From 4d4021b0bbd1fad7c72b9155863f5b3ccb43ae91 Mon Sep 17 00:00:00 2001 From: sheetal Date: Mon, 8 Dec 2025 10:50:40 +0530 Subject: [PATCH 01/43] ASoC: tegra: Fix uninitialized flat cache warning in tegra210_ahub The tegra210_ahub driver started triggering a warning after commit e062bdfdd6ad ("regmap: warn users about uninitialized flat cache"), which flags drivers using REGCACHE_FLAT without register defaults. Since the driver omits default definitions because its registers are zero initialized, the following warning is shown: WARNING KERN tegra210-ahub 2900800.ahub: using zero-initialized flat cache, this may cause unexpected behavior Switch to REGCACHE_FLAT_S which is the recommended cache type for sparse register maps without defaults. This cache type initializes entries on-demand from hardware, eliminating the warning while using memory efficiently. Signed-off-by: sheetal Link: https://patch.msgid.link/20251208052040.4025612-1-sheetal@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_ahub.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index e795907a3963..261d9067d27b 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -2077,7 +2077,7 @@ static const struct regmap_config tegra210_ahub_regmap_config = { .val_bits = 32, .reg_stride = 4, .max_register = TEGRA210_MAX_REGISTER_ADDR, - .cache_type = REGCACHE_FLAT, + .cache_type = REGCACHE_FLAT_S, }; static const struct regmap_config tegra186_ahub_regmap_config = { @@ -2085,7 +2085,7 @@ static const struct regmap_config tegra186_ahub_regmap_config = { .val_bits = 32, .reg_stride = 4, .max_register = TEGRA186_MAX_REGISTER_ADDR, - .cache_type = REGCACHE_FLAT, + .cache_type = REGCACHE_FLAT_S, }; static const struct regmap_config tegra264_ahub_regmap_config = { @@ -2094,7 +2094,7 @@ static const struct regmap_config tegra264_ahub_regmap_config = { .reg_stride = 4, .writeable_reg = tegra264_ahub_wr_reg, .max_register = TEGRA264_MAX_REGISTER_ADDR, - .cache_type = REGCACHE_FLAT, + .cache_type = REGCACHE_FLAT_S, }; static const struct tegra_ahub_soc_data soc_data_tegra210 = { From e2cb8ef0372665854fca6fa7b30b20dd35acffeb Mon Sep 17 00:00:00 2001 From: Andrew Elantsev Date: Wed, 10 Dec 2025 23:38:00 +0300 Subject: [PATCH 02/43] ASoC: amd: yc: Add quirk for Honor MagicBook X16 2025 Add a DMI quirk for the Honor MagicBook X16 2025 laptop fixing the issue where the internal microphone was not detected. Signed-off-by: Andrew Elantsev Link: https://patch.msgid.link/20251210203800.142822-1-elantsew.andrew@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index f210a253da9f..bf4d9d336561 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -661,6 +661,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 C7UCX"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HONOR"), + DMI_MATCH(DMI_PRODUCT_NAME, "GOH-X"), + } + }, {} }; From 20c734cb678332883d317b17bf8fe7361648e170 Mon Sep 17 00:00:00 2001 From: Robert Oscilowski Date: Sat, 15 Nov 2025 19:43:58 +0100 Subject: [PATCH 03/43] ASoC: qcom: sdm845: set quaternary MI2S codec DAI to I2S format We configure the codec DAI format for primary and secondary but not the quaternery MI2S path. Add the missing configuration to enable speaker codecs on the quaternary MI2S like the MAX9827 found on the OnePlus 6. Signed-off-by: Robert Oscilowski Signed-off-by: Casey Connolly Signed-off-by: David Heidelberg Reviewed-by: Alexey Klimov Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20251115-sdm845-quaternary-v3-1-c16bf19128ac@ixit.cz Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index e18a8e44f2db..0ce9dff4dc52 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -365,10 +365,12 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); break; case QUATERNARY_MI2S_RX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT, MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); break; case QUATERNARY_TDM_RX_0: From 9f4d0899efd9892fc7514c9488270e1bb7dedd2b Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Wed, 10 Dec 2025 15:21:09 +0900 Subject: [PATCH 04/43] ASoC: fsl_sai: Constrain sample rates from audio PLLs only in master mode If SAI works in master mode it will generate clocks for external codec from audio PLLs. Thus sample rates should be constrained according to audio PLL clocks. While SAI works in slave mode which means clocks are generated externally then constraints are independent of audio PLLs. Fixes: 4edc98598be4 ("ASoC: fsl_sai: Add sample rate constraint") Signed-off-by: Chancel Liu Link: https://patch.msgid.link/20251210062109.2577735-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 72bfc91e21b9..86730c214914 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -917,8 +917,14 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, tx ? sai->dma_params_tx.maxburst : sai->dma_params_rx.maxburst); - ret = snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &sai->constraint_rates); + if (sai->is_consumer_mode[tx]) + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &fsl_sai_rate_constraints); + else + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &sai->constraint_rates); return ret; } From cb0ae6f22790ead71a866f94c7a5a70ad56af16a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 12 Dec 2025 20:11:12 +0800 Subject: [PATCH 05/43] ASoC: sdw_utils: subtract the endpoint that is not present MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When asoc_sdw_count_sdw_endpoints() count the num_ends, it doesn't skip the unpresented endpoints. But, asoc_sdw_parse_sdw_endpoints() will skip the unpresented endpoints either by quirk or the SDCA function doesn't show up the endpoint. The endpoint number mismatches between count and parse and the machine driver will show up a warning about it. Fixes: 26ee34d2f5c7 ("ASoC: sdw_utils: Add codec_conf for every DAI") Closes: https://github.com/thesofproject/linux/issues/5620 Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Vijendar Mukunda Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20251212121112.3313017-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sdw_utils/soc_sdw_utils.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 6c656b2f7f3a..f169d95895ea 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1534,8 +1534,10 @@ int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card, * endpoint check is not necessary */ if (dai_info->quirk && - !(dai_info->quirk_exclude ^ !!(dai_info->quirk & ctx->mc_quirk))) + !(dai_info->quirk_exclude ^ !!(dai_info->quirk & ctx->mc_quirk))) { + (*num_devs)--; continue; + } } else { /* Check SDCA codec endpoint if there is no matching quirk */ ret = is_sdca_endpoint_present(dev, codec_info, adr_link, i, j); @@ -1543,8 +1545,10 @@ int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card, return ret; /* The endpoint is not present, skip */ - if (!ret) + if (!ret) { + (*num_devs)--; continue; + } } dev_dbg(dev, From 2a03b40deacbd293ac9aed0f9b11197dad54fe5f Mon Sep 17 00:00:00 2001 From: Haotian Zhang Date: Mon, 15 Dec 2025 12:26:52 +0800 Subject: [PATCH 06/43] ALSA: vxpocket: Fix resource leak in vxpocket_probe error path When vxpocket_config() fails, vxpocket_probe() returns the error code directly without freeing the sound card resources allocated by snd_card_new(), which leads to a memory leak. Add proper error handling to free the sound card and clear the allocation bit when vxpocket_config() fails. Fixes: 15b99ac17295 ("[PATCH] pcmcia: add return value to _config() functions") Signed-off-by: Haotian Zhang Link: https://patch.msgid.link/20251215042652.695-1-vulab@iscas.ac.cn Signed-off-by: Takashi Iwai --- sound/pcmcia/vx/vxpocket.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 2e09f2a513a6..9a5c9aa8eec4 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -284,7 +284,13 @@ static int vxpocket_probe(struct pcmcia_device *p_dev) vxp->p_dev = p_dev; - return vxpocket_config(p_dev); + err = vxpocket_config(p_dev); + if (err < 0) { + card_alloc &= ~(1 << i); + snd_card_free(card); + return err; + } + return 0; } static void vxpocket_detach(struct pcmcia_device *link) From 5032347c04ba7ff9ba878f262e075d745c06a2a8 Mon Sep 17 00:00:00 2001 From: Haotian Zhang Date: Mon, 15 Dec 2025 17:04:33 +0800 Subject: [PATCH 07/43] ALSA: pcmcia: Fix resource leak in snd_pdacf_probe error path When pdacf_config() fails, snd_pdacf_probe() returns the error code directly without freeing the sound card resources allocated by snd_card_new(), which leads to a memory leak. Add proper error handling to free the sound card and clear the card list entry when pdacf_config() fails. Fixes: 15b99ac17295 ("[PATCH] pcmcia: add return value to _config() functions") Suggested-by: Takashi Iwai Signed-off-by: Haotian Zhang Link: https://patch.msgid.link/20251215090433.211-1-vulab@iscas.ac.cn Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 13419837dfb7..a3291e626440 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -131,7 +131,13 @@ static int snd_pdacf_probe(struct pcmcia_device *link) link->config_index = 1; link->config_regs = PRESENT_OPTION; - return pdacf_config(link); + err = pdacf_config(link); + if (err < 0) { + card_list[i] = NULL; + snd_card_free(card); + return err; + } + return 0; } From 26e455064983e00013c0a63ffe0eed9e9ec2fa89 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 15 Dec 2025 14:06:47 +0200 Subject: [PATCH 08/43] ASoC: SOF: ipc4-topology: Prefer 32-bit DMIC blobs for 8-bit formats as well With the introduction