From f54d87dad7619c8026e95b848d6ef677b9f2b55f Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Mon, 1 Sep 2025 16:57:57 +0800 Subject: [PATCH 01/39] ASoC: rt712: avoid skipping the blind write Some devices might not use the DMIC function of the RT712VB. Therefore, this patch avoids skipping the blind write with RT712VB. Signed-off-by: Shuming Fan Link: https://patch.msgid.link/20250901085757.1287945-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt712-sdca.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c index 5b298db5f0f6..0ebaae426e73 100644 --- a/sound/soc/codecs/rt712-sdca.c +++ b/sound/soc/codecs/rt712-sdca.c @@ -1890,11 +1890,9 @@ int rt712_sdca_io_init(struct device *dev, struct sdw_slave *slave) rt712_sdca_va_io_init(rt712); } else { - if (!rt712->dmic_function_found) { - dev_err(&slave->dev, "%s RT712 VB detected but no SMART_MIC function exposed in ACPI\n", + if (!rt712->dmic_function_found) + dev_warn(&slave->dev, "%s RT712 VB detected but no SMART_MIC function exposed in ACPI\n", __func__); - goto suspend; - } /* multilanes and DMIC are supported by rt712vb */ prop->lane_control_support = true; From d05afb53c683ef7ed1228b593c3360f4d3126c58 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 21 Aug 2025 09:26:37 +0100 Subject: [PATCH 02/39] ASoC: wm8940: Correct PLL rate rounding Using a single value of 22500000 for both 48000Hz and 44100Hz audio will sometimes result in returning wrong dividers due to rounding. Update the code to use the actual value for both. Fixes: 294833fc9eb4 ("ASoC: wm8940: Rewrite code to set proper clocks") Reported-by: Ankur Tyagi Signed-off-by: Charles Keepax Tested-by: Ankur Tyagi Link: https://patch.msgid.link/20250821082639.1301453-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 401ee20897b1..46c16c9bc17a 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -693,7 +693,12 @@ static int wm8940_update_clocks(struct snd_soc_dai *dai) f = wm8940_get_mclkdiv(priv->mclk, fs256, &mclkdiv); if (f != priv->mclk) { /* The PLL performs best around 90MHz */ - fpll = wm8940_get_mclkdiv(22500000, fs256, &mclkdiv); + if (fs256 % 8000) + f = 22579200; + else + f = 24576000; + + fpll = wm8940_get_mclkdiv(f, fs256, &mclkdiv); } wm8940_set_dai_pll(dai, 0, 0, priv->mclk, fpll); From b4799520dcd6fe1e14495cecbbe9975d847cd482 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 21 Aug 2025 09:26:38 +0100 Subject: [PATCH 03/39] ASoC: wm8940: Correct typo in control name Fixes: 0b5e92c5e020 ("ASoC WM8940 Driver") Reported-by: Ankur Tyagi Signed-off-by: Charles Keepax Tested-by: Ankur Tyagi Link: https://patch.msgid.link/20250821082639.1301453-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 46c16c9bc17a..94873ea63014 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -220,7 +220,7 @@ static const struct snd_kcontrol_new wm8940_snd_controls[] = { SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL, 0, 255, 0, wm8940_adc_tlv), SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum), - SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST, + SOC_SINGLE_TLV("Capture Boost Volume", WM8940_ADCBOOST, 8, 1, 0, wm8940_capture_boost_vol_tlv), SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL, 0, 63, 0, wm8940_spk_vol_tlv), From 9b17d3724df55ecc2bc67978822585f2b023be48 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 21 Aug 2025 09:26:39 +0100 Subject: [PATCH 04/39] ASoC: wm8974: Correct PLL rate rounding Using a single value of 22500000 for both 48000Hz and 44100Hz audio will sometimes result in returning wrong dividers due to rounding. Update the code to use the actual value for both. Fixes: 51b2bb3f2568 ("ASoC: wm8974: configure pll and mclk divider automatically") Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20250821082639.1301453-4-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index bdf437a5403f..db16d893a235 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -419,10 +419,14 @@ static int wm8974_update_clocks(struct snd_soc_dai *dai) fs256 = 256 * priv->fs; f = wm8974_get_mclkdiv(priv->mclk, fs256, &mclkdiv); - if (f != priv->mclk) { /* The PLL performs best around 90MHz */ - fpll = wm8974_get_mclkdiv(22500000, fs256, &mclkdiv); + if (fs256 % 8000) + f = 22579200; + else + f = 24576000; + + fpll = wm8974_get_mclkdiv(f, fs256, &mclkdiv); } wm8974_set_dai_pll(dai, 0, 0, priv->mclk, fpll); From 78338108b5a856dc98223a335f147846a8a18c51 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 29 Aug 2025 15:57:34 +0300 Subject: [PATCH 05/39] ASoC: codec: sma1307: Fix memory corruption in sma1307_setting_loaded() The sma1307->set.header_size is how many integers are in the header (there are 8 of them) but instead of allocating space of 8 integers we allocate 8 bytes. This leads to memory corruption when we copy data it on the next line: memcpy(sma1307->set.header, data, sma1307->set.header_size * sizeof(int)); Also since we're immediately copying over the memory in ->set.header, there is no need to zero it in the allocator. Use devm_kmalloc_array() to allocate the memory instead. Fixes: 576c57e6b4c1 ("ASoC: sma1307: Add driver for Iron Device SMA1307") Signed-off-by: Dan Carpenter Link: https://patch.msgid.link/aLGjvjpueVstekXP@stanley.mountain Signed-off-by: Mark Brown --- sound/soc/codecs/sma1307.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sma1307.c b/sound/soc/codecs/sma1307.c index 6a601e7134ea..b683e676640d 100644 --- a/sound/soc/codecs/sma1307.c +++ b/sound/soc/codecs/sma1307.c @@ -1737,9 +1737,10 @@ static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *fil sma1307->set.checksum = data[sma1307->set.header_size - 2]; sma1307->set.num_mode = data[sma1307->set.header_size - 1]; num_mode = sma1307->set.num_mode; - sma1307->set.header = devm_kzalloc(sma1307->dev, - sma1307->set.header_size, - GFP_KERNEL); + sma1307->set.header = devm_kmalloc_array(sma1307->dev, + sma1307->set.header_size, + sizeof(int), + GFP_KERNEL); if (!sma1307->set.header) { sma1307->set.status = false; return; From f1d0260362d72f9f454dc1f9db2eeb80cb801f28 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Thu, 21 Aug 2025 11:15:47 +0530 Subject: [PATCH 06/39] ASoC: amd: acp: Adjust pdm gain value Set pdm gain value by setting PDM_MISC_CTRL_MASK value. To avoid low pdm gain value. Signed-off-by: Venkata Prasad Potturu Reviewed-by: Mario Limonciello (AMD) Link: https://patch.msgid.link/20250821054606.1279178-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/amd.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index cb8d97122f95..73a028e67246 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -130,7 +130,7 @@ #define PDM_DMA_INTR_MASK 0x10000 #define PDM_DEC_64 0x2 #define PDM_CLK_FREQ_MASK 0x07 -#define PDM_MISC_CTRL_MASK 0x10 +#define PDM_MISC_CTRL_MASK 0x18 #define PDM_ENABLE 0x01 #define PDM_DISABLE 0x00 #define DMA_EN_MASK 0x02 From 28edfaa10ca1b370b1a27fde632000d35c43402c Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Mon, 1 Sep 2025 16:15:07 +0100 Subject: [PATCH 07/39] ASoC: SDCA: Add quirk for incorrect function types for 3 systems Certain systems have CS42L43 DisCo that claims to conform to version 0.6.28 but uses the function types from the 1.0 spec. Add a quirk as a workaround. Closes: https://github.com/thesofproject/linux/issues/5515 Cc: stable@vger.kernel.org Signed-off-by: Maciej Strozek Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20250901151518.3197941-1-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/sdca.h | 1 + sound/soc/sdca/sdca_device.c | 20 ++++++++++++++++++++ sound/soc/sdca/sdca_functions.c | 13 ++++++++----- 3 files changed, 29 insertions(+), 5 deletions(-) diff --git a/include/sound/sdca.h b/include/sound/sdca.h index 5a5d6de78d72..9c6a351c9d47 100644 --- a/include/sound/sdca.h +++ b/include/sound/sdca.h @@ -46,6 +46,7 @@ struct sdca_device_data { enum sdca_quirk { SDCA_QUIRKS_RT712_VB, + SDCA_QUIRKS_SKIP_FUNC_TYPE_PATCHING, }; #if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_SOC_SDCA) diff --git a/sound/soc/sdca/sdca_device.c b/sound/soc/sdca/sdca_device.c index 0244cdcdd109..4798ce2c8f0b 100644 --- a/sound/soc/sdca/sdca_device.c +++ b/sound/soc/sdca/sdca_device.c @@ -7,6 +7,7 @@ */ #include +#include #include #include #include @@ -55,11 +56,30 @@ static bool