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

[PATCH] ASoC: dwc: Add Single DMA mode support
by Myunguk Kim 13 Sep '23

13 Sep '23

There is a SoC between dwc and DMA block (ie. PL330) that does not have a burst signal and supports only single. So write not-support-burst property on dts, it support single DMA mode. Signed-off-by: Myunguk Kim <mwkim(a)gaonchips.com> --- .../bindings/sound/snps,designware-i2s.yaml | 3 +++ include/sound/designware_i2s.h | 1 + sound/soc/dwc/dwc-i2s.c | 12 ++++++++++-- 3 files changed, 14 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml index a48d040b0a4f..43a46ba2a70c 100644 --- a/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml @@ -86,6 +86,9 @@ properties: The phandle to System Register Controller syscon node and the I2S-rx(ADC) enabled control offset and mask of SYS_SYSCONSAIF__SYSCFG register. + no-burst: + description: Use single transaction only between DesignWare I2S and DMA IP. + allOf: - $ref: dai-common.yaml# - if: diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index f6803205a9fb..f0207f21f09d 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -36,6 +36,7 @@ struct i2s_platform_data { #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) #define DW_I2S_QUIRK_16BIT_IDX_OVERRIDE (1 << 2) + #define DW_I2S_QUIRK_NO_BURST (1 << 3) unsigned int quirks; unsigned int i2s_reg_comp1; unsigned int i2s_reg_comp2; diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 5ab1b3eb2d28..71ff894035a4 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -713,7 +713,10 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; dev->play_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2]) >> 8; - dev->play_dma_data.dt.maxburst = 16; + if (dev->quirks & DW_I2S_QUIRK_NO_BURST) + dev->play_dma_data.dt.maxburst = 1; + else + dev->play_dma_data.dt.maxburst = 16; } if (COMP1_RX_ENABLED(comp1)) { idx2 = COMP2_RX_WORDSIZE_0(comp2); @@ -722,7 +725,10 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; dev->capture_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2] >> 8); - dev->capture_dma_data.dt.maxburst = 16; + if (dev->quirks & DW_I2S_QUIRK_NO_BURST) + dev->capture_dma_data.dt.maxburst = 1; + else + dev->capture_dma_data.dt.maxburst = 16; } return 0; @@ -979,6 +985,8 @@ static int dw_i2s_probe(struct platform_device *pdev) ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata); } else { clk_id = "i2sclk"; + if (of_get_property(pdev->dev.of_node, "no-burst", NULL)) + dev->quirks |= DW_I2S_QUIRK_NO_BURST; ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res); } if (ret < 0) -- 2.34.1

3 7

AIF2 ADC Stereo Capture Route is uninitialised in alsa-ucm-conf
by GitHub issues - edited 13 Sep '23

13 Sep '23

alsa-project/alsa-ucm-conf issue #351 was edited from broukema: DESCRIPTION: `AIF2 ADC Stereo Capture Route` is not initialised in `alsa-ucm-conf`. This implies that if its setting is significant, any accidental change by the user (using any of the various `alsa` tools) to a wrong setting will not be corrected by any of the three `PinePhone/*.conf` files, leading, in particular, to audio errors during a phone call. This was discovered by testing variations on the Mobian version of the PinePhone alsa-ucm-conf files. CODING ARGUMENT: All parameters that can significantly affect the audio system should be either initialised or overridden in configuration files that the user expects to necessarily work, such as those for phone calls. In `alsa-ucm-conf` commit b68aa52 (one beyond v1.2.10): ```` $ grep -ri "AIF2 ADC Stereo Capture" . $ ```` shows that `AIF2 ADC Stereo Capture Route` is not set for any device at all (PinePhone or other). HARDWARE ARGUMENT: For the PinePhone (PP), Megi recommends in https://xnux.eu/devices/feature/audio-pp.html that `AIF2 ADC Stereo Capture Route` is set to `Mix Mono`. TESTING ARGUMENT: Experimentation with different stored alsa settings (from ```` sudo alsactl store -f - > YYYYMMDD_example_alsa_settings ```` ) showed that settings files whose only substantial difference was in the setting of `AIF2 ADC Stereo Capture Route` had the effect that audio from the PP was heard on the remote phone when the value was `Mix Mono` and was not heard with the value `Stereo`. Testing was done with the equivalent of https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… with the settings file https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… that has the `Mix Mono` value selected. RECOMMENDATIONS: * `AIF2 ADC Stereo Capture Route` should be initialised or set to *something* in one of the .conf files (in the PP case, most likely in `VoiceCall.conf`) to prevent users who experiment and have difficulty remembering which of the 50 PP alsa parameters should be set in which way from accidentally sabotaging their otherwise working audio calls; * Based on Megi's 2020 page and my testing, it seems that the correct value should be `Mix Mono`. COMMENT: While setting all 50 of the alsa settings ```` $ sudo alsactl store -f - |grep "name " |wc 50 258 1788 ```` in the .conf files is most likely unnecessary, there could be others that are necessary too. Ideally, someone who really understands how the PP audio system works (including Megi's two diagrams at https://xnux.eu/devices/feature/audio-pp.html) should be able to make these recommendations. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/351 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: dwc: fix typo in comment
by Myunguk Kim 13 Sep '23

