAP_IRQ to what the core driver selects in order to provide this API. This fixes a build error: aarch64-linux-ld: drivers/mfd/cs42l43.o: in function `cs42l43_boot_work': cs42l43.c:(.text+0x19d4): undefined reference to `devm_regmap_add_irq_chip' Fixes: ace6d1448138 ("mfd: cs42l43: Add support for cs42l43 core driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: James Schulman <james.schulman(a)cirrus.com> Cc: David Rhodes <david.rhodes(a)cirrus.com> Cc: Richard Fitzgerald <rf(a)opensource.cirrus.com> Cc: alsa-devel(a)alsa-project.org Cc: patches(a)opensource.cirrus.com Cc: Lee Jones <lee(a)kernel.org> --- drivers/mfd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff -- a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -241,6 +241,7 @@ config MFD_CS42L43 tristate select MFD_CORE select REGMAP + select REGMAP_IRQ config MFD_CS42L43_I2C tristate "Cirrus Logic CS42L43 (I2C)"

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[PATCH] ALSA: usb-audio: scarlett_gen2: Fix another -Wformat-truncation warning
by Peter Ujfalusi 19 Sep '23

19 Sep '23

The recent enablement of -Wformat-truncation leads to a false-positive warning for mixer_scarlett_gen2.c. For suppressing the warning, replace snprintf() with scnprintf(). As stated in the above, truncation doesn't matter. Fixes: 78bd8f5126f8 ("ALSA: usb-audio: scarlett_gen2: Fix -Wformat-truncation warning") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/usb/mixer_scarlett_gen2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index e0242b38b3f7..23aab1947958 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -3195,8 +3195,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /* Add input phantom controls */ if (info->inputs_per_phantom == 1) { for (i = 0; i < info->phantom_count; i++) { - snprintf(s, sizeof(s), fmt, i + 1, - "Phantom Power", "Switch"); + scnprintf(s, sizeof(s), fmt, i + 1, + "Phantom Power", "Switch"); err = scarlett2_add_new_ctl( mixer, &scarlett2_phantom_ctl, i, 1, s, &private->phantom_ctls[i]); -- 2.42.0

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Gapless support
by GitHub pull_request - opened 19 Sep '23

19 Sep '23

alsa-project/tinycompress pull request #23 was opened from mohsRafi: This branch will add gapless playback support while switching audio track files. For gapless playback use case where each track differs in encoding new codec parameters that need to be propagated to kernel is done via set codec parameters for next track API Request URL : https://github.com/alsa-project/tinycompress/pull/23 Patch URL : https://github.com/alsa-project/tinycompress/pull/23.patch Repository URL: https://github.com/alsa-project/tinycompress

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[PATCH] ASoC: simple-card-utils: fixup simple_util_startup() error handling
by Kuninori Morimoto 19 Sep '23

19 Sep '23

It should use "goto" instead of "return" Fixes: 5ca2ab459817 ("ASoC: simple-card-utils: Add new system-clock-fixed flag") Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Closes: https://lore.kernel.org/all/202309141205.ITZeDJxV-lkp@intel.com/ Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- I have mentioned that different point when reply, but Dan's report was correct. sound/soc/generic/simple-card-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 36ce3a4343f9..348fe6f8a537 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -310,7 +310,7 @@ int simple_util_startup(struct snd_pcm_substream *substream) if (fixed_sysclk % props->mclk_fs) { dev_err(rtd->dev, "fixed sysclk %u not divisible by mclk_fs %u\n", fixed_sysclk, props->mclk_fs); - return -EINVAL; + goto codec_err; } ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, fixed_rate, fixed_rate); -- 2.25.1

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[PATCH v6 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 19 Sep '23

