From 2b974284aa073d6e2936f9032e8ad7b99480b5b8 Mon Sep 17 00:00:00 2001 From: "Hendrik v. Raven" Date: Thu, 14 Nov 2024 12:01:25 +0100 Subject: [PATCH 01/16] ASoc: simple-mux: add idle-state support So far the mux changes it state immediately, even when not in use. Allow overriding this behaviour by specifying an optional idle-state. This state is used whenever the mux is powered down, only switching to the selected state on power up. If unspecified it defaults to as-is, maintaining the previous behaviour. Signed-off-by: Hendrik v. Raven Link: https://patch.msgid.link/20241114-simple-mux-idle-state-v2-1-a30cb37d2be2@merzmedtech.de Signed-off-by: Mark Brown --- sound/soc/codecs/simple-mux.c | 39 ++++++++++++++++++++++++++++++++++- 1 file changed, 38 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/simple-mux.c b/sound/soc/codecs/simple-mux.c index 240af0563283..390696440155 100644 --- a/sound/soc/codecs/simple-mux.c +++ b/sound/soc/codecs/simple-mux.c @@ -6,6 +6,7 @@ #include #include +#include #include #include @@ -16,6 +17,7 @@ struct simple_mux { struct gpio_desc *gpiod_mux; unsigned int mux; const char *mux_texts[MUX_TEXT_SIZE]; + unsigned int idle_state; struct soc_enum mux_enum; struct snd_kcontrol_new mux_mux; struct snd_soc_dapm_widget mux_widgets[MUX_WIDGET_SIZE]; @@ -57,6 +59,9 @@ static int simple_mux_control_put(struct snd_kcontrol *kcontrol, priv->mux = ucontrol->value.enumerated.item[0]; + if (priv->idle_state != MUX_IDLE_AS_IS && dapm->bias_level < SND_SOC_BIAS_PREPARE) + return 0; + gpiod_set_value_cansleep(priv->gpiod_mux, priv->mux); return snd_soc_dapm_mux_update_power(dapm, kcontrol, @@ -75,10 +80,33 @@ static unsigned int simple_mux_read(struct snd_soc_component *component, static const struct snd_kcontrol_new simple_mux_mux = SOC_DAPM_ENUM_EXT("Muxer", simple_mux_enum, simple_mux_control_get, simple_mux_control_put); +static int simple_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct simple_mux *priv = snd_soc_component_get_drvdata(c); + + if (priv->idle_state != MUX_IDLE_AS_IS) { + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + gpiod_set_value_cansleep(priv->gpiod_mux, priv->mux); + break; + case SND_SOC_DAPM_POST_PMD: + gpiod_set_value_cansleep(priv->gpiod_mux, priv->idle_state); + break; + default: + break; + } + } + + return 0; +} + static const struct snd_soc_dapm_widget simple_mux_dapm_widgets[MUX_WIDGET_SIZE] = { SND_SOC_DAPM_INPUT("IN1"), SND_SOC_DAPM_INPUT("IN2"), - SND_SOC_DAPM_MUX("MUX", SND_SOC_NOPM, 0, 0, &simple_mux_mux), // see simple_mux_probe() + SND_SOC_DAPM_MUX_E("MUX", SND_SOC_NOPM, 0, 0, &simple_mux_mux, // see simple_mux_probe() + simple_mux_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUTPUT("OUT"), }; @@ -93,6 +121,7 @@ static int simple_mux_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; struct simple_mux *priv; + int ret; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -121,6 +150,14 @@ static int simple_mux_probe(struct platform_device *pdev) /* Overwrite text ("Input 1", "Input 2") if property exists */ of_property_read_string_array(np, "state-labels", priv->mux_texts, MUX_TEXT_SIZE); + ret = of_property_read_u32(np, "idle-state", &priv->idle_state); + if (ret < 0) { + priv->idle_state = MUX_IDLE_AS_IS; + } else if (priv->idle_state != MUX_IDLE_AS_IS && priv->idle_state >= 2) { + dev_err(dev, "invalid idle-state %u\n", priv->idle_state); + return -EINVAL; + } + /* switch to use priv data instead of default */ priv->mux_enum.texts = priv->mux_texts; priv->mux_mux.private_value = (unsigned long)&priv->mux_enum; -- 2.51.0 From 3b7e11a0116c30848d44429650ad80f9cc3bd963 Mon Sep 17 00:00:00 2001 From: "Hendrik v. Raven" Date: Thu, 14 Nov 2024 12:01:26 +0100 Subject: [PATCH 02/16] ASoC: dt-bindings: simple-mux: add idle-state property simple-mux immediately activates the new output, even when it is powered down. This can be undesirable in some cases, for example when a mechanical relais is used. Adds "idle-state" property from the mux controller to select the output state to be used when the mux is powered down. Signed-off-by: Hendrik v. Raven Link: https://patch.msgid.link/20241114-simple-mux-idle-state-v2-2-a30cb37d2be2@merzmedtech.de Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/simple-audio-mux.yaml | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml index 194ac1d4f4f5..9b1bda4852e1 100644 --- a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -29,6 +29,10 @@ properties: $ref: /schemas/types.yaml#/definitions/string-array maxItems: 2 + idle-state: + description: If present specifies the state when the mux is powered down + $ref: /schemas/mux/mux-controller.yaml#/properties/idle-state + sound-name-prefix: true required: @@ -43,4 +47,5 @@ examples: compatible = "simple-audio-mux"; mux-gpios = <&gpio 3 0>; state-labels = "Label_A", "Label_B"; + idle-state = <0>; }; -- 2.51.0 From c48a4497356f701f94f1951626637ae240af909e Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 13 Nov 2024 18:57:13 +0100 Subject: [PATCH 03/16] ASoC: sma1307: fix uninitialized variable refence When firmware loading is disabled, gcc warns that the local 'fw' variable fails to get initialized: sound/soc/codecs/sma1307.c: In function 'sma1307_setting_loaded.isra': sound/soc/codecs/sma1307.c:1717:12: error: 'fw' is used uninitialized [-Werror=uninitialized] 1717 | if (!fw) { | ^ sound/soc/codecs/sma1307.c:1712:32: note: 'fw' was declared here 1712 | const struct firmware *fw; Check the return code from request_firmware() to ensure that the firmware is correctly set, and drop the incorrect release_firmware() on that uninitialized data. Fixes: 576c57e6b4c1 ("ASoC: sma1307: Add driver for Iron Device SMA1307") Signed-off-by: Arnd Bergmann Link: https://patch.msgid.link/20241113175734.2443315-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/sma1307.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/sma1307.c b/sound/soc/codecs/sma1307.c index 81638768ac12..f2cea6186d98 100644 --- a/sound/soc/codecs/sma1307.c +++ b/sound/soc/codecs/sma1307.c @@ -1711,13 +1711,13 @@ static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *fil { const struct firmware *fw; int *data, size, offset, num_mode; + int ret; - request_firmware(&fw, file, sma1307->dev); + ret = request_firmware(&fw, file, sma1307->dev); - if (!fw) { - dev_err(sma1307->dev, "%s: failed to read \"%s\"\n", - __func__, setting_file); - release_firmware(fw); + if (ret) { + dev_err(sma1307->dev, "%s: failed to read \"%s\": %pe\n", + __func__, setting_file, ERR_PTR(ret)); sma1307->set.status = false; return; } else if ((fw->size) < SMA1307_SETTING_HEADER_SIZE) { -- 2.51.0 From ba888450828befb0607219f34c03aa8645625447 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Thu, 14 Nov 2024 11:28:51 +0100 Subject: [PATCH 04/16] ASoC: stm32: dfsdm: change rate upper limits Increase rate upper limit to 192kHz to reflect the rate range actually supported by the STM32 DFSDM peripheral. Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20241114102851.2497942-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 78bd817af839..c914d1c46850 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -142,7 +142,7 @@ static const struct snd_soc_dai_driver stm32_adfsdm_dai = { SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, - .rate_max = 48000, + .rate_max = 192000, }, .ops = &stm32_adfsdm_dai_ops, }; -- 2.51.0 From a59360466a712d416f8cddfa4e52e118c53aa3a3 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 13 Nov 2024 14:44:17 +0800 Subject: [PATCH 05/16] ASoC: Intel: soc-acpi-intel-lnl-match: add rt712_vb + rt1320 support MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Realtek Gen6 AIOC supports rt712_vb on SoundWire link 2 and rt1320 on SoundWire link 1. Signed-off-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Liam Girdwood Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20241113064418.162592-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-lnl-match.c | 65 +++++++++++++++++++ 1 file changed, 65 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 094ed4b27cb0..98a9c36d7a4c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -8,6 +8,7 @@ #include #include +#include "soc-acpi-intel-sdca-quirks.h" #include "soc-acpi-intel-sdw-mockup-match.h" struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_machines[] = { @@ -90,6 +91,30 @@ static const struct snd_soc_acpi_endpoint rt722_endpoints[] = { }, }; +static const struct snd_soc_acpi_endpoint