From d979b783d61f7f1f95664031b71a33afc74627b2 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 23 May 2025 17:58:14 +0200 Subject: firmware: cs_dsp: Fix OOB memory read access in KUnit test (wmfw info) KASAN reported out of bounds access - cs_dsp_mock_wmfw_add_info(), because the source string length was rounded up to the allocation size. Cc: Simon Trimmer Cc: Charles Keepax Cc: Richard Fitzgerald Cc: patches@opensource.cirrus.com Cc: stable@vger.kernel.org Signed-off-by: Jaroslav Kysela Reviewed-by: Richard Fitzgerald Link: https://patch.msgid.link/20250523155814.1256762-1-perex@perex.cz Signed-off-by: Mark Brown --- drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c b/drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c index 934d40a4d709..5e1d5a810afe 100644 --- a/drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c +++ b/drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c @@ -133,10 +133,11 @@ void cs_dsp_mock_wmfw_add_info(struct cs_dsp_mock_wmfw_builder *builder, if (info_len % 4) { /* Create a padded string with length a multiple of 4 */ + size_t copy_len = info_len; info_len = round_up(info_len, 4); tmp = kunit_kzalloc(builder->test_priv->test, info_len, GFP_KERNEL); KUNIT_ASSERT_NOT_ERR_OR_NULL(builder->test_priv->test, tmp); - memcpy(tmp, info, info_len); + memcpy(tmp, info, copy_len); info = tmp; } -- cgit v1.2.3-59-g8ed1b From f4ba2ea57da51d616b689c4b8826c517ff5a8523 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 23 May 2025 17:41:51 +0200 Subject: firmware: cs_dsp: Fix OOB memory read access in KUnit test (ctl cache) KASAN reported out of bounds access - cs_dsp_ctl_cache_init_multiple_offsets(). The code uses mock_coeff_template.length_bytes (4 bytes) for register value allocations. But later, this length is set to 8 bytes which causes test code failures. As fix, just remove the lenght override, keeping the original value 4 for all operations. Cc: Simon Trimmer Cc: Charles Keepax Cc: Richard Fitzgerald Cc: patches@opensource.cirrus.com Cc: stable@vger.kernel.org Signed-off-by: Jaroslav Kysela Reviewed-by: Richard Fitzgerald Link: https://patch.msgid.link/20250523154151.1252585-1-perex@perex.cz Signed-off-by: Mark Brown --- drivers/firmware/cirrus/test/cs_dsp_test_control_cache.c | 1 - 1 file changed, 1 deletion(-) diff --git a/drivers/firmware/cirrus/test/cs_dsp_test_control_cache.c b/drivers/firmware/cirrus/test/cs_dsp_test_control_cache.c index 83386cc978e3..ebca3a4ab0f1 100644 --- a/drivers/firmware/cirrus/test/cs_dsp_test_control_cache.c +++ b/drivers/firmware/cirrus/test/cs_dsp_test_control_cache.c @@ -776,7 +776,6 @@ static void cs_dsp_ctl_cache_init_multiple_offsets(struct kunit *test) "dummyalg", NULL); /* Create controls identical except for offset */ - def.length_bytes = 8; def.offset_dsp_words = 0; def.shortname = "CtlA"; cs_dsp_mock_wmfw_add_coeff_desc(local->wmfw_builder, &def); -- cgit v1.2.3-59-g8ed1b From 9079db287fc3e38e040b0edeb0a25770bb679c8e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 26 May 2025 11:47:01 +0200 Subject: ASoC: codecs: wcd9335: Fix missing free of regulator supplies Driver gets and enables all regulator supplies in probe path (wcd9335_parse_dt() and wcd9335_power_on_reset()), but does not cleanup in final error paths and in unbind (missing remove() callback). This leads to leaked memory and unbalanced regulator enable count during probe errors or unbind. Fix this by converting entire code into devm_regulator_bulk_get_enable() which also greatly simplifies the code. Fixes: 20aedafdf492 ("ASoC: wcd9335: add support to wcd9335 codec") Cc: stable@vger.kernel.org Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20250526-b4-b4-asoc-wcd9395-vdd-px-fixes-v1-1-0b8a2993b7d3@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 25 +++++++------------------ 1 file changed, 7 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 8ee4360aff92..5e19e813748d 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -332,7 +332,6 @@ struct wcd9335_codec { int intr1; struct gpio_desc *reset_gpio; - struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY]; unsigned int rx_port_value[WCD9335_RX_MAX]; unsigned int tx_port_value[WCD9335_TX_MAX]; @@ -355,6 +354,10 @@ struct wcd9335_irq { char *name; }; +static const char * const wcd9335_supplies[] = { + "vdd-buck", "vdd-buck-sido", "vdd-tx", "vdd-rx", "vdd-io", +}; + static const struct wcd9335_slim_ch wcd9335_tx_chs[WCD9335_TX_MAX] = { WCD9335_SLIM_TX_CH(0), WCD9335_SLIM_TX_CH(1), @@ -4989,30 +4992,16 @@ static int wcd9335_parse_dt(struct wcd9335_codec *wcd) if (IS_ERR(wcd->native_clk)) return dev_err_probe(dev, PTR_ERR(wcd->native_clk), "slimbus clock not found\n"); - wcd->supplies[0].supply = "vdd-buck"; - wcd->supplies[1].supply = "vdd-buck-sido"; - wcd->supplies[2].supply = "vdd-tx"; - wcd->supplies[3].supply = "vdd-rx"; - wcd->supplies[4].supply = "vdd-io"; - - ret = regulator_bulk_get(dev, WCD9335_MAX_SUPPLY, wcd->supplies); + ret = devm_regulator_bulk_get_enable(dev, ARRAY_SIZE(wcd9335_supplies), + wcd9335_supplies); if (ret) - return dev_err_probe(dev, ret, "Failed to get supplies\n"); + return dev_err_probe(dev, ret, "Failed to get and enable supplies\n"); return 0; } static int wcd9335_power_on_reset(struct wcd9335_codec *wcd) { - struct device *dev = wcd->dev; - int