13 Sep '23

This fixes typo in comment Signed-off-by: Myunguk Kim <mwkim(a)gaonchips.com> --- sound/soc/dwc/dwc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 71ff894035a4..45e55e550924 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -138,7 +138,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) irq_valid = true; } - /* Error Handling: TX */ + /* Error Handling: RX */ if (isr[i] & ISR_RXFO) { dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; -- 2.34.1

4 4

[PATCH] ASoC: imx-rpmsg: Set ignore_pmdown_time for dai_link
by Chancel Liu 13 Sep '23

13 Sep '23

i.MX rpmsg sound cards work on codec slave mode. MCLK will be disabled by CPU DAI driver in hw_free(). Some codec requires MCLK present at power up/down sequence. So need to set ignore_pmdown_time to power down codec immediately before MCLK is turned off. Take WM8962 as an example, if MCLK is disabled before DAPM power down playback stream, FIFO error will arise in WM8962 which will have bad impact on playback next. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/imx-rpmsg.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 3c7b95db2eac..d2ecd5e821b1 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -89,6 +89,13 @@ static int imx_rpmsg_probe(struct platform_device *pdev) SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + /* i.MX rpmsg sound cards work on codec slave mode. MCLK will be + * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK + * present at power up/down sequence. So need to set ignore_pmdown_time + * to power down codec immediately before MCLK is turned off. + */ + data->dai.ignore_pmdown_time = 1; + /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); if (ret) { -- 2.25.1

2 1

time to remove COPYING.GPL from alsa-plugins?
by GitHub issues - opened 13 Sep '23

13 Sep '23

alsa-project/alsa-plugins issue #54 was opened from michael-mw-wang: It seems the only thing in alsa-plugins that could have a GPL license is rate/rate_samplerate.c, which has this at the top of the file: * This plugin code is supposed to be used and distributed primarily * under GPL v2 or later, in order to follow the license of libsamplerate. * However, if you already own a commercial license to use libsamplerate * for dynamic linking, this plugin code can be used and distributed also * under LGPL v2.1 or later. In 2016, libsamplerate dropped its commercial license requirement and was released with the 2-Clause BSD license. (See http://libsndfile.github.io/libsamplerate//license.html) So I was wondering if the license for rate_samplerate.c can be updated to remove the GPLv2, which means the COPYING.GPL file can also be removed from the alsa-plugins package? Thank you! Issue URL : https://github.com/alsa-project/alsa-plugins/issues/54 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v5 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 13 Sep '23