jack_amp_g1_dmic_endpoints_endpoints[] = { + /* Jack Endpoint */ + { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + /* Amp Endpoint, work as spk_l_endpoint */ + { + .num = 1, + .aggregated = 1, + .group_position = 0, + .group_id = 1, + }, + /* DMIC Endpoint */ + { + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { { /* Jack Playback Endpoint */ .num = 0, @@ -198,6 +223,15 @@ static const struct snd_soc_acpi_adr_device rt1712_3_single_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt712_vb_2_group1_adr[] = { + { + .adr = 0x000230025D071201ull, + .num_endpoints = ARRAY_SIZE(jack_amp_g1_dmic_endpoints_endpoints), + .endpoints = jack_amp_g1_dmic_endpoints_endpoints, + .name_prefix = "rt712" + } +}; + static const struct snd_soc_acpi_adr_device rt722_0_single_adr[] = { { .adr = 0x000030025d072201ull, @@ -252,6 +286,15 @@ static const struct snd_soc_acpi_adr_device rt1318_2_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1320_1_group1_adr[] = { + { + .adr = 0x000130025D132001ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "rt1320-1" + } +}; + static const struct snd_soc_acpi_adr_device rt713_0_adr[] = { { .adr = 0x000031025D071301ull, @@ -410,6 +453,21 @@ static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_l0_rt1318_l1[] = { {} }; +static const struct snd_soc_acpi_link_adr lnl_sdw_rt712_vb_l2_rt1320_l1[] = { + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt712_vb_2_group1_adr), + .adr_d = rt712_vb_2_group1_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1320_1_group1_adr), + .adr_d = rt1320_1_group1_adr, + }, + {} +}; + +/* this table is used when there is no I2S codec present */ /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { /* mockup tests need to be first */ @@ -485,6 +543,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt713-l0-rt1318-l1.tplg" }, + { + .link_mask = BIT(1) | BIT(2), + .links = lnl_sdw_rt712_vb_l2_rt1320_l1, + .drv_name = "sof_sdw", + .machine_check = snd_soc_acpi_intel_sdca_is_device_rt712_vb, + .sof_tplg_filename = "sof-lnl-rt712-l2-rt1320-l1.tplg" + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines); -- 2.51.0 From 82ff5abc2edcfba0c0f1a1be807795e2876f46e9 Mon Sep 17 00:00:00 2001 From: Jonas Karlman Date: Fri, 15 Nov 2024 04:43:44 +0000 Subject: [PATCH 06/16] ASoC: hdmi-codec: reorder channel allocation list The ordering in hdmi_codec_get_ch_alloc_table_idx() results in wrong channel allocation for a number of cases, e.g. when ELD reports FL|FR|LFE|FC|RL|RR or FL|FR|LFE|FC|RL|RR|RC|RLC|RRC: ca_id 0x01 with speaker mask FL|FR|LFE is selected instead of ca_id 0x03 with speaker mask FL|FR|LFE|FC for 4 channels and ca_id 0x04 with speaker mask FL|FR|RC gets selected instead of ca_id 0x0b with speaker mask FL|FR|LFE|FC|RL|RR for 6 channels Fix this by reordering the channel allocation list with most specific speaker masks at the top. Signed-off-by: Jonas Karlman Signed-off-by: Christian Hewitt Link: https://patch.msgid.link/20241115044344.3510979-1-christianshewitt@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 140 +++++++++++++++++++--------------- 1 file changed, 77 insertions(+), 63 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 74caae52e127..d9df29a26f4f 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -185,84 +185,97 @@ static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { /* * hdmi_codec_channel_alloc: speaker configuration available for CEA * - * This is an ordered list that must match with hdmi_codec_8ch_chmaps struct + * This is an ordered list where ca_id must exist in hdmi_codec_8ch_chmaps * The preceding ones have better chances to be selected by * hdmi_codec_get_ch_alloc_table_idx(). */ static const struct hdmi_codec_cea_spk_alloc hdmi_codec_channel_alloc[] = { { .ca_id = 0x00, .n_ch = 2, - .mask = FL | FR}, - /* 2.1 */ - { .ca_id = 0x01, .n_ch = 4, - .mask = FL | FR | LFE}, - /* Dolby Surround */ + .mask = FL | FR }, + { .ca_id = 0x03, .n_ch = 4, + .mask = FL | FR | LFE | FC }, { .ca_id = 0x02, .n_ch = 4, .mask = FL | FR | FC }, - /* surround51 */ + { .ca_id = 0x01, .n_ch = 4, + .mask = FL | FR | LFE }, { .ca_id = 0x0b, .n_ch = 6, - .mask = FL | FR | LFE | FC | RL | RR}, - /* surround40 */ - { .ca_id = 0x08, .n_ch = 6, - .mask = FL | FR | RL | RR }, - /* surround41 */ - { .ca_id = 0x09, .n_ch = 6, - .mask = FL | FR | LFE | RL | RR }, - /* surround50 */ + .mask = FL | FR | LFE | FC | RL | RR }, { .ca_id = 0x0a, .n_ch = 6, .mask = FL | FR | FC | RL | RR }, - /* 6.1 */ - { .ca_id = 0x0f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | RC }, - /* surround71 */ + { .ca_id = 0x09, .n_ch = 6, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 6, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 6, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 6, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 6, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 6, + .mask = FL | FR | RC }, { .ca_id = 0x13, .n_ch = 8, .mask = FL | FR | LFE | FC | RL | RR | RLC | RRC }, - /* others */ - { .ca_id = 0x03, .n_ch = 8, - .mask = FL | FR | LFE | FC }, - { .ca_id = 0x04, .n_ch = 8, - .mask = FL | FR | RC}, - { .ca_id = 0x05, .n_ch = 8, - .mask = FL | FR | LFE | RC }, - { .ca_id = 0x06, .n_ch = 8, - .mask = FL | FR | FC | RC }, - { .ca_id = 0x07, .n_ch = 8, - .mask = FL | FR | LFE | FC | RC }, - { .ca_id = 0x0c, .n_ch = 8, - .mask = FL | FR | RC | RL | RR }, - { .ca_id = 0x0d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RC }, - { .ca_id = 0x0e, .n_ch = 8, - .mask = FL | FR | FC | RL | RR | RC }, - { .ca_id = 0x10, .n_ch = 8, - .mask = FL | FR | RL | RR | RLC | RRC }, - { .ca_id = 0x11, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, { .ca_id = 0x12, .n_ch = 8, .mask = FL | FR | FC | RL | RR | RLC | RRC }, - { .ca_id = 0x14, .n_ch = 8, - .mask = FL | FR | FLC | FRC }, - { .ca_id = 0x15, .n_ch = 8, - .mask = FL | FR | LFE | FLC | FRC }, - { .ca_id = 0x16, .n_ch = 8, - .mask = FL | FR | FC | FLC | FRC }, - { .ca_id = 0x17, .n_ch = 8, - .mask = FL | FR | LFE | FC | FLC | FRC }, - { .ca_id = 0x18, .n_ch = 8, - .mask = FL | FR | RC | FLC | FRC }, - { .ca_id = 0x19, .n_ch = 8, - .mask = FL | FR | LFE | RC | FLC | FRC }, - { .ca_id = 0x1a, .n_ch = 8, - .mask = FL | FR | RC | FC | FLC | FRC }, - { .ca_id = 0x1b, .n_ch = 8, - .mask = FL | FR | LFE | RC | FC | FLC | FRC }, - { .ca_id = 0x1c, .n_ch = 8, - .mask = FL | FR | RL | RR | FLC | FRC }, - { .ca_id = 0x1d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | FLC | FRC }, { .ca_id = 0x1e, .n_ch = 8, .mask = FL | FR | FC | RL | RR | FLC | FRC }, - { .ca_id = 0x1f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, + { .ca_id = 0x11, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | FLC | FRC }, + { .ca_id = 0x10, .n_ch = 8, + .mask = FL | FR | RL | RR | RLC | RRC }, + { .ca_id = 0x1c, .n_ch = 8, + .mask = FL | FR | RL | RR | FLC | FRC }, + { .ca_id = 0x0f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | RC }, + { .ca_id = 0x1b, .n_ch = 8, + .mask = FL | FR | LFE | RC | FC | FLC | FRC }, + { .ca_id = 0x0e, .n_ch = 8, + .mask = FL | FR | FC | RL | RR | RC }, + { .ca_id = 0x1a, .n_ch = 8, + .mask = FL | FR | RC | FC | FLC | FRC }, + { .ca_id = 0x0d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RC }, + { .ca_id = 0x19, .n_ch = 8, + .mask = FL | FR | LFE | RC | FLC | FRC }, + { .ca_id = 0x0c, .n_ch = 8, + .mask = FL | FR | RC | RL | RR }, + { .ca_id = 0x18, .n_ch = 8, + .mask = FL | FR | RC | FLC | FRC }, + { .ca_id = 0x17, .n_ch = 8, + .mask = FL | FR | LFE | FC | FLC | FRC }, + { .ca_id = 0x16, .n_ch = 8, + .mask = FL | FR | FC | FLC | FRC }, + { .ca_id = 0x15, .n_ch = 8, + .mask = FL | FR | LFE | FLC | FRC }, + { .ca_id = 0x14, .n_ch = 8, + .mask = FL | FR | FLC | FRC }, + { .ca_id = 0x0b, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR }, + { .ca_id = 0x0a, .n_ch = 8, + .mask = FL | FR | FC | RL | RR }, + { .ca_id = 0x09, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 8, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 8, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 8, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 8, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 8, + .mask = FL | FR | RC }, + { .ca_id = 0x03, .n_ch = 8, + .mask = FL | FR | LFE | FC }, + { .ca_id = 0x02, .n_ch = 8, + .mask = FL | FR | FC }, + { .ca_id = 0x01, .n_ch = 8, + .mask = FL | FR | LFE }, }; struct hdmi_codec_priv { @@ -371,7 +384,8 @@ static int hdmi_codec_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdmi_codec_priv *hcp = info->private_data; - map = info->chmap[hcp->chmap_idx].map; + if (hcp->chmap_idx != HDMI_CODEC_CHMAP_IDX_UNKNOWN) + map = info->chmap[hcp->chmap_idx].map; for (i = 0; i < info->max_channels; i++) { if (hcp->chmap_idx == HDMI_CODEC_CHMAP_IDX_UNKNOWN) -- 2.51.0 From 67a0463d339059eeeead9cd015afa594659cfdaf Mon Sep 17 00:00:00 2001 From: Alex Far Date: Sat, 16 Nov 2024 21:58:45 +0300 Subject: [PATCH 07/16] ASoC: amd: yc: fix internal mic on Redmi G 2022 This laptop model requires an additional detection quirk to enable the internal microphone Signed-off-by: Alex Far Link: https://patch.msgid.link/ZzjrZY3sImcqTtGx@RedmiG 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 2436e8deb2be..1b9834ee5d46 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -409,6 +409,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Xiaomi Book Pro 14 2022"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "TIMI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Redmi G 2022"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.51.0 From 0109ee00788e0ad7b888a799c26b5a93b343876b Mon Sep 17 00:00:00 2001 From: Mario Limonciello Date: Sun, 17 Nov 2024 20:55:27 -0600 Subject: [PATCH 08/16] ASoC: amd: Fix build dependencies for `SND_SOC_AMD_PS` The pci-ps module now must have `SND_SOC_ACPI_AMD_MATCH` selected to reference the `snd_soc_acpi_amd_acp63_sdw_machines` symbol. Fixes: 56d540befd59 ("ASoC: amd: ps: add soundwire machines for acp6.3 platform") Cc: Vijendar.Mukunda@amd.com Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202411180658.mIeWje2V-lkp@intel.com/ Signed-off-by: Mario Limonciello Link: https://patch.msgid.link/20241118025527.3318493-1-superm1@kernel.org Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6dec44f516c1..c7590d4989bb 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -163,6 +163,7 @@ config SND_SOC_AMD_SOUNDWIRE config SND_SOC_AMD_PS tristate "AMD Audio Coprocessor-v6.3 Pink Sardine support" select SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE + select SND_SOC_ACPI_AMD_MATCH depends on X86 && PCI && ACPI help This option enables Audio Coprocessor i.e ACP v6.3 support on -- 2.51.0 From e038f43edaf0083f6aa7c9415d86cf28dfd152f9 Mon Sep 17 00:00:00 2001 From: Charles Han Date: Mon, 18 Nov 2024 16:45:53 +0800 Subject: [PATCH 09/16] ASoC: imx-audmix: Add NULL check in imx_audmix_probe devm_kasprintf() can return a NULL pointer on failure,but this returned value in imx_audmix_probe() is not checked. Add NULL check in imx_audmix_probe(), to handle kernel NULL pointer dereference error. Fixes: 05d996e11348 ("ASoC: imx-audmix: Split capture device for audmix") Signed-off-by: Charles Han Link: https://patch.msgid.link/20241118084553.4195-1-hanchunchao@inspur.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmix.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index dcf770b55c4b..231400661c90 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -274,6 +274,9 @@ static int imx_audmix_probe(struct platform_device *pdev) /* Add AUDMIX Backend */ be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "audmix-%d", i); + if (!be_name) + return -ENOMEM; + priv->dai[num_dai + i].cpus = &dlc[1]; priv->dai[num_dai + i].codecs = &snd_soc_dummy_dlc; -- 2.51.0 From f32c3f01c21cdd6a354988006aaca5e3dfe478f9 Mon Sep 17 00:00:00 2001 From: liujing Date: Wed, 13 Nov 2024 09:57:58 +0800 Subject: [PATCH 10/16] ASoC: apple: Fix the wrong format specifier In the mca_fe_hw_params(), the variable tdm_slot_width is of type unsigned int, so the output should be %u Signed-off-by: liujing Link: https://patch.msgid.link/20241113015758.5441-1-liujing@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/apple/mca.