ret; - - ret = regulator_bulk_enable(WCD9335_MAX_SUPPLY, wcd->supplies); - if (ret) { - dev_err(dev, "Failed to get supplies: err = %d\n", ret); - return ret; - } - /* * For WCD9335, it takes about 600us for the Vout_A and * Vout_D to be ready after BUCK_SIDO is powered up. -- cgit v1.2.3-59-g8ed1b From dc59189d32fc3dbddcf418fd4b418fb61f24ade6 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 26 May 2025 11:47:02 +0200 Subject: ASoC: codecs: wcd937x: Drop unused buck_supply Last user of wcd937x_priv->buck_supply was removed in commit 216d04139a6d ("ASoC: codecs: wcd937x: Remove separate handling for vdd-buck supply"). Fixes: 216d04139a6d ("ASoC: codecs: wcd937x: Remove separate handling for vdd-buck supply") Cc: stable@vger.kernel.org Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20250526-b4-b4-asoc-wcd9395-vdd-px-fixes-v1-2-0b8a2993b7d3@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index 3b1a1518e764..a3a4b1f53e88 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -91,7 +91,6 @@ struct wcd937x_priv { struct regmap_irq_chip *wcd_regmap_irq_chip; struct regmap_irq_chip_data *irq_chip; struct regulator_bulk_data supplies[WCD937X_MAX_BULK_SUPPLY]; - struct regulator *buck_supply; struct snd_soc_jack *jack; unsigned long status_mask; s32 micb_ref[WCD937X_MAX_MICBIAS]; -- cgit v1.2.3-59-g8ed1b From 63fe298652d4eda07d738bfcbbc59d1343a675ef Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 26 May 2025 11:47:03 +0200 Subject: ASoC: codecs: wcd9375: Fix double free of regulator supplies Driver gets regulator supplies in probe path with devm_regulator_bulk_get(), so should not call regulator_bulk_free() in error and remove paths to avoid double free. Fixes: 216d04139a6d ("ASoC: codecs: wcd937x: Remove separate handling for vdd-buck supply") Cc: stable@vger.kernel.org Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20250526-b4-b4-asoc-wcd9395-vdd-px-fixes-v1-3-0b8a2993b7d3@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index a3a4b1f53e88..b9df58b86ce9 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -2944,10 +2944,8 @@ static int wcd937x_probe(struct platform_device *pdev) return dev_err_probe(dev, ret, "Failed to get supplies\n"); ret = regulator_bulk_enable(WCD937X_MAX_BULK_SUPPLY, wcd937x->supplies); - if (ret) { - regulator_bulk_free(WCD937X_MAX_BULK_SUPPLY, wcd937x->supplies); + if (ret) return dev_err_probe(dev, ret, "Failed to enable supplies\n"); - } wcd937x_dt_parse_micbias_info(dev, wcd937x); @@ -2983,7 +2981,6 @@ static int wcd937x_probe(struct platform_device *pdev) err_disable_regulators: regulator_bulk_disable(WCD937X_MAX_BULK_SUPPLY, wcd937x->supplies); - regulator_bulk_free(WCD937X_MAX_BULK_SUPPLY, wcd937x->supplies); return ret; } @@ -3000,7 +2997,6 @@ static void wcd937x_remove(struct platform_device *pdev) pm_runtime_dont_use_autosuspend(dev); regulator_bulk_disable(WCD937X_MAX_BULK_SUPPLY, wcd937x->supplies); - regulator_bulk_free(WCD937X_MAX_BULK_SUPPLY, wcd937x->supplies); } #if defined(CONFIG_OF) -- cgit v1.2.3-59-g8ed1b From 16719d48197bbd8cff121b32acec67d954335437 Mon Sep 17 00:00:00 2001 From: Qasim Ijaz Date: Mon, 26 May 2025 20:18:20 +0100 Subject: ASoC: codecs: fix out-of-bounds access on invalid clock config MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit get_coeff() returns –EINVAL when no table entry matches. The driver then uses that value as an index into coeff_div[], causing an OOB access. To fix lets abort the hw_params call instead. Fixes: de2b3119f9f7 ("ASoC: codecs: add support for ES8375") Signed-off-by: Qasim Ijaz Link: https://patch.msgid.link/20250526191820.72577-1-qasdev00@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8375.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/es8375.c b/sound/soc/codecs/es8375.c index decc86c92427..009259632107 100644 --- a/sound/soc/codecs/es8375.c +++ b/sound/soc/codecs/es8375.c @@ -319,6 +319,7 @@ static int es8375_hw_params(struct snd_pcm_substream *substream, coeff = get_coeff(es8375->vddd, dmic_enable, es8375->mclk_freq, params_rate(params)); if (coeff < 0) { dev_warn(component->dev, "Clock coefficients do not match"); + return coeff; } regmap_write(es8375->regmap, ES8375_CLK_MGR4, coeff_div[coeff].Reg0x04); -- cgit v1.2.3-59-g8ed1b From e3de7984e45155888eebbca5a32c1cc5f29fa859 Mon Sep 17 00:00:00 2001 From: Bram Vlerick Date: Tue, 27 May 2025 13:08:49 +0200 Subject: ASoC: tas571x: add separate tas5733 controls The controls between the tas5717 and tas5733 should not be shared since the biquad and register setup is not identical. For example, writing to 0x5c on the tas5717 modifies ch2_bq[10] while on the tas5733 this is ch1_cross_bq[3]. see https://www.ti.com/lit/ds/symlink/tas5733l.pdf and https://www.ti.com/lit/ds/symlink/tas5717.pdf for more details on the register maps. Signed-off-by: Bram Vlerick Link: https://patch.msgid.link/20250527-tas5733-biquad-fix-v1-1-0d3d941700bb@openpixelsystems.