March 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH v1] slimbus: qcom-ngd-ctrl: Reduce auto suspend delay
by Viken Dadhaniya 08 Mar '24

08 Mar '24

Currently we have auto suspend delay of 1s which is very high and it takes long time to driver for runtime suspend after use case is done. Hence to optimize runtime PM ops, reduce auto suspend delay to 100ms. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index efeba8275a66..5de45a0e3da5 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -81,7 +81,7 @@ #define SLIM_USR_MC_DISCONNECT_PORT 0x2E #define SLIM_USR_MC_REPEAT_CHANGE_VALUE 0x0 -#define QCOM_SLIM_NGD_AUTOSUSPEND MSEC_PER_SEC +#define QCOM_SLIM_NGD_AUTOSUSPEND (MSEC_PER_SEC / 10) #define SLIM_RX_MSGQ_TIMEOUT_VAL 0x10000 #define SLIM_LA_MGR 0xFF -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 2

[PATCH v1 0/2] Add PDM controller for StarFive JH8100 SoC
by Xingyu Wu 08 Mar '24

08 Mar '24

The Pulse Density Modulation (PDM) controller is a digital PDM microphone interface controller and decoder that supports both mono/stereo PDM format, and a Inter-IC Sound (I2S) transmitter that outputs standard stereo audio data to another device. The PDM controller includes two PDM modules, each PDM module can drive one bitstream sampling clock and two bitstream coming data with sampling clock rising and falling edge. On the JH8100 SoC, PDM and I2S are fixedly connected as follow: PDM module 0 --> I2S channel 0 PDM module 1 --> I2S channel 1 The first patch adds documentation to describe PDM controller bindings. And the second patch adds PDM driver for the StarFive JH8100 SoC. The DTS patch of PDM controller will be submitted after the patchs of JH8100 DTS are accepted. Xingyu Wu (2): dt-bindings: ASoC: Add PDM controller for the StarFive JH8100 SoC ASoC: starfive: Add PDM controller support .../bindings/sound/starfive,jh8100-pdm.yaml | 84 ++++ MAINTAINERS | 7 + sound/soc/starfive/Kconfig | 7 + sound/soc/starfive/Makefile | 1 + sound/soc/starfive/jh8100_pdm.c | 395 ++++++++++++++++++ 5 files changed, 494 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh8100-pdm.yaml create mode 100644 sound/soc/starfive/jh8100_pdm.c -- 2.25.1

3 10

[PATCH v2 00/22] ASoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 08 Mar '24

08 Mar '24

Changes in v2: - Bindings: Changed link-name/codec/clk-provider to remove `items` and leave just the enum - Moved .*-dai-link pattern additionalProperties after `type: object` - Added ref to sound-card-common.yaml - Fixed dai-link-xxx -> xxx-dai-link typo in example comment This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (22): ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: Commonize ADDA rate transform functions and enums ASoC: mediatek: Assign dummy when codec not specified for a DAI link ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 112 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 102 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 124 +- .../bindings/sound/mt8195-mt6359.yaml | 115 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- sound/soc/mediatek/common/Makefile | 2 + .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dai-adda-common.c | 70 + .../soc/mediatek/common/mtk-dai-adda-common.h | 45 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 205 ++- .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +- sound/soc/mediatek/mt8186/Makefile | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 19 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 94 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 114 +- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 35 files changed, 1714 insertions(+), 2771 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

2 26

[GIT PULL] ASoC fixes for v6.8-rc7
by Mark Brown 08 Mar '24

08 Mar '24

The following changes since commit ed00a6945dc32462c2d3744a3518d2316da66fcc: ASoC: amd: yc: Fix non-functional mic on Lenovo 21J2 (2024-02-28 12:24:59 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-rc7 for you to fetch changes up to 96e202f8c52ac49452f83317cf3b34cd1ad81e18: ASoC: wm8962: Fix up incorrect error message in wm8962_set_fll (2024-03-06 17:44:22 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 Some more driver specific fixes for v6.8, plus one new x86 platform quirk. All good fixes to have if you have systems that use the relevant hardware. ---------------------------------------------------------------- Al Raj Hassain (1): ASoC: amd: yc: Add HP Pavilion Aero Laptop 13-be2xxx(8BD6) into DMI quirk table Alban Boyé (1): ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet Andreas Pape (1): ASoC: rcar: adg: correct TIMSEL setting for SSI9 Rob Herring (1): ASoC: dt-bindings: nvidia: Fix 'lge' vendor prefix Stuart Henderson (4): ASoC: madera: Fix typo in madera_set_fll_clks shift value ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC ASoC: wm8962: Enable both SPKOUTR_ENA and SPKOUTL_ENA in mono mode ASoC: wm8962: Fix up incorrect error message in wm8962_set_fll .../sound/nvidia,tegra-audio-max9808x.yaml | 2 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++++++ sound/soc/codecs/madera.c | 2 +- sound/soc/codecs/wm8962.c | 29 ++++++++++++++-------- sound/soc/intel/boards/bytcr_rt5640.c | 12 +++++++++ sound/soc/sh/rcar/adg.c | 7 ++++++ 6 files changed, 47 insertions(+), 12 deletions(-)

