September 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 00/13] Extend AHUB audio support for Tegra210 and later
by Sameer Pujar 04 Oct '21

04 Oct '21

Earlier as part of series [0], support for ADMAIF and I/O modules (such as I2S, DMIC and DSPK) was added. This series aims at exposing some of the AHUB internal modules (listed below), which can be used for audio pre or post processing. * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer These modules can be plugged into audio paths and relevant processing can be done. The MUX routes are extended to allow add or remove above modules in the path via mixer controls. This is similar to how specific ADMAIF channels are connected to relevant I/O module instances at the moment. Some of these modules can alter PCM parameters. Consider example of resampler (44.1 -> 48 kHz) in the path. aplay(44.1 kHz) -> ADMAIF -> SFC -> (48 kHz) I2S -> (48kHz) Codec The modules following SFC should be using converted sample rate and DAIs need to be configured accordingly. The audio-graph driver provides a mechanism to fixup the new parameters which can be specified in DT for a given DAI. Then core uses these new values via fixup callback and then pass it to respective DAIs hw_param() callback. The "convert-rate", described in [1], property can be used when there is rate conversion in the audio path. Similarly "convert-channels" can be used when there is channel conversion in the path. There is no "convert-xxx" property for sample size conversions. It can be added if necessary. [0] https://www.lkml.org/lkml/2020/7/21/1357 [1] Documentation/devicetree/bindings/sound/audio-graph-port.yaml Sameer Pujar (13): ASoC: soc-pcm: Don't reconnect an already active BE ASoC: simple-card-utils: Increase maximum DAI links limit to 512 ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules ASoC: tegra: Add routes for few AHUB modules ASoC: tegra: Add Tegra210 based MVC driver ASoC: tegra: Add Tegra210 based SFC driver ASoC: tegra: Add Tegra210 based AMX driver ASoC: tegra: Add Tegra210 based ADX driver ASoC: tegra: Add Tegra210 based Mixer driver arm64: defconfig: Enable few Tegra210 based AHUB drivers arm64: tegra: Add few AHUB devices for Tegra210 and later arm64: tegra: Extend APE audio support on Jetson platforms .../bindings/sound/nvidia,tegra210-adx.yaml | 74 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 20 + .../bindings/sound/nvidia,tegra210-amx.yaml | 72 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 67 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 79 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 76 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 1554 ++++++++- arch/arm64/boot/dts/nvidia/tegra186.dtsi | 120 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 1493 ++++++++- .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 1520 ++++++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 116 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 77 + arch/arm64/configs/defconfig | 5 + include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/soc-pcm.c | 4 + sound/soc/tegra/Kconfig | 48 + sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra210_adx.c | 527 +++ sound/soc/tegra/tegra210_adx.h | 72 + sound/soc/tegra/tegra210_ahub.c | 511 ++- sound/soc/tegra/tegra210_amx.c | 595 ++++ sound/soc/tegra/tegra210_amx.h | 93 + sound/soc/tegra/tegra210_mixer.c | 667 ++++ sound/soc/tegra/tegra210_mixer.h | 100 + sound/soc/tegra/tegra210_mvc.c | 629 ++++ sound/soc/tegra/tegra210_mvc.h | 117 + sound/soc/tegra/tegra210_sfc.c | 3542 ++++++++++++++++++++ sound/soc/tegra/tegra210_sfc.h | 78 + 31 files changed, 13601 insertions(+), 423 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml create mode 100644 sound/soc/tegra/tegra210_adx.c create mode 100644 sound/soc/tegra/tegra210_adx.h create mode 100644 sound/soc/tegra/tegra210_amx.c create mode 100644 sound/soc/tegra/tegra210_amx.h create mode 100644 sound/soc/tegra/tegra210_mixer.c create mode 100644 sound/soc/tegra/tegra210_mixer.h create mode 100644 sound/soc/tegra/tegra210_mvc.c create mode 100644 sound/soc/tegra/tegra210_mvc.h create mode 100644 sound/soc/tegra/tegra210_sfc.c create mode 100644 sound/soc/tegra/tegra210_sfc.h -- 2.7.4

5 33

[PATCH] ASoC: fsl_spdif: implement bypass mode from in to out
by Shengjiu Wang 02 Oct '21