of 8-bit formats the DMIC blob lookup also needs to be modified to prefer the 32-bit blob when 8-bit format is used on FE. At the same time we also need to make sure that in case 8-bit format is used, but only 16-bit blob is available for DMIC then we will not try to look for 8-bit blob (which is invalid) as fallback, but for a 16-bit one. Fixes: c04c2e829649 ("ASoC: SOF: ipc4-topology: Add support for 8-bit formats") Cc: stable@vger.kernel.org Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Seppo Ingalsuo Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20251215120648.4827-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 221e9d4052b8..47959f182f4b 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1752,11 +1752,9 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai channel_count = params_channels(params); sample_rate = params_rate(params); bit_depth = params_width(params); - /* - * Look for 32-bit blob first instead of 16-bit if copier - * supports multiple formats - */ - if (bit_depth == 16 && !single_bitdepth) { + + /* Prefer 32-bit blob if copier supports multiple formats */ + if (bit_depth <= 16 && !single_bitdepth) { dev_dbg(sdev->dev, "Looking for 32-bit blob first for DMIC\n"); format_change = true; bit_depth = 32; @@ -1799,10 +1797,18 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai if (format_change) { /* * The 32-bit blob was not found in NHLT table, try to - * look for one based on the params + * look for 16-bit for DMIC or based on the params for + * SSP */ - bit_depth = params_width(params); - format_change = false; + if (linktype == SOF_DAI_INTEL_DMIC) { + bit_depth = 16; + if (params_width(params) == 16) + format_change = false; + } else { + bit_depth = params_width(params); + format_change = false; + } + get_new_blob = true; } else if (linktype == SOF_DAI_INTEL_DMIC && !single_bitdepth) { /* From 816f291fc23f325d31509d0e97873249ad75ae9a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 15 Dec 2025 14:06:48 +0200 Subject: [PATCH 09/43] ASoC: SOF: ipc4-topology: Convert FLOAT to S32 during blob selection SSP/DMIC blobs have no support for FLOAT type, they are using S32 on data bus. Convert the format from FLOAT_LE to S32_LE to make sure that the correct format is used within the path. FLOAT conversion will be done on the host side (or within the path). Fixes: f7c41911ad74 ("ASoC: SOF: ipc4-topology: Add support for float sample type") Cc: stable@vger.kernel.org Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Seppo Ingalsuo Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20251215120648.4827-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 47959f182f4b..32b628e2fe29 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1843,7 +1843,7 @@ out: *len = cfg->size >> 2; *dst = (u32 *)cfg->caps; - if (format_change) { + if (format_change || params_format(params) == SNDRV_PCM_FORMAT_FLOAT_LE) { /* * Update the params to reflect that different blob was loaded * instead of the requested bit depth (16 -> 32 or 32 -> 16). From 84085139290a38c5f8a14e5bba60936392c17c7f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 15 Dec 2025 15:07:41 +0200 Subject: [PATCH 10/43] ASoC: SOF: topology: Add context when sink or source widget is missing Add some context to the error prints when sink or source widget is not found by printing the name of the other side of the connection. Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Link: https://patch.msgid.link/20251215130741.31106-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index c1083ea4624a..6b09b8cdf1cb 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2106,8 +2106,8 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, /* source component */ source_swidget = snd_sof_find_swidget(scomp, (char *)route->source); if (!source_swidget) { - dev_err(scomp->dev, "error: source %s not found\n", - route->source); + dev_err(scomp->dev, "source %s for sink %s is not found\n", + route->source, route->sink); ret = -EINVAL; goto err; } @@ -2125,8 +2125,8 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, /* sink component */ sink_swidget = snd_sof_find_swidget(scomp, (char *)route->sink); if (!sink_swidget) { - dev_err(scomp->dev, "error: sink %s not found\n", - route->sink); + dev_err(scomp->dev, "sink %s for source %s is not found\n", + route->sink, route->source); ret = -EINVAL; goto err; } From da230e232352750a80c8fc883eac1c87c8849027 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 15 Dec 2025 15:07:23 +0200 Subject: [PATCH 11/43] ASoC: SOF: ipc4-topology: set playback channel mask MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Currently, we send all channels to all amps and copy the channel_mask to all ALH DMAs in playback. However, the amp may not have the capability to run any process and SOF may need to split the channels and send specific data channel to each amp. In that case, we need to split the channel_mask in ALH DMA. Copy the channel mask only if the widget channel count is the same the FE channels for playback, otherwise, split the channels among the aggregated DAIs. Like what we did in capture. Signed-off-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Liam Girdwood Signed-off-by: Peter Ujfalusi Link: https://patch.msgid.link/20251215130723.31081-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 221e9d4052b8..588defd3eec9 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -2280,8 +2280,19 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, ch_map >>= 4; } - step = ch_count / blob->alh_cfg.device_count; - mask = GENMASK(step - 1, 0); + if (swidget->id == snd_soc_dapm_dai_in && ch_count == out_ref_channels) { + /* + * For playback DAI widgets where the channel number is equal to + * the output reference channels, set the step = 0 to ensure all + * the ch_mask is applied to all alh mappings. + */ + mask = ch_mask; + step = 0; + } else { + step = ch_count / blob->alh_cfg.device_count; + mask = GENMASK(step - 1, 0); + } + /* * Set each gtw_cfg.node_id to blob->alh_cfg.mapping[] * for all widgets with the same stream name @@ -2316,8 +2327,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, } /* - * Set the same channel mask for playback as the audio data is - * duplicated for all speakers. For capture, split the channels + * Set the same channel mask if the widget channel count is the same + * as the FE channels for playback as the audio data is duplicated + * for all speakers in this case. Otherwise, split the channels * among the aggregated DAIs. For example, with 4 channels on 2 * aggregated DAIs, the channel_mask should be 0x3 and 0xc for the * two DAI's. @@ -2326,10 +2338,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, * the tables in soc_acpi files depending on the _ADR and devID * registers for each codec. */ - if (w->id == snd_soc_dapm_dai_in) - blob->alh_cfg.mapping[i].channel_mask = ch_mask; - else - blob->alh_cfg.mapping[i].channel_mask = mask << (step * i); + blob->alh_cfg.mapping[i].channel_mask = mask << (step * i); i++; } From 544c0494cdb3732281e1f2e279cfa561724355db Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 15 Dec 2025 15:08:05 +0200 Subject: [PATCH 12/43] ASoC: SOF: Intel: pci-mtl: Change the topology path to intel/sof-ipc4-tplg The default topology path for IPC4 is intel/sof-ipc4-tplg with a symlink to it as intel/sof-ace-tplg to support old kernels. sof-bin has been released in this manner for almost two years now, it is time to change the default path for MTL family. Link: https://thesofproject.github.io/latest/getting_started/intel_debug/introduction.html#topology-file Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Link: https://patch.msgid.link/20251215130805.31146-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-mtl.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index 7b2533999195..23adc5d765b4 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -47,7 +47,7 @@ static const struct sof_dev_desc mtl_desc = { [SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/mtl", }, .default_tplg_path = { - [SOF_IPC_TYPE_4] = "intel/sof-ace-tplg", + [SOF_IPC_TYPE_4] = "intel/sof-ipc4-tplg", }, .default_fw_filename = { [SOF_IPC_TYPE_4] = "sof-mtl.ri", @@ -77,7 +77,7 @@ static const struct sof_dev_desc arl_desc = { [SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/arl", }, .default_tplg_path = { - [SOF_IPC_TYPE_4] = "intel/sof-ace-tplg", + [SOF_IPC_TYPE_4] = "intel/sof-ipc4-tplg", }, .default_fw_filename = { [SOF_IPC_TYPE_4] = "sof-arl.ri", @@ -107,7 +107,7 @@ static const struct sof_dev_desc arl_s_desc = { [SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/arl-s", }, .default_tplg_path = { - [SOF_IPC_TYPE_4] = "intel/sof-ace-tplg", + [SOF_IPC_TYPE_4] = "intel/sof-ipc4-tplg", }, .default_fw_filename = { [SOF_IPC_TYPE_4] = "sof-arl-s.ri", From c8f3c9fa75ff3822b56b47d5cfa0aaa484040ea8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 15 Dec 2025 12:10:35 +0200 Subject: [PATCH 13/43] ASoC: soc-acpi / SOF: Add best_effort flag to get_function_tplg_files op When there is no fallback possibility available for the function topology use it is better to try to create a profile for the card in best effort manner, leaving out non supported links for example. As an example: some laptops present SSPx-BT link but we don't have fragment yet to support this. If we only have support for functional topology without monolithic fallback then we would fail the card creation. The reason why the monolithic topology works on the same device is that it does not have the SSPx-BT link handled, it is ignored. In case when there is no fallback possibility we should try to create the card with links that we support as best effort instead of failing and leaving the user without a card. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Link: https://patch.msgid.link/20251215101036.9370-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 5 ++++- .../intel/common/sof-function-topology-lib.c | 5 ++++- .../intel/common/sof-function-topology-lib.h | 2 +- sound/soc/sof/topology.c | 18 +++++++++++++++++- 4 files changed, 26 insertions(+), 4 deletions(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 90d73b9bddab..0519afd7217f 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -203,6 +203,8 @@ struct snd_soc_acpi_link_adr { * @mach: the pointer of the machine driver * @prefix: the prefix of the topology file name. Typically, it is the path. * @tplg_files: the pointer of the array of the topology file names. + * @best_effort: ignore non supported links and try to build the card in best effort + * with supported links */ /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { @@ -224,7 +226,8 @@ struct snd_soc_acpi_mach { const u32 tplg_quirk_mask; int (*get_function_tplg_files)(struct snd_soc_card *card, const struct snd_soc_acpi_mach *mach, - const char *prefix, const char ***tplg_files); + const char *prefix, const char ***tplg_files, + bool best_effort); }; #define SND_SOC_ACPI_MAX_CODECS 3 diff --git a/sound/soc/intel/common/sof-function-topology-lib.c b/sound/soc/intel/common/sof-function-topology-lib.c index b10d4794159a..0daa7d83808b 100644 --- a/sound/soc/intel/common/sof-function-topology-lib.c +++ b/sound/soc/intel/common/sof-function-topology-lib.c @@ -28,7 +28,7 @@ enum tplg_device_id { #define SOF_INTEL_PLATFORM_NAME_MAX 4 int sof_sdw_get_tplg_files(struct snd_soc_card *card, const struct snd_soc_acpi_mach *mach, - const char *prefix, const char ***tplg_files) + const char *prefix, const char ***tplg_files, bool best_effort) { struct snd_soc_acpi_mach_params mach_params = mach->mach_params; struct snd_soc_dai_link *dai_link; @@ -87,6 +87,9 @@ int sof_sdw_get_tplg_files(struct snd_soc_card *card, const struct snd_soc_acpi_ dev_dbg(card->dev, "dai_link %s is not supported by separated tplg yet\n", dai_link->name); + if (best_effort) + continue; + return 0; } if (tplg_mask & BIT(tplg_dev)) diff --git a/sound/soc/intel/common/sof-function-topology-lib.h b/sound/soc/intel/common/sof-function-topology-lib.h index e7d0c39d0788..f358f8c52d78 100644 --- a/sound/soc/intel/common/sof-function-topology-lib.h +++ b/sound/soc/intel/common/sof-function-topology-lib.h @@ -10,6 +10,6 @@ #define _SND_SOC_ACPI_INTEL_GET_TPLG_H int sof_sdw_get_tplg_files(struct snd_soc_card *card, const struct snd_soc_acpi_mach *mach, - const char *prefix, const char ***tplg_files); + const char *prefix, const char ***tplg_files, bool best_effort); #endif diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index c1083ea4624a..c76545e70860 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2506,12 +2506,28 @@ int snd_sof_load_topology(struct snd_soc_component *scomp, const char *file) if (!tplg_files) return -ENOMEM; + /* Try to use function topologies if possible */ if (!sof_pdata->disable_function_topology && !disable_function_topology && sof_pdata->machine && sof_pdata->machine->get_function_tplg_files) { + /* + * When the topology name contains 'dummy' word, it means that + * there is no fallback option to monolithic topology in case + * any of the function topologies might be missing. + * In this case we should use best effort to form the card, + * ignoring functionalities that we are missing a fragment for. + * + * Note: monolithic topologies also ignore these possibly + * missing functions, so the functionality of the card would be + * identical to the case if there would be a fallback monolithic + * topology created for the configuration. + */ + bool no_fallback = strstr(file, "dummy"); + tplg_cnt = sof_pdata->machine->get_function_tplg_files(scomp->card, sof_pdata->machine, tplg_filename_prefix, - &tplg_files); + &tplg_files, + no_fallback); if (tplg_cnt < 0) { kfree(tplg_files); return tplg_cnt; From 91b7f7d0eedaaa8993e662c4c6db9b3cfe8a2faf Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 15 Dec 2025 12:10:36 +0200 Subject: [PATCH 14/43] ASoC: Intel: soc-acpi-intel-nvl-match: Drop rt722 l3 from the match table Revert "ASoC: Intel: soc-acpi-intel-nvl-match: add rt722 l3 support" NVL should be only using functional topologies for products, no monolithic topologies are planned to be released. In parallel a feature has been landed [1] which allows to remove the entries from the match table for sdca codecs to rely solely on function fragments. This reverts commit 41566e3de40616375e8dfe5455344558b79f9354. Link: https://lore.kernel.org/linux-sound/20251014071335.3844631-1-yung-chuan.liao@linux.intel.com/ Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Link: https://patch.msgid.link/20251215101036.9370-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-nvl-match.c | 49 ------------------- 1 file changed, 49 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-nvl-match.c b/sound/soc/intel/common/soc-acpi-intel-nvl-match.c index 2768dd10aaa0..b8695d47e55b 100644 --- a/sound/soc/intel/common/soc-acpi-intel-nvl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-nvl-match.c @@ -15,49 +15,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_nvl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_nvl_machines); -/* - * Multi-function codecs with three endpoints created for - * headset, amp and dmic functions. - */ -static const struct snd_soc_acpi_endpoint rt_mf_endpoints[] = { - { - .num = 0, - .aggregated = 0, - .group_position = 0, - .group_id = 0, - }, - { - .num = 1, - .aggregated = 0, - .group_position = 0, - .group_id = 0, - }, - { - .num = 2, - .aggregated = 0, - .group_position = 0, - .group_id = 0, - }, -}; - -static const struct snd_soc_acpi_adr_device rt722_3_single_adr[] = { - { - .adr = 0x000330025d072201ull, - .num_endpoints = ARRAY_SIZE(rt_mf_endpoints), - .endpoints = rt_mf_endpoints, - .name_prefix = "rt722" - } -}; - -static const struct snd_soc_acpi_link_adr nvl_rt722_l3[] = { - { - .mask = BIT(3), - .num_adr = ARRAY_SIZE(rt722_3_single_adr), - .adr_d = rt722_3_single_adr, - }, - {} -}; - /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_nvl_sdw_machines[] = { /* mockup tests need to be first */ @@ -79,12 +36,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_nvl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-nvl-rt715-rt711-rt1308-mono.tplg", }, - { - .link_mask = BIT(3), - .links = nvl_rt722_l3, - .drv_name = "sof_sdw", - .sof_tplg_filename = "sof-nvl-rt722.tplg", - }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_nvl_sdw_machines); From 5526c1c6ba1d0913c7dfcbbd6fe1744ea7c55f1e Mon Sep 17 00:00:00 2001 From: Shipei Qu Date: Wed, 17 Dec 2025 10:46:30 +0800 Subject: [PATCH 15/43] ALSA: usb-mixer: us16x08: validate meter packet indices get_meter_levels_from_urb() parses the 64-byte meter packets sent by the device and fills the per-channel arrays meter_level[], comp_level[] and master_level[] in struct snd_us16x08_meter_store. Currently the function derives the channel index directly from the meter packet (MUB2(meter_urb, s) - 1) and uses it to index those arrays without validating the range. If the packet contains a negative or out-of-range channel number, the driver may write past the end of these arrays. Introduce a local channel variable and validate it before updating the arrays. We reject negative indices, limit meter_level[] and comp_level[] to SND_US16X08_MAX_CHANNELS, and guard master_level[] updates with ARRAY_SIZE(master_level). Fixes: d2bb390a2081 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk") Reported-by: DARKNAVY (@DarkNavyOrg) Closes: https://lore.kernel.org/tencent_21C112743C44C1A2517FF219@qq.com Signed-off-by: Shipei Qu Link: https://patch.msgid.link/20251217024630.59576-1-qu@darknavy.com Signed-off-by: Takashi Iwai --- sound/usb/mixer_us16x08.