2 1

[PATCH v1 1/1] ASoC: Intel: catpt: Carefully use PCI bitwise constants
by Andy Shevchenko 07 Mar '24

07 Mar '24

PM constants for PCI devices are defined with bitwise annotation. When used as is, sparse complains about that: .../catpt/dsp.c:390:9: warning: restricted pci_power_t degrades to integer .../catpt/dsp.c:414:9: warning: restricted pci_power_t degrades to integer Force them to be u32 in the driver. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/catpt/dsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c index 346bec000306..5454c6d9ab5b 100644 --- a/sound/soc/intel/catpt/dsp.c +++ b/sound/soc/intel/catpt/dsp.c @@ -387,7 +387,7 @@ int catpt_dsp_power_down(struct catpt_dev *cdev) mask = cdev->spec->d3srampgd_bit | cdev->spec->d3pgd_bit; catpt_updatel_pci(cdev, VDRTCTL0, mask, cdev->spec->d3pgd_bit); - catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, PCI_D3hot); + catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, (__force u32)PCI_D3hot); /* give hw time to drop off */ udelay(50); @@ -411,7 +411,7 @@ int catpt_dsp_power_up(struct catpt_dev *cdev) val = mask & (~CATPT_VDRTCTL2_DTCGE); catpt_updatel_pci(cdev, VDRTCTL2, mask, val); - catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, PCI_D0); + catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, (__force u32)PCI_D0); /* SRAM power gating none */ mask = cdev->spec->d3srampgd_bit | cdev->spec->d3pgd_bit; -- 2.43.0.rc1.1.gbec44491f096

3 2

[PATCH] ASoC: wm_adsp: Fix missing mutex_lock in wm_adsp_write_ctl()
by Richard Fitzgerald 07 Mar '24

07 Mar '24

wm_adsp_write_ctl() must hold the pwr_lock mutex when calling cs_dsp_get_ctl(). This was previously partially fixed by commit 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()") but this only put locking around the call to cs_dsp_coeff_write_ctrl(), missing the call to cs_dsp_get_ctl(). Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()") --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e451c009f2d9..7d5c096e06cd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -683,11 +683,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); + struct cs_dsp_coeff_ctl *cs_ctl; struct wm_coeff_ctl *ctl; int ret; mutex_lock(&dsp->cs_dsp.pwr_lock); + cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); mutex_unlock(&dsp->cs_dsp.pwr_lock); -- 2.30.2

2 1

[PATCH v2 0/2] ASoC: codecs: ES8326: change members of private structure
by Zhang Yi 07 Mar '24

07 Mar '24

We found that using 0x45 as the default value for interrupt-clk would cause a headset detection error.So we took 0x00 as the default value for interrupt-clk and passed the test. We removed mic1-src and mic2-src, which were not used. Zhang Yi (2): ASoC: codecs: ES8326: Changing members of private structure ASoC: codecs: ES8326: change support for ES8326 .../bindings/sound/everest,es8326.yaml | 8 ++++---- sound/soc/codecs/es8326.c | 18 +----------------- 2 files changed, 5 insertions(+), 21 deletions(-) -- 2.17.1

2 3

[PATCH 1/2] ALSA: hda: hda_component: Add missing #include guards
by Richard Fitzgerald 07 Mar '24

07 Mar '24

Add the conventional include guards around the content of the hda_component.h header file. This prevents double-declaration of struct hda_component if the header gets included multiple times. This isn't causing any problems with current code, so no need to backport to older kernels. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/pci/hda/hda_component.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index deae9dea01b4..a0fcc723483d 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -6,6 +6,9 @@ * Cirrus Logic International Semiconductor Ltd. */ +#ifndef __HDA_COMPONENT_H__ +#define __HDA_COMPONENT_H__ + #include <linux/acpi.h> #include <linux/component.h> @@ -82,3 +85,5 @@ static inline void hda_component_manager_unbind(struct hda_codec *cdc, { component_unbind_all(hda_codec_dev(cdc), comps); } + +#endif /* ifndef __HDA_COMPONENT_H__ */ -- 2.30.2

2 2

[PATCH] firmware: cirrus: cs_dsp: Remove non-existent member from kerneldoc
by Richard Fitzgerald 07 Mar '24

07 Mar '24

The kerneldoc for struct cs_dsp refers to a fw_file_name member but there's no such member. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- include/linux/firmware/cirrus/cs_dsp.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 29cd11d5a3cf..23384a54d575 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -123,7 +123,6 @@ struct cs_dsp_client_ops; * @sysclk_mask: Mask of frequency bits within sysclk register (ADSP1 only) * @sysclk_shift: Shift of frequency bits within sysclk register (ADSP1 only) * @alg_regions: List of currently loaded algorithm regions - * @fw_file_name: Filename of the current firmware * @fw_name: Name of the current firmware * @fw_id: ID of the current firmware, obtained from the wmfw * @fw_id_version: Version of the firmware, obtained from the wmfw -- 2.30.2