13 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 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 (29): 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: Add USB backend ASoC driver for Q6 sound: usb: card: Introduce USB SND platform op callbacks sound: usb: 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 sound: usb: qcom: Add USB QMI definitions sound: usb: Introduce QC USB SND offloading support sound: usb: card: Check for support for requested audio format sound: soc: soc-usb: Add PCM format check API for USB backend sound: soc: qcom: qusb6: Ensure PCM format is supported by USB audio device sound: usb: 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 sound: Pass USB SND card and PCM information to SOC USB sound: soc: qdsp6: Add SND kcontrol to select offload device sound: soc: qdsp6: Add SND kcontrol for fetching offload status sound: soc: qcom: q6usb: Add headphone jack for offload connection status sound: usb: qc_audio_offload: Use card and PCM index from QMI request sound: usb: card: Allow for rediscovery of connected USB SND devices sound: soc: soc-usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,q6usb-dais.yaml | 53 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 13 + .../devicetree/bindings/usb/usb-xhci.yaml | 4 + 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 | 55 + 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 | 452 ++++ sound/soc/soc-usb.c | 200 ++ sound/usb/Kconfig | 15 + sound/usb/Makefile | 2 +- sound/usb/card.c | 108 + sound/usb/card.h | 23 + sound/usb/endpoint.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 86 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1837 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 45 files changed, 4863 insertions(+), 192 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.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

4 49

[PATCH] ASoC: cs35l56: Disable low-power hibernation mode
by Richard Fitzgerald 13 Sep '23

13 Sep '23

Do not allow the CS35L56 to be put into its lowest power "hibernation" mode. This only affects I2C because "hibernation" is already disabled on SPI and SoundWire. Recent firmwares need a different wake-up sequence. Until that sequence has been specified, the chip "hibernation" mode must be disabled otherwise it can intermittently fail to wake. THIS WILL NOT APPLY CLEANLY TO 6.5 AND EARLIER: We will send a separate backport patch to stable. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-i2c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 9f4f2f4f23f5..d10e0e2380e8 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -27,7 +27,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client) return -ENOMEM; cs35l56->base.dev = dev; - cs35l56->base.can_hibernate = true; i2c_set_clientdata(client, cs35l56); cs35l56->base.regmap = devm_regmap_init_i2c(client, regmap_config); -- 2.30.2

2 1

[PATCH 0/6] ASoC: rt5640: Fix various IRQ handling issues
by Hans de Goede 13 Sep '23

13 Sep '23

Hi All, The recent(ish) rt5640 changes to add HDA header jack-detect support and the related suspend/resume handling fixes have introduced several issues with IRQ handling on boards not using the HDA header jack-detect support. This series fixes these issues, see the individual commit messages for details. Regards, Hans Hans de Goede (6): ASoC: rt5640: Revert "Fix sleep in atomic context" ASoC: rt5640: Fix sleep in atomic context ASoC: rt5640: Do not disable/enable IRQ twice on suspend/resume ASoC: rt5640: Enable the IRQ on resume after configuring jack-detect ASoC: rt5640: Fix IRQ not being free-ed for HDA jack detect mode ASoC: rt5640: Only cancel jack-detect work on suspend if active sound/soc/codecs/rt5640.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) -- 2.41.0

2 7

AIF2 ADC Stereo Capture Route is unitialised in alsa-ucm-conf
by GitHub issues - edited 12 Sep '23

