September 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v6 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 31 Oct '22

31 Oct '22

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V5: - Fix build issues reported by Intel Test Bot - Check devm_kasprintf return - Add ACPI dependency to HDA drivers V5: https://lkml.org/lkml/2021/12/16/430 Changes from V4: - Save index received from ACPI for latter use in DSP Support - Move CS35L41_DSP1_CCM_CORE_CTRL to regmap_write so doesn't affect DSP hibernation feature V4: https://lkml.org/lkml/2021/12/14/487 Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 29 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 527 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 146 ++++ sound/soc/codecs/Kconfig | 11 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 355 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2055 insertions(+), 1096 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

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[PATCH 00/12] ASoC: mediatek: Add support for MT8188 SoC
by Trevor Wu 29 Oct '22

29 Oct '22

This series of patches adds support for Mediatek AFE of MT8188 SoC. Patches are based on broonie tree "for-next" branch. Trevor Wu (12): ASoC: mediatek: common: add SMC ops ID ASoC: mediatek: mt8188: add common header ASoC: mediatek: mt8188: support audsys clock ASoC: mediatek: mt8188: support adda in platform driver ASoC: mediatek: mt8188: support etdm in platform driver ASoC: mediatek: mt8188: support pcmif in platform driver ASoC: mediatek: mt8188: support audio clock control ASoC: mediatek: mt8188: add platform driver ASoC: mediatek: mt8188: add control for timing select dt-bindings: mediatek: mt8188: add audio afe document ASoC: mediatek: mt8188: add machine driver with mt6359 dt-bindings: mediatek: mt8188: add mt8188-mt6359 document .../bindings/sound/mt8188-afe-pcm.yaml | 202 + .../bindings/sound/mt8188-mt6359.yaml | 70 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-base-afe.h | 16 + sound/soc/mediatek/mt8188/Makefile | 15 + sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 656 ++++ sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 114 + sound/soc/mediatek/mt8188/mt8188-afe-common.h | 151 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 3451 +++++++++++++++++ sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 206 + sound/soc/mediatek/mt8188/mt8188-audsys-clk.h | 15 + .../soc/mediatek/mt8188/mt8188-audsys-clkid.h | 83 + sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 639 +++ sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 2598 +++++++++++++ sound/soc/mediatek/mt8188/mt8188-dai-pcm.c | 362 ++ sound/soc/mediatek/mt8188/mt8188-mt6359.c | 810 ++++ sound/soc/mediatek/mt8188/mt8188-reg.h | 3300 ++++++++++++++++ 18 files changed, 12712 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8188-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8188-mt6359.yaml create mode 100644 sound/soc/mediatek/mt8188/Makefile create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-clk.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-clk.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-common.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-adda.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-mt6359.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-reg.h -- 2.18.0

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[PATCH 0/9] ASoC: qdsp6: audioreach: add multi-port, SAL and MFC support
by Srinivas Kandagatla 28 Oct '22

28 Oct '22

This patchset adds support to multi-port connections between AudioReach Modules which is required for sophisticated graphs like ECNS or Speaker Protection. Also as part of ECNS testing new module support for SAL and MFC are added. Tested on SM8450 with ECNS. Thanks, Srini Srinivas Kandagatla (9): ASoC: qdsp6: audioreach: topology use idr_alloc_u32 ASoC: qdsp6: audioreach: remove unused connection_list ASoC: qdsp6: audioreach: update dapm kcontrol private data ASoC: qdsp6: audioreach: Simplify handing FE and BE graph connections ASoC: qdsp6: audioreach: simplify module_list sz calculation ASoC: qdsp6: audioreach: add support for more port connections ASoC: qdsp6: audioreach: add support to enable SAL Module ASoC: qdsp6: audioreach: add support for MFC Module ASoC: qdsp6: audioreach: add support to enable module command include/uapi/sound/snd_ar_tokens.h | 27 +++ sound/soc/qcom/qdsp6/audioreach.c | 298 ++++++++++++++++++++--------- sound/soc/qcom/qdsp6/audioreach.h | 47 +++-- sound/soc/qcom/qdsp6/q6apm.c | 84 +------- sound/soc/qcom/qdsp6/q6apm.h | 6 +- sound/soc/qcom/qdsp6/topology.c | 242 +++++++++++++++++++---- 6 files changed, 480 insertions(+), 224 deletions(-) -- 2.21.0