2 1

[PATCH] firmware: cirrus: cs_dsp: Initialize debugfs_root to invalid
by Richard Fitzgerald 07 Mar '24

07 Mar '24

Initialize debugfs_root to -ENODEV so that if the client never sets a valid debugfs root the debugfs files will not be created. A NULL pointer passed to any of the debugfs_create_*() functions means "create in the root of debugfs". It doesn't mean "ignore". Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 79d4254d1f9b..9f3d665cfdcf 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -522,7 +522,7 @@ void cs_dsp_cleanup_debugfs(struct cs_dsp *dsp) { cs_dsp_debugfs_clear(dsp); debugfs_remove_recursive(dsp->debugfs_root); - dsp->debugfs_root = NULL; + dsp->debugfs_root = ERR_PTR(-ENODEV); } EXPORT_SYMBOL_NS_GPL(cs_dsp_cleanup_debugfs, FW_CS_DSP); #else @@ -2246,6 +2246,11 @@ static int cs_dsp_common_init(struct cs_dsp *dsp) mutex_init(&dsp->pwr_lock); +#ifdef CONFIG_DEBUG_FS + /* Ensure this is invalid if client never provides a debugfs root */ + dsp->debugfs_root = ERR_PTR(-ENODEV); +#endif + return 0; } -- 2.30.2

2 1

[PATCH 00/22] ASoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 07 Mar '24

07 Mar '24

This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (22): ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: Commonize ADDA rate transform functions and enums ASoC: mediatek: Assign dummy when codec not specified for a DAI link ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 112 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 112 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 129 +- .../bindings/sound/mt8195-mt6359.yaml | 123 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- sound/soc/mediatek/common/Makefile | 2 + .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dai-adda-common.c | 70 + .../soc/mediatek/common/mtk-dai-adda-common.h | 45 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 205 ++- .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +- sound/soc/mediatek/mt8186/Makefile | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 19 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 94 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 114 +- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 35 files changed, 1739 insertions(+), 2769 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

4 31

Strange trigger_htstamp after snd_pcm_drop
by GitHub issues - opened 07 Mar '24

07 Mar '24

alsa-project/alsa-lib issue #387 was opened from z-s-e: I have conceptually the following situation (actual code can be found [here](https://gitlab.com/z-s-e/alpe/-/blob/main/src/alpe_alsa.cpp?ref_type=…): ``` // sound is currently playing snd_pcm_status(pcm, pcm_status); snd_pcm_status_get_htstamp(pcm_status, &ts1); snd_pcm_drop(pcm); snd_pcm_status(pcm, pcm_status); snd_pcm_status_get_trigger_htstamp(status, &ts2); ``` On my hardware I get a `ts2` timestamp that is earlier than `ts1`. From my understanding that should be impossible, as `ts2` should contain the timestamp of when the drop happened, which should obviously be after `ts1`. Is that a bug or do I misunderstand the meaning of those timestamps? Issue URL : https://github.com/alsa-project/alsa-lib/issues/387 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

RE: [EXTERNAL] Re: [PATCH v10] The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp sp
by Ding, Shenghao 07 Mar '24