12 Sep '23

alsa-project/alsa-ucm-conf issue #351 was edited from broukema: DESCRIPTION: `AIF2 ADC Stereo Capture Route` is not initialised in `alsa-ucm-conf`. This implies that if its setting is significant, any accidental change by the user (using any of the various `alsa` tools) to a wrong setting will not be corrected by any of the three `PinePhone/*.conf` files, leading, in particular, to audio errors during a phone call. This was discovered by testing variations on the Mobian version of the PinePhone alsa-ucm-conf files. CODING ARGUMENT: All parameters that can significantly affect the audio system should be either initialised or overridden in configuration files that the user expects to necessarily work, such as those for phone calls. In `alsa-ucm-conf` commit b68aa52 (one beyond v1.2.10): ```` $ grep -ri "AIF2 ADC Stereo Capture" . $ ```` shows that `AIF2 ADC Stereo Capture Route` is not set for any device at all (PinePhone or other). HARDWARE ARGUMENT: For the PinePhone (PP), Megi recommends in https://xnux.eu/devices/feature/audio-pp.html that `AIF2 ADC Stereo Capture Route` is set to `Mix Mono`. TESTING ARGUMENT: Experimentation with different stored alsa settings (from ```` sudo alsactl store -f - > YYYYMMDD_example_alsa_settings ```` ) showed that settings files whose only substantial difference was in the setting of `AIF2 ADC Stereo Capture Route` had the effect that audio from the PP was heard on the remote phone when the value was `Mix Mono` and was not heard with the value `Stereo`. Testing was done with the equivalent of https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… with the settings file https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… that has the `Mix Mono` value selected. RECOMMENDATIONS: * `AIF2 ADC Stereo Capture Route` should be initialised or set to *something* in one of the .conf files (in the PP case, most likely in `VoiceCall.conf`) to prevent users who experiment and have difficulty remembering which of the 50 PP alsa parameters should be set in which way from accidentally sabotaging their otherwise working audio calls; * Based on Megi's 2020 page and my testing, it seems that the correct value should be `Mix Mono`. COMMENT: While setting all 50 of the alsa settings ```` $ sudo alsactl store -f - |grep "name " |wc 50 258 1788 ```` in the .conf files is most likely unnecessary, there could be others that are necessary too. Ideally, someone who really understands how the PP audio system works (including Megi's two diagrams at https://xnux.eu/devices/feature/audio-pp.html) should be able to make these recommendations. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/351 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Bug: AIF2 ADC Stereo Capture Route is unitialised in alsa-ucm-conf
by GitHub issues - opened 12 Sep '23

12 Sep '23

alsa-project/alsa-ucm-conf issue #351 was opened from broukema: DESCRIPTION: `AIF2 ADC Stereo Capture Route` is not initialised in `alsa-ucm-conf`. This implies that if its setting is significant, any accidental change by the user (using any of the various `alsa` tools) to a wrong setting will not be corrected by any of the three `PinePhone/*.conf` files, leading, in particular, to audio errors during a phone call. This was discovered by testing variations on the Mobian version of the PinePhone alsa-ucm-conf files. CODING ARGUMENT: All parameters that can significantly affect the audio system should be either initialised or overridden in configuration files that the user expects to necessarily work, such as those for phone calls. In `alsa-ucm-conf` commit b68aa52 (one beyond v1.2.10): ```` $ grep -ri "AIF2 ADC Stereo Capture" . $ ```` shows that `AIF2 ADC Stereo Capture Route` is not set for any device at all (PinePhone or other). HARDWARE ARGUMENT: For the PinePhone (PP), Megi recommends in https://xnux.eu/devices/feature/audio-pp.html that `AIF2 ADC Stereo Capture Route` is set to `Mix Mono`. TESTING ARGUMENT: Experimentation with different stored alsa settings (from ```` sudo alsactl store -f - > YYYYMMDD_example_alsa_settings ```` ) showed that settings files whose only substantial difference was in the setting of `AIF2 ADC Stereo Capture Route` had the effect that audio from the PP was heard on the remote phone when the value was `Mix Mono` and was not heard with the value `Stereo`. Testing was done with the equivalent of https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… with the settings file https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… that has the `Mix Mono` value selected. RECOMMENDATIONS: * `AIF2 ADC Stereo Capture Route` should be initialised or set to *something* in one of the .conf files (in the PP case, most likely in `VoiceCall.conf`) to prevent users who experiment and have difficulty remembering which of the 50 PP alsa parameters should be set in which way from accidentally sabotaging their otherwise working audio calls; * Based on Megi's 2020 page and my testing, it seems that the correct value should be `Mix Mono`. COMMENT: While setting all 50 of the alsa settings ```` $ sudo alsactl store -f - |grep "name " |wc 50 258 1788 ```` in the .conf files is most likely unnecessary, there could be others that are necessary too. Ideally, someone who really understands how the PP audio system works (including Megi's two diagrams at https://xnux.eu/devices/feature/audio-pp.html) should be able to make these recommendations. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/351 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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