02 Oct '21

From: Viorel Suman <viorel.suman(a)nxp.com> Implement SPDIF bypass mode. It implies internal SoC routing of SPDIF input signal to SPDIF output signal. The test bed requires two boards: B1 configured in bypass mode, and B2 to feed B1 SPDIF RX port and read B1 SPDIF TX port: B2 TX -> B1 RX, B2 RX <- B1 TX. The test procedure: a) Boot both boards b) B2: start "arecord <spdifcard> -r 48kHz | aplay <local DAC>" c) B2: start "aplay <spdifcard> -r 48kHz <2ch 48kHz audio file>" d) B1: enable bypass mode: amixer -cimxspdif cset numid=8,iface=PCM,name='Bypass Mode' on e) B2: check DAC audio, make sure the same sample rate is used at steps b) and c), in example above the rate is 48kHz. f) B1: try to run "aplay" or "arecord" on imxspdif card while in bypass mode - both must fail until bypass mode is disabled g) B1: disable bypass mode: amixer -cimxspdif cset numid=8,iface=PCM,name='Bypass Mode' off h) B1: check the usual playback and capture on imxspdif card. During this test try to set bypass mode - must not be allowed while playback or capture is running. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 74 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 74 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 928b59069283..d178b479c8bd 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -111,6 +111,7 @@ struct spdif_mixer_control { * @dma_params_tx: DMA parameters for transmit channel * @dma_params_rx: DMA parameters for receive channel * @regcache_srpc: regcache for SRPC + * @bypass: status of bypass input to output */ struct fsl_spdif_priv { const struct fsl_spdif_soc_data *soc; @@ -133,6 +134,7 @@ struct fsl_spdif_priv { struct snd_dmaengine_dai_dma_data dma_params_rx; /* regcache for SRPC */ u32 regcache_srpc; + bool bypass; }; static struct fsl_spdif_soc_data fsl_spdif_vf610 = { @@ -905,6 +907,69 @@ static int fsl_spdif_rx_rcm_put(struct snd_kcontrol *kcontrol, return 0; } +static int fsl_spdif_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *priv = snd_soc_dai_get_drvdata(dai); + + ucontrol->value.integer.value[0] = priv->bypass ? 1 : 0; + + return 0; +} + +static int fsl_spdif_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *priv = snd_soc_dai_get_drvdata(dai); + struct snd_soc_card *card = dai->component->card; + bool set = (ucontrol->value.integer.value[0] != 0); + struct regmap *regmap = priv->regmap; + struct snd_soc_pcm_runtime *rtd; + u32 scr, mask; + int stream; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link); + + if (priv->bypass == set) + return 0; /* nothing to do */ + + if (snd_soc_dai_active(dai)) { + dev_err(dai->dev, "Cannot change BYPASS mode while stream is running.\n"); + return -EBUSY; + } + + pm_runtime_get_sync(dai->dev); + + if (set) { + /* Disable interrupts */ + regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0); + + /* Configure BYPASS mode */ + scr = SCR_TXSEL_RX | SCR_RXFIFO_OFF; + mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK | + SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK | SCR_TXSEL_MASK; + /* Power up SPDIF module */ + mask |= SCR_LOW_POWER; + } else { + /* Power down SPDIF module, disable TX */ + scr = SCR_LOW_POWER | SCR_TXSEL_OFF; + mask = SCR_LOW_POWER | SCR_TXSEL_MASK; + } + + regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); + + /* Disable playback & capture if BYPASS mode is enabled, enable otherwise */ + for_each_pcm_streams(stream) + rtd->pcm->streams[stream].substream_count = (set ? 0 : 1); + + priv->bypass = set; + pm_runtime_put_sync(dai->dev); + + return 0; +} + /* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1075,6 +1140,15 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .info = fsl_spdif_rxrate_info, .get = fsl_spdif_rxrate_get, }, + /* RX bypass controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "Bypass Mode", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = fsl_spdif_bypass_get, + .put = fsl_spdif_bypass_put, + }, /* User bit sync mode set/get controller */ { .iface = SNDRV_CTL_ELEM_IFACE_PCM, -- 2.17.1

2 1

[PATCH] ASoC: soc-component: Remove conditional definition of debugfs data members
by Simon Trimmer 02 Oct '21

02 Oct '21

This simplification allows the use of the standard kernel pattern of static inline dummy functions for debugfs code. Most systems will only have a small number of snd_soc_components so the memory impact is minimal. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Suggested-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-component.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index e09a2d108e8c..3a35d149e92f 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -227,10 +227,8 @@ struct snd_soc_component { struct snd_compr_stream *mark_compr_open; void *mark_pm; -#ifdef CONFIG_DEBUG_FS struct dentry *debugfs_root; const char *debugfs_prefix; -#endif }; #define for_each_component_dais(component, dai)\ -- 2.33.0