07 Mar '24

> -----Original Message----- > From: Mark Brown <broonie(a)kernel.org> > Sent: Wednesday, March 6, 2024 3:19 AM > To: Ding, Shenghao <shenghao-ding(a)ti.com> > Cc: andriy.shevchenko(a)linux.intel.com; lgirdwood(a)gmail.com; > perex(a)perex.cz; pierre-louis.bossart(a)linux.intel.com; > 13916275206(a)139.com; alsa-devel(a)alsa-project.org; linux- > kernel(a)vger.kernel.org; liam.r.girdwood(a)intel.com; bard.liao(a)intel.com; > mengdong.lin(a)intel.com; yung-chuan.liao(a)linux.intel.com; Xu, Baojun > <baojun.xu(a)ti.com>; Lu, Kevin <kevin-lu(a)ti.com>; tiwai(a)suse.de; > soyer(a)irl.hu; Baojun.Xu(a)fpt.com; Navada Kanyana, Mukund > <navada(a)ti.com> > Subject: [EXTERNAL] Re: [PATCH v10] The tas2783 is a smart audio amplifier > with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and > I2S/TDM interfaces designed for portable applications. An on-chip DSP > supports Texas Instruments SmartAmp spe... > > On Tue, Mar 05, 2024 at 04:43:35PM +0800, Shenghao Ding wrote: > > The ASoC component provides the majority of the functionality of the > > device, all the audio functions. > > > +static const struct reg_default tas2783_reg_defaults[] = { > > + /* Default values for ROM mode. Activated. */ > > + { 0x8002, 0x1a }, /* AMP inactive. */ > > + { 0x8097, 0xc8 }, > > + { 0x80b5, 0x74 }, > > + { 0x8099, 0x20 }, > > + { 0xfe8d, 0x0d }, > > + { 0xfebe, 0x4a }, > > + { 0x8230, 0x00 }, > > + { 0x8231, 0x00 }, > > + { 0x8232, 0x00 }, > > + { 0x8233, 0x01 }, > > + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ > > + { 0x8419, 0x00 }, > > + { 0x841a, 0x00 }, > > + { 0x841b, 0x00 }, > > + { 0x8428, 0x40 }, /* Unmute channel */ > > + { 0x8429, 0x00 }, > > + { 0x842a, 0x00 }, > > + { 0x842b, 0x00 }, > > + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ > > + { 0x8549, 0x00 }, > > + { 0x854a, 0x00 }, > > + { 0x854b, 0x00 }, > > + { 0x8558, 0x40 }, /* Unmute channel */ > > + { 0x8559, 0x00 }, > > + { 0x855a, 0x00 }, > > + { 0x855b, 0x00 }, > > + { 0x800a, 0x3a }, /* Enable both channel */ > > These comments sound like this register default table is not actually the > physical default register values that the chip has... > > > +static const struct regmap_config tasdevice_regmap = { > > + .reg_bits = 32, > > + .val_bits = 8, > > + .readable_reg = tas2783_readable_register, > > + .volatile_reg = tas2783_volatile_register, > > + .max_register = 0x44ffffff, > > + .reg_defaults = tas2783_reg_defaults, > > + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), > > ...but this is set as the register defaults. This will cause problems with things > like cache sync where we don't write values out if they're not the defaults. > We also try to keep default settings from the silicon except in the most > obvious cases, it avoids issues with trying to work out what to set and > accomodate different use cases for different systems. > > If you do need to set non-default values then either just regular writes during > probe or a regmap patch would do it. So, can I remove ".reg_defaults = tas2783_reg_defaults," and tas2783_reg_defaults from the code? > > > + .cache_type = REGCACHE_RBTREE, > > + .use_single_read = true, > > + .use_single_write = true, > > REGCACHE_MAPLE is generally the most sensible choice for modern devices > - it's a newer and fancier data structure underlying it and there's only a few > cases with low end devices, mostly doing block I/O which this doesn't > support, where the RBTREE cache is still better. Accept > > > + u16 dev_info; > > + int ret; > > + > > + if (!tas_dev->sdw_peripheral) { > > + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", > > + __func__); > > + return; > > + } > > + > > + dev_info = tas_dev->sdw_peripheral->bus->link_id | > > + tas_dev->sdw_peripheral->id.unique_id << 16; > > I'm kind of surprised dev_info works as a variable name without something > getting upset that it aliases the function of the same name. Accept

2 1

[PATCH 0/4] ASoC: fsl: Support register and unregister rpmsg sound card through remoteproc
by Chancel Liu 07 Mar '24

07 Mar '24

echo /lib/firmware/fw.elf > /sys/class/remoteproc/remoteproc0/firmware (A) echo start > /sys/class/remoteproc/remoteproc0/state (B) echo stop > /sys/class/remoteproc/remoteproc0/state The rpmsg sound card is registered in (A) and unregistered in (B). After "start", imx-audio-rpmsg registers devices for ASoC platform driver and machine driver. Then sound card is registered. After "stop", imx-audio-rpmsg unregisters devices for ASoC platform driver and machine driver. Then sound card is unregistered. Chancel Liu (4): ASoC: fsl: imx_pcm_rpmsg: Register component with rpmsg channel name ASoC: fsl: imx-audio-rpmsg: Register device with rpmsg channel name ASoC: fsl: Let imx-audio-rpmsg register platform device for card ASoC: fsl: imx-rpmsg: Update to correct DT node sound/soc/fsl/fsl_rpmsg.c | 11 ----------- sound/soc/fsl/imx-audio-rpmsg.c | 22 +++++++++++++++++++--- sound/soc/fsl/imx-pcm-rpmsg.c | 11 ++++++++--- sound/soc/fsl/imx-rpmsg.c | 21 ++++++++++++++++++--- 4 files changed, 45 insertions(+), 20 deletions(-) -- 2.43.0

2 6

[PATCH v1 2/2] ASoC: codecs: ES8326: change support for ES8326
by Zhang Yi 07 Mar '24

07 Mar '24

> > On Wed, Mar 06, 2024 at 09:34:14AM +0800, Zhang Yi wrote: > > > Removed mic1-src and mic2-src. and changed default value > > > of interrupt-clk > > > > We could do with a better changelog here. > > > > > - everest,mic1-src: > > > - $ref: /schemas/types.yaml#/definitions/uint8 > > > - description: > > > - the value of reg 2A when headset plugged. > > > - minimum: 0x00 > > > - maximum: 0x77 > > > - default: 0x22 > > > - > > > - everest,mic2-src: > > > - $ref: /schemas/types.yaml#/definitions/uint8 > > > - description: > > > - the value of reg 2A when headset unplugged. > > > - minimum: 0x00 > > > - maximum: 0x77 > > > - default: 0x44 > > > > This will make any DTs using the properties instantly buggy. I believe > > there's a way of marking properties as deprecated, you should use that > > instead. Or we could just leave the properties there and ignore them at > > runtime. > 'deprecated: true' is the way. OK.I will modify the properties, thank you for your support