sdca_device_quirk_rt712_vb(struct sdw_slave *slave) return false; } +static bool sdca_device_quirk_skip_func_type_patching(struct sdw_slave *slave) +{ + const char *vendor, *sku; + + vendor = dmi_get_system_info(DMI_SYS_VENDOR); + sku = dmi_get_system_info(DMI_PRODUCT_SKU); + + if (vendor && sku && + !strcmp(vendor, "Dell Inc.") && + (!strcmp(sku, "0C62") || !strcmp(sku, "0C63") || !strcmp(sku, "0C6B")) && + slave->sdca_data.interface_revision == 0x061c && + slave->id.mfg_id == 0x01fa && slave->id.part_id == 0x4243) + return true; + + return false; +} + bool sdca_device_quirk_match(struct sdw_slave *slave, enum sdca_quirk quirk) { switch (quirk) { case SDCA_QUIRKS_RT712_VB: return sdca_device_quirk_rt712_vb(slave); + case SDCA_QUIRKS_SKIP_FUNC_TYPE_PATCHING: + return sdca_device_quirk_skip_func_type_patching(slave); default: break; } diff --git a/sound/soc/sdca/sdca_functions.c b/sound/soc/sdca/sdca_functions.c index f26f597dca9e..13f68f7b6dd6 100644 --- a/sound/soc/sdca/sdca_functions.c +++ b/sound/soc/sdca/sdca_functions.c @@ -90,6 +90,7 @@ static int find_sdca_function(struct acpi_device *adev, void *data) { struct fwnode_handle *function_node = acpi_fwnode_handle(adev); struct sdca_device_data *sdca_data = data; + struct sdw_slave *slave = container_of(sdca_data, struct sdw_slave, sdca_data); struct device *dev = &adev->dev; struct fwnode_handle *control5; /* used to identify function type */ const char *function_name; @@ -137,11 +138,13 @@ static int find_sdca_function(struct acpi_device *adev, void *data) return ret; } - ret = patch_sdca_function_type(sdca_data->interface_revision, &function_type); - if (ret < 0) { - dev_err(dev, "SDCA version %#x invalid function type %d\n", - sdca_data->interface_revision, function_type); - return ret; + if (!sdca_device_quirk_match(slave, SDCA_QUIRKS_SKIP_FUNC_TYPE_PATCHING)) { + ret = patch_sdca_function_type(sdca_data->interface_revision, &function_type); + if (ret < 0) { + dev_err(dev, "SDCA version %#x invalid function type %d\n", + sdca_data->interface_revision, function_type); + return ret; + } } function_name = get_sdca_function_name(function_type); From 0c28431f6fe13f3a3be0978f79c1a7ae8a93d028 Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Tue, 2 Sep 2025 13:21:00 +0300 Subject: [PATCH 08/39] ASoC: SOF: imx: Fix devm_ioremap_resource check devm_ioremap_resource does not return NULL on error but an error pointer so we need to use IS_ERR to check the return code. While at it also pass the error code to dev_err_probe to improve logging. Fixes: bc163baef570 ("ASoC: Use of_reserved_mem_region_to_resource() for "memory-region"") Signed-off-by: Daniel Baluta Link: https://patch.msgid.link/20250902102101.378809-1-daniel.baluta@nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx-common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c index f00b381cec3b..d66c198b861a 100644 --- a/sound/soc/sof/imx/imx-common.c +++ b/sound/soc/sof/imx/imx-common.c @@ -316,9 +316,9 @@ static int imx_parse_ioremap_memory(struct snd_sof_dev *sdev) } sdev->bar[blk_type] = devm_ioremap_resource(sdev->dev, res); - if (!sdev->bar[blk_type]) + if (IS_ERR(sdev->bar[blk_type])) return dev_err_probe(sdev->dev, - -ENOMEM, + PTR_ERR(sdev->bar[blk_type]), "failed to ioremap %s region\n", chip_info->memory[i].name); } From 35fc531a59694f24a2456569cf7d1a9c6436841c Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 2 Sep 2025 13:06:39 +0100 Subject: [PATCH 09/39] ASoC: SOF: Intel: hda-stream: Fix incorrect variable used in error message The dev_err message is reporting an error about capture streams however it is using the incorrect variable num_playback instead of num_capture. Fix this by using the correct variable num_capture. Fixes: a1d1e266b445 ("ASoC: SOF: Intel: Add Intel specific HDA stream operations") Signed-off-by: Colin Ian King Acked-by: Peter Ujfalusi Link: https://patch.msgid.link/20250902120639.2626861-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index aa6b0247d5c9..a34f472ef175 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -890,7 +890,7 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) if (num_capture >= SOF_HDA_CAPTURE_STREAMS) { dev_err(sdev->dev, "error: too many capture streams %d\n", - num_playback); + num_capture); return -EINVAL; } From ad64c073c9a031850de1542e6e976b0249e7e650 Mon Sep 17 00:00:00 2001 From: Paul Menzel Date: Wed, 3 Sep 2025 12:08:41 +0200 Subject: [PATCH 10/39] ALSA: docs: Remove 3rd person singular s in *to indicate* Fixes: 78811dd56def ("ALSA: docs: Add documents for recently changes in snd-usb-audio") Signed-off-by: Paul Menzel Link: https://patch.msgid.link/20250903100842.267194-1-pmenzel@molgen.mpg.de Signed-off-by: Takashi Iwai --- Documentation/sound/alsa-configuration.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 062b86522e4d..accaebbdd642 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2293,7 +2293,7 @@ delayed_register notice the need. skip_validation Skip unit descriptor validation (default: no). - The option is used to ignores the validation errors with the hexdump + The option is used to ignore the validation errors with the hexdump of the unit descriptor instead of a driver probe error, so that we can check its details. quirk_flags From 3254959b4dd065eae396cf78ccc1361460b2f53e Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Wed, 3 Sep 2025 22:47:47 +0530 Subject: [PATCH 11/39] ASoC: amd: amd_sdw: Add quirks for some new Dell laptops Add a quirk to include the codec amplifier function for Dell SKU's listed in quirk table. Note: In these SKU's, the RT722 codec amplifier is excluded, and an external amplifier is used instead. Signed-off-by: Syed Saba Kareem Message-ID: <20250903171817.2549507-1-syed.sabakareem@amd.com> Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/amd/acp/acp-sdw-legacy-mach.c b/sound/soc/amd/acp/acp-sdw-legacy-mach.c index c2197b75a7dd..5a3cfedacbaf 100644 --- a/sound/soc/amd/acp/acp-sdw-legacy-mach.c +++ b/sound/soc/amd/acp/acp-sdw-legacy-mach.c @@ -79,6 +79,22 @@ static const struct dmi_system_id soc_sdw_quirk_table[] = { }, .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), }, + { + .callback = soc_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0DD3"), + }, + .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), + }, + { + .callback = soc_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0DD4"), + }, + .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), + }, {} }; From 68f27f7c7708183e7873c585ded2f1b057ac5b97 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 4 Sep 2025 12:18:50 +0200 Subject: [PATCH 12/39] ASoC: qcom: q6apm-lpass-dais: Fix NULL pointer dereference if source graph failed If earlier opening of source graph fails (e.g. ADSP rejects due to incorrect audioreach topology), the graph is closed and "dai_data->graph[dai->id]" is assigned NULL. Preparing the DAI for sink graph continues though and next call to q6apm_lpass_dai_prepare() receives dai_data->graph[dai->id]=NULL leading to NULL pointer exception: qcom-apm gprsvc:service:2:1: Error (1) Processing 0x01001002 cmd qcom-apm gprsvc:service:2:1: DSP returned error[1001002] 1 q6apm-lpass-dais 30000000.remoteproc:glink-edge:gpr:service@1:bedais: fail to start APM port 78 q6apm-lpass-dais 30000000.remoteproc:glink-edge:gpr:service@1:bedais: ASoC: error at snd_soc_pcm_dai_prepare on TX_CODEC_DMA_TX_3: -22 Unable to handle kernel NULL pointer dereference at virtual address 00000000000000a8 ... Call trace: q6apm_graph_media_format_pcm+0x48/0x120 (P) q6apm_lpass_dai_prepare+0x110/0x1b4 snd_soc_pcm_dai_prepare+0x74/0x108 __soc_pcm_prepare+0x44/0x160 dpcm_be_dai_prepare+0x124/0x1c0 Fixes: 30ad723b93ad ("ASoC: qdsp6: audioreach: add q6apm lpass dai support") Cc: stable@vger.kernel.org Signed-off-by: Krzysztof Kozlowski Reviewed-by: Srinivas Kandagatla Message-ID: <20250904101849.121503-2-krzysztof.kozlowski@linaro.org> Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index a0d90462fd6a..20974f10406b 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -213,8 +213,10 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s return 0; err: - q6apm_graph_close(dai_data->graph[dai->id]); - dai_data->graph[dai->id] = NULL; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + q6apm_graph_close(dai_data->graph[dai->id]); + dai_data->graph[dai->id] = NULL; + } return rc; } From f81e63047600d023cbfda372b6de8f2821ff6839 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Aug 2025 17:37:15 +0100 Subject: [PATCH 13/39] ASoC: SDCA: Fix return value in sdca_regmap_mbq_size() The MBQ size function returns an integer representing the size of a Control. Currently if the Control is not found the function will return false which makes little sense. Correct this typo to return -EINVAL. Fixes: e3f7caf74b79 ("ASoC: SDCA: Add generic regmap SDCA helpers") Signed-off-by: Charles Keepax Message-ID: <20250820163717.1095846-2-ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown --- sound/soc/sdca/sdca_regmap.