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index c9e7d40c47cc..b4f4696809dd 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -616,7 +616,7 @@ static int mca_fe_hw_params(struct snd_pcm_substream *substream, tdm_slot_width = 32; if (tdm_slot_width < params_width(params)) { - dev_err(dev, "TDM slots too narrow (tdm=%d params=%d)\n", + dev_err(dev, "TDM slots too narrow (tdm=%u params=%d)\n", tdm_slot_width, params_width(params)); return -EINVAL; } -- 2.51.0 From 8697ecc3274214c65ff271a58e7abb601216f2a8 Mon Sep 17 00:00:00 2001 From: anish kumar Date: Thu, 21 Nov 2024 15:29:58 -0800 Subject: [PATCH 11/16] ASoC: doc: dapm: Add location information for dapm-graph tool To help developers debug DAPM issues and visualize widget connectivity, the dapm-graph tool provides a graphical representation of how widgets and routes are connected. This commit adds the location information for the tool to the documentation, making it easier for users to find and use it for troubleshooting DAPM-related problems. Signed-off-by: anish kumar Link: https://patch.msgid.link/20241121232958.46179-1-yesanishhere@gmail.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 14c4dc026e6b..73a42d5a9f30 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -35,6 +35,9 @@ The graph for the STM32MP1-DK1 sound card is shown in picture: :alt: Example DAPM graph :align: center +You can also generate compatible graph for your sound card using +`tools/sound/dapm-graph` utility. + DAPM power domains ================== -- 2.51.0 From cbc86dd0a4fe9f8c41075328c2e740b68419d639 Mon Sep 17 00:00:00 2001 From: =?utf8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Fri, 22 Nov 2024 08:56:05 +0100 Subject: [PATCH 12/16] ASoC: amd: yc: Add quirk for microphone on Lenovo Thinkpad T14s Gen 6 21M1CTO1WW MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Add a quirk for Tova's Lenovo Thinkpad T14s with product name 21M1. Suggested-by: Tova Link: https://bugs.debian.org/1087673 Signed-off-by: Uwe Kleine-König Link: https://patch.msgid.link/20241122075606.213132-2-ukleinek@debian.org 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 1b9834ee5d46..6439c175552a 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -220,6 +220,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21M1"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.51.0 From 4095cf872084ecfdfdb0e681f3e9ff9745acfa75 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 27 Nov 2024 16:52:25 +0530 Subject: [PATCH 13/16] ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry Add condition check to register ACP PDM sound card by reading _WOV acpi entry. Fixes: 5426f506b584 ("ASoC: amd: Add support for enabling DMIC on acp6x via _DSD") Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20241127112227.227106-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 6439c175552a..facd82f0f251 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -551,8 +551,14 @@ static int acp6x_probe(struct platform_device *pdev) struct acp6x_pdm *machine = NULL; struct snd_soc_card *card; struct acpi_device *adev; + acpi_handle handle; + acpi_integer dmic_status; int ret; + bool is_dmic_enable, wov_en; + /* IF WOV entry not found, enable dmic based on AcpDmicConnected entry*/ + is_dmic_enable = false; + wov_en = true; /* check the parent device's firmware node has _DSD or not */ adev = ACPI_COMPANION(pdev->dev.parent); if (adev) { @@ -560,9 +566,19 @@ static int acp6x_probe(struct platform_device *pdev) if (!acpi_dev_get_property(adev, "AcpDmicConnected", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == 1) - platform_set_drvdata(pdev, &acp6x_card); + is_dmic_enable = true; } + handle = ACPI_HANDLE(pdev->dev.parent); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; + + if (is_dmic_enable && wov_en) + platform_set_drvdata(pdev, &acp6x_card); + else + return 0; + /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.51.0 From 2f2020327cc8561d7c520d2f2d9acea84fa7b3a3 Mon Sep 17 00:00:00 2001 From: =?utf8?q?N=C3=ADcolas=20F=2E=20R=2E=20A=2E=20Prado?= Date: Tue, 26 Nov 2024 15:09:43 -0500 Subject: [PATCH 14/16] ASoC: mediatek: Check num_codecs is not zero to avoid panic during probe MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Following commit 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()"), COMP_DUMMY() became an array with zero length, and only gets populated with the dummy struct after the card is registered. Since the sound card driver's probe happens before the card registration, accessing any of the members of a dummy component during probe will result in undefined behavior. This can be observed in the mt8188 and mt8195 machine sound drivers. By omitting a dai link subnode in the sound card's node in the Devicetree, the default uninitialized