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[PATCH v2] ASoC: dt-bindings: Document audio OF graph dai-tdm-slot-num dai-tdm-slot-width props
by Marek Vasut 27 Oct '22

27 Oct '22

Document dai-tdm-slot-num and dai-tdm-slot-width props as those are parsed by simple graph card and may therefore appear in audio OF graph node. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: devicetree(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org To: linux-arm-kernel(a)lists.infradead.org --- V2: Drop the definition refs and just fill in type and description --- .../devicetree/bindings/sound/audio-graph-port.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index bc46a95ed8400..64654ceef2089 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -74,6 +74,12 @@ patternProperties: convert-sample-format: $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + dai-tdm-slot-num: + description: Number of slots in use. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: Width in bits for each slot. + $ref: /schemas/types.yaml#/definitions/uint32 dai-tdm-slot-width-map: description: Mapping of sample widths to slot widths. For hardware that cannot support a fixed slot width or a slot width always -- 2.35.1

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[PATCH v4 00/11] ASoC: cleanups and improvements for jz4740-i2s
by Aidan MacDonald 25 Oct '22

25 Oct '22

This series is a preparatory cleanup of the jz4740-i2s driver before adding support for a new SoC. The two improvements are lifting unnecessary restrictions on sample rates and formats -- the existing ones appear to be derived from the limitations of the JZ4740's internal codec and don't reflect the actual capabilities of the I2S controller. I'm unable to test the series on any JZ47xx SoCs, but I have tested on an X1000 (which is the SoC I'll be adding in a followup series). Changes in v2: * Drop two patches already in sound for-next. * Squash two removal patches into the regmap fields patch. * Remove the unused 'mem' resource in the driver private struct. * Use regmap_set_bits() and regmap_clear_bits() to improve readability. * Add fix for SoCs with independent FIFO flush bits (ie. most of them). * Update sample formats patch with a more informative commit message. * Add two new patches to refactor DAI/component probing. Changes in v3: * Fix missing 'ret' in patch 11 (yes, that was pretty silly of me) Changes in v4: * Refactor FIFO flush bits fix so it doesn't depend on regmap conversion. Aidan MacDonald (11): ASoC: jz4740-i2s: Handle independent FIFO flush bits ASoC: jz4740-i2s: Remove unused 'mem' resource ASoC: jz4740-i2s: Convert to regmap API ASoC: jz4740-i2s: Simplify using regmap fields ASoC: jz4740-i2s: Use FIELD_PREP() macros in hw_params callback ASoC: jz4740-i2s: Align macro values and sort includes ASoC: jz4740-i2s: Make the PLL clock name SoC-specific ASoC: jz4740-i2s: Support S20_LE and S24_LE sample formats ASoC: jz4740-i2s: Support continuous sample rate ASoC: jz4740-i2s: Move component functions near the component driver ASoC: jz4740-i2s: Refactor DAI probe/remove ops as component ops sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 461 ++++++++++++++++++---------------- 2 files changed, 248 insertions(+), 214 deletions(-) -- 2.35.1

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[PATCH v2] ASoC: core: clarify the driver name initialization
by Jaroslav Kysela 24 Oct '22