1 0

[PATCH 1/5] ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC
by Stuart Henderson 06 Mar '24

06 Mar '24

Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> --- sound/soc/codecs/wm8962.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index fb90ae6a8a34..6d7bb696b135 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2914,8 +2914,12 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: + fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + break; case WM8962_FLL_OSC: fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + snd_soc_component_update_bits(component, WM8962_PLL2, + WM8962_OSC_ENA, WM8962_OSC_ENA); break; case WM8962_FLL_INT: snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1, -- 2.39.2

2 8

[PATCH v1 0/2] ASoC: codecs: ES8326: change members of private structure
by Zhang Yi 06 Mar '24

06 Mar '24

We found that using 0x45 as the default value for interrupt-clk would cause a headset detection error.So we took 0x00 as the default value for interrupt-clk and passed the test. We removed mic1-src and mic2-src, which were not used. *** BLURB HERE *** Zhang Yi (2): ASoC: codecs: ES8326: Changing members of private structure ASoC: codecs: ES8326: change support for ES8326 .../bindings/sound/everest,es8326.yaml | 22 ++----------------- sound/soc/codecs/es8326.c | 18 +-------------- 2 files changed, 3 insertions(+), 37 deletions(-) -- 2.17.1

3 4

[PATCH 0/5] ASoC: Harden DAPM route checks and Intel fixes
by Cezary Rojewski 06 Mar '24

06 Mar '24

Set of loosely connected patches. Most impactful change is dropping any permisiveness when snd_soc_dapm_add_routes() fails in soc-topology.c To do it safely, disable route checks for all skylake-driver boards. Relevant background: Since commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe failures for Intel boards have been partially addressed by commit a22ae72b86a4 ("ASoC: soc-core: isable route checks for legacy devices") and its follow up but only skl_nau88l25_ssm4567.c is patched. The rest of the boards still need fixing. After that, removal of copy-pastas found in ssm4567.c and redundant code in i2s_test.c for avs-boards. Amadeusz Sławiński (1): ASoC: Intel: avs: i2s_test: Remove redundant dapm routes Cezary Rojewski (4): ASoC: Intel: Disable route checks for Skylake boards ASoC: topology: Do not ignore route checks when parsing graphs ASoC: Intel: avs: ssm4567: Do not ignore route checks ASoC: Intel: avs: ssm4567: Board cleanup sound/soc/intel/avs/boards/i2s_test.c | 79 ------------------- sound/soc/intel/avs/boards/ssm4567.c | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 4 + sound/soc/intel/boards/kbl_rt5660.c | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 + .../intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + .../soc/intel/boards/skl_nau88l25_max98357a.c | 1 + sound/soc/intel/boards/skl_rt286.c | 1 + sound/soc/soc-topology.c | 5 +- 14 files changed, 19 insertions(+), 85 deletions(-) -- 2.25.1

2 13

[PATCH] ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()
by Richard Fitzgerald 06 Mar '24