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 1c5712c31f5e..f9df40730eff 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -655,17 +655,25 @@ static void get_meter_levels_from_urb(int s, u8 *meter_urb) { int val = MUC2(meter_urb, s) + (MUC3(meter_urb, s) << 8); + int ch = MUB2(meter_urb, s) - 1; + + if (ch < 0) + return; if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 && MUA2(meter_urb, s) == 0x04 && MUB0(meter_urb, s) == 0x62) { - if (MUC0(meter_urb, s) == 0x72) - store->meter_level[MUB2(meter_urb, s) - 1] = val; - if (MUC0(meter_urb, s) == 0xb2) - store->comp_level[MUB2(meter_urb, s) - 1] = val; + if (ch < SND_US16X08_MAX_CHANNELS) { + if (MUC0(meter_urb, s) == 0x72) + store->meter_level[ch] = val; + if (MUC0(meter_urb, s) == 0xb2) + store->comp_level[ch] = val; + } } if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 && - MUA2(meter_urb, s) == 0x02 && MUB0(meter_urb, s) == 0x62) - store->master_level[MUB2(meter_urb, s) - 1] = val; + MUA2(meter_urb, s) == 0x02 && MUB0(meter_urb, s) == 0x62) { + if (ch < ARRAY_SIZE(store->master_level)) + store->master_level[ch] = val; + } } /* Function to retrieve current meter values from the device. From 720eebd514c0c56f1e9da25666b746edf82ff227 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 16 Dec 2025 16:48:12 +0000 Subject: [PATCH 16/43] ALSA: hda/realtek: Add support for HP Trekker Laptop Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with I2C Signed-off-by: Stefan Binding Link: https://patch.msgid.link/20251216164830.832148-2-sbinding@opensource.cirrus.com 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 c8a9b9b15cb4..ec57c075757c 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6795,6 +6795,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8f40, "HP ZBook 8 G2a 14", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8f41, "HP ZBook 8 G2a 16", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8f42, "HP ZBook 8 G2a 14W", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x8f57, "HP Trekker G7JC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8f62, "HP ZBook 8 G2a 16W", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1032, "ASUS VivoBook X513EA", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1034, "ASUS GU605C", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), From 108c422c495dc3232aebad837408cd74d23b1794 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 16 Dec 2025 16:48:13 +0000 Subject: [PATCH 17/43] ALSA: hda/realtek: Add support for HP Clipper Laptop Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with I2C Signed-off-by: Stefan Binding Link: https://patch.msgid.link/20251216164830.832148-3-sbinding@opensource.cirrus.com 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 ec57c075757c..e8f3cdcff0f3 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6771,6 +6771,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8e61, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e62, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e8a, "HP NexusX", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x8e9c, "HP 16 Clipper OmniBook X X360", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e9d, "HP 17 Turbine OmniBook X UMA", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e9e, "HP 17 Turbine OmniBook X UMA", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8eb6, "HP Abe A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), From 70d6df5cb599d92ded120ce4b6ace5d59aa1f817 Mon Sep 17 00:00:00 2001 From: Dirk Su Date: Wed, 17 Dec 2025 10:52:44 +0800 Subject: [PATCH 18/43] ALSA: hda/realtek: fix micmute LED reversed on HP Abe and Bantie Quirk ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO make mute/micmute LEDs on HP Abe and Bantie functional. But the micmute LED's function is reversed, LED will be on when Mic enabled and off when Mic disabled. Create a new function to fix the micmute LED reversed issue. Fixes: b72a6ddf6af2 ("ALSA: hda/realtek: fix mute/micmute LEDs don't work for HP 200 G2i") Signed-off-by: Dirk Su Link: https://patch.msgid.link/20251217025257.44600-1-dirk.su@canonical.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index e8f3cdcff0f3..2bc99a8755c9 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -1656,6 +1656,18 @@ static void alc236_fixup_hp_mute_led_micmute_vref(struct hda_codec *codec, alc236_fixup_hp_micmute_led_vref(codec, fix, action); } +static void alc236_fixup_hp_mute_led_micmute_gpio(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->micmute_led_polarity = 1; + + alc236_fixup_hp_mute_led_coefbit2(codec, fix, action); + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x01); +} + static inline void alc298_samsung_write_coef_pack(struct hda_codec *codec, const unsigned short coefs[2]) { @@ -5326,9 +5338,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO] = { .type = HDA_FIXUP_FUNC, - .v.func = alc236_fixup_hp_mute_led_coefbit2, - .chained = true, - .chain_id = ALC236_FIXUP_HP_GPIO_LED, + .v.func = alc236_fixup_hp_mute_led_micmute_gpio, }, [ALC236_FIXUP_LENOVO_INV_DMIC] = { .type = HDA_FIXUP_FUNC, From 9f5f3583ba423e6eed0a96e4d4b7d808d618f3aa Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Tue, 16 Dec 2025 10:40:42 +0100 Subject: [PATCH 19/43] ASoC: fsl_easrc: fix duplicate debugfs directory error This driver registers two components: asrc and easrc, both attached using the device name as component name. Eventually debugfs directories with identical name are created in soc_init_component_debugfs(), leading to error message: debugfs: '30c90000.easrc' already exists in 'tqm-tlv320aic32' Fix this by adding the debugfs_prefix. Signed-off-by: Alexander Stein Reviewed-by: Fabio Estevam Link: https://patch.msgid.link/20251216094045.623184-2-alexander.stein@ew.tq-group.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_easrc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index f404a39009e1..e64a0d97afd0 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1577,6 +1577,9 @@ static const struct snd_soc_component_driver fsl_easrc_component = { .controls = fsl_easrc_snd_controls, .num_controls = ARRAY_SIZE(fsl_easrc_snd_controls), .legacy_dai_naming = 1, +#ifdef CONFIG_DEBUG_FS + .debugfs_prefix = "easrc", +#endif }; static const struct reg_default fsl_easrc_reg_defaults[] = { From 4de6cea0d8e10c9e3f38ccff7edd45891976e67a Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Tue, 16 Dec 2025 10:40:43 +0100 Subject: [PATCH 20/43] ASoC: fsl_asrc_dma: fix duplicate debugfs directory error This driver registers a component for asrc. This is also used together with easrc, both attached using the device name as component name. Eventually debugfs directories with identical name are created in soc_init_component_debugfs(), leading to error message: debugfs: '30c90000.easrc' already exists in 'tqm-tlv320aic32' Fix this by adding the debugfs_prefix. Signed-off-by: Alexander Stein Reviewed-by: Fabio Estevam Link: https://patch.msgid.link/20251216094045.623184-3-alexander.stein@ew.tq-group.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc_dma.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 1bba48318e2d..7dacc06b2f02 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -473,5 +473,8 @@ struct snd_soc_component_driver fsl_asrc_component = { .pointer = fsl_asrc_dma_pcm_pointer, .pcm_construct = fsl_asrc_dma_pcm_new, .legacy_dai_naming = 1, +#ifdef CONFIG_DEBUG_FS + .debugfs_prefix = "asrc", +#endif }; EXPORT_SYMBOL_GPL(fsl_asrc_component); From 00b960a83c764208b0623089eb70af3685e3906f Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 16 Dec 2025 15:02:01 +0800 Subject: [PATCH 21/43] ASoC: ak4458: remove the reset operation in probe and remove The reset_control handler has the reference count for usage, as there is reset operation in runtime suspend and resume, then reset operation in probe() would cause the reference count of reset not balanced. Previously add reset operation in probe and remove is to fix the compile issue with !CONFIG_PM, as the driver has been update to use RUNTIME_PM_OPS(), so that change can be reverted. Fixes: 1e0dff741b0a ("ASoC: ak4458: remove "reset-gpios" property handler") Signed-off-by: Shengjiu Wang Link: https://patch.msgid.link/20251216070201.358477-1-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/ak4458.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 783d2ef21c11..f81cd8cebdd8 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -783,16 +783,12 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) pm_runtime_enable(&i2c->dev); regcache_cache_only(ak4458->regmap, true); - ak4458_reset(ak4458, false); return 0; } static void ak4458_i2c_remove(struct i2c_client *i2c) { - struct ak4458_priv *ak4458 = i2c_get_clientdata(i2c); - - ak4458_reset(ak4458, true); pm_runtime_disable(&i2c->dev); } From 90ed688792a6b7012b3e8a2f858bc3fe7454d0eb Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Tue, 16 Dec 2025 11:22:45 +0100 Subject: [PATCH 22/43] ASoC: fsl_sai: Add missing registers to cache default Drivers does cache sync during runtime resume, setting all writable registers. Not all writable registers are set in cache default, resulting in the erorr message: fsl-sai 30c30000.sai: using zero-initialized flat cache, this may cause unexpected behavior Fix this by adding missing writable register defaults. Signed-off-by: Alexander Stein Link: https://patch.msgid.link/20251216102246.676181-1-alexander.stein@ew.tq-group.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 86730c214914..2fa14fbdfe1a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1081,6 +1081,7 @@ static const struct reg_default fsl_sai_reg_defaults_ofs0[] = { {FSL_SAI_TDR6, 0}, {FSL_SAI_TDR7, 0}, {FSL_SAI_TMR, 0}, + {FSL_SAI_TTCTL, 0}, {FSL_SAI_RCR1(0), 0}, {FSL_SAI_RCR2(0), 0}, {FSL_SAI_RCR3(0), 0}, @@ -1104,12 +1105,14 @@ static const struct reg_default fsl_sai_reg_defaults_ofs8[] = { {FSL_SAI_TDR6, 0}, {FSL_SAI_TDR7, 0}, {FSL_SAI_TMR, 0}, + {FSL_SAI_TTCTL, 0}, {FSL_SAI_RCR1(8), 0}, {FSL_SAI_RCR2(8), 0}, {FSL_SAI_RCR3(8), 0}, {FSL_SAI_RCR4(8), 0}, {FSL_SAI_RCR5(8), 0}, {FSL_SAI_RMR, 0}, + {FSL_SAI_RTCTL, 0}, {FSL_SAI_MCTL, 0}, {FSL_SAI_MDIV, 0}, }; From 08fd332eeb88515af4f1892d91f6ef4ea7558b71 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Tue, 16 Dec 2025 09:49:30 +0100 Subject: [PATCH 23/43] ASoC: fsl_xcvr: provide regmap names This driver uses multiple regmaps, which will causes name conflicts in debugfs like: debugfs: '30cc0000.xcvr' already exists in 'regmap' Fix this by adding a name for the non-core regmap configurations. Signed-off-by: Alexander Stein Link: https://patch.msgid.link/20251216084931.553328-1-alexander.stein@ew.tq-group.