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

[PATCH v2 0/4] Add missing dt-binding properties to rt5682(s)
by Nícolas F. R. A. Prado 29 Oct '22

29 Oct '22

This series adds the missing sound-dai-cells to rt5682s and supplies for both rt5682s and rt5682. These properties are already used by sc7180-trogdor.dtsi (and derived DTs like sc7180-trogdor-kingoftown.dtsi), so the commits in this series are really just documenting their usage. v1: https://lore.kernel.org/all/20221021190908.1502026-1-nfraprado@collabora.co… Changes in v2: - Changed sound-dai-cells to 1 on rt5682s - Added commit fixing sound-dai-cells on rt5682 - Added mention to commit messages that properties are already being used by DTs Nícolas F. R. A. Prado (4): ASoC: dt-bindings: realtek,rt5682s: Add #sound-dai-cells ASoC: dt-bindings: realtek,rt5682s: Add AVDD and MICVDD supplies ASoC: dt-bindings: rt5682: Set sound-dai-cells to 1 ASoC: dt-bindings: rt5682: Add AVDD, MICVDD and VBAT supplies .../devicetree/bindings/sound/realtek,rt5682s.yaml | 7 +++++++ Documentation/devicetree/bindings/sound/rt5682.txt | 8 +++++++- 2 files changed, 14 insertions(+), 1 deletion(-) -- 2.38.1

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[PATCH v3 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 Changes since v2: Fixed two spelling mistakes in commit message No code changes. 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 | 310 ++++++++++++++++++++--------- 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 | 243 +++++++++++++++++++--- 6 files changed, 489 insertions(+), 228 deletions(-) -- 2.21.0

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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 00/21] ASoC: SOF: Intel: HDA: refactor codec and multi-link suport
by Pierre-Louis Bossart 28 Oct '22

28 Oct '22

Existing HDaudio controllers expose an HDAudio DMA which is used to interface with HDaudio codecs. All other interfaces supported by Intel (SoundWire, SSP, DMIC) rely for data transfers on another GP-DMA managed by the DSP firmware - the HDaudio DMA is only used for memory-to-DSP transfers. New HDaudio extensions will enable the use of this HDaudio DMA for all of SoundWire, SSP, DMIC. These extensions will be backwards-compatible for HDaudio and iDISP codecs, but will require new programming sequences and DAI callbacks for SoundWire, SSP and DMIC. Before we add support for 'extended audio links' and the programming sequences for the DMA, we need to refactor the code. All HDaudio codec support needs to be well identified in a separate file, and likewise all the 'multi-link' handling needs to be better split. This patchset removes a number of 'old' Kconfig dependencies and options, adds helpers with a fallback to remove IS_ENABLED checks in the code and tries to simplify programming sequences when possible. One indirect benefit from this refactoring is that developers can switch with a kernel parameter from HDaudio support to a variant of 'nocodec' support. This proves extremely useful to test on existing Intel RVPs and Up boards, where the same build can be used to check 3 interfaces (HDaudio, SSP, DMIC) by just removing modules, setting the kernel parameter and reloading modules. Pierre-Louis Bossart (21): ASoC: SOF: Intel: remove option to disable the common_hdmi handling ASoC: SOF: Intel: remove all dependencies on SND_SOC_HDAC_HDMI ASoC: SOF: Intel: hda-codec: simplify SND_SOC_SOF_HDA_AUDIO_CODEC handling ASoC: SOF: Intel: move codec state change to hda-codec.c ASoC: SOF: Intel: start moving multi-link handling in dedicated file ASoC: SOF: Intel: hda: add multi-link helper for LOSVID ASoC: SOF: Intel: move all RIRB/CMD_IO helpers to hda-codec.c ASoC: SOF: Intel: hda-ctrl: add codec wakeup helper ASoC: SOF: Intel: hda-codec: add hda_codec_device_remove() helper ASoC: SOF: Intel: hda-codec: add stop_cmd_io helper ASoC: SOF: Intel: hda-stream: always allocate CORB/RIRB buffer ASoC: SOF: Intel: hda-codec: add hda_codec_check_rirb_status() helper ASoC: SOF: Intel: hda-ctrl: use helper to clear RIRB status ASoC: SOF: Intel: hda-dsp: clarify dependencies on SND_SOC_SOF_HDA ASoC: SOF: Intel: hda-codec: add helpers to suspend and resume cmd_io ASoC: SOF: Intel: clarify bus_init and bus_exit sequences ASoC: SOF: Intel: hda-mlink: add helpers to suspend/resume links ASoC: SOF: Intel: add hda_bus_ml_free helper ASoC: SOF: Intel: hda: clarify Kconfig dependencies ASoC: SOF: Intel: hda-codec: use GPL-2.0-only license ASoC: SOF: introduce new DEBUG_NOCODEC mode sound/soc/sof/Kconfig | 18 ++- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/hda-bus.c | 23 ++- sound/soc/sof/intel/hda-codec.c | 252 +++++++++++++++++++++++++------ sound/soc/sof/intel/hda-ctrl.c | 63 ++------ sound/soc/sof/intel/hda-dai.c | 10 +- sound/soc/sof/intel/hda-dsp.c | 44 ++---- sound/soc/sof/intel/hda-mlink.c | 89 +++++++++++ sound/soc/sof/intel/hda-stream.c | 30 +--- sound/soc/sof/intel/hda.c | 62 ++------ sound/soc/sof/intel/hda.h | 63 ++++++-- sound/soc/sof/sof-audio.c | 5 + sound/soc/sof/sof-priv.h | 3 + 14 files changed, 438 insertions(+), 228 deletions(-) create mode 100644 sound/soc/sof/intel/hda-mlink.c -- 2.34.1