dummy codec is used, and when its dai_name pointer gets passed to strcmp() it results in a null pointer dereference and a kernel panic. In addition to that, set_card_codec_info() in the generic helpers file, mtk-soundcard-driver.c, will populate a dai link with a dummy codec when a dai link node is present in DT but with no codec property. The result is that at probe time, a dummy codec can either be uninitialized with num_codecs = 0, or be an initialized dummy codec, with num_codecs = 1 and dai_name = "snd-soc-dummy-dai". In order to accommodate for both situations, check that num_codecs is not zero before accessing the codecs' fields but still check for the codec's dai name against "snd-soc-dummy-dai" as needed. While at it, also drop the check that dai_name is not null in the mt8192 driver, introduced in commit 4d4e1b6319e5 ("ASoC: mediatek: mt8192: Check existence of dai_name before dereferencing"), as it is actually redundant given the preceding num_codecs != 0 check. Fixes: 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") Signed-off-by: Nícolas F. R. A. Prado Reviewed-by: AngeloGioacchino Del Regno Acked-by: Kuninori Morimoto Reviewed-by: Fei Shao Acked-by: Trevor Wu Link: https://patch.msgid.link/20241126-asoc-mtk-dummy-panic-v1-1-42d53e168d2e@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 9 +++++++-- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 ++-- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 9 +++++++-- 3 files changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index 84abdba9ddb6..e04b88a57535 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -1277,10 +1277,12 @@ static int mt8188_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, for_each_card_prelinks(card, i, dai_link) { if (strcmp(dai_link->name, "DPTX_BE") == 0) { - if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + if (dai_link->num_codecs && + strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8188_dptx_codec_init; } else if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { - if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + if (dai_link->num_codecs && + strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8188_hdmi_codec_init; } else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 || strcmp(dai_link->name, "UL_SRC_BE") == 0) { @@ -1292,6 +1294,9 @@ static int mt8188_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, strcmp(dai_link->name, "ETDM2_OUT_BE") == 0 || strcmp(dai_link->name, "ETDM1_IN_BE") == 0 || strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { + if (!dai_link->num_codecs) + continue; + if (!strcmp(dai_link->codecs->dai_name, MAX98390_CODEC_DAI)) { /* * The TDM protocol settings with fixed 4 slots are defined in diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 1aba9c75594e..b1598cc5587e 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -1091,7 +1091,7 @@ static int mt8192_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data) dai_link->ignore = 0; } - if (dai_link->num_codecs && dai_link->codecs[0].dai_name && + if (dai_link->num_codecs && strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0) dai_link->ops = &mt8192_rt1015_i2s_ops; } @@ -1119,7 +1119,7 @@ static int mt8192_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, int i; for_each_card_prelinks(card, i, dai_link) - if (dai_link->num_codecs && dai_link->codecs[0].dai_name && + if (dai_link->num_codecs && strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0) dai_link->ops = &mt8192_rt1015_i2s_ops; } diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 56b9d2433a1e..2b9cb3248795 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -1378,10 +1378,12 @@ static int mt8195_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, for_each_card_prelinks(card, i, dai_link) { if (strcmp(dai_link->name, "DPTX_BE") == 0) { - if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + if (dai_link->num_codecs && + strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8195_dptx_codec_init; } else if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { - if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + if (dai_link->num_codecs && + strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8195_hdmi_codec_init; } else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 || strcmp(dai_link->name, "UL_SRC1_BE") == 0 || @@ -1394,6 +1396,9 @@ static int mt8195_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, strcmp(dai_link->name, "ETDM2_OUT_BE") == 0 || strcmp(dai_link->name, "ETDM1_IN_BE") == 0 || strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { + if (!dai_link->num_codecs) + continue; + if (!strcmp(dai_link->codecs->dai_name, MAX98390_CODEC_DAI)) { if (!