2 1

[PATCH v3 00/12] Add support for on demand pipeline setup/destroy
by Peter Ujfalusi 02 Oct '21

02 Oct '21

Hi, The previous, v2 of this series was sent by Daniel Baluta: https://lore.kernel.org/alsa-devel/20210917143659.401102-1-daniel.baluta@os… We have agreed that it might be better that someone from Intel is going to take it from here as we already have the infrastructure up to test and verify the dynamic pipelines support. Changes since v2 (sent by Daniel Baluta): - patch 10: Fix NULL point dereference in hda_dai_update_config() - I have kept Daniel's SoB for the series. Changes since v1: - Signed-off-by tag added by Daniel This series implements initial support for dynamic pipelines to setup/teardown pipeline as needed when a PCM is open/closed. Initially dynamic pipelines are only supported with single core setup which will be expanded with a follow-up series. Review with SOF community at https://github.com/thesofproject/linux/pull/2794 The feature has been merged on 1st of April to sof-dev, all issues found since has been fixed and squashed to this upstream series. Regards, Peter --- Ranjani Sridharan (12): ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/hda-dai.c | 174 +++--- sound/soc/sof/intel/hda.c | 177 ++++-- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/ipc.c | 22 + sound/soc/sof/pcm.c | 58 +- sound/soc/sof/pm.c | 4 +- sound/soc/sof/sof-audio.c | 709 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 32 +- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 362 +++++-------- 17 files changed, 1032 insertions(+), 538 deletions(-) -- 2.33.0

2 13

[PATCH] ASoC: SOF: Intel: hda-stream: limit PROCEN workaround
by Peter Ujfalusi 01 Oct '21

01 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The work-around enabled in hda-stream.c is only required on earlier versions of SOCs/PCH (Skylake, KabyLake, ApolloLake, GeminiLake). Before setting the format on the host DMA, it is required to couple the host and link DMA - which as a consequence shall use the same format. This patch introduces a quirk field in the platform descriptor and makes the work-around conditional. Newer platforms have no limitations on the use of host and link DMA, which can use different formats. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/hda-stream.c | 18 ++++++++++++------ sound/soc/sof/intel/shim.h | 4 ++++ 3 files changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index e6a1f6532547..16b7d8b1adff 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -143,5 +143,6 @@ const struct sof_intel_dsp_desc apl_chip_info = { .rom_init_timeout = 150, .ssp_count = APL_SSP_COUNT, .ssp_base_offset = APL_SSP_BASE_OFFSET, + .quirks = SOF_INTEL_PROCEN_FMT_QUIRK, }; EXPORT_SYMBOL_NS(apl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 63c367478f1c..47a465b06da5 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -405,6 +405,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct snd_pcm_hw_params *params) { + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); struct hdac_bus *bus = sof_to_bus(sdev); struct hdac_stream *hstream = &stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); @@ -538,6 +539,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, /* * Recommended hardware programming sequence for HDAudio DMA format + * on earlier platforms - this is not needed on newer platforms * * 1. Put DMA into coupled mode by clearing PPCTL.PROCEN bit * for corresponding stream index before the time of writing @@ -547,9 +549,11 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, * enable decoupled mode */ - /* couple host and link DMA, disable DSP features */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - mask, 0); + if (chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK) { + /* couple host and link DMA, disable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, 0); + } /* program stream format */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -557,9 +561,11 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, SOF_HDA_ADSP_REG_CL_SD_FORMAT, 0xffff, hstream->format_val); - /* decouple host and link DMA, enable DSP features */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - mask, mask); + if (chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK) { + /* decouple host and link DMA, enable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, mask); + } /* program last valid index */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index e9f7d4d7fcce..8e2a0f4c0f66 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -151,6 +151,9 @@ #define PCI_PMCS 0x84 #define PCI_PMCS_PS_MASK 0x3 +/* Intel quirks */ +#define SOF_INTEL_PROCEN_FMT_QUIRK BIT(0) + /* DSP hardware descriptor */ struct sof_intel_dsp_desc { int cores_num; @@ -166,6 +169,7 @@ struct sof_intel_dsp_desc { int ssp_base_offset; /* base address of the SSPs */ u32 sdw_shim_base; u32 sdw_alh_base; + u32 quirks; bool (*check_sdw_irq)(struct snd_sof_dev *sdev); }; -- 2.33.0