24 Oct '22

The driver field in the struct snd_ctl_card_info is a valid user space identifier. Actually, many ASoC drivers do not care and let to initialize this field using a standard wrapping method. Unfortunately, in this way, this field becomes unusable and unreadable for the drivers with longer card names. Also, there is a possibility to have clashes (driver field has only limit of 15 characters). This change will print an error when the wrapping is used. The developers of the affected drivers should fix the problem. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> v1..v2: - remove the wrong DMI condition per Mark's review --- sound/soc/soc-core.c | 37 +++++++++++++++++++++++-------------- 1 file changed, 23 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e824ff1a9fc0..590143ebf6df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1840,21 +1840,22 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) } } -#define soc_setup_card_name(name, name1, name2, norm) \ - __soc_setup_card_name(name, sizeof(name), name1, name2, norm) -static void __soc_setup_card_name(char *name, int len, - const char *name1, const char *name2, - int normalization) +#define soc_setup_card_name(card, name, name1, name2) \ + __soc_setup_card_name(card, name, sizeof(name), name1, name2) +static void __soc_setup_card_name(struct snd_soc_card *card, + char *name, int len, + const char *name1, const char *name2) { + const char *src = name1 ? name1 : name2; int i; - snprintf(name, len, "%s", name1 ? name1 : name2); + snprintf(name, len, "%s", src); - if (!normalization) + if (name != card->snd_card->driver) return; /* - * Name normalization + * Name normalization (driver field) * * The driver name is somewhat special, as it's used as a key for * searches in the user-space. @@ -1874,6 +1875,14 @@ static void __soc_setup_card_name(char *name, int len, break; } } + + /* + * The driver field should contain a valid string from the user view. + * The wrapping usually does not work so well here. Set a smaller string + * in the specific ASoC driver. + */ + if (strlen(src) > len - 1) + dev_err(card->dev, "ASoC: driver name too long '%s' -> '%s'\n", src, name); } static void soc_cleanup_card_resources(struct snd_soc_card *card) @@ -2041,12 +2050,12 @@ static int snd_soc_bind_card(struct snd_soc_card *card) /* try to set some sane longname if DMI is available */ snd_soc_set_dmi_name(card, NULL); - soc_setup_card_name(card->snd_card->shortname, - card->name, NULL, 0); - soc_setup_card_name(card->snd_card->longname, - card->long_name, card->name, 0); - soc_setup_card_name(card->snd_card->driver, - card->driver_name, card->name, 1); + soc_setup_card_name(card, card->snd_card->shortname, + card->name, NULL); + soc_setup_card_name(card, card->snd_card->longname, + card->long_name, card->name); + soc_setup_card_name(card, card->snd_card->driver, + card->driver_name, card->name); if (card->components) { /* the current implementation of snd_component_add() accepts */ -- 2.35.3

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[PATCH] ASoC: amd: acp: Add setbias level for rt5682s codec in machine driver
by Venkata Prasad Potturu 19 Oct '22