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sdca/sdca_regmap.c b/sound/soc/sdca/sdca_regmap.c index 5cb3048ea8cf..72f893e00ff5 100644 --- a/sound/soc/sdca/sdca_regmap.c +++ b/sound/soc/sdca/sdca_regmap.c @@ -196,7 +196,7 @@ int sdca_regmap_mbq_size(struct sdca_function_data *function, unsigned int reg) control = function_find_control(function, reg); if (!control) - return false; + return -EINVAL; return clamp_val(control->nbits / BITS_PER_BYTE, sizeof(u8), sizeof(u32)); } From 16c912ec34edc4d7daecde58e40d480858830a12 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Aug 2025 17:37:16 +0100 Subject: [PATCH 14/39] ASoC: SDCA: Fix return value in detected_mode_handler() The detected mode IRQ handler should return an irqreturn_t not a regular error code. Correct the return value in detected_mode_handler(). Fixes: b9ab3b618241 ("ASoC: SDCA: Add some initial IRQ handlers") Signed-off-by: Charles Keepax Message-ID: <20250820163717.1095846-3-ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown --- sound/soc/sdca/sdca_interrupts.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sdca/sdca_interrupts.c b/sound/soc/sdca/sdca_interrupts.c index 8018773ee426..79bf3042f57d 100644 --- a/sound/soc/sdca/sdca_interrupts.c +++ b/sound/soc/sdca/sdca_interrupts.c @@ -155,7 +155,7 @@ static irqreturn_t detected_mode_handler(int irq, void *data) SDCA_CTL_SELECTED_MODE_NAME); if (!name) - return -ENOMEM; + return IRQ_NONE; kctl = snd_soc_component_get_kcontrol(component, name); if (!kctl) { From ec630c2c8ce215dd365b8c3644f004f645714a0f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Aug 2025 17:37:17 +0100 Subject: [PATCH 15/39] ASoC: SDCA: Reorder members of hide struct to remove holes Remove some padding holes in the sdca_entity_hide struct by reordering the members. Signed-off-by: Charles Keepax Message-ID: <20250820163717.1095846-4-ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown --- include/sound/sdca_function.h | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/include/sound/sdca_function.h b/include/sound/sdca_function.h index 06ec126cdcc3..ea68856e4c8c 100644 --- a/include/sound/sdca_function.h +++ b/include/sound/sdca_function.h @@ -1063,27 +1063,30 @@ struct sdca_entity_ge { /** * struct sdca_entity_hide - information specific to HIDE Entities * @hid: HID device structure - * @hidtx_ids: HIDTx Report ID * @num_hidtx_ids: number of HIDTx Report ID - * @hidrx_ids: HIDRx Report ID * @num_hidrx_ids: number of HIDRx Report ID - * @hide_reside_function_num: indicating which Audio Function Numbers within this Device - * @max_delay: the maximum time in microseconds allowed for the Device to change the ownership from Device to Host - * @af_number_list: which Audio Function Numbers within this Device are sending/receiving the messages in this HIDE - * @hid_desc: HID descriptor for the HIDE Entity + * @hidtx_ids: HIDTx Report ID + * @hidrx_ids: HIDRx Report ID + * @af_number_list: which Audio Function Numbers within this Device are + * sending/receiving the messages in this HIDE + * @hide_reside_function_num: indicating which Audio Function Numbers + * within this Device + * @max_delay: the maximum time in microseconds allowed for the Device + * to change the ownership from Device to Host * @hid_report_desc: HID Report Descriptor for the HIDE Entity + * @hid_desc: HID descriptor for the HIDE Entity */ struct sdca_entity_hide { struct hid_device *hid; unsigned int *hidtx_ids; - int num_hidtx_ids; unsigned int *hidrx_ids; + int num_hidtx_ids; int num_hidrx_ids; + unsigned int af_number_list[SDCA_MAX_FUNCTION_COUNT]; unsigned int hide_reside_function_num; unsigned int max_delay; - unsigned int af_number_list[SDCA_MAX_FUNCTION_COUNT]; - struct hid_descriptor hid_desc; unsigned char *hid_report_desc; + struct hid_descriptor hid_desc; }; /** From d0f61658db58583b3acd1ada70a5352c39cd0388 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 1 Sep 2025 09:44:04 +0200 Subject: [PATCH 16/39] ASoC: codecs: lpass-rx-macro: Fix playback quality distortion Commit bb4a0f497bc1 ("ASoC: codecs: lpass: Drop unused AIF_INVALID first DAI identifier") removed first entry in enum with DAI identifiers, because it looked unused. Turns out that there is a relation between DAI ID and "RX_MACRO RX0 MUX"-like kcontrols which use "rx_macro_mux_text" array. That "rx_macro_mux_text" array used first three entries of DAI IDs enum, with value '0' being invalid. The value passed tp "RX_MACRO RX0 MUX"-like kcontrols was used as DAI ID and set to configure active channel count and mask, which are arrays indexed by DAI ID. After removal of first AIF_INVALID DAI identifier, this kcontrol was updating wrong entries in active channel count and mask arrays which was visible in reduced quality (distortions) during headset playback on the Qualcomm SM8750 MTP8750 board. It seems it also fixes recording silence (instead of actual sound) via headset, even though that's different macro codec. Reported-by: Alexey Klimov Fixes: bb4a0f497bc1 ("ASoC: codecs: lpass: Drop unused AIF_INVALID first DAI identifier") Signed-off-by: Krzysztof Kozlowski Reviewed-by: Srinivas Kandagatla Tested-by: Srinivas Kandagatla Message-ID: <20250901074403.137263-2-krzysztof.kozlowski@linaro.org> Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-rx-macro.c | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 238dbdb46c18..a8fc842cc94e 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -618,6 +618,7 @@ static struct interp_sample_rate sr_val_tbl[] = { {176400, 0xB}, {352800, 0xC}, }; +/* Matches also rx_macro_mux_text */ enum { RX_MACRO_AIF1_PB, RX_MACRO_AIF2_PB, @@ -722,6 +723,7 @@ static const char * const rx_int2_2_interp_mux_text[] = { "ZERO", "RX INT2_2 MUX", }; +/* Order must match RX_MACRO_MAX_DAIS enum (offset by 1) */ static const char *const rx_macro_mux_text[] = { "ZERO", "AIF1_PB", "AIF2_PB", "AIF3_PB", "AIF4_PB" }; @@ -2474,6 +2476,7 @@ static int rx_macro_mux_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct snd_soc_dapm_update *update = NULL; u32 rx_port_value = ucontrol->value.enumerated.item[0]; + unsigned int dai_id; u32 aif_rst; struct rx_macro *rx = snd_soc_component_get_drvdata(component); @@ -2490,19 +2493,24 @@ static int rx_macro_mux_put(struct snd_kcontrol *kcontrol, switch (rx_port_value) { case 0: - if (rx->active_ch_cnt[aif_rst]) { - clear_bit(widget->shift, - &rx->active_ch_mask[aif_rst]); - rx->active_ch_cnt[aif_rst]--; + /* + * active_ch_cnt and active_ch_mask use DAI IDs (RX_MACRO_MAX_DAIS). + * active_ch_cnt == 0 was tested in if() above. + */ + dai_id = aif_rst - 1; + if (rx->active_ch_cnt[dai_id]) { + clear_bit(widget->shift, &rx->active_ch_mask[dai_id]); + rx->active_ch_cnt[dai_id]--; } break; case 1: case 2: case 3: case 4: - set_bit(widget->shift, - &rx->active_ch_mask[rx_port_value]); - rx->active_ch_cnt[rx_port_value]++; + /* active_ch_cnt and active_ch_mask use DAI IDs (WSA_MACRO_MAX_DAIS). */ + dai_id = rx_port_value - 1; + set_bit(widget->shift, &rx->active_ch_mask[dai_id]); + rx->active_ch_cnt[dai_id]++; break; default: dev_err(component->dev, From 9004a450fccbeb40a71cc173747da37a459fd4dc Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sun, 31 Aug 2025 17:14:02 +0200 Subject: [PATCH 17/39] ASoC: codecs: lpass-wsa-macro: Fix speaker quality distortion Commit bb4a0f497bc1 ("ASoC: codecs: lpass: Drop unused AIF_INVALID first DAI identifier") removed first entry in enum with DAI identifiers, because it looked unused. Turns out that there is a relation between DAI ID and "WSA RX0 Mux"-like kcontrols (which use "rx_mux_text" array). That "rx_mux_text" array used first three entries of DAI IDs enum, with value '0' being invalid. The value passed tp "WSA RX0 Mux"-like kcontrols was used as DAI ID and set to configure active channel count and mask, which are arrays indexed by DAI ID. After removal of first AIF_INVALID DAI identifier, this kcontrol was updating wrong entries in active channel count and mask arrays which was visible in reduced quality (distortions) during speaker playback on several boards like Lenovo T14s laptop and Qualcomm SM8550-based boards. Reported-by: Alexey Klimov Fixes: bb4a0f497bc1 ("ASoC: codecs: lpass: Drop unused AIF_INVALID first DAI identifier") Signed-off-by: Krzysztof Kozlowski Reviewed-by: Srinivas Kandagatla Tested-by: Srinivas Kandagatla Message-ID: <20250831151401.30897-2-krzysztof.kozlowski@linaro.org> Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-wsa-macro.c | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index da6adb3de21d..d7eec9fdaf9c 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -368,6 +368,7 @@ static struct interp_sample_rate int_mix_sample_rate_val[] = { {192000, 0x6}, /* 192K */ }; +/* Matches also rx_mux_text */ enum { WSA_MACRO_AIF1_PB, WSA_MACRO_AIF_MIX1_PB, @@ -465,6 +466,7 @@ static const char *const rx_mix_ec_text[] = { "ZERO", "RX_MIX_TX0", "RX_MIX_TX1" }; +/* Order must match WSA_MACRO_MAX_DAIS enum (offset by 1) */ static const char *const rx_mux_text[] = { "ZERO", "AIF1_PB", "AIF_MIX1_PB" }; @@ -2207,6 +2209,7 @@ static int wsa_macro_rx_mux_put(struct snd_kcontrol *kcontrol, u32 rx_port_value = ucontrol->value.integer.value[0]; u32 bit_input; u32 aif_rst; + unsigned int dai_id; struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); aif_rst = wsa->rx_port_value[widget->shift]; @@ -2224,17 +2227,22 @@ static int wsa_macro_rx_mux_put(struct snd_kcontrol *kcontrol, switch (rx_port_value) { case 0: - if (wsa->active_ch_cnt[aif_rst]) { - clear_bit(bit_input, - &wsa->active_ch_mask[aif_rst]); - wsa->active_ch_cnt[aif_rst]--; + /* + * active_ch_cnt and active_ch_mask use DAI IDs (WSA_MACRO_MAX_DAIS). + * active_ch_cnt == 0 was tested in if() above. + */ + dai_id = aif_rst - 1; + if (wsa->active_ch_cnt[dai_id]) { + clear_bit(bit_input, &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } break; case 1: case 2: - set_bit(bit_input, - &wsa->active_ch_mask[rx_port_value]); - wsa->active_ch_cnt[rx_port_value]++; + /* active_ch_cnt and active_ch_mask use DAI IDs (WSA_MACRO_MAX_DAIS). */ + dai_id = rx_port_value - 1; + set_bit(bit_input, &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; break; default: dev_err(component->dev, From 71d2893a235bf3b95baccead27b3d47f2f2cdc4c Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Mon, 8 Sep 2025 06:27:27 +0800 Subject: [PATCH 18/39] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low and insert it between R0 and InvR0. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20250907222728.988-1-shenghao-ding@ti.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/tas2781_hda.c | 25 +++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..536940c78f00 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,23 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing function is a bit different from the + * order in UEFI. Here is the conversion to match the order of calibrated-data + * writing function. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + struct cali_reg reg_data; + + memcpy(®_data, &data[base], sizeof(reg_data)); + /* the data order has to be swapped between r0_low_reg and inv0_reg */ + swap(reg_data.r0_low_reg, reg_data.invr0_reg); + + cpu_to_be32_array((__force __be32 *)(data + offset + 1), + (u32 *)®_data, TASDEV_CALIB_N); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -103,8 +120,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -130,9 +146,8 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = p->ndev - 1; j >= 0; j--) { l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + cali_cnv(data, cali_data->cali_dat_sz_per_dev * j, l); + data[l] = j; } } From 690aa09b1845c0d5c3c29dabd50a9d0488c97c48 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Tue, 9 Sep 2025 11:28:29 +0200 Subject: [PATCH 19/39] ASoC: Intel: catpt: Expose correct bit depth to userspace MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Currently wrong bit depth is exposed in hw params, causing clipped volume during playback. Expose correct parameters. Fixes: a126750fc865 ("ASoC: Intel: catpt: PCM operations") Reported-by: Andy Shevchenko Tested-by: Andy Shevchenko Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Message-ID: <20250909092829.375953-1-amadeuszx.slawinski@linux.intel.com> Signed-off-by: Mark Brown --- sound/soc/intel/catpt/pcm.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index 46acb7fdc547..bf734c69c4e0 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -568,8 +568,9 @@ static const struct snd_pcm_hardware catpt_pcm_hardware = { SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, .period_bytes_min = PAGE_SIZE, .period_bytes_max = CATPT_BUFFER_MAX_SIZE / CATPT_PCM_PERIODS_MIN, .periods_min = CATPT_PCM_PERIODS_MIN, @@ -698,14 +699,18 @@ static struct snd_soc_dai_driver dai_drivers[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, .capture = { .stream_name = "Analog Capture", .channels_min = 2, .channels_max = 4, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, }, { @@ -717,7 +722,9 @@ static struct snd_soc_dai_driver dai_drivers[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, }, { @@ -729,7 +736,9 @@ static struct snd_soc_dai_driver dai_drivers[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, }, { @@ -741,7 +750,9 @@ static struct snd_soc_dai_driver dai_drivers[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, }, { From 5f1af203ef964e7f7bf9d32716dfa5f332cc6f09 Mon Sep 17 00:00:00 2001 From: Mohammad Rafi Shaik Date: Mon, 8 Sep 2025 11:06:29 +0530 Subject: [PATCH 20/39] ASoC: qcom: audioreach: Fix lpaif_type configuration for the I2S interface Fix missing lpaif_type configuration for the I2S interface. The proper lpaif interface type required to allow DSP to vote appropriate clock setting for I2S interface. Fixes: 25ab80db6b133 ("ASoC: qdsp6: audioreach: add module configuration command helpers") Cc: stable@vger.kernel.org Reviewed-by: Srinivas Kandagatla Signed-off-by: Mohammad Rafi Shaik Message-ID: <20250908053631.70978-2-mohammad.rafi.shaik@oss.qualcomm.com> Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/audioreach.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 4ebaaf736fb9..3f5eed5afce5 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -971,6 +971,7 @@ static int audioreach_i2s_set_media_format(struct q6apm_graph *graph, param_data->param_id = PARAM_ID_I2S_INTF_CFG; param_data->param_size = ic_sz - APM_MODULE_PARAM_DATA_SIZE; + intf_cfg->cfg.lpaif_type = module->hw_interface_type; intf_cfg->cfg.intf_idx = module->hw_interface_idx; intf_cfg->cfg.sd_line_idx = module->sd_line_idx; From 33b55b94bca904ca25a9585e3cd43d15f0467969 Mon Sep 17 00:00:00 2001 From: Mohammad Rafi Shaik Date: Mon, 8 Sep 2025 11:06:30 +0530 Subject: [PATCH 21/39] ASoC: qcom: q6apm-lpass-dais: Fix missing set_fmt DAI op for I2S The q6i2s_set_fmt() function was defined but never linked into the I2S DAI operations, resulting DAI format settings is being ignored during stream setup. This change fixes the issue by properly linking the .set_fmt handler within the DAI ops. Fixes: 30ad723b93ade ("ASoC: qdsp6: audioreach: add q6apm lpass dai support") Cc: stable@vger.kernel.org Reviewed-by: Srinivas Kandagatla Signed-off-by: Mohammad Rafi Shaik Message-ID: <20250908053631.70978-3-mohammad.rafi.shaik@oss.qualcomm.com> Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index a0d90462fd6a..36c034819b38 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -260,6 +260,7 @@ static const struct snd_soc_dai_ops q6i2s_ops = { .shutdown = q6apm_lpass_dai_shutdown, .set_channel_map = q6dma_set_channel_map, .hw_params = q6dma_hw_params, + .set_fmt = q6i2s_set_fmt, }; static const struct snd_soc_dai_ops q6hdmi_ops = { From 596e8ba2faf0d2beb9bb68801622fa6461918c1d Mon Sep 17 00:00:00 2001 From: Mohammad Rafi Shaik Date: Mon, 8 Sep 2025 11:06:31 +0530 Subject: [PATCH 22/39] ASoC: qcom: sc8280xp: Enable DAI format configuration for MI2S interfaces Add support for configuring the DAI format on MI2S interfaces, this enhancement allows setting the appropriate bit clock and frame clock polarity, ensuring correct audio data transmission over MI2S. Reviewed-by: Srinivas Kandagatla Signed-off-by: Mohammad Rafi Shaik Rule: add Link: https://lore.kernel.org/stable/20250908053631.70978-4-mohammad.rafi.shaik%40oss.qualcomm.com Message-ID: <20250908053631.70978-4-mohammad.rafi.shaik@oss.qualcomm.com> Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 73f9f82c4e25..3067b95bcdbb 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -32,6 +32,10 @@ static int sc8280xp_snd_init(struct snd_soc_pcm_runtime *rtd) int dp_pcm_id = 0; switch (cpu_dai->id) { + case PRIMARY_MI2S_RX...QUATERNARY_MI2S_TX: + case QUINARY_MI2S_RX...QUINARY_MI2S_TX: + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_BP_FP); + break; case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: /* From b0035df56dcd210d735e90ecd16817693f1a2ed9 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 11 Sep 2025 15:11:31 +0800 Subject: [PATCH 23/39] ALSA: hda/tas2781: Fix a potential race condition that causes a NULL pointer in case no efi.get_variable exsits A a potential race condition reported by one of my customers that leads to a NULL pointer dereference, where the call to efi.get_variable should be guarded with efi_rt_services_supported() to ensure that function exists. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/tas2781_hda.c | 5 +++++ sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 5 +++++ 2 files changed, 10 insertions(+) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index 536940c78f00..96e6d82dc69e 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -193,6 +193,11 @@ int tas2781_save_calibration(struct tas2781_hda *hda) efi_status_t status; int i; + if (!efi_rt_services_supported(EFI_RT_SUPPORTED_GET_VARIABLE)) { + dev_err(p->dev, "%s: NO EFI FOUND!\n", __func__); + return -EINVAL; + } + if (hda->catlog_id < LENOVO) efi_guid = tasdev_fct_efi_guid[hda->catlog_id]; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 45a70fbf6205..b5b7a1e82b75 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -315,6 +315,11 @@ static int tas2563_save_calibration(struct tas2781_hda *h) unsigned int attr; int ret, i, j, k; + if (!efi_rt_services_supported(EFI_RT_SUPPORTED_GET_VARIABLE)) { + dev_err(p->dev, "%s: NO EFI FOUND!\n", __func__); + return -EINVAL; + } + cd->cali_dat_sz_per_dev = TAS2563_CAL_DATA_SIZE * TASDEV_CALIB_N; /* extra byte for each device is the device number */ From ec8f26092e525ee3cb1a56d455109fd40bfff3ef Mon Sep 17 00:00:00 2001 From: Donald Menig Date: Sun, 14 Sep 2025 09:43:33 +0200 Subject: [PATCH 24/39] ALSA: hda/realtek: Add ALC295 Dell TAS2781 I2C fixup This patch adds a new fixup for the ALC295 codec on some Dell laptops that use the TAS2781 I2C amplifier. The fixup correctly initializes the amplifier and pins, allowing sound to work on all speakers of these devices. The fixup chain is added to the relevant quirk entries for Dell Polaris models. [ adjusted for 6.17 kernel code by tiwai ] Fixes: 1e9c708dc3ae ("ALSA: hda/tas2781: Add new quirk for Lenovo, ASUS, Dell projects") Link: https://bugzilla.suse.com/show_bug.cgi?id=1249575 Signed-off-by: Donald Menig Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 85bb8c4d3b17..59cc01079fee 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -3702,6 +3702,7 @@ enum { ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, + ALC295_FIXUP_DELL_TAS2781_I2C, ALC245_FIXUP_TAS2781_SPI_2, ALC287_FIXUP_TXNW2781_I2C, ALC287_FIXUP_YOGA7_14ARB7_I2C, @@ -5167,6 +5168,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc294_fixup_gx502_hp, }, + [ALC295_FIXUP_DELL_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_tias_i2c, + .chained = true, + .chain_id = ALC289_FIXUP_DUAL_SPK + }, [ALC294_FIXUP_ASUS_GU502_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6289,8 +6296,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0c1e, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c28, "Dell Inspiron 16 Plus 7630", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0c4d, "Dell", ALC287_FIXUP_CS35L41_I2C_4), - SND_PCI_QUIRK(0x1028, 0x0c94, "Dell Polaris 3 metal", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x1028, 0x0c96, "Dell Polaris 2in1", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1028, 0x0c94, "Dell Polaris 3 metal", ALC295_FIXUP_DELL_TAS2781_I2C), + SND_PCI_QUIRK(0x1028, 0x0c96, "Dell Polaris 2in1", ALC295_FIXUP_DELL_TAS2781_I2C), SND_PCI_QUIRK(0x1028, 0x0cbd, "Dell Oasis 13 CS MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbe, "Dell Oasis 13 2-IN-1 MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbf, "Dell Oasis 13 Low Weight MTU-L", ALC289_FIXUP_DELL_CS35L41_SPI_2), From f205ed23f0687ea8b14c140e0af9643eed683691 Mon Sep 17 00:00:00 2001 From: Bou-Saan Che Date: Sun, 14 Sep 2025 22:15:37 +0300 Subject: [PATCH 25/39] ALSA: hda: cs35l41: Support Lenovo Thinkbook 13x Gen 5 This laptop does not contain _DSD so needs to be supported using the configuration table. Signed-off-by: Bou-Saan Che Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/cs35l41_hda_property.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/codecs/side-codecs/cs35l41_hda_property.c b/sound/hda/codecs/side-codecs/cs35l41_hda_property.c index d8249d997c2a..16d5ea77192f 100644 --- a/sound/hda/codecs/side-codecs/cs35l41_hda_property.c +++ b/sound/hda/codecs/side-codecs/cs35l41_hda_property.c @@ -135,6 +135,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "17AA38C8", 4, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT }, 0, 2, -1, 1000, 4500, 24 }, { "17AA38F9", 2, EXTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38FA", 2, EXTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, + { "17AA3929", 4, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT }, 0, 2, -1, 1000, 4500, 24 }, + { "17AA392B", 4, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT }, 0, 2, -1, 1000, 4500, 24 }, {} }; @@ -558,6 +560,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "17AA38C8", generic_dsd_config }, { "CSC3551", "17AA38F9", generic_dsd_config }, { "CSC3551", "17AA38FA", generic_dsd_config }, + { "CSC3551", "17AA3929", generic_dsd_config }, + { "CSC3551", "17AA392B", generic_dsd_config }, {} }; From c1d31894d892a88454eee6530c99f3c4fcbc9397 Mon Sep 17 00:00:00 2001 From: Bou-Saan Che Date: Sun, 14 Sep 2025 22:17:38 +0300 Subject: [PATCH 26/39] ALSA: hda/realtek: Support Lenovo Thinkbook 13x Gen 5 The laptop does not contain valid _DSD for these amps, so requires entries into the CS35L41 configuration table to function correctly. Signed-off-by: Bou-Saan Che Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 59cc01079fee..c86ba6ae4ef8 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7100,6 +7100,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), SND_PCI_QUIRK(0x17aa, 0x391f, "Yoga S990-16 pro Quad YC Quad", ALC287_FIXUP_TXNW2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_TXNW2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3929, "Thinkbook 13x Gen 5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x392b, "Thinkbook 13x Gen 5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), From d99c203034989c3652cc8e7330ba5750fc74c04f Mon Sep 17 00:00:00 2001 From: Bou-Saan Che Date: Sun, 14 Sep 2025 22:17:50 +0300 Subject: [PATCH 27/39] ALSA: hda/realtek: Fix volume control on Lenovo Thinkbook 13x Gen 4 The issue was caused by incorrect configuration in the driver, which prevented proper volume control on certain systems. Signed-off-by: Bou-Saan Che Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index c86ba6ae4ef8..618ec00d3f98 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7078,8 +7078,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x38c7, "Thinkbook 13x Gen 4", ALC287_FIXUP_CS35L41_I2C_4), - SND_PCI_QUIRK(0x17aa, 0x38c8, "Thinkbook 13x Gen 4", ALC287_FIXUP_CS35L41_I2C_4), + SND_PCI_QUIRK(0x17aa, 0x38c7, "Thinkbook 13x Gen 4", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38c8, "Thinkbook 13x Gen 4", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), From