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas571x.c | 65 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/tas571x.h | 34 ++++++++++++++++++++++++ 2 files changed, 98 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 6c6e7ae07d80..00b131294547 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -718,6 +718,69 @@ static const struct regmap_config tas5721_regmap_config = { .volatile_table = &tas571x_volatile_regs, }; +static const struct snd_kcontrol_new tas5733_controls[] = { + /* MVOL LSB is ignored - see comments in tas571x_i2c_probe() */ + SOC_SINGLE_TLV("Master Volume", + TAS571X_MVOL_REG, 1, 0x1ff, 1, + tas5717_volume_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", + TAS571X_CH1_VOL_REG, TAS571X_CH2_VOL_REG, + 1, 0x1ff, 1, tas5717_volume_tlv), + SOC_DOUBLE("Speaker Switch", + TAS571X_SOFT_MUTE_REG, + TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, + 1, 1), + + SOC_DOUBLE_R_RANGE("CH1 Mixer Volume", + TAS5717_CH1_LEFT_CH_MIX_REG, + TAS5717_CH1_RIGHT_CH_MIX_REG, + 16, 0, 0x80, 0), + + SOC_DOUBLE_R_RANGE("CH2 Mixer Volume", + TAS5717_CH2_LEFT_CH_MIX_REG, + TAS5717_CH2_RIGHT_CH_MIX_REG, + 16, 0, 0x80, 0), + + /* + * The biquads are named according to the register names. + * Please note that TI's TAS57xx Graphical Development Environment + * tool names them different. + */ + BIQUAD_COEFS("CH1 - Biquad 0", TAS5733_CH1_BQ0_REG), + BIQUAD_COEFS("CH1 - Biquad 1", TAS5733_CH1_BQ1_REG), + BIQUAD_COEFS("CH1 - Biquad 2", TAS5733_CH1_BQ2_REG), + BIQUAD_COEFS("CH1 - Biquad 3", TAS5733_CH1_BQ3_REG), + BIQUAD_COEFS("CH1 - Biquad 4", TAS5733_CH1_BQ4_REG), + BIQUAD_COEFS("CH1 - Biquad 5", TAS5733_CH1_BQ5_REG), + BIQUAD_COEFS("CH1 - Biquad 6", TAS5733_CH1_BQ6_REG), + BIQUAD_COEFS("CH1 - Biquad 7", TAS5733_CH1_BQ7_REG), + BIQUAD_COEFS("CH1 - Biquad 8", TAS5733_CH1_BQ8_REG), + BIQUAD_COEFS("CH1 - Biquad 9", TAS5733_CH1_BQ9_REG), + BIQUAD_COEFS("CH1 - Biquad 10", TAS5733_CH1_BQ10_REG), + + BIQUAD_COEFS("CH2 - Biquad 0", TAS5733_CH2_BQ0_REG), + BIQUAD_COEFS("CH2 - Biquad 1", TAS5733_CH2_BQ1_REG), + BIQUAD_COEFS("CH2 - Biquad 2", TAS5733_CH2_BQ2_REG), + BIQUAD_COEFS("CH2 - Biquad 3", TAS5733_CH2_BQ3_REG), + BIQUAD_COEFS("CH2 - Biquad 4", TAS5733_CH2_BQ4_REG), + BIQUAD_COEFS("CH2 - Biquad 5", TAS5733_CH2_BQ5_REG), + BIQUAD_COEFS("CH2 - Biquad 6", TAS5733_CH2_BQ6_REG), + BIQUAD_COEFS("CH2 - Biquad 7", TAS5733_CH2_BQ7_REG), + BIQUAD_COEFS("CH2 - Biquad 8", TAS5733_CH2_BQ8_REG), + BIQUAD_COEFS("CH2 - Biquad 9", TAS5733_CH2_BQ9_REG), + BIQUAD_COEFS("CH2 - Biquad 10", TAS5733_CH2_BQ10_REG), + + BIQUAD_COEFS("CH1 - Cross Biquad 0", TAS5733_CH1_CBQ0_REG), + BIQUAD_COEFS("CH1 - Cross Biquad 1", TAS5733_CH1_CBQ1_REG), + BIQUAD_COEFS("CH1 - Cross Biquad 2", TAS5733_CH1_CBQ2_REG), + BIQUAD_COEFS("CH1 - Cross Biquad 3", TAS5733_CH1_CBQ3_REG), + + BIQUAD_COEFS("CH2 - Cross Biquad 0", TAS5733_CH2_CBQ0_REG), + BIQUAD_COEFS("CH2 - Cross Biquad 1", TAS5733_CH2_CBQ1_REG), + BIQUAD_COEFS("CH2 - Cross Biquad 2", TAS5733_CH2_CBQ2_REG), + BIQUAD_COEFS("CH2 - Cross Biquad 3", TAS5733_CH2_CBQ3_REG), +}; + static const char *const tas5733_supply_names[] = { "AVDD", "DVDD", @@ -770,7 +833,7 @@ static const struct regmap_config tas5733_regmap_config = { static const struct tas571x_chip tas5733_chip = { .supply_names = tas5733_supply_names, .num_supply_names = ARRAY_SIZE(tas5733_supply_names), - .controls = tas5717_controls, + .controls = tas5733_controls, .num_controls = ARRAY_SIZE(tas5717_controls), .regmap_config = &tas5733_regmap_config, .vol_reg_size = 2, diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 5340d3bec31d..2b3eff4023b9 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -104,4 +104,38 @@ #define TAS5717_CH2_LEFT_CH_MIX_REG 0x76 #define TAS5717_CH2_RIGHT_CH_MIX_REG 0x77 +#define TAS5733_CH1_BQ0_REG 0x26 +#define TAS5733_CH1_BQ1_REG 0x27 +#define TAS5733_CH1_BQ2_REG 0x28 +#define TAS5733_CH1_BQ3_REG 0x29 +#define TAS5733_CH1_BQ4_REG 0x2a +#define TAS5733_CH1_BQ5_REG 0x2b +#define TAS5733_CH1_BQ6_REG 0x2c +#define TAS5733_CH1_BQ7_REG 0x2d +#define TAS5733_CH1_BQ8_REG 0x2e +#define TAS5733_CH1_BQ9_REG 0x2f + +#define TAS5733_CH2_BQ0_REG 0x30 +#define TAS5733_CH2_BQ1_REG 0x31 +#define TAS5733_CH2_BQ2_REG 0x32 +#define TAS5733_CH2_BQ3_REG 0x33 +#define TAS5733_CH2_BQ4_REG 0x34 +#define TAS5733_CH2_BQ5_REG 0x35 +#define TAS5733_CH2_BQ6_REG 0x36 +#define TAS5733_CH2_BQ7_REG 0x37 +#define TAS5733_CH2_BQ8_REG 0x38 +#define TAS5733_CH2_BQ9_REG 0x39 + +#define TAS5733_CH1_BQ10_REG 0x58 +#define TAS5733_CH1_CBQ0_REG 0x59 +#define TAS5733_CH1_CBQ1_REG 0x5a +#define TAS5733_CH1_CBQ2_REG 0x5b +#define TAS5733_CH1_CBQ3_REG 0x5c + +#define TAS5733_CH2_BQ10_REG 0x5d +#define TAS5733_CH2_CBQ0_REG 0x5e +#define TAS5733_CH2_CBQ1_REG 0x5f +#define TAS5733_CH2_CBQ2_REG 0x60 +#define TAS5733_CH2_CBQ3_REG 0x61 + #endif /* _TAS571X_H */ -- cgit v1.2.3-59-g8ed1b From bc4005ef43104da589951dba69291360c6a11ae7 Mon Sep 17 00:00:00 2001 From: Bram Vlerick Date: Wed, 28 May 2025 10:25:48 +0200 Subject: ASoC: tas571x: fix tas5733 num_controls Commit e3de7984e451 ("ASoC: tas571x: add separate tas5733 controls") introduces a separate struct for the tas5733 controls but did not update the num_controls with the correct ARRAY_SIZE. Fixes: e3de7984e451 ("ASoC: tas571x: add separate tas5733 controls") Signed-off-by: Bram Vlerick Acked-by: Peter Korsgaard Link: https://patch.msgid.link/20250528-tas5733-fix-controls-size-v1-1-5c70595accaf@openpixelsystems.