(codec_init & MAX98390_CODEC_INIT)) { dai_link->init = mt8195_max98390_init; -- 2.51.0 From e9db1b551774037ebe39dde4a658d89ba95e260b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 27 Nov 2024 17:29:54 +0800 Subject: [PATCH 15/16] ASoC: SOF: ipc3-topology: Convert the topology pin index to ALH dai index MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Intel SoundWire machine driver always uses Pin number 2 and above. Currently, the pin number is used as the FW DAI index directly. As a result, FW DAI 0 and 1 are never used. That worked fine because we use up to 2 DAIs in a SDW link. Convert the topology pin index to ALH dai index, the mapping is using 2-off indexing, iow, pin #2 is ALH dai #0. The issue exists since beginning. And the Fixes tag is the first commit that this commit can be applied. Fixes: b66bfc3a9810 ("ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP") Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Liam Girdwood Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20241127092955.20026-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 26 ++++++++++++++++++++++++-- 1 file changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index be61e377e59e..c2fce554a674 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -20,6 +20,9 @@ /* size of tplg ABI in bytes */ #define SOF_IPC3_TPLG_ABI_SIZE 3 +/* Base of SOF_DAI_INTEL_ALH, this should be aligned with SOC_SDW_INTEL_BIDIR_PDI_BASE */ +#define INTEL_ALH_DAI_INDEX_BASE 2 + struct sof_widget_data { int ctrl_type; int ipc_cmd; @@ -1594,6 +1597,17 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget) if (ret < 0) goto free; + /* Subtract the base to match the FW dai index. */ + if (comp_dai->type == SOF_DAI_INTEL_ALH) { + if (comp_dai->dai_index < INTEL_ALH_DAI_INDEX_BASE) { + dev_err(sdev->dev, + "Invalid ALH dai index %d, only Pin numbers >= %d can be used\n", + comp_dai->dai_index, INTEL_ALH_DAI_INDEX_BASE); + return -EINVAL; + } + comp_dai->dai_index -= INTEL_ALH_DAI_INDEX_BASE; + } + dev_dbg(scomp->dev, "dai %s: type %d index %d\n", swidget->widget->name, comp_dai->type, comp_dai->dai_index); sof_dbg_comp_config(scomp, &comp_dai->config); @@ -2167,8 +2181,16 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * case SOF_DAI_INTEL_ALH: if (data) { /* save the dai_index during hw_params and reuse it for hw_free */ - if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) - config->dai_index = data->dai_index; + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + /* Subtract the base to match the FW dai index. */ + if (data->dai_index < INTEL_ALH_DAI_INDEX_BASE) { + dev_err(sdev->dev, + "Invalid ALH dai index %d, only Pin numbers >= %d can be used\n", + config->dai_index, INTEL_ALH_DAI_INDEX_BASE); + return -EINVAL; + } + config->dai_index = data->dai_index - INTEL_ALH_DAI_INDEX_BASE; + } config->alh.stream_id = data->dai_data; } break; -- 2.51.0 From b682aa788e5f9f1ddacdfbb453e49fd3f4e83721 Mon Sep 17 00:00:00 2001 From: Ilya Zverev Date: Wed, 27 Nov 2024 15:44:20 +0200 Subject: [PATCH 16/16] ASoC: amd: yc: Add a quirk for microfone on Lenovo ThinkPad P14s Gen 5 21MES00B00 New ThinkPads need new quirk entries. Ilya has tested this one. Laptop product id is 21MES00B00, though the shorthand 21ME works. Closes: https://bugzilla.kernel.org/show_bug.cgi?id=219533 Cc: stable@vger.kernel.org Signed-off-by: Ilya Zverev Link: https://patch.msgid.link/20241127134420.14471-1-ilya@zverev.info 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 facd82f0f251..e38c5885dadf 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -248,6 +248,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21M5"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21ME"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.51.0