c7a321e4e90e1bd072697bc050b9426e04cffc6a Mon Sep 17 00:00:00 2001 From: Mohammad Rafi Shaik Date: Sun, 14 Sep 2025 18:45:49 +0530 Subject: [PATCH 28/39] ASoC: qcom: sc8280xp: Fix sound card driver name match data for QCS8275 The QCS8275 board is based on Qualcomm's QCS8300 SoC family, and all supported firmware files are located in the qcs8300 directory. The sound topology and ALSA UCM configuration files have also been migrated from the qcs8275 directory to the actual SoC qcs8300 directory in linux-firmware. With the current setup, the sound topology fails to load, resulting in sound card registration failure. This patch updates the driver match data to use the correct driver name qcs8300 for the qcs8275-sndcard, ensuring that the sound card driver correctly loads the sound topology and ALSA UCM configuration files from the qcs8300 directory. Fixes: 34d340d48e595 ("ASoC: qcom: sc8280xp: Add support for QCS8275") Cc: stable@vger.kernel.org Signed-off-by: Mohammad Rafi Shaik Reviewed-by: Srinivas Kandagatla Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20250914131549.1198740-1-mohammad.rafi.shaik@oss.qualcomm.com Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 73f9f82c4e25..db48168b7d3f 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -186,7 +186,7 @@ static int sc8280xp_platform_probe(struct platform_device *pdev) static const struct of_device_id snd_sc8280xp_dt_match[] = { {.compatible = "qcom,qcm6490-idp-sndcard", "qcm6490"}, {.compatible = "qcom,qcs6490-rb3gen2-sndcard", "qcs6490"}, - {.compatible = "qcom,qcs8275-sndcard", "qcs8275"}, + {.compatible = "qcom,qcs8275-sndcard", "qcs8300"}, {.compatible = "qcom,qcs9075-sndcard", "qcs9075"}, {.compatible = "qcom,qcs9100-sndcard", "qcs9100"}, {.compatible = "qcom,sc8280xp-sndcard", "sc8280xp"}, From 73caf2bcf3f0a3843c091b78b0711e356f8a2ef9 Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Mon, 15 Sep 2025 10:54:56 +0800 Subject: [PATCH 29/39] ASoC: Intel: sof_sdw: use PRODUCT_FAMILY for Fatcat series PRODUCT_NAME is machine-specific. Use PRODUCT_FAMILY to ensure the machine quirk is applied with consistent audio configurations across Fatcat series products. Signed-off-by: Mac Chiang Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao Link: https://patch.msgid.link/20250915025456.1154200-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f997b2dc221b..28f03a5f29f7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -761,7 +761,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .callback = sof_sdw_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), - DMI_MATCH(DMI_PRODUCT_NAME, "Fatcat"), + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Fatcat"), }, .driver_data = (void *)(SOC_SDW_PCH_DMIC | SOF_BT_OFFLOAD_SSP(2) | From d7871f400cad1da376f1d7724209a1c49226c456 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 10 Sep 2025 22:43:59 +0530 Subject: [PATCH 30/39] ASoC: amd: acp: Fix incorrect retrival of acp_chip_info Use dev_get_drvdata(dev->parent) instead of dev_get_platdata(dev) to correctly obtain acp_chip_info members in the acp I2S driver. Previously, some members were not updated properly due to incorrect data access, which could potentially lead to null pointer dereferences. This issue was missed in the earlier commit ("ASoC: amd: acp: Fix NULL pointer deref in acp_i2s_set_tdm_slot"), which only addressed set_tdm_slot(). This change ensures that all relevant functions correctly retrieve acp_chip_info, preventing further null pointer dereference issues. Fixes: e3933683b25e ("ASoC: amd: acp: Remove redundant acp_dev_data structure") Signed-off-by: Venkata Prasad Potturu Reviewed-by: Cezary Rojewski Link: https://patch.msgid.link/20250910171419.3682468-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 617690362ad7..4ba0a66981ea 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -73,7 +73,7 @@ static int acp_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct device *dev = cpu_dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); int mode; mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; @@ -199,7 +199,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ u32 reg_val, fmt_reg, tdm_fmt; u32 lrclk_div_val, bclk_div_val; - chip = dev_get_platdata(dev); + chip = dev_get_drvdata(dev->parent); rsrc = chip->rsrc; /* These values are as per Hardware Spec */ @@ -386,7 +386,7 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct { struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; u32 val, period_bytes, reg_val, ier_val, water_val, buf_size, buf_reg; @@ -516,14 +516,13 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; struct acp_stream *stream = substream->runtime->private_data; u32 reg_dma_size = 0, reg_fifo_size = 0, reg_fifo_addr = 0; u32 phy_addr = 0, acp_fifo_addr = 0, ext_int_ctrl; unsigned int dir = substream->stream; - chip = dev_get_platdata(dev); switch (dai->driver->id) { case I2S_SP_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { @@ -632,7 +631,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d { struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; unsigned int dir = substream->stream; unsigned int irq_bit = 0; From cc648f4dde2ffbb74b9c1626e3eaaac89c6fbe16 Mon Sep 17 00:00:00 2001 From: Balamurugan C Date: Mon, 15 Sep 2025 10:56:54 +0800 Subject: [PATCH 31/39] ASoC: Intel: PTL: Add entry for HDMI-In capture support to non-I2S codec boards. Adding HDMI-In capture support for the PTL products which doesn't have onboard I2S codec. But need to support HDMI-In capture via I2S and audio playback through HDMI/DP monitor. Signed-off-by: Balamurugan C Signed-off-by: Bard Liao Link: https://patch.msgid.link/20250915025655.1154279-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 6 ++++++ sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 6 ++++++ 2 files changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 48ee5353bdf1..729c0cd7c19c 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -216,6 +216,12 @@ static const struct platform_device_id board_ids[] = { /* SSP 0 and SSP 2 are used for HDMI IN */ SOF_HDMI_PLAYBACK_PRESENT), }, + { + .name = "ptl_lt6911_hdmi_ssp", + .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | + /* SSP 0 and SSP 2 are used for HDMI IN */ + SOF_HDMI_PLAYBACK_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c index e292701dfcfe..3c8b10e21ceb 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -61,6 +61,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + /* place amp-only boards in the end of table */ + { + .id = "INTC10B0", + .drv_name = "ptl_lt6911_hdmi_ssp", + .sof_tplg_filename = "sof-ptl-hdmi-ssp02.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_ptl_machines); From 1dd28fd86c3fa4e395031dd6f2ba920242107010 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Wed, 17 Sep 2025 08:11:43 +0000 Subject: [PATCH 32/39] ASoC: rt5682s: Adjust SAR ADC button mode to fix noise issue Adjust register settings for SAR adc button detection mode to fix noise issue in headset. Signed-off-by: Jack Yu Link: https://patch.msgid.link/766cd1d2dd7a403ba65bb4cc44845f71@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682s.