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[PATCH] ASoC: fsl_asrc fsl_esai fsl_sai: allow CONFIG_PM=N
by Maarten Zanders 28 Oct '22

28 Oct '22

When CONFIG_PM=N, pm_runtime_put_sync() returns -ENOSYS which breaks the probe function of these drivers. Other users of pm_runtime_put_sync() typically don't check the return value. In order to keep the program flow as intended, check for -ENOSYS. This commit is similar to commit 0434d3f (omap-mailbox.c). This commit fixes: cab04ab (ASoC: fsl_asrc: Don't use devm_regmap_init_mmio_clk) 203773e (ASoC: fsl_esai: Don't use devm_regmap_init_mmio_clk) 2277e7e (ASoC: fsl_sai: Don't use devm_regmap_init_mmio_clk) Signed-off-by: Maarten Zanders <maarten.zanders(a)mind.be> --- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 44dcbf49456c..08ca410ef551 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1226,7 +1226,7 @@ static int fsl_asrc_probe(struct platform_device *pdev) } ret = pm_runtime_put_sync(&pdev->dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; ret = devm_snd_soc_register_component(&pdev->dev, &fsl_asrc_component, diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index bda66b30e063..763f5f0592af 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1070,7 +1070,7 @@ static int fsl_esai_probe(struct platform_device *pdev) regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); ret = pm_runtime_put_sync(&pdev->dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; /* diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 38f6362099d5..01f3463c0e47 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1149,7 +1149,7 @@ static int fsl_sai_probe(struct platform_device *pdev) } ret = pm_runtime_put_sync(dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; /*

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[PATCH v1 0/2] Modify documentation and machine driver for mt8186_rt1019_rt5682s sound card
by Ajye Huang 28 Oct '22

28 Oct '22

v1: - Documentation: Add dmic-gpios optional prop for two DMICs case. - dmic codec driver: - "dmic-gpios" property is used for amixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: mediatek: dt-bindings: modify machine bindings for two MICs case ASoC: mediatek: mt8186-rt5682: Modify machine driver for two DMICs case .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 6 + .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 104 +++++++++++++++++- 2 files changed, 109 insertions(+), 1 deletion(-) -- 2.25.1

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[PATCH v4 0/9] ASoC: Intel: avs: PCM power management
by Cezary Rojewski 28 Oct '22