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas571x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 00b131294547..6bf37c77f0a7 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -834,7 +834,7 @@ static const struct tas571x_chip tas5733_chip = { .supply_names = tas5733_supply_names, .num_supply_names = ARRAY_SIZE(tas5733_supply_names), .controls = tas5733_controls, - .num_controls = ARRAY_SIZE(tas5717_controls), + .num_controls = ARRAY_SIZE(tas5733_controls), .regmap_config = &tas5733_regmap_config, .vol_reg_size = 2, }; -- cgit v1.2.3-59-g8ed1b From df7996076b1e1ba8a0690542d0e40f703f2f9eb7 Mon Sep 17 00:00:00 2001 From: meowmeowbeanz Date: Wed, 28 May 2025 10:56:58 -0700 Subject: ASoC: amd: yc: Add support for Lenovo Yoga 7 16ARP8 Add DMI quirk entry for Lenovo Yoga 7 16ARP8 (83BS) to enable digital microphone support via ACP driver. Fixes microphone detection on this specific model which was previously falling back to non-functional generic audio paths. Tested-by: meowmeowbeanz Signed-off-by: meowmeowbeanz Link: https://patch.msgid.link/20250528-yoga-7-16arp8-microphone-fix-v1-1-bfeed2ecd0c2@gmx.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 3d9da93d22ee..7e62445e02c1 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -311,6 +311,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83AS"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83BS"), + } + }, { .driver_data = &acp6x_card, .matches = { -- cgit v1.2.3-59-g8ed1b From ff0045de4ee0288dec683690f66f2f369b7d3466 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:10:17 +0200 Subject: ASoC: codecs: hda: Fix RPM usage count underflow MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit RPM manipulation in hda_codec_probe_complete()'s error path is superfluous and leads to RPM usage count underflow if the build-controls operation fails. hda_codec_probe_complete() is called in: 1) hda_codec_probe() for all non-HDMI codecs 2) in card->late_probe() for HDMI codecs Error path for hda_codec_probe() takes care of bus' RPM already. For 2) if late_probe() fails, ASoC performs card cleanup what triggers hda_codec_remote() - same treatment is in 1). Fixes: b5df2a7dca1c ("ASoC: codecs: Add HD-Audio codec driver") Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/hda.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index ddc00927313c..dc7794c9ac44 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -152,7 +152,7 @@ int hda_codec_probe_complete(struct hda_codec *codec) ret = snd_hda_codec_build_controls(codec); if (ret < 0) { dev_err(&hdev->dev, "unable to create controls %d\n", ret); - goto out; + return ret; } /* Bus suspended codecs as it does not manage their pm */ @@ -160,7 +160,7 @@ int hda_codec_probe_complete(struct hda_codec *codec) /* rpm was forbidden in snd_hda_codec_device_new() */ snd_hda_codec_set_power_save(codec, 2000); snd_hda_codec_register(codec); -out: + /* Complement pm_runtime_get_sync(bus) in probe */ pm_runtime_mark_last_busy(bus->dev); pm_runtime_put_autosuspend(bus->dev); -- cgit v1.2.3-59-g8ed1b From 9ad1f3cd0d60444c69948854c7e50d2a61b63755 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:10:18 +0200 Subject: ASoC: Intel: avs: Fix deadlock when the failing IPC is SET_D0IX MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The procedure handling IPC timeouts and EXCEPTION_CAUGHT notification shall cancel any D0IX work before proceeding with DSP recovery. If SET_D0IX called from delayed_work is the failing IPC the procedure will deadlock. Conditionally skip cancelling the work to fix that. Fixes: 335c4cbd201d ("ASoC: Intel: avs: D0ix power state support") Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/ipc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c index 08ed9d96738a..0314f9d4ea5f 100644 --- a/sound/soc/intel/avs/ipc.c +++ b/sound/soc/intel/avs/ipc.c @@ -169,7 +169,9 @@ static void avs_dsp_exception_caught(struct avs_dev *adev, union avs_notify_msg dev_crit(adev->dev, "communication severed, rebooting dsp..\n"); - cancel_delayed_work_sync(&ipc->d0ix_work); + /* Avoid deadlock as the exception may be the response to SET_D0IX. */ + if (current_work() != &ipc->d0ix_work.work) + cancel_delayed_work_sync(&ipc->d0ix_work); ipc->in_d0ix = false; /* Re-enabled on recovery completion. */ pm_runtime_disable(adev->dev); -- cgit v1.2.3-59-g8ed1b From 347c8d6db7c9d65d93ef226849b273823f54eaea Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:10:19 +0200 Subject: ASoC: Intel: avs: Fix PPLCxFMT calculation MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit HDAudio transfer types utilize SDxFMT for front-end (HOST) and PPLCxFMT for back-end (LINK) side when setting up the stream. BE's substream->runtime duplicates FE runtime so switch to using BE's hw_params to address incorrect format values on the LINK side when FE and BE formats differ. The problem is introduced with commit d070002a20fc ("ASoC: Intel: avs: HDA PCM BE operations") but the code has been shuffled around since then so direct 'Fixes:' tag does not apply. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 405cfc1ab0cb..d1acc636add7 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -450,9 +450,10 @@ static int avs_dai_hda_be_hw_free(struct snd_pcm_substream *substream, struct sn static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *be = snd_soc_substream_to_rtd(substream); const struct snd_soc_pcm_stream *stream_info; struct hdac_ext_stream *link_stream; + const struct snd_pcm_hw_params *p; struct avs_dma_data *data; unsigned int format_val; unsigned int bits; @@ -460,14 +461,15 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn data = snd_soc_dai_get_dma_data(dai, substream); link_stream = data->link_stream; + p = &be->dpcm[substream->stream].hw_params; if (link_stream->link_prepared) return 0; stream_info = snd_soc_dai_get_pcm_stream(dai, substream->stream); - bits = snd_hdac_stream_format_bits(runtime->format, runtime->subformat, + bits = snd_hdac_stream_format_bits(params_format(p), params_subformat(p), stream_info->sig_bits); - format_val = snd_hdac_stream_format(runtime->channels, bits, runtime->rate); + format_val = snd_hdac_stream_format(params_channels(p), bits, params_rate(p)); snd_hdac_ext_stream_decouple(&data->adev->base.core, link_stream, true); snd_hdac_ext_stream_reset(link_stream); -- cgit v1.2.3-59-g8ed1b From 2f78724d4f0c665c83e202e3989d5333a2cb1036 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:10:20 +0200 Subject: ASoC: Intel: avs: Fix possible null-ptr-deref when initing hw MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Search result of avs_dai_find_path_template() shall be verified before being used. As 'template' is already known when avs_hw_constraints_init() is fired, drop the search entirely. Fixes: f2f847461fb7 ("ASoC: Intel: avs: Constrain path based on BE capabilities") Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index d1acc636add7..ccf90428126d 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -83,10 +83,8 @@ void avs_period_elapsed(struct snd_pcm_substream *substream) static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule); static int avs_hw_constraints_init(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hw_constraint_list *r, *c, *s; - struct avs_tplg_path_template *template; struct avs_dma_data *data; int ret; @@ -99,8 +97,7 @@ static int avs_hw_constraints_init(struct snd_pcm_substream *substream, struct s c = &(data->channels_list); s = &(data->sample_bits_list); - template = avs_dai_find_path_template(dai, !rtd->dai_link->no_pcm, substream->stream); - ret = avs_path_set_constraint(data->adev, template, r, c, s); + ret = avs_path_set_constraint(data->adev, data->template, r, c, s); if (ret <= 0) return ret; -- cgit v1.2.3-59-g8ed1b From 9e3285be55e6c0829e451b4a341e3059da47ec9d Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 30 May 2025 16:10:21 +0200 Subject: ASoC: Intel: avs: Fix paths in MODULE_FIRMWARE hints MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The binaries for cAVS architecture are located in "intel/avs" subdirectory, not "intel". Fixes: 94aa347d34e0 ("ASoC: Intel: avs: Add MODULE_FIRMWARE to inform about FW") Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/core.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 485842838025..ec1b3f55cb5c 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -945,14 +945,14 @@ MODULE_AUTHOR("Cezary Rojewski "); MODULE_AUTHOR("Amadeusz Slawinski "); MODULE_DESCRIPTION("Intel cAVS sound driver"); MODULE_LICENSE("GPL"); -MODULE_FIRMWARE("intel/skl/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/apl/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/cnl/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/icl/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/jsl/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/lkf/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/tgl/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/ehl/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/adl/dsp_basefw.bin"); -MODULE_FIRMWARE("intel/adl_n/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/skl/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/apl/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/cnl/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/icl/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/jsl/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/lkf/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/tgl/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/ehl/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/adl/dsp_basefw.bin"); +MODULE_FIRMWARE("intel/avs/adl_n/dsp_basefw.bin"); MODULE_FIRMWARE("intel/fcl/dsp_basefw.bin"); -- cgit v1.2.3-59-g8ed1b From 5f342aeee2724d31046172eb5caab8e0e8afd57d Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:10:22 +0200 Subject: ASoC: Intel: avs: Verify kcalloc() status when setting constraints MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit All memory operations shall be checked. Fixes: f2f847461fb7 ("ASoC: Intel: avs: Constrain path based on BE capabilities") Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/path.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index ed8f0ea0e10d..e8e6b1c7fc90 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -134,6 +134,8 @@ int avs_path_set_constraint(struct avs_dev *adev, struct avs_tplg_path_template rlist = kcalloc(i, sizeof(*rlist), GFP_KERNEL); clist = kcalloc(i, sizeof(*clist), GFP_KERNEL); slist = kcalloc(i, sizeof(*slist), GFP_KERNEL); + if (!rlist || !clist || !slist) + return -ENOMEM; i = 0; list_for_each_entry(path_template, &template->path_list, node) { -- cgit v1.2.3-59-g8ed1b From 93e246b6769bdacb09cfff4ea0f00fe5ab4f0d7a Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:10:23 +0200 Subject: ASoC: Intel: avs: Verify content returned by parse_int_array() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The first element of the returned array stores its length. If it is 0, any manipulation beyond the element at index 0 ends with null-ptr-deref. Fixes: 5a565ba23abe ("ASoC: Intel: avs: Probing and firmware tracing over debugfs") Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/debugfs.