06 Mar '24

snd_soc_card_get_kcontrol() must be holding a read lock on card->controls_rwsem while walking the controls list. Compare with snd_ctl_find_numid(). The existing function is renamed snd_soc_card_get_kcontrol_locked() so that it can be called from contexts that are already holding card->controls_rwsem (for example, control get/put functions). There are few direct or indirect callers of snd_soc_card_get_kcontrol(), and most are safe. Three require changes, which have been included in this patch: codecs/cs35l45.c: cs35l45_activate_ctl() is called from a control put() function so is changed to call snd_soc_card_get_kcontrol_locked(). codecs/cs35l56.c: cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from control get()/put() functions so is changed to call snd_soc_card_get_kcontrol_locked(). fsl/fsl_xcvr.c: fsl_xcvr_activate_ctl() is called from three places, one of which already holds card->controls_rwsem: 1. fsl_xcvr_mode_put(), a control put function, which will already be holding card->controls_rwsem. 2. fsl_xcvr_startup(), a DAI startup function. 3. fsl_xcvr_shutdown(), a DAI shutdown function. To fix this, fsl_xcvr_activate_ctl() has been changed to call snd_soc_card_get_kcontrol_locked() so that it is safe to call directly from fsl_xcvr_mode_put(). The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been changed to take a read lock on card->controls_rsem() around calls to fsl_xcvr_activate_ctl(). While this is not very elegant, it keeps the change small, to avoid this patch creating a large collateral churn in fsl/fsl_xcvr.c. Analysis of other callers of snd_soc_card_get_kcontrol() is that they do not need any changes, they are not holding card->controls_rwsem when they call snd_soc_card_get_kcontrol(). Direct callers of snd_soc_card_get_kcontrol(): fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler Indirect callers via soc_component_notify_control(): codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler codecs/ak4118.c: ak4118_irq_handler() - IRQ handler codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used Indirect callers via snd_soc_limit_volume(): qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function ti/rx51.c: rx51_aic34_init() - DAI init function I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c and ak4118.c changes. Backport note: The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added since the Fixes commit so won't all be present on older kernels. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 209c6cdfd283 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card") --- It would be great if people could test the fsl/, qcom/, ti/rx51 and ak4418 drivers. --- include/sound/soc-card.h | 2 ++ sound/soc/codecs/cs35l45.c | 2 +- sound/soc/codecs/cs35l56.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 12 +++++++++++- sound/soc/soc-card.c | 24 ++++++++++++++++++++++-- 5 files changed, 37 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-card.h b/include/sound/soc-card.h index ecc02e955279..1f4c39922d82 100644 --- a/include/sound/soc-card.h +++ b/include/sound/soc-card.h @@ -30,6 +30,8 @@ static inline void snd_soc_card_mutex_unlock(struct snd_soc_card *card) struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); +struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, + const char *name); int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, struct snd_soc_jack *jack); int snd_soc_card_jack_new_pins(struct snd_soc_card *card, const char *id, diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 44c221745c3b..2392c6effed8 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -184,7 +184,7 @@ static int cs35l45_activate_ctl(struct snd_soc_component *component, else snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s", ctl_name); - kcontrol = snd_soc_card_get_kcontrol(component->card, name); + kcontrol = snd_soc_card_get_kcontrol_locked(component->card, name); if (!kcontrol) { dev_err(component->dev, "Can't find kcontrol %s\n", name); return -EINVAL; diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 2c1313e34cce..6dd0319bc843 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -114,7 +114,7 @@ static int cs35l56_sync_asp1_mixer_widgets_with_firmware(struct cs35l56_private name = full_name; } - kcontrol = snd_soc_card_get_kcontrol(dapm->card, name); + kcontrol = snd_soc_card_get_kcontrol_locked(dapm->card, name); if (!kcontrol) { dev_warn(cs35l56->base.dev, "Could not find control %s\n", name); continue; diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index f0fb33d719c2..c46f64557a7f 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -174,7 +174,9 @@ static int fsl_xcvr_activate_ctl(struct snd_soc_dai *dai, const char *name, struct snd_kcontrol *kctl; bool enabled; - kctl = snd_soc_card_get_kcontrol(card, name); + lockdep_assert_held(&card->snd_card->controls_rwsem); + + kctl = snd_soc_card_get_kcontrol_locked(card, name); if (kctl == NULL) return -ENOENT; @@ -576,10 +578,14 @@ static int fsl_xcvr_startup(struct snd_pcm_substream *substream, xcvr->streams |= BIT(substream->stream); if (!xcvr->soc_data->spdif_only) { + struct snd_soc_card *card = dai->component->card; + /* Disable XCVR controls if there is stream started */ + down_read(&card->snd_card->controls_rwsem); fsl_xcvr_activate_ctl(dai, fsl_xcvr_mode_kctl.name, false); fsl_xcvr_activate_ctl(dai, fsl_xcvr_arc_mode_kctl.name, false); fsl_xcvr_activate_ctl(dai, fsl_xcvr_earc_capds_kctl.name, false); + up_read(&card->snd_card->controls_rwsem); } return 0; @@ -598,11 +604,15 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, /* Enable XCVR controls if there is no stream started */ if (!xcvr->streams) { if (!xcvr->soc_data->spdif_only) { + struct snd_soc_card *card = dai->component->card; + + down_read(&card->snd_card->controls_rwsem); fsl_xcvr_activate_ctl(dai, fsl_xcvr_mode_kctl.name, true); fsl_xcvr_activate_ctl(dai, fsl_xcvr_arc_mode_kctl.name, (xcvr->mode == FSL_XCVR_MODE_ARC)); fsl_xcvr_activate_ctl(dai, fsl_xcvr_earc_capds_kctl.name, (xcvr->mode == FSL_XCVR_MODE_EARC)); + up_read(&card->snd_card->controls_rwsem); } ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, FSL_XCVR_IRQ_EARC_ALL, 0); diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c index 285ab4c9c716..8a2f163da6bc 100644 --- a/sound/soc/soc-card.c +++ b/sound/soc/soc-card.c @@ -5,6 +5,9 @@ // Copyright (C) 2019 Renesas Electronics Corp. // Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> // + +#include <linux/lockdep.h> +#include <linux/rwsem.h> #include <sound/soc.h> #include <sound/jack.h> @@ -26,12 +29,15 @@ static inline int _soc_card_ret(struct snd_soc_card *card, return ret; } -struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, - const char *name) +struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, + const char *name) { struct snd_card *card = soc_card->snd_card; struct snd_kcontrol *kctl; + /* must be held read or write */ + lockdep_assert_held(&card->controls_rwsem); + if (unlikely(!name)) return NULL; @@ -40,6 +46,20 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, return kctl; return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol_locked); + +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + down_read(&card->controls_rwsem); + kctl = snd_soc_card_get_kcontrol_locked(soc_card, name); + up_read(&card->controls_rwsem); + + return kctl; +} EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); static int jack_new(struct snd_soc_card *card, const char *id, int type, -- 2.30.2