3 4

ALSA kernel projects - for academic purposes
by Muni Sekhar 01 Oct '21

01 Oct '21

Hi all, What small projects would you suggest to a novice with the ALSA kernel. The aim is to develop a familiarity with the ALSA kernel source code, and also to submit it for academic purposes. -- Thanks, Sekhar

7 15

[PATCH] ALSA: usb-audio: disable implicit feedback sync for Behringer UFX1204 and UFX1604
by Geraldo Nascimento 01 Oct '21

01 Oct '21

Hi Takashi, Hopefully you will be able to apply this cleanly to branch master of your sound.git. Thanks! Geraldo Nascimento --- Behringer UFX1204 and UFX1604 have Synchronous endpoints to which current ALSA code applies implicit feedback sync as if they were Asynchronous endpoints. This breaks UAC compliance and is unneeded. The commit 5e35dc0338d85ccebacf3f77eca1e5dea73155e8 and subsequent 1a15718b41df026cffd0e42cfdc38a1384ce19f9 were meant to clear up noise. Unfortunately, noise persisted for those using higher sample rates and this was only solved by commit d2e8f641257d0d3af6e45d6ac2d6f9d56b8ea964 Since there are no more reports of noise, let's get rid of the implicit-fb quirks breaking UAC compliance. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 23767a14d126..70319c822c10 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -54,8 +54,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Fixed EP */ /* FIXME: check the availability of generic matching */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0001, 0x81, 1), /* Behringer UFX1604 */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0002, 0x81, 1), /* Behringer UFX1204 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0001, 0x81, 2), /* Solid State Logic SSL2 */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */

2 1

[PATCH v2] soundwire: debugfs: use controller id and link_id for debugfs
by Srinivas Kandagatla 01 Oct '21

01 Oct '21

link_id can be zero and if we have multiple controller instances in a system like Qualcomm debugfs will end-up with duplicate namespace resulting in incorrect debugfs entries. Using bus-id and link-id combination should give a unique debugfs directory entry and should fix below warning too. "debugfs: Directory 'master-0' with parent 'soundwire' already present!" Fixes: bf03473d5bcc ("soundwire: add debugfs support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: Added Link ID along with bus id. drivers/soundwire/debugfs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/debugfs.c b/drivers/soundwire/debugfs.c index b6cad0d59b7b..49900cd207bc 100644 --- a/drivers/soundwire/debugfs.c +++ b/drivers/soundwire/debugfs.c @@ -19,7 +19,7 @@ void sdw_bus_debugfs_init(struct sdw_bus *bus) return; /* create the debugfs master-N */ - snprintf(name, sizeof(name), "master-%d", bus->link_id); + snprintf(name, sizeof(name), "master-%d-%d", bus->id, bus->link_id); bus->debugfs = debugfs_create_dir(name, sdw_debugfs_root); } -- 2.21.0

3 2

[PATCH] ASoC: max98373: Mark cache dirty before entering sleep
by Ryan Lee 01 Oct '21

01 Oct '21

Amp lose its register values in case amp power loss or 'ForceReset' over Soundwire SCP_ctrl register(0x0044) or HW_RESET pin control during the audio suspend and resume. Mark cache dirty before audio suspension to restore existing values when audio resume. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98373-sdw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index dc520effc61c..a7e4a6e880b0 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -259,6 +259,7 @@ static __maybe_unused int max98373_suspend(struct device *dev) regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); + regcache_mark_dirty(max98373->regmap); return 0; } -- 2.17.1

3 16

[PATCH regression fix] ASoC: nau8824: Fix headphone vs headset, button-press detection no longer working
by Hans de Goede 30 Sep '21

30 Sep '21

Commit 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling") replaced the nau8824_dapm_enable_pin() helper with direct calls to snd_soc_dapm_enable_pin(), but the helper was using snd_soc_dapm_force_enable_pin() and not forcing the MICBIAS + SAR supplies on breaks headphone vs headset and button-press detection. Replace the snd_soc_dapm_enable_pin() calls with snd_soc_dapm_force_enable_pin() to fix this. Cc: stable(a)vger.kernel.org Fixes: 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/nau8824.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cea1d43e0cb8..77f647529354 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -874,8 +874,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_enable_pin(dapm, "SAR"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); -- 2.31.1

2 1