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

[PATCH V2 0/4] ASoC: codecs: Add aw88399 amplifier driver
by wangweidong.a＠awinic.com 25 Oct '23

25 Oct '23

From: Weidong Wang <wangweidong.a(a)awinic.com> Add the awinic,aw88399 property to the awinic,aw88395.yaml file. Add i2c and amplifier registration for aw88399 and their associated operation functions. v1 -> v2: Modify the reset mode Delete AW88399_DSP_I2C_WRITES macros and related debugging statements Change the value of max_register in aw88399 Change the value of max_register in aw88261 Delete the judgment of unnecessary pointers Modify the judgment of the ret return value Weidong Wang (4): ASoC: dt-bindings: Add schema for "awinic,aw88399" ASoC: codecs: Modify max_register usage error ASoC: codecs: Add code for bin parsing compatible with aw88399 ASoC: codecs: Add aw88399 amplifier driver .../bindings/sound/awinic,aw88395.yaml | 1 + sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw88261.c | 2 +- sound/soc/codecs/aw88395/aw88395_lib.c | 3 + sound/soc/codecs/aw88395/aw88395_reg.h | 1 + sound/soc/codecs/aw88399.c | 1911 +++++++++++++++++ sound/soc/codecs/aw88399.h | 599 ++++++ 8 files changed, 2532 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/aw88399.c create mode 100644 sound/soc/codecs/aw88399.h base-commit: ce55c22ec8b223a90ff3e084d842f73cfba35588 -- 2.41.0

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[PATCH] sound: add quirk flag to enable native DSD for McIntosh devices
by McCarthy, Max 25 Oct '23

25 Oct '23

From: Max McCarthy <mmccarthy(a)mcintoshlabs.com> McIntosh devices supporting native DSD require the feature to be explicitly exposed. Add a flag that fixes an issue where DSD audio was defaulting to DSD over PCM instead of delivering raw DSD data. Signed-off-by: Max McCarthy <mmccarthy(a)mcintoshlabs.com> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4e64842245e1..ab2b938502eb 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2220,6 +2220,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x2ab6, /* T+A devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x2afd, /* McIntosh Laboratory, Inc. */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x2d87, /* Cayin device */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3336, /* HEM devices */ ---

2 1

[bug report] ASoC: mediatek: mt8186_mt6366_rt1019_rt5682s: add rt5650 support
by Dan Carpenter 25 Oct '23

25 Oct '23

Hello xiazhengqiao, The patch d88c43383101: "ASoC: mediatek: mt8186_mt6366_rt1019_rt5682s: add rt5650 support" from Oct 19, 2023 (linux-next), leads to the following Smatch static checker warning: sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c:198 mt8186_rt5682s_init() warn: does endianness matter for 'type'? sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c 161 static int mt8186_rt5682s_init(struct snd_soc_pcm_runtime *rtd) 162 { 163 struct snd_soc_component *cmpnt_afe = 164 snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); 165 struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe); 166 struct mtk_soc_card_data *soc_card_data = 167 snd_soc_card_get_drvdata(rtd->card); 168 struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv; 169 struct snd_soc_jack *jack = &priv->headset_jack; 170 struct snd_soc_component *cmpnt_codec = 171 snd_soc_rtd_to_codec(rtd, 0)->component; 172 int ret; 173 int type; 174 175 ret = mt8186_dai_i2s_set_share(afe, "I2S1", "I2S0"); 176 if (ret) { 177 dev_err(rtd->dev, "Failed to set up shared clocks\n"); 178 return ret; 179 } 180 181 ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", 182 SND_JACK_HEADSET | SND_JACK_BTN_0 | 183 SND_JACK_BTN_1 | SND_JACK_BTN_2 | 184 SND_JACK_BTN_3, 185 jack, mt8186_jack_pins, 186 ARRAY_SIZE(mt8186_jack_pins)); 187 if (ret) { 188 dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); 189 return ret; 190 } 191 192 snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); 193 snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); 194 snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); 195 snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); 196 197 type = SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3; --> 198 return snd_soc_component_set_jack(cmpnt_codec, jack, (void *)&type); This is an unpublished Smatch check where I manually review casts to see if they are correct. Quite often they aren't because of an endian bug or a 64 bit vs 32 bit issue. Here it's not clear to me what's happening. Normally with this sort of pass a void pointer code, you can tie it very easily to the same driver. But in this case it's much more difficult. There are two functions which use the void *data pointer, rt5640_set_jack() and rt5645_component_set_jack(). One takes an int and the other takes a struct rt5640_set_jack_data pointer. So presumably we know that the cmpnt_codec->driver->set_jack points to rt5645_component_set_jack(). But how do we know that? Is there a trick for me as a reviewer to use? 199 } regards, dan carpenter

2 2

[PATCH v9 00/34] Introduce QC USB SND audio offloading support
by Wesley Cheng 25 Oct '23