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_xcvr.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 06434b2c9a0f..a268fb81a2f8 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1323,6 +1323,7 @@ static const struct reg_default fsl_xcvr_phy_reg_defaults[] = { }; static const struct regmap_config fsl_xcvr_regmap_phy_cfg = { + .name = "phy", .reg_bits = 8, .reg_stride = 4, .val_bits = 32, @@ -1335,6 +1336,7 @@ static const struct regmap_config fsl_xcvr_regmap_phy_cfg = { }; static const struct regmap_config fsl_xcvr_regmap_pllv0_cfg = { + .name = "pllv0", .reg_bits = 8, .reg_stride = 4, .val_bits = 32, @@ -1345,6 +1347,7 @@ static const struct regmap_config fsl_xcvr_regmap_pllv0_cfg = { }; static const struct regmap_config fsl_xcvr_regmap_pllv1_cfg = { + .name = "pllv1", .reg_bits = 8, .reg_stride = 4, .val_bits = 32, From d05d125f4e123e23c89d002e9922f83cee7716e1 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 16 Dec 2025 17:06:01 +0800 Subject: [PATCH 24/43] ASoC: rt1320: update VC blind write settings This patch updates blind write settings for VC version. Signed-off-by: Shuming Fan Link: https://patch.msgid.link/20251216090601.3955252-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1320-sdw.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt1320-sdw.c b/sound/soc/codecs/rt1320-sdw.c index e3f9b03df3aa..feecef258b65 100644 --- a/sound/soc/codecs/rt1320-sdw.c +++ b/sound/soc/codecs/rt1320-sdw.c @@ -115,7 +115,8 @@ static const struct reg_sequence rt1320_blind_write[] = { static const struct reg_sequence rt1320_vc_blind_write[] = { { 0xc003, 0xe0 }, { 0xe80a, 0x01 }, - { 0xc5c3, 0xf3 }, + { 0xc5c3, 0xf2 }, + { 0xc5c8, 0x03 }, { 0xc057, 0x51 }, { 0xc054, 0x35 }, { 0xca05, 0xd6 }, @@ -126,8 +127,6 @@ static const struct reg_sequence rt1320_vc_blind_write[] = { { 0xc609, 0x40 }, { 0xc046, 0xff }, { 0xc045, 0xff }, - { 0xda81, 0x14 }, - { 0xda8d, 0x14 }, { 0xc044, 0xff }, { 0xc043, 0xff }, { 0xc042, 0xff }, @@ -136,8 +135,8 @@ static const struct reg_sequence rt1320_vc_blind_write[] = { { 0xcc10, 0x01 }, { 0xc700, 0xf0 }, { 0xc701, 0x13 }, - { 0xc901, 0x09 }, - { 0xc900, 0xd0 }, + { 0xc901, 0x04 }, + { 0xc900, 0x73 }, { 0xde03, 0x05 }, { 0xdd0b, 0x0d }, { 0xdd0a, 0xff }, @@ -153,6 +152,7 @@ static const struct reg_sequence rt1320_vc_blind_write[] = { { 0xf082, 0xff }, { 0xf081, 0xff }, { 0xf080, 0xff }, + { 0xe801, 0x01 }, { 0xe802, 0xf8 }, { 0xe803, 0xbe }, { 0xc003, 0xc0 }, @@ -202,7 +202,7 @@ static const struct reg_sequence rt1320_vc_blind_write[] = { { 0x3fc2bfc3, 0x00 }, { 0x3fc2bfc2, 0x00 }, { 0x3fc2bfc1, 0x00 }, - { 0x3fc2bfc0, 0x03 }, + { 0x3fc2bfc0, 0x07 }, { 0x0000d486, 0x43 }, { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x00 }, { 0x1000db00, 0x07 }, @@ -241,9 +241,7 @@ static const struct reg_sequence rt1320_vc_blind_write[] = { { 0x1000db21, 0x00 }, { 0x1000db22, 0x00 }, { 0x1000db23, 0x00 }, - { 0x0000d540, 0x01 }, - { 0x0000c081, 0xfc }, - { 0x0000f01e, 0x80 }, + { 0x0000d540, 0x21 }, { 0xc01b, 0xfc }, { 0xc5d1, 0x89 }, { 0xc5d8, 0x0a }, From fa43ab13c59f4c047c479673792ed033ab567c65 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Tue, 16 Dec 2025 16:16:56 +0900 Subject: [PATCH 25/43] ASoC: fsl-asoc-card: Use of_property_present() for non-boolean properties The use of of_property_read_bool() for non-boolean properties is deprecated in favor of of_property_present() when testing for property presence. Otherwise there'll be kernel warning: [ 29.018081] OF: /sound-wm8962: Read of boolean property 'hp-det-gpios' with a value. Signed-off-by: Chancel Liu Link: https://patch.msgid.link/20251216071656.648412-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 2c7eb0baa0f3..70a6159430ed 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -1045,8 +1045,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) * The notifier is initialized in snd_soc_card_jack_new(), then * snd_soc_jack_notifier_register can be called. */ - if (of_property_read_bool(np, "hp-det-gpios") || - of_property_read_bool(np, "hp-det-gpio") /* deprecated */) { + if (of_property_present(np, "hp-det-gpios") || + of_property_present(np, "hp-det-gpio") /* deprecated */) { ret = simple_util_init_jack(&priv->card, &priv->hp_jack, 1, NULL, "Headphone Jack"); if (ret) @@ -1055,8 +1055,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb); } - if (of_property_read_bool(np, "mic-det-gpios") || - of_property_read_bool(np, "mic-det-gpio") /* deprecated */) { + if (of_property_present(np, "mic-det-gpios") || + of_property_present(np, "mic-det-gpio") /* deprecated */) { ret = simple_util_init_jack(&priv->card, &priv->mic_jack, 0, NULL, "Mic Jack"); if (ret) From e43aefb771e82f2e13a435c37ef55813f4140f93 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 17 Dec 2025 16:32:26 +0000 Subject: [PATCH 26/43] ASoC: Intel: soc-acpi-intel-mtl-match: Add 6 amp CS35L56 with feedback Add a match for 6x CS35L56, 3x on link 0 and 3x on link 1. To support the CDB35L56-EIGHT-C board using 6 amps. This is the same as the existing 8-amp configuration mtl_cs35l56_x8_link0_link1_fb, but reduced to 6 amps. Signed-off-by: Stefan Binding Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20251217163227.1186373-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 42 +++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index ec9fd8486c05..f0cf956ffb82 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -699,6 +699,27 @@ static const struct snd_soc_acpi_adr_device cs35l56_1_fb_adr[] = { }, }; +static const struct snd_soc_acpi_adr_device cs35l56_6amp_1_fb_adr[] = { + { + .adr = 0x00013701FA355601ull, + .num_endpoints = ARRAY_SIZE(cs35l56_r_fb_endpoints), + .endpoints = cs35l56_r_fb_endpoints, + .name_prefix = "AMP6" + }, + { + .adr = 0x00013601FA355601ull, + .num_endpoints = ARRAY_SIZE(cs35l56_3_fb_endpoints), + .endpoints = cs35l56_3_fb_endpoints, + .name_prefix = "AMP5" + }, + { + .adr = 0x00013501FA355601ull, + .num_endpoints = ARRAY_SIZE(cs35l56_5_fb_endpoints), + .endpoints = cs35l56_5_fb_endpoints, + .name_prefix = "AMP4" + }, +}; + static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = { { .adr = 0x00023201FA355601ull, @@ -1069,6 +1090,21 @@ static const struct snd_soc_acpi_link_adr mtl_cs35l56_x8_link0_link1_fb[] = { {} }; +static const struct snd_soc_acpi_link_adr mtl_cs35l56_x6_link0_link1_fb[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs35l56_6amp_1_fb_adr), + .adr_d = cs35l56_6amp_1_fb_adr, + }, + { + .mask = BIT(0), + /* First 3 amps in cs35l56_0_fb_adr */ + .num_adr = 3, + .adr_d = cs35l56_0_fb_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr mtl_cs35l63_x2_link1_link3_fb[] = { { .mask = BIT(3), @@ -1189,6 +1225,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .sof_tplg_filename = "sof-mtl-cs35l56-l01-fb8.tplg", .get_function_tplg_files = sof_sdw_get_tplg_files, }, + { + .link_mask = BIT(0) | BIT(1), + .links = mtl_cs35l56_x6_link0_link1_fb, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-cs35l56-l01-fb6.tplg" + }, { .link_mask = BIT(0), .links = mtl_cs42l43_l0, From 26f637fa08879152b9c82417d0d4096019b386ff Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 17 Dec 2025 16:32:27 +0000 Subject: [PATCH 27/43] ASoC: Intel: soc-acpi-intel-mtl-match: Add 6 amp CS35L63 with feedback Add match for 6x CS35L63, 3x on link 2 and 3x on link 3. This is to support 6 amps on the CDB35L63-CB8 board. Signed-off-by: Stefan Binding Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20251217163227.1186373-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 62 +++++++++++++++++++ 1 file changed, 62 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index f0cf956ffb82..1270ee21ee72 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -720,6 +720,48 @@ static const struct snd_soc_acpi_adr_device cs35l56_6amp_1_fb_adr[] = { }, }; +static const struct snd_soc_acpi_adr_device cs35l63_6amp_3_fb_adr[] = { + { + .adr = 0x00033001FA356301ull, + .num_endpoints = ARRAY_SIZE(cs35l56_l_fb_endpoints), + .endpoints = cs35l56_l_fb_endpoints, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033201FA356301ull, + .num_endpoints = ARRAY_SIZE(cs35l56_2_fb_endpoints), + .endpoints = cs35l56_2_fb_endpoints, + .name_prefix = "AMP3" + }, + { + .adr = 0x00033401FA356301ull, + .num_endpoints = ARRAY_SIZE(cs35l56_4_fb_endpoints), + .endpoints = cs35l56_4_fb_endpoints, + .name_prefix = "AMP5" + }, +}; + +static const struct snd_soc_acpi_adr_device cs35l63_6amp_2_fb_adr[] = { + { + .adr = 0x00023101FA356301ull, + .num_endpoints = ARRAY_SIZE(cs35l56_r_fb_endpoints), + .endpoints = cs35l56_r_fb_endpoints, + .name_prefix = "AMP2" + }, + { + .adr = 0x00023301FA356301ull, + .num_endpoints = ARRAY_SIZE(cs35l56_3_fb_endpoints), + .endpoints = cs35l56_3_fb_endpoints, + .name_prefix = "AMP4" + }, + { + .adr = 0x00023501FA356301ull, + .num_endpoints = ARRAY_SIZE(cs35l56_5_fb_endpoints), + .endpoints = cs35l56_5_fb_endpoints, + .name_prefix = "AMP6" + }, +}; + static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = { { .adr = 0x00023201FA355601ull, @@ -1105,6 +1147,20 @@ static const struct snd_soc_acpi_link_adr mtl_cs35l56_x6_link0_link1_fb[] = { {} }; +static const struct snd_soc_acpi_link_adr mtl_cs35l63_x6_link2_link3_fb[] = { + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l63_6amp_3_fb_adr), + .adr_d = cs35l63_6amp_3_fb_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l63_6amp_2_fb_adr), + .adr_d = cs35l63_6amp_2_fb_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr mtl_cs35l63_x2_link1_link3_fb[] = { { .mask = BIT(3), @@ -1244,6 +1300,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-cs35l56-l01-fb8.tplg", }, + { + .link_mask = BIT(2) | BIT(3), + .links = mtl_cs35l63_x6_link2_link3_fb, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-cs35l56-l01-fb6.tplg", + }, { .link_mask = GENMASK(3, 0), .links = mtl_3_in_1_sdca, From f7cede182c963720edd1e5fb50ea4f1c7eafa30e Mon Sep 17 00:00:00 2001 From: Antheas Kapenekakis Date: Tue, 16 Dec 2025 22:17:14 +0100 Subject: [PATCH 28/43] ALSA: hda/realtek: Add Asus quirk for TAS amplifiers By default, these devices use the quirk ALC294_FIXUP_ASUS_SPK. Not using it causes the headphone jack to stop working. Therefore, introduce a new quirk ALC287_FIXUP_TXNW2781_I2C_ASUS that binds to the TAS amplifier while using that quirk. Cc: stable@kernel.org Fixes: 18a4895370a7 ("ALSA: hda/realtek: Add match for ASUS Xbox Ally projects") Signed-off-by: Antheas Kapenekakis Link: https://patch.msgid.link/20251216211714.1116898-1-lkml@antheas.dev 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 2bc99a8755c9..355f11827531 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -3765,6 +3765,7 @@ enum { ALC295_FIXUP_DELL_TAS2781_I2C, ALC245_FIXUP_TAS2781_SPI_2, ALC287_FIXUP_TXNW2781_I2C, + ALC287_FIXUP_TXNW2781_I2C_ASUS, ALC287_FIXUP_YOGA7_14ARB7_I2C, ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT, @@ -6063,6 +6064,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, + [ALC287_FIXUP_TXNW2781_I2C_ASUS] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_txnw_i2c, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_SPK, + }, [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { .type = HDA_FIXUP_FUNC, .v.func = yoga7_14arb7_fixup_i2c, @@ -6839,8 +6846,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1314, "ASUS GA605K", ALC285_FIXUP_ASUS_GA605K_HEADSET_MIC), - SND_PCI_QUIRK(0x1043, 0x1384, "ASUS RC73XA", ALC287_FIXUP_TXNW2781_I2C), - SND_PCI_QUIRK(0x1043, 0x1394, "ASUS RC73YA", ALC287_FIXUP_TXNW2781_I2C), + SND_PCI_QUIRK(0x1043, 0x1384, "ASUS RC73XA", ALC287_FIXUP_TXNW2781_I2C_ASUS), + SND_PCI_QUIRK(0x1043, 0x1394, "ASUS RC73YA", ALC287_FIXUP_TXNW2781_I2C_ASUS), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650PY/PZ/PV/PU/PYV/PZV/PIV/PVV", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), From 095d621141826a2841dae85b52c784c147ea99d3 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 16 Dec 2025 13:49:20 +0000 Subject: [PATCH 29/43] ASoC: ops: fix snd_soc_get_volsw for sx controls SX controls are currently broken, since the clamp introduced in commit a0ce874cfaaa ("ASoC: ops: improve snd_soc_get_volsw") does not handle SX controls, for example where the min value in the clamp is greater than the max value in the clamp. Add clamp parameter to prevent clamping in SX controls. The nature of SX controls mean that it wraps around 0, with a variable number of bits, therefore clamping the value becomes complicated and prone to error. Fixes 35 kunit tests for soc_ops_test_access. Fixes: a0ce874cfaaa ("ASoC: ops: improve snd_soc_get_volsw") Co-developed-by: Charles Keepax Signed-off-by: Stefan Binding Tested-by: Peter Ujfalusi Link: https://patch.msgid.link/20251216134938.788625-1-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 32 ++++++++++++++++++++------------ 1 file changed, 20 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index ce86978c158d..624e9269fc25 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -111,7 +111,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); static int sdca_soc_q78_reg_to_ctl(struct soc_mixer_control *mc, unsigned int reg_val, - unsigned int mask, unsigned int shift, int max) + unsigned int mask, unsigned int shift, int max, + bool sx) { int val = reg_val; @@ -141,20 +142,26 @@ static unsigned int sdca_soc_q78_ctl_to_reg(struct soc_mixer_control *mc, int va } static int soc_mixer_reg_to_ctl(struct soc_mixer_control *mc, unsigned int reg_val, - unsigned int mask, unsigned int shift, int max) + unsigned int mask, unsigned int shift, int max, + bool sx) { int val = (reg_val >> shift) & mask; if (mc->sign_bit) val = sign_extend32(val, mc->sign_bit); - val = clamp(val, mc->min, mc->max); - val -= mc->min; + if (sx) { + val -= mc->min; // SX controls intentionally can overflow here + val = min_t(unsigned int, val & mask, max); + } else { + val = clamp(val, mc->min, mc->max); + val -= mc->min; + } if (mc->invert) val = max - val; - return val & mask; + return val; } static unsigned int soc_mixer_ctl_to_reg(struct soc_mixer_control *mc, int val, @@ -280,9 +287,10 @@ static int soc_put_volsw(struct snd_kcontrol *kcontrol, static int soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc, int mask, int max) + struct soc_mixer_control *mc, int mask, int max, bool sx) { - int (*reg_to_ctl)(struct soc_mixer_control *, unsigned int, unsigned int, unsigned int, int); + int (*reg_to_ctl)(struct soc_mixer_control *, unsigned int, unsigned int, + unsigned int, int, bool); struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); unsigned int reg_val; int val; @@ -293,16 +301,16 @@ static int soc_get_volsw(struct snd_kcontrol *kcontrol, reg_to_ctl = soc_mixer_reg_to_ctl; reg_val = snd_soc_component_read(component, mc->reg); - val = reg_to_ctl(mc, reg_val, mask, mc->shift, max); + val = reg_to_ctl(mc, reg_val, mask, mc->shift, max, sx); ucontrol->value.integer.value[0] = val; if (snd_soc_volsw_is_stereo(mc)) { if (mc->reg == mc->rreg) { - val = reg_to_ctl(mc, reg_val, mask, mc->rshift, max); + val = reg_to_ctl(mc, reg_val, mask, mc->rshift, max, sx); } else { reg_val = snd_soc_component_read(component, mc->rreg); - val = reg_to_ctl(mc, reg_val, mask, mc->shift, max); + val = reg_to_ctl(mc, reg_val, mask, mc->shift, max, sx); } ucontrol->value.integer.value[1] = val; @@ -371,7 +379,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; unsigned int mask = soc_mixer_mask(mc); - return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max - mc->min); + return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max - mc->min, false); } EXPORT_SYMBOL_GPL(snd_soc_get_volsw); @@ -413,7 +421,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; unsigned int mask = soc_mixer_sx_mask(mc); - return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max); + return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max, true); } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); From 17753d1755a589659433ff4ead595f2bb7f695a8 Mon Sep 17 00:00:00 2001 From: Chris Chiu Date: Thu, 18 Dec 2025 06:22:51 +0000 Subject: [PATCH 30/43] ALSA: hda/realtek: fix PCI SSID for one of the HP 200 G2i laptop The PCI subsystem ID of the HP machine Abe A6U should be 0x8ee7 instead of 0x8eb7. Fixes: a30fa8122222 ("ALSA: hda/realtek: fix mute/micmute LEDs don't work for more HP laptops") Signed-off-by: Chris Chiu Link: https://patch.msgid.link/20251218062251.2039592-1-chris.chiu@canonical.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 355f11827531..1de46c06f8c2 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6792,7 +6792,6 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8e9d, "HP 17 Turbine OmniBook X UMA", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e9e, "HP 17 Turbine OmniBook X UMA", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8eb6, "HP Abe A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), - SND_PCI_QUIRK(0x103c, 0x8eb7, "HP Abe A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8eb8, "HP Abe A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8ec1, "HP 200 G2i", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8ec4, "HP Bantie I6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), @@ -6808,6 +6807,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8eda, "HP ZBook Firefly 16W", ALC245_FIXUP_HP_TAS2781_SPI_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8ee4, "HP Bantie A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8ee5, "HP Bantie A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), + SND_PCI_QUIRK(0x103c, 0x8ee7, "HP Abe A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8f0c, "HP ZBook X G2i 16W", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8f0e, "HP ZBook X G2i 16W", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8f40, "HP ZBook 8 G2a 14", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), From e4ca5ecc3c411f2fe970369f55bb72ac96adea85 Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Fri, 19 Dec 2025 11:49:02 +0800 Subject: [PATCH 31/43] ASoC: Intel: sof_sdw: shift SSP BT mask bits. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The SSP BT mask bits overlapped with SOC_SDW_CODEC_SPKR, SOC_SDW_SIDECAR_AMPS, and SOC_SDW_CODEC_MIC BIT[15–17] in sdw_utils.h. Shift the SSP BT mask bits to a higher range to eliminate the conflict. Signed-off-by: Mac Chiang Signed-off-by: Bard Liao Link: https://patch.msgid.link/20251219034902.3630537-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_common.