3 3

[PATCH v1 1/1] ASoC: Intel: atom: sst_ipc: Remove unused intel-mid.h
by Andy Shevchenko 06 Mar '24

06 Mar '24

intel-mid.h is providing some core parts of the South Complex PM, which are usually are not used by individual drivers. In particular, this driver doesn't use it, so simply remove the unused header. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3fc2c9a6c44d..0630e58b9d6b 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -19,8 +19,9 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/compress_driver.h> -#include <asm/intel-mid.h> + #include <asm/platform_sst_audio.h> + #include "../sst-mfld-platform.h" #include "sst.h" -- 2.43.0.rc1.1.gbec44491f096

2 1

[PATCH] ASoC: dt-bindings: rt1015: Convert to dtschema
by Javier García 06 Mar '24

06 Mar '24

Convert the RT1015 Mono Class D Audio Amplifier to DT schema. Signed-off-by: Javier García <javier.gar.tab(a)gmail.com> --- .../bindings/sound/realtek,rt1015.yaml | 41 +++++++++++++++++++ .../devicetree/bindings/sound/rt1015.txt | 23 ----------- 2 files changed, 41 insertions(+), 23 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1015.txt diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml new file mode 100644 index 000000000000..880196081a60 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1015 Mono Class D Audio Amplifier + +maintainers: + - Jack Yu <jack.yu(a)realtek.com> + +properties: + compatible: + enum: + - realtek,rt1015 + + reg: + maxItems: 1 + + realtek,power-up-delay-ms: + description: Set a delay time for flush work to be completed, + this vlaue is adjustable depending on platform. + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@28 { + compatible = "realtek,rt1015"; + reg = <0x28>; + realtek,power-up-delay-ms = <50>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt deleted file mode 100644 index e498966d436f..000000000000 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ /dev/null @@ -1,23 +0,0 @@ -RT1015 Mono Class D Audio Amplifier - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt1015". - -- reg : The I2C address of the device. - -Optional properties: - -- realtek,power-up-delay-ms - Set a delay time for flush work to be completed, - this value is adjustable depending on platform. - -Example: - -rt1015: codec@28 { - compatible = "realtek,rt1015"; - reg = <0x28>; - realtek,power-up-delay-ms = <50>; -}; -- 2.41.0

3 2

Alsamixer is exiting upon receiving some keys, such as volume control ones
by GitHub issues - opened 06 Mar '24

06 Mar '24

alsa-project/alsa-utils issue #255 was opened from Porkepix: When in an environment not intercepting those keys (usually managed by your compositor/window manager), upon receiving some keycodes such as volume up or down, alsamixer is exiting and writing the UTF-8 keycode inside the terminal while it shouldn't. Examples: Mute key > 440u Volume down key > 438u Volume up key > 439u Curiously enough, the microphone mute one doesn't do anything. These should at the very least not exit alsamixer. A bonus would be, when they're not intercepted, to actually use them to apply mutes and volumes up/down. Issue URL : https://github.com/alsa-project/alsa-utils/issues/255 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] slimbus: stream: Add null pointer check to prevent crash
by Viken Dadhaniya 06 Mar '24