25 Oct '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 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 (4): xhci: split free interrupter into separate remove and free parts 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: 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 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 | 4 + .../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-mem.c | 136 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 53 +- drivers/usb/host/xhci-sideband.c | 372 ++++ drivers/usb/host/xhci.c | 114 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 67 + 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 | 201 ++ 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 | 1867 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 43 files changed, 5041 insertions(+), 89 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-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 v5 0/5] ASoC: makes CPU/Codec channel connection map more generic
by Kuninori Morimoto 25 Oct '23

25 Oct '23

Hi Mark Cc Bard, Pierre-Louis, Jerome, DT-ML This is v5 patch-set. Current ASoC is supporting CPU/Codec = N:M (N < M) connection by using ch_map idea. This patch-set expands it that all connection uses this idea, and no longer N < M limit [1][2]. Link: https://lore.kernel.org/r/87fs6wuszr.wl-kuninori.morimoto.gx@renesas.com [1] Link: https://lore.kernel.org/r/878r7yqeo4.wl-kuninori.morimoto.gx@renesas.com [2] This patch is tested on Audio-Graph-Card2 with sample dtsi, but needs Tested-by, at least from Intel. [PATCH 1/5] patch got Tested-by from Pierre-Louis / Jerome before, but v5 is using different idea from v4. Thus I didn't add below tag. Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Jerome Brunet <jbrunet(a)baylibre.com> v4 -> v5 - use cpu/codec index on ch_maps - separate card2 sample DT patch into prepare and new feature - ch-maps -> ch-map-idx on DT v3 -> v4 - add Jerome on To - add "description" on "ch-maps" v2 -> v3 - tidyup comment - use more clear connection image on DT - "ch_maps" -> "ch-maps" on DT - Add DT maintainer on "To:" for all patches v1 -> v2 - makes CPU/Codec connection relation clear on comment/sample - fixup type "connction" -> "connection" - makes error message clear Kuninori Morimoto (4): ASoC: makes CPU/Codec channel connection map more generic ASoC: audio-graph-card2: add CPU:Codec = N:M support ASoC: audio-graph-card2-custom-sample: add CPU/Codec = N:M sample dt-bindings: audio-graph-port: add ch-maps property Kuninori Morimoto (5): ASoC: makes CPU/Codec channel connection map more generic ASoC: audio-graph-card2: add CPU:Codec = N:M support ASoC: audio-graph-card2-custom-sample: tidyup comment / numbering ASoC: audio-graph-card2-custom-sample: add CPU/Codec = N:M sample dt-bindings: audio-graph-port: add ch-map-idx property .../bindings/sound/audio-graph-port.yaml | 7 +- include/sound/soc.h | 55 ++++++- .../audio-graph-card2-custom-sample.dtsi | 136 +++++++++++++++--- sound/soc/generic/audio-graph-card2.c | 49 +++++++ sound/soc/intel/boards/sof_sdw.c | 28 ++-- sound/soc/soc-core.c | 97 ++++++++++++- sound/soc/soc-dapm.c | 45 ++---- sound/soc/soc-pcm.c | 44 ++---- 8 files changed, 360 insertions(+), 101 deletions(-) -- 2.25.1

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[PATCH v2 0/3] ASoC: mediatek: Remove redundant code and add sample rate checker of MT7986 SoC
by Maso Huang 24 Oct '23

24 Oct '23

Changes in v2: - devide v1 patch [1/2] into v2 [1/3] [2/3] with appropriate title and description - refine error log based on reviewer's suggestions [3/3] 1. Drop the remove callback of mt7986_wm8960. 2. Remove the mt7986_wm8960_priv structure 3. Add sample rate checker. Patches are based on broonie tree "for-next" branch. Maso Huang (3): ASoC: mediatek: mt7986: drop the remove callback of mt7986_wm8960 ASoC: mediatek: mt7986: remove the mt7986_wm8960_priv structure ASoC: mediatek: mt7986: add sample rate checker sound/soc/mediatek/mt7986/mt7986-dai-etdm.c | 23 +++++++++-- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 43 ++++++--------------- 2 files changed, 31 insertions(+), 35 deletions(-) -- 2.18.0

3 6

[PATCH 1/6] ASoC: Intel: soc-acpi-cht: Add Lenovo Yoga Tab 3 Pro YT3-X90 quirk
by Hans de Goede 24 Oct '23