19 Sep '23

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 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 (30): 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 dt-bindings: usb: dwc3: Add snps,num-hc-interrupters definition dt-bindings: usb: xhci: Add num-hc-interrupters definition usb: dwc3: Add DT parameter to specify maximum number of 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 ASoC: dt-bindings: Add Q6USB backend ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: q6afe: Split USB AFE dev_token param into separate API ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: qcom: qdsp6: Add SND kcontrol to select offload device ASoC: qcom: qdsp6: Add SND kcontrol for fetching offload status ASoC: qcom: qdsp6: Add headphone jack for offload connection status 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 .../devicetree/bindings/sound/qcom,q6usb.yaml | 55 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 13 + .../devicetree/bindings/usb/usb-xhci.yaml | 6 + 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 | 4 + drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-hub.c | 29 +- drivers/usb/host/xhci-mem.c | 102 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 48 +- drivers/usb/host/xhci-sideband.c | 195 ++ drivers/usb/host/xhci.c | 111 +- drivers/usb/host/xhci.h | 105 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 86 + include/linux/usb/xhci-sideband.h | 57 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 51 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 56 + sound/soc/qcom/qdsp6/q6afe.c | 206 +- 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 | 450 ++++ sound/soc/soc-usb.c | 200 ++ sound/usb/Kconfig | 15 + sound/usb/Makefile | 2 +- sound/usb/card.c | 123 ++ sound/usb/card.h | 23 + sound/usb/endpoint.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 94 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1854 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 45 files changed, 4902 insertions(+), 196 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb.yaml create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-intr.h 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/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 AUTOSEL 6.4 04/31] ASoC: amd: vangogh: Use dmi_first_match() for DMI quirk handling
by Sasha Levin 19 Sep '23

19 Sep '23

From: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> [ Upstream commit 3dd26e27ccb4f18b4d25c0a49e1888eca9c6a724 ] In preparation for supporting ACPI probing, move DMI quirk handling logic at the probe's top, to be able to return as quickly as possible in case there is no DMI matching. Additionally, simplify the code by replacing dmi_check_system() and related callback with dmi_first_match(). While at it, also drop a few unnecessary empty lines. Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> Link: https://lore.kernel.org/r/20230705214800.193244-3-cristian.ciocaltea@collab… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/vangogh/acp5x-mach.c | 28 +++++++--------------------- 1 file changed, 7 insertions(+), 21 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-mach.c b/sound/soc/amd/vangogh/acp5x-mach.c index e5bcd1e6eb73a..7040809b24729 100644 --- a/sound/soc/amd/vangogh/acp5x-mach.c +++ b/sound/soc/amd/vangogh/acp5x-mach.c @@ -22,7 +22,6 @@ #define DRV_NAME "acp5x_mach" #define DUAL_CHANNEL 2 -#define VG_JUPITER 1 #define ACP5X_NAU8821_BCLK 3072000 #define ACP5X_NAU8821_FREQ_OUT 12288000 #define ACP5X_NAU8821_COMP_NAME "i2c-NVTN2020:00" @@ -31,7 +30,6 @@ #define ACP5X_CS35L41_COMP_RNAME "spi-VLV1776:01" #define ACP5X_CS35L41_DAI_NAME "cs35l41-pcm" -static unsigned long acp5x_machine_id; static struct snd_soc_jack vg_headset; SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("acp5x_i2s_dma.0"))); @@ -242,7 +240,6 @@ static int acp5x_cs35l41_hw_params(struct snd_pcm_substream *substream, } return 0; - } static const struct snd_soc_ops acp5x_cs35l41_play_ops = { @@ -292,8 +289,6 @@ static struct snd_soc_dai_link acp5x_8821_35l41_dai[] = { }, }; - - static const struct snd_soc_dapm_widget acp5x_8821_35l41_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -331,16 +326,8 @@ static struct snd_soc_card acp5x_8821_35l41_card = { .num_controls = ARRAY_SIZE(acp5x_8821_controls), }; -static int acp5x_vg_quirk_cb(const struct dmi_system_id *id) -{ - acp5x_machine_id = VG_JUPITER; - - return 1; -} - static const struct dmi_system_id acp5x_vg_quirk_table[] = { { - .callback = acp5x_vg_quirk_cb, .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Valve"), DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Jupiter"), @@ -351,23 +338,22 @@ static const struct dmi_system_id acp5x_vg_quirk_table[] = { static int acp5x_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct acp5x_platform_info *machine; struct device *dev = &pdev->dev; struct snd_soc_card *card; int ret; + dmi_id = dmi_first_match(acp5x_vg_quirk_table); + if (!dmi_id) + return -ENODEV; + + card = &acp5x_8821_35l41_card; + machine = devm_kzalloc(dev, sizeof(*machine), GFP_KERNEL); if (!machine) return -ENOMEM; - dmi_check_system(acp5x_vg_quirk_table); - switch (acp5x_machine_id) { - case VG_JUPITER: - card = &acp5x_8821_35l41_card; - break; - default: - return -ENODEV; - } card->dev = dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); -- 2.40.1

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[PATCH 1/2] ALSA: docs: Add Marian M2 driver documentation
by Ivan Orlov 18 Sep '23