06 Mar '24

We are facing crash due to null pointer dereference of stream in slim_stream_disable(). there is a possible scenario where client driver is calling slimbus stream APIs in incorrect sequence and it might lead to null pointer access of the stream in slimbus enable/disable/prepare/unprepare/free functions. Fix this issue by adding null pointer check of the stream before accessing in all stream API’s exposed to client. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/stream.c | 37 +++++++++++++++++++++++++++++++++---- 1 file changed, 33 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 1d6b38657917..c5a436fd0952 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -202,10 +202,16 @@ static int slim_get_prate_code(int rate) int slim_stream_prepare(struct slim_stream_runtime *rt, struct slim_stream_config *cfg) { - struct slim_controller *ctrl = rt->dev->ctrl; + struct slim_controller *ctrl; struct slim_port *port; int num_ports, i, port_id, prrate; + if (!rt || !cfg) { + pr_err("%s: Stream or cfg is NULL, Check from client side\n", __func__); + return -EINVAL; + } + + ctrl = rt->dev->ctrl; if (rt->ports) { dev_err(&rt->dev->dev, "Stream already Prepared\n"); return -EINVAL; @@ -358,9 +364,15 @@ int slim_stream_enable(struct slim_stream_runtime *stream) { DEFINE_SLIM_BCAST_TXN(txn, SLIM_MSG_MC_BEGIN_RECONFIGURATION, 3, SLIM_LA_MANAGER, NULL); - struct slim_controller *ctrl = stream->dev->ctrl; + struct slim_controller *ctrl; int ret, i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + + ctrl = stream->dev->ctrl; if (ctrl->enable_stream) { ret = ctrl->enable_stream(stream); if (ret) @@ -411,12 +423,18 @@ int slim_stream_disable(struct slim_stream_runtime *stream) { DEFINE_SLIM_BCAST_TXN(txn, SLIM_MSG_MC_BEGIN_RECONFIGURATION, 3, SLIM_LA_MANAGER, NULL); - struct slim_controller *ctrl = stream->dev->ctrl; + struct slim_controller *ctrl; int ret, i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + if (!stream->ports || !stream->num_ports) return -EINVAL; + ctrl = stream->dev->ctrl; if (ctrl->disable_stream) ctrl->disable_stream(stream); @@ -448,6 +466,11 @@ int slim_stream_unprepare(struct slim_stream_runtime *stream) { int i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + if (!stream->ports || !stream->num_ports) return -EINVAL; @@ -476,8 +499,14 @@ EXPORT_SYMBOL_GPL(slim_stream_unprepare); */ int slim_stream_free(struct slim_stream_runtime *stream) { - struct slim_device *sdev = stream->dev; + struct slim_device *sdev; + + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + sdev = stream->dev; spin_lock(&sdev->stream_list_lock); list_del(&stream->node); spin_unlock(&sdev->stream_list_lock); -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

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[PATCH v1] slimbus: qcom-ngd-ctrl: Add timeout for wait operation
by Viken Dadhaniya 06 Mar '24

06 Mar '24

In current driver qcom_slim_ngd_up_worker() indefinitely waiting for ctrl->qmi_up completion object. This is resulting in workqueue lockup on Kthread. Added wait_for_completion_interruptible_timeout to allow the thread to wait for specific timeout period and bail out instead waiting infinitely. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index efeba8275a66..c27076d8b7db 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -83,6 +83,7 @@ #define QCOM_SLIM_NGD_AUTOSUSPEND MSEC_PER_SEC #define SLIM_RX_MSGQ_TIMEOUT_VAL 0x10000 +#define SLIM_QMI_TIMEOUT_MS 1000 #define SLIM_LA_MGR 0xFF #define SLIM_ROOT_FREQ 24576000 @@ -1451,7 +1452,11 @@ static void qcom_slim_ngd_up_worker(struct work_struct *work) ctrl = container_of(work, struct qcom_slim_ngd_ctrl, ngd_up_work); /* Make sure qmi service is up before continuing */ - wait_for_completion_interruptible(&ctrl->qmi_up); + if (!wait_for_completion_interruptible_timeout(&ctrl->qmi_up, + msecs_to_jiffies(SLIM_QMI_TIMEOUT_MS))) { + dev_err(ctrl->dev, "QMI wait timeout\n"); + return; + } mutex_lock(&ctrl->ssr_lock); qcom_slim_ngd_enable(ctrl, true); -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

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[PATCH] ASoC: da7219: read fmw property to get mclk for non-dts systems
by cujomalainey＠chromium.org 05 Mar '24

05 Mar '24

From: Akshu Agrawal <akshu.agrawal(a)amd.com> Non-dts based systems can use ACPI DSDT to pass on the mclk to da7219. This enables da7219 mclk to be linked to system clock. Enable/Disable of the mclk is already handled in the codec so platform drivers don't have to explicitly do handling of mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Karthik Ramasubramanian <kramasub(a)google.com> --- include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 8 +++++++- 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index dde4542b084f3..4cf4c52be3f7f 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -40,6 +40,8 @@ struct da7219_pdata { const char *dai_clk_names[DA7219_DAI_NUM_CLKS]; + const char *mclk_name; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 311ea7918b312..800f61d993043 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1782,6 +1782,8 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct device *dev) pdata->dai_clk_names[DA7219_DAI_WCLK_IDX], pdata->dai_clk_names[DA7219_DAI_BCLK_IDX]); + device_property_read_string(dev, "dlg,mclk-name", &pdata->mclk_name); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -2521,7 +2523,11 @@ static int da7219_probe(struct snd_soc_component *component) da7219_handle_pdata(component); /* Check if MCLK provided */ - da7219->mclk = clk_get(component->dev, "mclk"); + if (da7219->pdata->mclk_name) + da7219->mclk = clk_get(NULL, da7219->pdata->mclk_name); + if (!da7219->mclk) + da7219->mclk = clk_get(component->dev, "mclk"); + if (IS_ERR(da7219->mclk)) { if (PTR_ERR(da7219->mclk) != -ENOENT) { ret = PTR_ERR(da7219->mclk); -- 2.44.0.rc1.240.g4c46232300-goog

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