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 3aa1dcec5172..5390f0a749d6 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -46,11 +46,11 @@ enum { #define SOC_SDW_NO_AGGREGATION BIT(14) /* BT audio offload: reserve 3 bits for future */ -#define SOF_BT_OFFLOAD_SSP_SHIFT 15 -#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(17, 15)) +#define SOF_BT_OFFLOAD_SSP_SHIFT 18 +#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(20, 18)) #define SOF_BT_OFFLOAD_SSP(quirk) \ (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) -#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(18) +#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(21) struct intel_mc_ctx { struct sof_hdmi_private hdmi; From a0c8ee09f94ba5a29ee5f7557eb2bc100d5e739a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 19 Dec 2025 11:49:37 +0800 Subject: [PATCH 32/43] ASoC: SOF: Intel: add -bt tplg suffix if BT is present MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We need to distinguish the topologies with and without BT PCM. Signed-off-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20251219034937.3630569-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index c1518dbee1b7..0bb85f92e106 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1549,6 +1549,7 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) * name string if quirk flag is set. */ if (mach) { + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); bool tplg_fixup = false; bool dmic_fixup = false; @@ -1598,6 +1599,18 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) sof_pdata->tplg_filename = tplg_filename; } + if (tplg_fixup && mach->mach_params.bt_link_mask && + chip->hw_ip_version >= SOF_INTEL_ACE_4_0) { + int bt_port = fls(mach->mach_params.bt_link_mask) - 1; + + tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, "%s-ssp%d-bt", + sof_pdata->tplg_filename, bt_port); + if (!tplg_filename) + return NULL; + + sof_pdata->tplg_filename = tplg_filename; + } + if (mach->link_mask) { mach->mach_params.links = mach->links; mach->mach_params.link_mask = mach->link_mask; @@ -1609,7 +1622,6 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) if (tplg_fixup && mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER && mach->mach_params.i2s_link_mask) { - const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); int ssp_num; int mclk_mask; From 54fa71f5f965fa3ec8846cef9d1154bcb2ba2850 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Fri, 19 Dec 2025 15:27:13 +0100 Subject: [PATCH 33/43] ASoC: codecs: pm4125: drop bogus container_of() error handling The dev_to_sdw_dev() helper uses container_of() to return the containing soundwire device structure of its pointer argument and will never return NULL. Fixes: 8ad529484937 ("ASoC: codecs: add new pm4125 audio codec driver") Cc: Alexey Klimov Signed-off-by: Johan Hovold Acked-by: Alexey Klimov Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20251219142715.19254-2-johan@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/pm4125.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/pm4125.c b/sound/soc/codecs/pm4125.c index 8bc3b9994019..4798cd8be9f8 100644 --- a/sound/soc/codecs/pm4125.c +++ b/sound/soc/codecs/pm4125.c @@ -1537,13 +1537,7 @@ static int pm4125_bind(struct device *dev) pm4125->sdw_priv[AIF1_CAP] = dev_get_drvdata(pm4125->txdev); pm4125->sdw_priv[AIF1_CAP]->pm4125 = pm4125; - pm4125->tx_sdw_dev = dev_to_sdw_dev(pm4125->txdev); - if (!pm4125->tx_sdw_dev) { - dev_err(dev, "could not get txslave with matching of dev\n"); - ret = -EINVAL; - goto error_put_tx; - } /* * As TX is the main CSR reg interface, which should not be suspended first. From 61a50d6f58b41e8a7e68d8fc8fc6bfbe30d790d8 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Fri, 19 Dec 2025 15:27:14 +0100 Subject: [PATCH 34/43] ASoC: codecs: wcd937x: drop bogus container_of() error handling The dev_to_sdw_dev() helper uses container_of() to return the containing soundwire device structure of its pointer argument and will never return NULL. Fixes: 9be3ec196da4 ("ASoC: codecs: wcd937x: add wcd937x codec driver") Cc: Prasad Kumpatla Signed-off-by: Johan Hovold Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20251219142715.19254-3-johan@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index f4dbcf04be49..10a2d598caa7 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -2763,11 +2763,6 @@ static int wcd937x_bind(struct device *dev) wcd937x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd937x->txdev); wcd937x->sdw_priv[AIF1_CAP]->wcd937x = wcd937x; wcd937x->tx_sdw_dev = dev_to_sdw_dev(wcd937x->txdev); - if (!wcd937x->tx_sdw_dev) { - dev_err(dev, "could not get txslave with matching of dev\n"); - ret = -EINVAL; - goto err_put_txdev; - } /* * As TX is the main CSR reg interface, which should not be suspended first. From 870b10f61d527fec594e0888f18cc4f32c47433d Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Fri, 19 Dec 2025 15:27:15 +0100 Subject: [PATCH 35/43] ASoC: soc_sdw_utils: drop bogus container_of() error handling The dev_to_sdw_dev() helper uses container_of() to return the containing soundwire device structure of its pointer argument and will never return NULL. Fixes: 4f8ef33dd44a ("ASoC: soc_sdw_utils: skip the endpoint that doesn't present") Cc: Bard Liao Signed-off-by: Johan Hovold Reviewed-by: Konrad Dybcio Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20251219142715.19254-4-johan@kernel.org Signed-off-by: Mark Brown --- sound/soc/sdw_utils/soc_sdw_utils.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index f169d95895ea..bf382aa07e92 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1414,10 +1414,6 @@ static int is_sdca_endpoint_present(struct device *dev, } slave = dev_to_sdw_dev(sdw_dev); - if (!slave) { - ret = -EINVAL; - goto put_device; - } /* Make sure BIOS provides SDCA properties */ if (!slave->sdca_data.interface_revision) { From 97af54473f2a79f663bd14d7c75e97d04bd0e283 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Fri, 19 Dec 2025 15:24:12 +0100 Subject: [PATCH 36/43] ASoC: codecs: pm4125: clean up bind() device reference handling A recent change fixed a couple of device leaks on component bind failure and on unbind but did so in a confusing way by adding misleading initialisations at bind() and bogus NULL checks at unbind(). Cc: Ma Ke Signed-off-by: Johan Hovold Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20251219142412.19043-1-johan@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/pm4125.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/pm4125.c b/sound/soc/codecs/pm4125.c index 8bc3b9994019..43dcafff6c77 100644 --- a/sound/soc/codecs/pm4125.c +++ b/sound/soc/codecs/pm4125.c @@ -1505,10 +1505,6 @@ static int pm4125_bind(struct device *dev) struct device_link *devlink; int ret; - /* Initialize device pointers to NULL for safe cleanup */ - pm4125->rxdev = NULL; - pm4125->txdev = NULL; - /* Give the soundwire subdevices some more time to settle */ usleep_range(15000, 15010); @@ -1624,11 +1620,8 @@ static void pm4125_unbind(struct device *dev) device_link_remove(dev, pm4125->rxdev); device_link_remove(pm4125->rxdev, pm4125->txdev); - /* Release device references acquired in bind */ - if (pm4125->txdev) - put_device(pm4125->txdev); - if (pm4125->rxdev) - put_device(pm4125->rxdev); + put_device(pm4125->txdev); + put_device(pm4125->rxdev); component_unbind_all(dev, pm4125); } From 6c11aa2b4cf767f5ccfe290b2572d53102dbe5ea Mon Sep 17 00:00:00 2001 From: Chen-Yu Tsai Date: Sun, 21 Dec 2025 11:57:13 +0800 Subject: [PATCH 37/43] ASoC: sun4i-spdif: Add missing kerneldoc fields for sun4i_spdif_quirks When sun4i_spdif_quirks was recently expanded, the kerneldoc covering the structure was not expanded to match. This ends up causing a warning when the documents are built. Add the missing fields. Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202501311953.0Ox9CW5w-lkp@intel.com/ Closes: https://lore.kernel.org/oe-kbuild-all/202503060947.QKUUR62l-lkp@intel.com/ Fixes: 0a2319308de8 ("ASoC: sun4i-spdif: Add clock multiplier settings") Fixes: 4a5ac6cd05a7 ("ASoC: sun4i-spdif: Support SPDIF output on A523 family") Signed-off-by: Chen-Yu Tsai Reviewed-by: Marcus Cooper Acked-by: Jernej Skrabec Link: https://patch.msgid.link/20251221035715.1722584-1-wens@kernel.org Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-spdif.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index 2e7ac8ab71bb..1e755a716c63 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -171,6 +171,8 @@ * @reg_dac_txdata: TX FIFO offset for DMA config. * @has_reset: SoC needs reset deasserted. * @val_fctl_ftx: TX FIFO flush bitmask. + * @mclk_multiplier: ratio of internal MCLK divider + * @tx_clk_name: name of TX module clock if split clock design */ struct sun4i_spdif_quirks { unsigned int reg_dac_txdata; From 830988b6cf197e6dcffdfe2008c5738e6c6c3c0f Mon Sep 17 00:00:00 2001 From: Haoxiang Li Date: Sat, 20 Dec 2025 00:28:45 +0800 Subject: [PATCH 38/43] ALSA: ac97: fix a double free in snd_ac97_controller_register() If ac97_add_adapter() fails, put_device() is the correct way to drop the device reference. kfree() is not required. Add kfree() if idr_alloc() fails and in ac97_adapter_release() to do the cleanup. Found by code review. Fixes: 74426fbff66e ("ALSA: ac97: add an ac97 bus") Cc: stable@vger.kernel.org Signed-off-by: Haoxiang Li Link: https://patch.msgid.link/20251219162845.657525-1-lihaoxiang@isrc.iscas.ac.cn Signed-off-by: Takashi Iwai --- sound/ac97/bus.