28 Oct '22

Goal of the series is implementation of suspend/resume operations for a PCM stream along with all the collaterals connected to the subject. Start with splitting avs_dai_fe_hw_free() as ideally we would like to reuse as much of existing code as possible but snd_pcm_lib_free_pages() is not desired part of the function when speaking of suspend operation. The actual implementation of suspend/resume() for component drivers follows. For most scenarios, the PM flow is similar to standard streaming one, except for the part where the position register are being saved and the lack of PCM pages freeing. To reduce code duplication, all avs_dai_suspend_XXX() and avs_dai_resume_XXX() functions reuse their non-PM equivalents. Order of operations is affected by the fact that path binding/unbinding happens only in FE part of the stream. Above essentially unlocks SX+streaming scenarios i.e.: power transitions with an ongoing stream. As some streams are allowed to run in low power state, support is provided for S0iX state. The handlers check ACPI capabilities and the number of active low-power paths before deciding between SX and S0iX flows. The last portion of the patchset is addition of power/clock gating overrides. There is no single set of registers that ensures AudioDSP firmware loads 100% of time on every single configuration. By having them exposed, user can have the loading procedure behavior adjusted for their configuration without having to recompile the kernel. Changes in v4: - replaced readb_poll_timeout() with read_poll_timeout() for the DRSM polling function as the register is u32 wide, not u8 Changes in v3: - fixed unused-but-set-variable warnings reported by the test robot Changes in v2: - moved DRSM reg polling to separate function, snd_hdac_stream_wait_drsm() as suggested by Pierre - moved the acpi_gbl_FADT under AVS_S0IX_SUPPORTED macro and relocated the checks from pcm.c to topology.c so that the low-power streams are filtered as early as possible - fixed compilation when CONFIG_ACPI is disabled as reported by the test robot - simplified all the avs_dai_resume/suspend_xxx() by moving ->ignore_suspend check one level up, to avs_component_pm_op() (patch 0002) Amadeusz Sławiński (1): ASoC: Intel: avs: Handle SUSPEND and RESUME triggers Cezary Rojewski (7): ASoC: Intel: avs: Split pcm pages freeing operation from hw_free() ASoC: Intel: avs: Introduce PCM power management routines ALSA: hda: Introduce snd_hdac_stream_wait_drsm() ASoC: Intel: avs: Restart instead of resuming HDA capture streams ASoC: Intel: avs: Count low power streams ASoC: Intel: avs: Power and clock gating policy overriding ASoC: Intel: avs: Enact power gating policy Piotr Maziarz (1): ASoC: Intel: avs: Standby power-state support include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 5 + sound/hda/hdac_stream.c | 22 ++ sound/soc/intel/avs/avs.h | 8 + sound/soc/intel/avs/core.c | 97 +++++++-- sound/soc/intel/avs/loader.c | 10 + sound/soc/intel/avs/pcm.c | 368 +++++++++++++++++++++++++++++---- sound/soc/intel/avs/topology.c | 10 + 8 files changed, 461 insertions(+), 60 deletions(-) -- 2.25.1

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[PATCH] ASoC: fsl_xcvr: Add Counter registers
by Shengjiu Wang 28 Oct '22