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 80b921695e7d..1d80a4b862e2 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -653,14 +653,15 @@ static void rt5682s_sar_power_mode(struct snd_soc_component *component, int mode switch (mode) { case SAR_PWR_SAVING: snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, - RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_DIS); + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_EN); snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, - RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, - RT5682S_CTRL_MB1_REG | RT5682S_CTRL_MB2_REG); + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK | + RT5682S_VREF_POW_MASK, RT5682S_CTRL_MB1_FSM | + RT5682S_CTRL_MB2_FSM | RT5682S_VREF_POW_FSM); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | - RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + RT5682S_SAR_BUTDET_POW_NORM | RT5682S_SAR_SEL_MB1_2_MANU); usleep_range(5000, 5500); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK, RT5682S_SAR_BUTDET_EN); @@ -688,7 +689,7 @@ static void rt5682s_sar_power_mode(struct snd_soc_component *component, int mode snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | - RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + RT5682S_SAR_BUTDET_POW_NORM | RT5682S_SAR_SEL_MB1_2_MANU); break; default: dev_err(component->dev, "Invalid SAR Power mode: %d\n", mode); @@ -725,7 +726,7 @@ static void rt5682s_disable_push_button_irq(struct snd_soc_component *component) snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | - RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + RT5682S_SAR_BUTDET_POW_NORM | RT5682S_SAR_SEL_MB1_2_MANU); } /** @@ -786,7 +787,7 @@ static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_ jack_type = SND_JACK_HEADSET; snd_soc_component_write(component, RT5682S_SAR_IL_CMD_3, 0x024c); snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, - RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_EN); + RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_DIS); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_SEL_MB1_2_MASK, val << RT5682S_SAR_SEL_MB1_2_SFT); rt5682s_enable_push_button_irq(component); @@ -966,7 +967,7 @@ static int rt5682s_set_jack_detect(struct snd_soc_component *component, RT5682S_EMB_JD_MASK | RT5682S_DET_TYPE | RT5682S_POL_FAST_OFF_MASK | RT5682S_MIC_CAP_MASK, RT5682S_EMB_JD_EN | RT5682S_DET_TYPE | - RT5682S_POL_FAST_OFF_HIGH | RT5682S_MIC_CAP_HS); + RT5682S_POL_FAST_OFF_LOW | RT5682S_MIC_CAP_HS); regmap_update_bits(rt5682s->regmap, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_POW_MASK, RT5682S_SAR_POW_EN); regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, From 44499ecb4f2817743c37d861bdb3e95f37d3d9cd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Sep 2025 15:09:01 +0200 Subject: [PATCH 33/39] ALSA: usb: qcom: Fix false-positive address space check The sanity check previously added to uaudio_transfer_buffer_setup() assumed the allocated buffer being linear-mapped. But the buffer allocated via usb_alloc_coherent() isn't always so, rather to be used with (SG-)DMA API. This leaded to a false-positive warning and the driver failed to work. Actually uaudio_transfer_buffer_setup() deals only with the DMA-API addresses for MEM_XFER_BUF type, while other callers of uaudio_iommu_map() are with pages with physical addresses for MEM_EVENT_RING and MEM_XFER_RING types. So this patch splits the mapping helper function to two different ones, uaudio_iommu_map() for the DMA pages and uaudio_iommu_map_pa() for the latter, in order to handle mapping differently for each type. Along with it, the unnecessary address check that caused probe error is dropped, too. Fixes: 3335a1bbd624 ("ALSA: qc_audio_offload: try to reduce address space confusion") Suggested-by: Arnd Bergmann Acked-by: Arnd Bergmann Reported-and-tested-by: Luca Weiss Closes: https://lore.kernel.org/DBR2363A95M1.L9XBNC003490@fairphone.com Signed-off-by: Takashi Iwai --- sound/usb/qcom/qc_audio_offload.c | 92 ++++++++++++++++--------------- 1 file changed, 48 insertions(+), 44 deletions(-) diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c index a25c5a531690..9ad76fff741b 100644 --- a/sound/usb/qcom/qc_audio_offload.c +++ b/sound/usb/qcom/qc_audio_offload.c @@ -538,38 +538,33 @@ static void uaudio_iommu_unmap(enum mem_type mtype, unsigned long iova, umap_size, iova, mapped_iova_size); } +static int uaudio_iommu_map_prot(bool dma_coherent) +{ + int prot = IOMMU_READ | IOMMU_WRITE; + + if (dma_coherent) + prot |= IOMMU_CACHE; + return prot; +} + /** - * uaudio_iommu_map() - maps iommu memory for adsp + * uaudio_iommu_map_pa() - maps iommu memory for adsp * @mtype: ring type * @dma_coherent: dma coherent * @pa: physical address for ring/buffer * @size: size of memory region - * @sgt: sg table for memory region * * Maps the XHCI related resources to a memory region that is assigned to be * used by the adsp. This will be mapped to the domain, which is created by * the ASoC USB backend driver. * */ -static unsigned long uaudio_iommu_map(enum mem_type mtype, bool dma_coherent, - phys_addr_t pa, size_t size, - struct sg_table *sgt) +static unsigned long uaudio_iommu_map_pa(enum mem_type mtype, bool dma_coherent, + phys_addr_t pa, size_t size) { - struct scatterlist *sg; unsigned long iova = 0; - size_t total_len = 0; - unsigned long iova_sg; - phys_addr_t pa_sg; bool map = true; - size_t sg_len; - int prot; - int ret; - int i; - - prot = IOMMU_READ | IOMMU_WRITE; - - if (dma_coherent) - prot |= IOMMU_CACHE; + int prot = uaudio_iommu_map_prot(dma_coherent); switch (mtype) { case MEM_EVENT_RING: @@ -583,20 +578,41 @@ static unsigned long uaudio_iommu_map(enum mem_type mtype, bool dma_coherent, &uaudio_qdev->xfer_ring_iova_size, &uaudio_qdev->xfer_ring_list, size); break; - case MEM_XFER_BUF: - iova = uaudio_get_iova(&uaudio_qdev->curr_xfer_buf_iova, - &uaudio_qdev->xfer_buf_iova_size, - &uaudio_qdev->xfer_buf_list, size); - break; default: dev_err(uaudio_qdev->data->dev, "unknown mem type %d\n", mtype); } if (!iova || !map) - goto done; + return 0; - if (!sgt) - goto skip_sgt_map; + iommu_map(uaudio_qdev->data->domain, iova, pa, size, prot, GFP_KERNEL); + + return iova; +} + +static unsigned long uaudio_iommu_map_xfer_buf(bool dma_coherent, size_t size, + struct sg_table *sgt) +{ + struct scatterlist *sg; + unsigned long iova = 0; + size_t total_len = 0; + unsigned long iova_sg; + phys_addr_t pa_sg; + size_t sg_len; + int prot = uaudio_iommu_map_prot(dma_coherent); + int ret; + int i; + + prot = IOMMU_READ | IOMMU_WRITE; + + if (dma_coherent) + prot |= IOMMU_CACHE; + + iova = uaudio_get_iova(&uaudio_qdev->curr_xfer_buf_iova, + &uaudio_qdev->xfer_buf_iova_size, + &uaudio_qdev->xfer_buf_list, size); + if (!iova) + goto done; iova_sg = iova; for_each_sg(sgt->sgl, sg, sgt->nents, i) { @@ -618,11 +634,6 @@ static unsigned long uaudio_iommu_map(enum mem_type mtype, bool dma_coherent, uaudio_iommu_unmap(MEM_XFER_BUF, iova, size, total_len); iova = 0; } - return iova; - -skip_sgt_map: - iommu_map(uaudio_qdev->data->domain, iova, pa, size, prot, GFP_KERNEL); - done: return iova; } @@ -1020,7 +1031,6 @@ static int uaudio_transfer_buffer_setup(struct snd_usb_substream *subs, struct sg_table xfer_buf_sgt; dma_addr_t xfer_buf_dma; void *xfer_buf; - phys_addr_t xfer_buf_pa; u32 len = xfer_buf_len; bool dma_coherent; dma_addr_t xfer_buf_dma_sysdev; @@ -1051,18 +1061,12 @@ static int uaudio_transfer_buffer_setup(struct snd_usb_substream *subs, if (!xfer_buf) return -ENOMEM; - /* Remapping is not possible if xfer_buf is outside of linear map */ - xfer_buf_pa = virt_to_phys(xfer_buf); - if (WARN_ON(!page_is_ram(PFN_DOWN(xfer_buf_pa)))) { - ret = -ENXIO; - goto unmap_sync; - } dma_get_sgtable(subs->dev->bus->sysdev, &xfer_buf_sgt, xfer_buf, xfer_buf_dma, len); /* map the physical buffer into sysdev as well */ - xfer_buf_dma_sysdev = uaudio_iommu_map(MEM_XFER_BUF, dma_coherent, - xfer_buf_pa, len, &xfer_buf_sgt); + xfer_buf_dma_sysdev = uaudio_iommu_map_xfer_buf(dma_coherent, + len, &xfer_buf_sgt); if (!xfer_buf_dma_sysdev) { ret = -ENOMEM; goto unmap_sync; @@ -1143,8 +1147,8 @@ uaudio_endpoint_setup(struct snd_usb_substream *subs, sg_free_table(sgt); /* data transfer ring */ - iova = uaudio_iommu_map(MEM_XFER_RING, dma_coherent, tr_pa, - PAGE_SIZE, NULL); + iova = uaudio_iommu_map_pa(MEM_XFER_RING, dma_coherent, tr_pa, + PAGE_SIZE); if (!iova) { ret = -ENOMEM; goto clear_pa; @@ -1207,8 +1211,8 @@ static int uaudio_event_ring_setup(struct snd_usb_substream *subs, mem_info->dma = sg_dma_address(sgt->sgl); sg_free_table(sgt); - iova = uaudio_iommu_map(MEM_EVENT_RING, dma_coherent, er_pa, - PAGE_SIZE, NULL); + iova = uaudio_iommu_map_pa(MEM_EVENT_RING, dma_coherent, er_pa, + PAGE_SIZE); if (!iova) { ret = -ENOMEM; goto clear_pa; From 7dd670db9afdca6aed4a9f99776cb72e6ed5a1f5 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 17 Sep 2025 17:06:09 +0100 Subject: [PATCH 34/39] ALSA: hda: intel-dsp-config: Prevent SEGFAULT if ACPI_HANDLE() is NULL Check in snd_intel_dsp_check_soundwire() that the pointer returned by ACPI_HANDLE() is not NULL, before passing it on to other functions. The original code assumed a non-NULL return, but if it was unexpectedly NULL it would end up passed to acpi_walk_namespace() as the start point, and would result in [ 3.219028] BUG: kernel NULL pointer dereference, address: 0000000000000018 [ 3.219029] #PF: supervisor read access in kernel mode [ 3.219030] #PF: error_code(0x0000) - not-present page [ 3.219031] PGD 0 P4D 0 [ 3.219032] Oops: Oops: 0000 [#1] SMP NOPTI [ 3.219035] CPU: 2 UID: 0 PID: 476 Comm: (udev-worker) Tainted: G S AW E 6.17.0-rc5-test #1 PREEMPT(voluntary) [ 3.219038] Tainted: [S]=CPU_OUT_OF_SPEC, [A]=OVERRIDDEN_ACPI_TABLE, [W]=WARN, [E]=UNSIGNED_MODULE [ 3.219040] RIP: 0010:acpi_ns_walk_namespace+0xb5/0x480 This problem was triggered by a bugged DSDT that the kernel couldn't parse. But it shouldn't be possible to SEGFAULT the kernel just because of some bugs in ACPI. Fixes: 0650857570d1 ("ALSA: hda: add autodetection for SoundWire") Signed-off-by: Richard Fitzgerald Signed-off-by: Takashi Iwai --- sound/hda/core/intel-dsp-config.