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index f4254703d29f..bb9b795e0226 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -298,6 +298,7 @@ static void ac97_adapter_release(struct device *dev) idr_remove(&ac97_adapter_idr, ac97_ctrl->nr); dev_dbg(&ac97_ctrl->adap, "adapter unregistered by %s\n", dev_name(ac97_ctrl->parent)); + kfree(ac97_ctrl); } static const struct device_type ac97_adapter_type = { @@ -319,7 +320,9 @@ static int ac97_add_adapter(struct ac97_controller *ac97_ctrl) ret = device_register(&ac97_ctrl->adap); if (ret) put_device(&ac97_ctrl->adap); - } + } else + kfree(ac97_ctrl); + if (!ret) { list_add(&ac97_ctrl->controllers, &ac97_controllers); dev_dbg(&ac97_ctrl->adap, "adapter registered by %s\n", @@ -361,14 +364,11 @@ struct ac97_controller *snd_ac97_controller_register( ret = ac97_add_adapter(ac97_ctrl); if (ret) - goto err; + return ERR_PTR(ret); ac97_bus_reset(ac97_ctrl); ac97_bus_scan(ac97_ctrl); return ac97_ctrl; -err: - kfree(ac97_ctrl); - return ERR_PTR(ret); } EXPORT_SYMBOL_GPL(snd_ac97_controller_register); From e340663bbf2a75dae5d4fddf90b49281f5c9df3f Mon Sep 17 00:00:00 2001 From: August Wikerfors Date: Mon, 22 Dec 2025 20:47:04 +0100 Subject: [PATCH 39/43] ALSA: hda/tas2781: properly initialize speaker_id for TAS2563 After speaker id retrieval was refactored to happen in tas2781_read_acpi, devices that do not use a speaker id need a negative speaker_id value instead of NULL, but no initialization was added to the TAS2563 code path. This causes the driver to attempt to load a non-existent firmware file name with a speaker id of 0 ("TAS2XXX38700.bin") instead of the correct file name without a speaker id ("TAS2XXX3870.bin"), resulting in low volume and these dmesg errors: tas2781-hda i2c-INT8866:00: Direct firmware load for TAS2XXX38700.bin failed with error -2 tas2781-hda i2c-INT8866:00: tasdevice_dsp_parser: load TAS2XXX38700.bin error tas2781-hda i2c-INT8866:00: dspfw load TAS2XXX38700.bin error [...] tas2781-hda i2c-INT8866:00: tasdevice_prmg_load: Firmware is NULL Fix this by setting speaker_id to -1 as is done for other models. Fixes: 945865a0ddf3 ("ALSA: hda/tas2781: fix speaker id retrieval for multiple probes") Cc: stable@vger.kernel.org Signed-off-by: August Wikerfors Link: https://patch.msgid.link/20251222194704.87232-1-git@augustwikerfors.se Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index c8619995b1d7..f7a7f216d586 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -111,8 +111,10 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); if (IS_ERR(sub)) { /* No subsys id in older tas2563 projects. */ - if (!strncmp(hid, "INT8866", sizeof("INT8866"))) + if (!strncmp(hid, "INT8866", sizeof("INT8866"))) { + p->speaker_id = -1; goto end_2563; + } dev_err(p->dev, "Failed to get SUBSYS ID.\n"); ret = PTR_ERR(sub); goto err; From 9be25402d8522e16e5ebe84f2b1b6c5de082a388 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Matou=C5=A1=20L=C3=A1nsk=C3=BD?= Date: Wed, 31 Dec 2025 18:12:07 +0100 Subject: [PATCH 40/43] ALSA: hda/realtek: Add quirk for Acer Nitro AN517-55 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add headset mic quirk for Acer Nitro AN517-55. This laptop uses the same audio configuration as the AN515-58 model. Signed-off-by: Matouš Lánský Link: https://patch.msgid.link/20251231171207.76943-1-matouslansky@post.cz 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 1de46c06f8c2..67baf04551cb 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6321,6 +6321,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1025, 0x1534, "Acer Predator PH315-54", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x159c, "Acer Nitro 5 AN515-58", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1597, "Acer Nitro 5 AN517-55", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x1826, "Acer Helios ZPC", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), SND_PCI_QUIRK(0x1025, 0x182c, "Acer Helios ZPD", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), From 9ed7a28225af02b74f61e7880d460db49db83758 Mon Sep 17 00:00:00 2001 From: Ruslan Krupitsa Date: Fri, 2 Jan 2026 02:53:36 +0300 Subject: [PATCH 41/43] ALSA: hda/realtek: add HP Laptop 15s-eq1xxx mute LED quirk HP Laptop 15s-eq1xxx with ALC236 codec does not enable the mute LED automatically. This patch adds a quirk entry for subsystem ID 0x8706 using the ALC236_FIXUP_HP_MUTE_LED_COEFBIT2 fixup, enabling correct mute LED behavior. Signed-off-by: Ruslan Krupitsa Link: https://patch.msgid.link/AS8P194MB112895B8EC2D87D53A876085BBBAA@AS8P194MB1128.EURP194.PROD.OUTLOOK.COM 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 67baf04551cb..61c7372e6307 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6509,6 +6509,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x863e, "HP Spectre x360 15-df1xxx", ALC285_FIXUP_HP_SPECTRE_X360_DF1), SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x86f9, "HP Spectre x360 13-aw0xxx", ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x8706, "HP Laptop 15s-eq1xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), From 47c27c9c9c720bc93fdc69605d0ecd9382e99047 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 7 Jan 2026 22:36:42 +0100 Subject: [PATCH 42/43] ALSA: pcm: Improve the fix for race of buffer access at PCM OSS layer Handle the error code from snd_pcm_buffer_access_lock() in snd_pcm_runtime_buffer_set_silence() function. Found by Alexandros Panagiotou Fixes: 93a81ca06577 ("ALSA: pcm: Fix race of buffer access at PCM OSS layer") Cc: stable@vger.kernel.org # 6.15 Signed-off-by: Jaroslav Kysela Link: https://patch.msgid.link/20260107213642.332954-1-perex@perex.cz Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 +- sound/core/oss/pcm_oss.c | 4 +++- sound/core/pcm_native.c | 9 +++++++-- 3 files changed, 11 insertions(+), 4 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 58fd6e84f961..a7860c047503 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1402,7 +1402,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif -void snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime); +int snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime); /** * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index a82dd155e1d3..b12df5b5ddfc 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1074,7 +1074,9 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) runtime->oss.params = 0; runtime->oss.prepare = 1; runtime->oss.buffer_used = 0; - snd_pcm_runtime_buffer_set_silence(runtime); + err = snd_pcm_runtime_buffer_set_silence(runtime); + if (err < 0) + goto failure; runtime->oss.period_frames = snd_pcm_alsa_frames(substream, oss_period_size); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 68bee40c9ada..932a9bf98cbc 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -730,13 +730,18 @@ static void snd_pcm_buffer_access_unlock(struct snd_pcm_runtime *runtime) } /* fill the PCM buffer with the current silence format; called from pcm_oss.c */ -void snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime) +int snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime) { - snd_pcm_buffer_access_lock(runtime); + int err; + + err = snd_pcm_buffer_access_lock(runtime); + if (err < 0) + return err; if (runtime->dma_area) snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes)); snd_pcm_buffer_access_unlock(runtime); + return 0; } EXPORT_SYMBOL_GPL(snd_pcm_runtime_buffer_set_silence); From b7e26c8bdae70832d7c4b31ec2995b1812a60169 Mon Sep 17 00:00:00 2001 From: Matthew Schwartz Date: Thu, 8 Jan 2026 01:36:50 -0800 Subject: [PATCH 43/43] ALSA: hda/tas2781: Skip UEFI calibration on ASUS ROG Xbox Ally X There is currently an issue with UEFI calibration data parsing for some TAS devices, like the ASUS ROG Xbox Ally X (RC73XA), that causes audio quality issues such as gaps in playback. Until the issue is root caused and fixed, add a quirk to skip using the UEFI calibration data and fall back to using the calibration data provided by the DSP firmware, which restores full speaker functionality on affected devices. Cc: stable@vger.kernel.org # 6.18 Link: https://lore.kernel.org/all/160aef32646c4d5498cbfd624fd683cc@ti.com/ Closes: https://lore.kernel.org/all/0ba100d0-9b6f-4a3b-bffa-61abe1b46cd5@linux.dev/ Suggested-by: Baojun Xu Signed-off-by: Matthew Schwartz Reviewed-by: Antheas Kapenekakis Link: https://patch.msgid.link/20260108093650.1142176-1-matthew.schwartz@linux.dev Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index f7a7f216d586..0e4bda3a544e 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -60,6 +60,7 @@ struct tas2781_hda_i2c_priv { int (*save_calibration)(struct tas2781_hda *h); int hda_chip_id; + bool skip_calibration; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -491,7 +492,8 @@ static void tasdevice_dspfw_init(void *context) /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ - hda_priv->save_calibration(tas_hda); + if (!hda_priv->skip_calibration) + hda_priv->save_calibration(tas_hda); } static void tasdev_fw_ready(const struct firmware *fmw, void *context) @@ -548,6 +550,7 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, void *master_data) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; struct hda_component_parent *parent = master_data; struct hda_component *comp; struct hda_codec *codec; @@ -573,6 +576,14 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, break; } + /* + * Using ASUS ROG Xbox Ally X (RC73XA) UEFI calibration data + * causes audio dropouts during playback, use fallback data + * from DSP firmware as a workaround. + */ + if (codec->core.subsystem_id == 0x10431384) + hda_priv->skip_calibration = true; + pm_runtime_get_sync(dev); comp->dev = dev;