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/debugfs.c b/sound/soc/intel/avs/debugfs.c index 8c4edda97f75..0e826ca20619 100644 --- a/sound/soc/intel/avs/debugfs.c +++ b/sound/soc/intel/avs/debugfs.c @@ -373,7 +373,10 @@ static ssize_t trace_control_write(struct file *file, const char __user *from, s return ret; num_elems = *array; - resource_mask = array[1]; + if (!num_elems) { + ret = -EINVAL; + goto free_array; + } /* * Disable if just resource mask is provided - no log priority flags. @@ -381,6 +384,7 @@ static ssize_t trace_control_write(struct file *file, const char __user *from, s * Enable input format: mask, prio1, .., prioN * Where 'N' equals number of bits set in the 'mask'. */ + resource_mask = array[1]; if (num_elems == 1) { ret = disable_logs(adev, resource_mask); } else { -- cgit v1.2.3-59-g8ed1b From 38b1befc7a35a475d90ec32bfbe319f4412880a1 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:10:25 +0200 Subject: ASoC: Intel: avs: Include missing string.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit File loader.c utilizes strscpy(). Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/loader.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c index 138e4e9de5e3..353e343b1d28 100644 --- a/sound/soc/intel/avs/loader.c +++ b/sound/soc/intel/avs/loader.c @@ -9,6 +9,7 @@ #include #include #include +#include #include #include #include "avs.h" -- cgit v1.2.3-59-g8ed1b From 9adf2de86611ac108d07e769a699556d87f052e2 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Mon, 2 Jun 2025 16:58:51 +0800 Subject: ASoC: rt1320: fix speaker noise when volume bar is 100% This patch updates the settings to fix the speaker noise. Signed-off-by: Shuming Fan Link: https://patch.msgid.link/20250602085851.4081886-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1320-sdw.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1320-sdw.c b/sound/soc/codecs/rt1320-sdw.c index f51ba345a16e..015cc710e6dc 100644 --- a/sound/soc/codecs/rt1320-sdw.c +++ b/sound/soc/codecs/rt1320-sdw.c @@ -204,7 +204,7 @@ static const struct reg_sequence rt1320_vc_blind_write[] = { { 0x3fc2bfc0, 0x03 }, { 0x0000d486, 0x43 }, { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x00 }, - { 0x1000db00, 0x04 }, + { 0x1000db00, 0x07 }, { 0x1000db01, 0x00 }, { 0x1000db02, 0x11 }, { 0x1000db03, 0x00 }, @@ -225,6 +225,21 @@ static const struct reg_sequence rt1320_vc_blind_write[] = { { 0x1000db12, 0x00 }, { 0x1000db13, 0x00 }, { 0x1000db14, 0x45 }, + { 0x1000db15, 0x0d }, + { 0x1000db16, 0x01 }, + { 0x1000db17, 0x00 }, + { 0x1000db18, 0x00 }, + { 0x1000db19, 0xbf }, + { 0x1000db1a, 0x13 }, + { 0x1000db1b, 0x09 }, + { 0x1000db1c, 0x00 }, + { 0x1000db1d, 0x00 }, + { 0x1000db1e, 0x00 }, + { 0x1000db1f, 0x12 }, + { 0x1000db20, 0x09 }, + { 0x1000db21, 0x00 }, + { 0x1000db22, 0x00 }, + { 0x1000db23, 0x00 }, { 0x0000d540, 0x01 }, { 0x0000c081, 0xfc }, { 0x0000f01e, 0x80 }, -- cgit v1.2.3-59-g8ed1b From 59a4d9a9efd921ae2b722ffa52217124bcbc0acf Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:12:31 +0200 Subject: ASoC: pcm: Do not open FEs with no BEs connected MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The check is performed in prepare-step, that is dpcm_fe_dai_prepare() but that is very late - code operates on invalid configuration from dpcm_fe_dai_open() till it gets there. Relocate the check to the open-step to avoid any invalid scenarios. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141231.2943351-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 43835197d1fe..2c21fd528afd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2510,17 +2510,6 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - /* there is no point preparing this FE if there are no BEs */ - if (list_empty(&fe->dpcm[stream].be_clients)) { - /* dev_err_once() for visibility, dev_dbg() for debugging UCM profiles */ - dev_err_once(fe->dev, "ASoC: no backend DAIs enabled for %s, possibly missing ALSA mixer-based routing or UCM profile\n", - fe->dai_link->name); - dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n", - fe->dai_link->name); - ret = -EINVAL; - goto out; - } - ret = dpcm_be_dai_prepare(fe, stream); if (ret < 0) goto out; @@ -2776,11 +2765,23 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) /* calculate valid and active FE <-> BE dpcms */ dpcm_add_paths(fe, stream, &list); + /* There is no point starting up this FE if there are no BEs. */ + if (list_empty(&fe->dpcm[stream].be_clients)) { + /* dev_err_once() for visibility, dev_dbg() for debugging UCM profiles. */ + dev_err_once(fe->dev, "ASoC: no backend DAIs enabled for %s, possibly missing ALSA mixer-based routing or UCM profile\n", + fe->dai_link->name); + dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n", fe->dai_link->name); + + ret = -EINVAL; + goto put_path; + } + ret = dpcm_fe_dai_startup(fe_substream); if (ret < 0) dpcm_fe_dai_cleanup(fe_substream); dpcm_clear_pending_state(fe, stream); +put_path: dpcm_path_put(&list); open_end: snd_soc_dpcm_mutex_unlock(fe); -- cgit v1.2.3-59-g8ed1b From bae071aa7bcd034054cec91666c80f812adeccd9 Mon Sep 17 00:00:00 2001 From: Yuuki NAGAO Date: Sat, 31 May 2025 23:13:41 +0900 Subject: ASoC: ti: omap-hdmi: Re-add dai_link->platform to fix card init The removed dai_link->platform component cause a fail which is exposed at runtime. (ex: when a sound tool is used) This patch re-adds the dai_link->platform component to have a full card registered. Before this patch: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 1: HDMI [HDMI], device 0: HDMI snd-soc-dummy-dai-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ speaker-test -D plughw:1,0 -t sine speaker-test 1.2.8 Playback device is plughw:1,0 Stream parameters are 48000Hz, S16_LE, 1 channels Sine wave rate is 440.0000Hz Playback open error: -22,Invalid argument After this patch which restores the platform component: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDMI], device 0: HDMI snd-soc-dummy-dai-0 [HDMI snd-soc-dummy-dai-0] Subdevices: 0/1 Subdevice #0: subdevice #0 -> Resolve the playback error. Fixes: 3b0db249cf8f ("ASoC: ti: remove unnecessary dai_link->platform") Signed-off-by: Yuuki NAGAO Link: https://patch.msgid.link/20250531141341.81164-1-wf.yn386@gmail.com Signed-off-by: Mark Brown --- sound/soc/ti/omap-hdmi.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index cf43ac19c4a6..55e7cb96858f 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -361,17 +361,20 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card->dai_link) return -ENOMEM; - compnent = devm_kzalloc(dev, sizeof(*compnent), GFP_KERNEL); + compnent = devm_kzalloc(dev, 2 * sizeof(*compnent), GFP_KERNEL); if (!compnent) return -ENOMEM; - card->dai_link->cpus = compnent; + card->dai_link->cpus = &compnent[0]; card->dai_link->num_cpus = 1; card->dai_link->codecs = &snd_soc_dummy_dlc; card->dai_link->num_codecs = 1; + card->dai_link->platforms = &compnent[1]; + card->dai_link->num_platforms = 1; card->dai_link->name = card->name; card->dai_link->stream_name = card->name; card->dai_link->cpus->dai_name = dev_name(ad->dssdev); + card->dai_link->platforms->name = dev_name(ad->dssdev); card->num_links = 1; card->dev = dev; -- cgit v1.2.3-59-g8ed1b From 4919353c7789b8047e06a9b2b943f775a8f72883 Mon Sep 17 00:00:00 2001 From: Mario Limonciello Date: Tue, 27 May 2025 12:26:56 -0500 Subject: ALSA: usb-audio: Add a quirk for Lenovo Thinkpad Thunderbolt 3 dock The audio controller in the Lenovo Thinkpad Thunderbolt 3 dock doesn't support reading the sampling rate. Add a quirk for it. Suggested-by: Takashi Iwai Signed-off-by: Mario Limonciello Link: https://patch.msgid.link/20250527172657.1972565-1-superm1@kernel.org Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5ed523b13fad..bd24f3a78ea9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2285,6 +2285,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DISABLE_AUTOSUSPEND), DEVICE_FLG(0x17aa, 0x104d, /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */ QUIRK_FLAG_DISABLE_AUTOSUSPEND), + DEVICE_FLG(0x17ef, 0x3083, /* Lenovo TBT3 dock */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x1852, 0x5062, /* Luxman D-08u */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), DEVICE_FLG(0x1852, 0x5065, /* Luxman DA-06 */ -- cgit v1.2.3-59-g8ed1b From 9a4e17b59631e55c79ce0efbfc9f9eb85d377c6a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 28 May 2025 21:44:03 +0200 Subject: ALSA: hda: cs35l41: Constify regmap_irq_chip Static 'struct regmap_irq_chip' is not modified so can be changed to const for more safety. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20250528194402.567062-2-krzysztof.kozlowski@linaro.org Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index d9c8872b1866..d5bc81099d0d 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1603,7 +1603,7 @@ static const struct regmap_irq cs35l41_reg_irqs[] = { CS35L41_REG_IRQ(IRQ1_STATUS1, AMP_SHORT_ERR), }; -static struct regmap_irq_chip cs35l41_regmap_irq_chip = { +static const struct regmap_irq_chip cs35l41_regmap_irq_chip = { .name = "cs35l41 IRQ1 Controller", .status_base = CS35L41_IRQ1_STATUS1, .mask_base = CS35L41_IRQ1_MASK1, -- cgit v1.2.3-59-g8ed1b From ab72bfce7647522e01a181e3600c3d14ff5c143e Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 29 May 2025 11:08:13 +0530 Subject: ALSA: hda: Add new pci id for AMD GPU display HD audio controller Add new pci id for AMD GPU display HD audio controller(device id- 0xab40). Signed-off-by: Vijendar Mukunda Reviewed-by: Alex Deucher Link: https://patch.msgid.link/20250529053838.2350071-1-Vijendar.Mukunda@amd.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e6df706f740d..e5210ed48ddf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2724,6 +2724,9 @@ static const struct pci_device_id azx_ids[] = { { PCI_VDEVICE(ATI, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + { PCI_VDEVICE(ATI, 0xab40), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, /* GLENFLY */ { PCI_DEVICE(PCI_VENDOR_ID_GLENFLY, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- cgit v1.2.3-59-g8ed1b From 3f100f524e75586537e337b34d18c8d604b398e7 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:13:09 +0200 Subject: ALSA: hda: Ignore unsol events for cards being shut down MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For the classic snd_hda_intel driver, codec->card and bus->card point to the exact same thing. When snd_card_diconnect() fires, bus->shutdown is set thanks to azx_dev_disconnect(). card->shutdown is already set when that happens but both provide basically the same functionality. For the DSP snd_soc_avs driver where multiple codecs are located on multiple cards, bus->shutdown 'shortcut' is not sufficient. One codec card may be unregistered while other codecs are still operational. Proper check in form of card->shutdown must be used to verify whether the codec's card is being shut down. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141309.2943404-1-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 1fef350d821e..df8f88beddd0 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -44,7 +44,7 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) struct hda_codec *codec = container_of(dev, struct hda_codec, core); /* ignore unsol events during shutdown */ - if (codec->bus->shutdown) + if (codec->card->shutdown || codec->bus->shutdown) return; /* ignore unsol events during system suspend/resume */ -- cgit v1.2.3-59-g8ed1b From 6a3439a417b910e662c666993798e0691bc81147 Mon Sep 17 00:00:00 2001 From: David Heimann Date: Sun, 1 Jun 2025 12:41:16 -0400 Subject: ALSA: usb-audio: Add implicit feedback quirk for RODE AI-1 The RODE AI-1 audio interface requires implicit feedback sync between playback endpoint 0x03 and feedback endpoint 0x84 on interface 3, but doesn't advertise this in its USB descriptors. Without this quirk, the device receives audio data but produces no output. Signed-off-by: David Heimann Cc: Link: https://patch.msgid.link/084dc88c-1193-4a94-a002-5599adff936c@app.fastmail.com Signed-off-by: Takashi Iwai --- sound/usb/implicit.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 4727043fd745..77f06da93151 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -57,6 +57,7 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */ IMPLICIT_FB_FIXED_DEV(0x0499, 0x172f, 0x81, 2), /* Steinberg UR22C */ IMPLICIT_FB_FIXED_DEV(0x0d9a, 0x00df, 0x81, 2), /* RTX6001 */ + IMPLICIT_FB_FIXED_DEV(0x19f7, 0x000a, 0x84, 3), /* RODE AI-1 */ IMPLICIT_FB_FIXED_DEV(0x22f0, 0x0006, 0x81, 3), /* Allen&Heath Qu-16 */ IMPLICIT_FB_FIXED_DEV(0x1686, 0xf029, 0x82, 2), /* Zoom UAC-2 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8003, 0x86, 2), /* Fractal Audio Axe-Fx II */ -- cgit v1.2.3-59-g8ed1b From 065a651e2fb35209cbfe6fc5f1ababf92b66d4a4 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 30 May 2025 16:10:24 +0200 Subject: ASoC: Intel: avs: Simplify verification of parse_int_array() result MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The function return either success or an error code, no need to involve '<' operator. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20250530141025.2942936-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/board_selection.c | 2 +- sound/soc/intel/avs/debugfs.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 636315060eb4..673ccf162023 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -548,7 +548,7 @@ static int avs_register_i2s_test_boards(struct avs_dev *adev) u32 *array, num_elems; ret = parse_int_array(i2s_test, strlen(i2s_test), (int **)&array); - if (ret < 0) { + if (ret) { dev_err(adev->dev, "failed to parse i2s_test parameter\n"); return ret; } diff --git a/sound/soc/intel/avs/debugfs.c b/sound/soc/intel/avs/debugfs.c index 0e826ca20619..c625cf879f17 100644 --- a/sound/soc/intel/avs/debugfs.c +++ b/sound/soc/intel/avs/debugfs.c @@ -144,7 +144,7 @@ static ssize_t probe_points_write(struct file *file, const char __user *from, si int ret; ret = parse_int_array_user(from, count, (int **)&array); - if (ret < 0) + if (ret) return ret; num_elems = *array; @@ -181,7 +181,7 @@ static ssize_t probe_points_disconnect_write(struct file *file, const char __use int ret; ret = parse_int_array_user(from, count, (int **)&array); - if (ret < 0) + if (ret) return ret; num_elems = *array; @@ -369,7 +369,7 @@ static ssize_t trace_control_write(struct file *file, const char __user *from, s int ret; ret = parse_int_array_user(from, count, (int **)&array); - if (ret < 0) + if (ret) return ret; num_elems = *array; -- cgit v1.2.3-59-g8ed1b From b7188a1c0d2d7b04f4558e10293651d49fcb2398 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Tue, 3 Jun 2025 16:52:44 +0200 Subject: ASoC: Intel: avs: boards: Fix rt5663 front end name MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix copy paste mistake in board name. Fixes: f1e282c333ac ("ASoC: Intel: avs: boards: Change rt5663 card name") Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://patch.msgid.link/20250603145244.871239-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt5663.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c index 122b6c48fd80..51648801710a 100644 --- a/sound/soc/intel/avs/boards/rt5663.c +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -228,7 +228,7 @@ static int avs_rt5663_probe(struct platform_device *pdev) card->name = "avs_rt5663"; } else { card->driver_name = "avs_rt5663"; - card->long_name = card->name = "AVS I2S ALC5640"; + card->long_name = card->name = "AVS I2S ALC5663"; } card->dev = dev; card->owner = THIS_MODULE; -- cgit v1.2.3-59-g8ed1b