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/core/intel-dsp-config.c b/sound/hda/core/intel-dsp-config.c index c15284742899..2a9e35cddcf7 100644 --- a/sound/hda/core/intel-dsp-config.c +++ b/sound/hda/core/intel-dsp-config.c @@ -650,6 +650,8 @@ static int snd_intel_dsp_check_soundwire(struct pci_dev *pci) int ret; handle = ACPI_HANDLE(&pci->dev); + if (!handle) + return -ENODEV; ret = sdw_intel_acpi_scan(handle, &info); if (ret < 0) From d33c3471047fc54966621d19329e6a23ebc8ec50 Mon Sep 17 00:00:00 2001 From: Praful Adiga Date: Thu, 18 Sep 2025 12:40:18 -0400 Subject: [PATCH 35/39] ALSA: hda/realtek: Fix mute led for HP Laptop 15-dw4xx This laptop uses the ALC236 codec with COEF 0x7 and idx 1 to control the mute LED. Enable the existing quirk for this device. Signed-off-by: Praful Adiga Cc: Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 618ec00d3f98..f267437c9698 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6476,6 +6476,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8992, "HP EliteBook 845 G9", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8994, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8995, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x89a0, "HP Laptop 15-dw4xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x89a4, "HP ProBook 440 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89a6, "HP ProBook 450 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89aa, "HP EliteBook 630 G9", ALC236_FIXUP_HP_GPIO_LED), From d1d6ad7f6686e208aba06b7af3feef7a7cba61cf Mon Sep 17 00:00:00 2001 From: Roy Vegard Ovesen Date: Mon, 22 Sep 2025 20:54:10 +0200 Subject: [PATCH 36/39] ALSA: usb-audio: don't apply interface quirk to Presonus S1824c Testing with a Presonus STUDIO 1824c together with a Behringer ultragain digital ADAT device shows that using all 3 altno settings works fine. When selecting sample rate, the driver sets the interface to the correct altno setting and the correct number of channels is set. Selecting the correct altno setting via Ardour, Reaper or whatever other way to set the sample rate is more convenient than re-loading the driver module with device_setup to set altno. Signed-off-by: Roy Vegard Ovesen Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 766db7d00cbc..4a35f962527e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1599,9 +1599,6 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, /* presonus studio 1810c: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x194f, 0x010c)) return s1810c_skip_setting_quirk(chip, iface, altno); - /* presonus studio 1824c: skip altsets incompatible with device_setup */ - if (chip->usb_id == USB_ID(0x194f, 0x010d)) - return s1810c_skip_setting_quirk(chip, iface, altno); return 0; } From 50a098e3e9b1bb30cefc43cdfba3c7b9b32e14a7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 27 Sep 2025 11:47:23 +0200 Subject: [PATCH 37/39] ALSA: hda/realtek: Add quirk for HP Spectre 14t-ea100 HP-Spectre 14t-ea100 model has no speaker output unless booting previously from Windows on dual boot, a reboot while on Linux will stop the speakers working. Applying the existing quirk for HP Spectre X360 EU0xxx seems fixing this speaker problem. Reported-by: Kaden Berger Cc: Link: https://lore.kernel.org/aMxdGAmfOQ6VPNU8@archlinux Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index f267437c9698..07ea76efa5de 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6487,6 +6487,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x89da, "HP Spectre x360 14t-ea100", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x89e7, "HP Elite x2 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a0f, "HP Pavilion 14-ec1xxx", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a20, "HP Laptop 15s-fq5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), From 659169c4eb21f8d9646044a4f4e1bc314f6f9d0c Mon Sep 17 00:00:00 2001 From: Roy Vegard Ovesen Date: Sat, 27 Sep 2025 17:27:30 +0200 Subject: [PATCH 38/39] ALSA: usb-audio: add mono main switch to Presonus S1824c The 1824c does not have the A/B switch that the 1810c has, but instead it has a mono main switch that sums the two main output channels to mono. Signed-off-by: Roy Vegard Ovesen Signed-off-by: Takashi Iwai --- sound/usb/mixer_s1810c.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index fac4bbc6b275..bd24556f6a7f 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -93,6 +93,7 @@ struct s1810c_ctl_packet { #define SC1810C_CTL_LINE_SW 0 #define SC1810C_CTL_MUTE_SW 1 +#define SC1824C_CTL_MONO_SW 2 #define SC1810C_CTL_AB_SW 3 #define SC1810C_CTL_48V_SW 4 @@ -123,6 +124,7 @@ struct s1810c_state_packet { #define SC1810C_STATE_48V_SW 58 #define SC1810C_STATE_LINE_SW 59 #define SC1810C_STATE_MUTE_SW 60 +#define SC1824C_STATE_MONO_SW 61 #define SC1810C_STATE_AB_SW 62 struct s1810_mixer_state { @@ -502,6 +504,15 @@ static const struct snd_kcontrol_new snd_s1810c_mute_sw = { .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8) }; +static const struct snd_kcontrol_new snd_s1824c_mono_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mono Main Out Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1824C_STATE_MONO_SW | SC1824C_CTL_MONO_SW << 8) +}; + static const struct snd_kcontrol_new snd_s1810c_48v_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "48V Phantom Power On Mic Inputs Switch", @@ -588,8 +599,17 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) if (ret < 0) return ret; - ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); - if (ret < 0) - return ret; + // The 1824c has a Mono Main switch instead of a + // A/B select switch. + if (mixer->chip->usb_id == USB_ID(0x194f, 0x010d)) { + ret = snd_s1810c_switch_init(mixer, &snd_s1824c_mono_sw); + if (ret < 0) + return ret; + } else if (mixer->chip->usb_id == USB_ID(0x194f, 0x010c)) { + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); + if (ret < 0) + return ret; + } + return ret; } From 9f2c0ac1423d5f267e7f1d1940780fc764b0fee3 Mon Sep 17 00:00:00 2001 From: Jeongjun Park Date: Sun, 28 Sep 2025 02:39:24 +0900 Subject: [PATCH 39/39] ALSA: usb-audio: fix race condition to UAF in snd_usbmidi_free The previous commit 0718a78f6a9f ("ALSA: usb-audio: Kill timer properly at removal") patched a UAF issue caused by the error timer. However, because the error timer kill added in this patch occurs after the endpoint delete, a race condition to UAF still occurs, albeit rarely. Additionally, since kill-cleanup for urb is also missing, freed memory can be accessed in interrupt context related to urb, which can cause UAF. Therefore, to prevent this, error timer and urb must be killed before freeing the heap memory. Cc: Reported-by: syzbot+f02665daa2abeef4a947@syzkaller.appspotmail.com Closes: https://syzkaller.appspot.com/bug?extid=f02665daa2abeef4a947 Fixes: 0718a78f6a9f ("ALSA: usb-audio: Kill timer properly at removal") Signed-off-by: Jeongjun Park Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index acb3bf92857c..97e7e7662b12 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1522,15 +1522,14 @@ static void snd_usbmidi_free(struct snd_usb_midi *umidi) { int i; + if (!umidi->disconnected) + snd_usbmidi_disconnect(&umidi->list); + for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; - if (ep->out) - snd_usbmidi_out_endpoint_delete(ep->out); - if (ep->in) - snd_usbmidi_in_endpoint_delete(ep->in); + kfree(ep->out); } mutex_destroy(&umidi->mutex); - timer_shutdown_sync(&umidi->error_timer); kfree(umidi); }