28 Oct '22

These counter registers are part of register list, add them to complete the register map - DMAC counter control registers - Data path Timestamp counter register - Data path bit counter register - Data path bit count timestamp register - Data path bit read timestamp register Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 40 ++++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 21 +++++++++++++++++++++ 2 files changed, 61 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index c043efe4548d..2a6802fb2a8b 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -934,6 +934,14 @@ static const struct reg_default fsl_xcvr_reg_defaults[] = { { FSL_XCVR_RX_DPTH_CTRL_SET, 0x00002C89 }, { FSL_XCVR_RX_DPTH_CTRL_CLR, 0x00002C89 }, { FSL_XCVR_RX_DPTH_CTRL_TOG, 0x00002C89 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_SET, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG, 0x00000000 }, + { FSL_XCVR_RX_DPTH_TSCR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCTR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCRR, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL_SET, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL_CLR, 0x00000000 }, @@ -944,6 +952,14 @@ static const struct reg_default fsl_xcvr_reg_defaults[] = { { FSL_XCVR_TX_CS_DATA_3, 0x00000000 }, { FSL_XCVR_TX_CS_DATA_4, 0x00000000 }, { FSL_XCVR_TX_CS_DATA_5, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_SET, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG, 0x00000000 }, + { FSL_XCVR_TX_DPTH_TSCR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCTR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCRR, 0x00000000 }, { FSL_XCVR_DEBUG_REG_0, 0x00000000 }, { FSL_XCVR_DEBUG_REG_1, 0x00000000 }, }; @@ -975,6 +991,14 @@ static bool fsl_xcvr_readable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_RX_DPTH_CTRL_SET: case FSL_XCVR_RX_DPTH_CTRL_CLR: case FSL_XCVR_RX_DPTH_CTRL_TOG: + case FSL_XCVR_RX_DPTH_CNTR_CTRL: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG: + case FSL_XCVR_RX_DPTH_TSCR: + case FSL_XCVR_RX_DPTH_BCR: + case FSL_XCVR_RX_DPTH_BCTR: + case FSL_XCVR_RX_DPTH_BCRR: case FSL_XCVR_TX_DPTH_CTRL: case FSL_XCVR_TX_DPTH_CTRL_SET: case FSL_XCVR_TX_DPTH_CTRL_CLR: @@ -985,6 +1009,14 @@ static bool fsl_xcvr_readable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_TX_CS_DATA_3: case FSL_XCVR_TX_CS_DATA_4: case FSL_XCVR_TX_CS_DATA_5: + case FSL_XCVR_TX_DPTH_CNTR_CTRL: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG: + case FSL_XCVR_TX_DPTH_TSCR: + case FSL_XCVR_TX_DPTH_BCR: + case FSL_XCVR_TX_DPTH_BCTR: + case FSL_XCVR_TX_DPTH_BCRR: case FSL_XCVR_DEBUG_REG_0: case FSL_XCVR_DEBUG_REG_1: return true; @@ -1017,6 +1049,10 @@ static bool fsl_xcvr_writeable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_RX_DPTH_CTRL_SET: case FSL_XCVR_RX_DPTH_CTRL_CLR: case FSL_XCVR_RX_DPTH_CTRL_TOG: + case FSL_XCVR_RX_DPTH_CNTR_CTRL: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG: case FSL_XCVR_TX_DPTH_CTRL_SET: case FSL_XCVR_TX_DPTH_CTRL_CLR: case FSL_XCVR_TX_DPTH_CTRL_TOG: @@ -1026,6 +1062,10 @@ static bool fsl_xcvr_writeable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_TX_CS_DATA_3: case FSL_XCVR_TX_CS_DATA_4: case FSL_XCVR_TX_CS_DATA_5: + case FSL_XCVR_TX_DPTH_CNTR_CTRL: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG: return true; default: return false; diff --git a/sound/soc/fsl/fsl_xcvr.h b/sound/soc/fsl/fsl_xcvr.h index 7f2853c60085..4769b0fca21d 100644 --- a/sound/soc/fsl/fsl_xcvr.h +++ b/sound/soc/fsl/fsl_xcvr.h @@ -49,6 +49,16 @@ #define FSL_XCVR_RX_DPTH_CTRL_CLR 0x188 #define FSL_XCVR_RX_DPTH_CTRL_TOG 0x18c +#define FSL_XCVR_RX_DPTH_CNTR_CTRL 0x1C0 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_SET 0x1C4 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR 0x1C8 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG 0x1CC + +#define FSL_XCVR_RX_DPTH_TSCR 0x1D0 +#define FSL_XCVR_RX_DPTH_BCR 0x1D4 +#define FSL_XCVR_RX_DPTH_BCTR 0x1D8 +#define FSL_XCVR_RX_DPTH_BCRR 0x1DC + #define FSL_XCVR_TX_DPTH_CTRL 0x220 /* TX datapath ctrl reg */ #define FSL_XCVR_TX_DPTH_CTRL_SET 0x224 #define FSL_XCVR_TX_DPTH_CTRL_CLR 0x228 @@ -59,6 +69,17 @@ #define FSL_XCVR_TX_CS_DATA_3 0x23C #define FSL_XCVR_TX_CS_DATA_4 0x240 #define FSL_XCVR_TX_CS_DATA_5 0x244 + +#define FSL_XCVR_TX_DPTH_CNTR_CTRL 0x260 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_SET 0x264 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR 0x268 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG 0x26C + +#define FSL_XCVR_TX_DPTH_TSCR 0x270 +#define FSL_XCVR_TX_DPTH_BCR 0x274 +#define FSL_XCVR_TX_DPTH_BCTR 0x278 +#define FSL_XCVR_TX_DPTH_BCRR 0x27C + #define FSL_XCVR_DEBUG_REG_0 0x2E0 #define FSL_XCVR_DEBUG_REG_1 0x2F0 -- 2.34.1

2 1

[PATCH 0/3] Add support for SAI on i.MX93 platform
by Chancel Liu 28 Oct '22

28 Oct '22

This patchset supports SAI on i.MX93 platform. Chancel Liu (3): ASoC: dt-bindings: fsl,sai: Add compatible string for i.MX93 platform ASoC: fsl_sai: Add support for i.MX93 platform ASoC: fsl_sai: Specify the maxburst to 8 on i.MX93 platform .../devicetree/bindings/sound/fsl,sai.yaml | 1 + sound/soc/fsl/fsl_sai.c | 23 +++++++++++++++---- sound/soc/fsl/fsl_sai.h | 1 + 3 files changed, 21 insertions(+), 4 deletions(-) -- 2.25.1

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[PATCH v1] ALSA: hda/realtek: Add quirk for ASUS Zenbook using CS35L41
by Stefan Binding 28 Oct '22

28 Oct '22

This Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using I2C with Internal Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 701a72ec5629a..b4f7ff8cfe41b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9404,6 +9404,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- 2.34.1

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