19 Oct '22

Add set_bais_level function for rt5682s codec to enable bclk and lrclk before codec widgets power on and disable bclk and lrclk after widgets power down, to avoid pop noise Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/acp-mach-common.c | 62 ++++++++++++++++++++++++++--- 1 file changed, 56 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 4c69cb6e3400..a78cf29387a7 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -167,11 +167,14 @@ static int acp_card_hs_startup(struct snd_pcm_substream *substream) &constraints_channels); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - if (!drvdata->soc_mclk) { - ret = acp_clk_enable(drvdata); - if (ret < 0) { - dev_err(rtd->card->dev, "Failed to enable HS clk: %d\n", ret); - return ret; + + if (strcmp(codec_dai->name, "rt5682s-aif1") && strcmp(codec_dai->name, "rt5682s-aif2")) { + if (!drvdata->soc_mclk) { + ret = acp_clk_enable(drvdata); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to enable HS clk: %d\n", ret); + return ret; + } } } @@ -280,7 +283,6 @@ static int acp_card_rt5682s_init(struct snd_soc_pcm_runtime *rtd) static const struct snd_soc_ops acp_card_rt5682s_ops = { .startup = acp_card_hs_startup, - .shutdown = acp_card_shutdown, }; static const unsigned int dmic_channels[] = { @@ -570,6 +572,52 @@ SND_SOC_DAILINK_DEF(sof_dmic, SND_SOC_DAILINK_DEF(pdm_dmic, DAILINK_COMP_ARRAY(COMP_CPU("acp-pdm-dmic"))); +static int acp_rtk_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct acp_card_drvdata *drvdata = card->drvdata; + int ret = 0; + + if (!component) + return 0; + + if (strncmp(component->name, "i2c-RTL5682", 11) && + strncmp(component->name, "i2c-10EC1019", 12)) + return 0; + + /* + * For Realtek's codec and amplifier components, + * the lrck and bclk must be enabled brfore their all dapms be powered on, + * and must be disabled after their all dapms be powered down + * to avoid any pop. + */ + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (snd_soc_dapm_get_bias_level(dapm) == SND_SOC_BIAS_OFF) { + clk_set_rate(drvdata->wclk, 48000); + clk_set_rate(drvdata->bclk, 48000 * 64); + + /* Increase bclk's enable_count */ + ret = clk_prepare_enable(drvdata->bclk); + if (ret < 0) + dev_err(component->dev, "Failed to enable bclk %d\n", ret); + } else { + /* + * Decrease bclk's enable_count. + * While the enable_count is 0, the bclk would be closed. + */ + clk_disable_unprepare(drvdata->bclk); + } + break; + default: + break; + } + + return ret; +} + int acp_sofdsp_dai_links_create(struct snd_soc_card *card) { struct snd_soc_dai_link *links; @@ -730,6 +778,7 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) card->dai_link = links; card->num_links = num_links; + card->set_bias_level = acp_rtk_set_bias_level; return 0; } @@ -907,6 +956,7 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) card->dai_link = links; card->num_links = num_links; + card->set_bias_level = acp_rtk_set_bias_level; return 0; } -- 2.25.1

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[PATCH -next] ASoC: soc-component: using pm_runtime_resume_and_get instead of pm_runtime_get_sync
by Zhang Qilong 19 Oct '22

19 Oct '22

Using the newest pm_runtime_resume_and_get is more appropriate for simplifing code here. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/soc-component.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index e12f8244242b..659b9ade4158 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -1213,11 +1213,9 @@ int snd_soc_pcm_component_pm_runtime_get(struct snd_soc_pcm_runtime *rtd, int i; for_each_rtd_components(rtd, i, component) { - int ret = pm_runtime_get_sync(component->dev); - if (ret < 0 && ret != -EACCES) { - pm_runtime_put_noidle(component->dev); + int ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0 && ret != -EACCES) return soc_component_ret(component, ret); - } /* mark stream if succeeded */ soc_component_mark_push(component, stream, pm); } -- 2.25.1

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[PATCH 0/2] ASoC: Intel: avs: Add support for max98927 codec
by Amadeusz Sławiński 18 Oct '22

18 Oct '22

This series adds machine board for max98927 codec present on some Chromebook devices. Amadeusz Sławiński (2): ASoC: Intel: avs: Add max98927 machine board ASoC: Intel: avs: Load max98927 on target platform sound/soc/intel/avs/board_selection.c | 8 + sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/max98927.c | 236 ++++++++++++++++++++++++++ 4 files changed, 256 insertions(+) create mode 100644 sound/soc/intel/avs/boards/max98927.c -- 2.25.1

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[PATCH -next 0/4] ASoC: Use DIV_ROUND_UP() instead of open-coding it
by Shang XiaoJing 18 Oct '22

18 Oct '22

Use DIV_ROUND_UP() instead of open-coding it, which intents and makes it more clear what is going on for the casual reviewer. Shang XiaoJing (4): ASoC: cs35l36: Use DIV_ROUND_UP() instead of open-coding it ASoC: wm8978: Use DIV_ROUND_UP() instead of open-coding it ASoC: rsnd: Use DIV_ROUND_UP() instead of open-coding it ASoC: ti: davinci-mcasp: Use DIV_ROUND_UP() instead of open-coding it sound/soc/codecs/cs35l36.c | 4 ++-- sound/soc/codecs/wm8978.c | 2 +- sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/ti/davinci-mcasp.c | 7 +++---- 4 files changed, 8 insertions(+), 9 deletions(-) -- 2.17.1

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