24 Oct '23

The Lenovo Yoga Tab 3 Pro YT3-X90 x86 tablet, which ships with Android with a custom kernel as factory OS, does not list the used WM5102 codec inside its DSDT. Workaround this with a new snd_soc_acpi_intel_baytrail_machines[] entry which matches on the SST id instead of the codec id like nocodec does, combined with using a machine_quirk callback which returns NULL on other machines to skip the new entry on other machines. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- .../intel/common/soc-acpi-intel-cht-match.c | 43 +++++++++++++++++++ 1 file changed, 43 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index cdcbf04b8832..5e2ec60e2954 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -75,6 +75,39 @@ static struct snd_soc_acpi_mach *cht_ess8316_quirk(void *arg) return arg; } +/* + * The Lenovo Yoga Tab 3 Pro YT3-X90, with Android factory OS has a buggy DSDT + * with the coded not being listed at all. + */ +static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { + { + /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), + DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), + }, + }, + { } +}; + +static struct snd_soc_acpi_mach cht_lenovo_yoga_tab3_x90_mach = { + .id = "10WM5102", + .drv_name = "bytcr_wm5102", + .fw_filename = "intel/fw_sst_22a8.bin", + .board = "bytcr_wm5102", + .sof_tplg_filename = "sof-cht-wm5102.tplg", +}; + +static struct snd_soc_acpi_mach *lenovo_yt3_x90_quirk(void *arg) +{ + if (dmi_check_system(lenovo_yoga_tab3_x90)) + return &cht_lenovo_yoga_tab3_x90_mach; + + /* Skip wildcard match snd_soc_acpi_intel_cherrytrail_machines[] entry */ + return NULL; +} + static const struct snd_soc_acpi_codecs rt5640_comp_ids = { .num_codecs = 2, .codecs = { "10EC5640", "10EC3276" }, @@ -175,6 +208,16 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "sof_pcm512x", .sof_tplg_filename = "sof-cht-src-50khz-pcm512x.tplg", }, + /* + * Special case for the Lenovo Yoga Tab 3 Pro YT3-X90 where the DSDT + * misses the codec. Match on the SST id instead, lenovo_yt3_x90_quirk() + * will return a YT3 specific mach or NULL when called on other hw, + * skipping this entry. + */ + { + .id = "808622A8", + .machine_quirk = lenovo_yt3_x90_quirk, + }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) /* -- 2.41.0

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[PATCH][next] ASoC: mediatek: mt8186: remove redundant assignments to variable tdm_con
by Colin Ian King 24 Oct '23

24 Oct '23

There are two occurrences where variable tdm_con is being initialized to zero and the next statement re-assigns tdm_con to a new value. The initializations are redundant and can be removed. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c index 4148dceb3a4c..ef2801f84d27 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c @@ -416,12 +416,10 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(afe->regmap, ETDM_IN1_CON1, ETDM_IN_CON1_CTRL_MASK, tdm_con); /* ETDM_IN1_CON3 */ - tdm_con = 0; tdm_con = ETDM_IN_CON3_FS(tran_rate); regmap_update_bits(afe->regmap, ETDM_IN1_CON3, ETDM_IN_CON3_CTRL_MASK, tdm_con); /* ETDM_IN1_CON4 */ - tdm_con = 0; tdm_con = ETDM_IN_CON4_FS(tran_relatch_rate); if (slave_mode) { if (lrck_inv) -- 2.39.2

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[PATCH][next] ASoC: codecs: rt298: remove redundant assignment to d_len_code
by Colin Ian King 24 Oct '23

24 Oct '23

Variable d_len_code is being initialized to zero and then re-assigned a different value in all the valid cases in the following switch statement. The only place it is not being assigned a value is on the return for a default case and in this case it does not need to be assigned. The initialization is redundant and can be removed. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/rt298.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 8fbd25ad9b47..ad3783ade1b5 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -789,7 +789,6 @@ static int rt298_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - d_len_code = 0; switch (params_width(params)) { /* bit 6:4 Bits per Sample */ case 16: -- 2.39.2

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[PATCH] ASoC: ti: ams-delta: Allow it to be test compiled
by Janusz Krzysztofik 24 Oct '23

24 Oct '23

The driver is now built only when MACH_AMS_DELTA is selected, which requires a very specific selection of ARCH settings. As a consequence, it gets very little attention from build-bots, if not none. Drop the driver dependency on <asm/mach-types.h>, no longer required since conversion to snd_soc_register_card() and drop of machine_is_ams_delta(). With that in place, allow the driver to be built in any environment as long as COMPILE_TEST is selected. Take care of not selecting SND_SOC_OMAP_MCBSP if COMMON_CLK is not selected. Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- sound/soc/ti/Kconfig | 5 +++-- sound/soc/ti/ams-delta.c | 2 -- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 593be22503b5..e22e41af3226 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -125,8 +125,9 @@ config SND_SOC_OMAP_ABE_TWL6040 config SND_SOC_OMAP_AMS_DELTA tristate "SoC Audio support for Amstrad E3 (Delta) videophone" - depends on MACH_AMS_DELTA && TTY - select SND_SOC_OMAP_MCBSP + depends on MACH_AMS_DELTA || COMPILE_TEST + depends on TTY + select SND_SOC_OMAP_MCBSP if COMMON_CLK select SND_SOC_CX20442 help Say Y or M if you want to add support for SoC audio device diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 371943350fdf..7436cca2d2e0 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -16,8 +16,6 @@ #include <sound/soc.h> #include <sound/jack.h> -#include <asm/mach-types.h> - #include <linux/platform_data/asoc-ti-mcbsp.h> #include "omap-mcbsp.h" -- 2.42.0

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