18 Sep '23

Add documentation for the new MARIAN Seraph M2 sound card. It covers current development status, available controls of the card and information about the integrated loopback. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/marian-m2.rst | 104 ++++++++++++++++++++++++ 2 files changed, 105 insertions(+) create mode 100644 Documentation/sound/cards/marian-m2.rst diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index e68bbb13c384..e873592d8d00 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -19,3 +19,4 @@ Card-Specific Information serial-u16550 img-spdif-in pcmtest + marian-m2 diff --git a/Documentation/sound/cards/marian-m2.rst b/Documentation/sound/cards/marian-m2.rst new file mode 100644 index 000000000000..a968feea79c0 --- /dev/null +++ b/Documentation/sound/cards/marian-m2.rst @@ -0,0 +1,104 @@ +.. SPDX-License-Identifier: GPL-2.0 + +======================= +MARIAN Seraph M2 Driver +======================= + +Sep 18, 2023 + +Ivan Orlov <ivan.orlov0322(a)gmail.com> + +STATE OF DEVELOPMENT +==================== + +This driver is based on initial driver written by Florian Faber in 2012, which seemed to work fine. +However, the initial code contained multiple issues, which had to be solved before sending the +driver upstream. + +The vendor lost the full documentation, so what we have here was recovered from drafts and found +after experiments with the card. + +What seems to be working fine: +- Playback and capture for all supported rates +- Integrated loopback (with some exceptions, see below) + +MEMORY MODEL +============ + +The hardware requires one huge contiguous DMA space to be allocated. After allocation, the bus address of +this buffer should be written to the hardware register. + +We can split this space into two parts: the first one contains samples for capture, another one contains +play samples: + +CAPTURE_CH_0, CAPTURE_CH_1, ..., CAPTURE_CH_127 | PLAY_CH_0, PLAY_CH_1, ..., PLAY_CH_127 + +The card supports the non-interleaved access mode only, so samples for each channel lay together: + +C0, C0, ..., C0, C1, C1, ..., C1, ..., C127 | C0, C0, ..., C0, C1, C1, ..., C1, ..., C127 + +The count of samples per each channel buffer needs to be set explicitly, so the address of the first +byte of the playback data depends on this value. The playback buffer starts where the capture buffer ends. +It makes the arbitrary period count/buffer size feature impossible to implement, and the driver supports only +2 periods per buffer. + +Controls +======== + +Input 1 Sync + 0 - No signal, 1 - valid MADI signal found, 2 - Synced with MADI signal +Input 2 Sync + 0 - No signal, 1 - valid MADI signal found, 2 - Synced with MADI signal + +Input 1 Channel Mode + 0 - 56 channels, 1 - 64 channels +Input 2 Channel Mode + 0 - 56 channels, 1 - 64 channels + +Input 1 Frame Mode + 0 - 48 kHz, 1 - 96 kHz +Input 2 Frame Mode + 0 - 48 kHz, 1 - 96 kHz + +Input 1 Frequency + Measured frequency on Input 1 +Input 2 Frequency + Measured frequency on Input 2 + +Output 1 Channel Mode + 0 - 56 channels, 1 - 64 channels +Output 2 Channel Mode + 0 - 56 channels, 1 - 64 channels + +Output 1 Frame Mode + 0 - 48 kHz, 1 - 96 kHz +Output 2 Frame Mode + 0 - 48 kHz, 1 - 96 kHz + +Clock Source + Internal/Sync Bus, Input Port 1, Input Port 2 + +Speed mode + VCO clock range (slow/fast) + +DCO Frequency (Hz) + +DCO Frequency (ms) + +DCO Detune + Range: -200...200 + +Loopback + Enable/Disable integrated loopback + + +Loopback +======== + +The card contains integrated loopback. When it is enabled, it sets the hardware’s DAW-in memory pointer +to the hardware’s DAW-out memory. So, what you play is what you record. + +You can enable the integrated loopback using the corresponding control. + +The loopback seems to work well on lower rates (like 28000). However, when the rate goes higher, I observe +multiple mistakes in recorded byte ordering. -- 2.34.1

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[PATCH 0/2] ASoC: da7213: add .auto_selectable_formats support
by Kuninori Morimoto 18 Sep '23

18 Sep '23

Hi Mark da7213 is still using M/S instead of P/C for SND_SOC_DAIFMT_CBx_CFx. [PATCH 1/2] will update it. [PATCH 2/2] will enable DAI format automatic select. Kuninori Morimoto (2): ASoC: da7213: tidyup SND_SOC_DAIFMT_xxx ASoC: da7213: add .auto_selectable_formats support sound/soc/codecs/da7213.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) -- 2.25.1

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