From eafae0fdd115a71b3a200ef1a31f86da04bac77f Mon Sep 17 00:00:00 2001 From: Evgeniy Harchenko Date: Fri, 15 Aug 2025 12:58:14 +0300 Subject: [PATCH 01/12] ALSA: hda/realtek: Add support for HP EliteBook x360 830 G6 and EliteBook 830 G6 The HP EliteBook x360 830 G6 and HP EliteBook 830 G6 have Realtek HDA codec ALC215. It needs the ALC285_FIXUP_HP_GPIO_LED quirk to enable the mute LED. Cc: Signed-off-by: Evgeniy Harchenko Link: https://patch.msgid.link/20250815095814.75845-1-evgeniyharchenko.dev@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index e90c4047ea62..db8e6352b942 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6368,6 +6368,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8548, "HP EliteBook x360 830 G6", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x854a, "HP EliteBook 830 G6", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x85c6, "HP Pavilion x360 Convertible 14-dy1xxx", ALC295_FIXUP_HP_MUTE_LED_COEFBIT11), SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360), SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), From c0ed3c2edc7692c6b8af7578b41012694dc8c671 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sat, 16 Aug 2025 12:27:41 +0800 Subject: [PATCH 02/12] ALSA: hda/tas2781: Add name prefix tas2781 for tas2781's dvc_tlv and amp_vol_tlv With some new devices adding into the driver, dvc_tlv and amp_vol_tlv will cause confusion for customers on which devices they support. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20250816042741.1659-1-shenghao-ding@ti.com Signed-off-by: Takashi Iwai --- include/sound/tas2781-tlv.h | 6 +++--- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- sound/hda/codecs/side-codecs/tas2781_hda_spi.c | 6 ++++-- sound/soc/codecs/tas2781-i2c.c | 4 ++-- 4 files changed, 10 insertions(+), 8 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index ef9b9f19d212..273224df9282 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, -10000, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 06c7bc2b9e9d..b91fff3fde97 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -267,7 +267,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 09a5d0f131b2..b9a55672bf15 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -494,9 +494,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new tas2781_snd_ctls[] = { ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0, - tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_getvol, tas2781_amp_putvol, + tas2781_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1, - tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), + tas2781_digital_getvol, tas2781_digital_putvol, + tas2781_dvc_tlv), ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 676130f4cf3e..0e09d794516f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -910,10 +910,10 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), + tas2781_digital_putvol, tas2781_dvc_tlv), }; static const struct snd_kcontrol_new tas2781_cali_controls[] = { From 89f0addeee3cb2dc49837599330ed9c4612f05b0 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 18 Aug 2025 12:59:45 +0300 Subject: [PATCH 03/12] ALSA: usb-audio: Fix size validation in convert_chmap_v3() The "p" pointer is void so sizeof(*p) is 1. The intent was to check sizeof(*cs_desc), which is 3, instead. Fixes: ecfd41166b72 ("ALSA: usb-audio: Validate UAC3 cluster segment descriptors") Signed-off-by: Dan Carpenter Link: https://patch.msgid.link/aKL5kftC1qGt6lpv@stanley.mountain Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index acf3dc2d79e0..5c235a5ba7e1 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -349,7 +349,7 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor u16 cs_len; u8 cs_type; - if (len < sizeof(*p)) + if (len < sizeof(*cs_desc)) break; cs_len = le16_to_cpu(cs_desc->wLength); if (len < cs_len) From af24c20c4633a667ac5b5e20cf9d96f6176a0ca3 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Fri, 15 Aug 2025 10:47:29 +0800 Subject: [PATCH 04/12] ASoC: codecs: ES9389: Modify the standby configuration Modify the standby configuration Signed-off-by: Zhang Yi Link: https://patch.msgid.link/20250815024729.3051-1-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8389.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8389.c b/sound/soc/codecs/es8389.c index ba1763f36f17..6e4c75d288ef 100644 --- a/sound/soc/codecs/es8389.c +++ b/sound/soc/codecs/es8389.c @@ -636,7 +636,7 @@ static int es8389_set_bias_level(struct snd_soc_component *component, regmap_write(es8389->regmap, ES8389_ANA_CTL1, 0x59); regmap_write(es8389->regmap, ES8389_ADC_EN, 0x00); regmap_write(es8389->regmap, ES8389_CLK_OFF1, 0x00); - regmap_write(es8389->regmap, ES8389_RESET, 0x7E); + regmap_write(es8389->regmap, ES8389_RESET, 0x3E); regmap_update_bits(es8389->regmap, ES8389_DAC_INV, 0x80, 0x80); usleep_range(8000, 8500); regmap_update_bits(es8389->regmap, ES8389_DAC_INV, 0x80, 0x00); From 018f659753fd38bb6fdba7fa8c751121b495e1f4 Mon Sep 17 00:00:00 2001 From: Vasiliy Kovalev Date: Mon, 18 Aug 2025 23:42:43 +0300 Subject: [PATCH 05/12] ALSA: hda/realtek: Fix headset mic on ASUS Zenbook 14 Add a PCI quirk to enable microphone input on the headphone jack on the ASUS Zenbook 14 UM3406HA laptop. This model uses an ALC294 codec with CS35L41 amplifiers over I2C, and the existing fixup for it did not enable the headset microphone. A new fix is introduced to get the mic working while keeping the amplifier settings correct. Fixes: 61cbc08fdb04 ("ALSA: hda/realtek: Add quirks for ASUS 2024 Zenbooks") Signed-off-by: Vasiliy Kovalev Link: https://patch.msgid.link/20250818204243.247297-1-kovalev@altlinux.org Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index db8e6352b942..6c78a286172c 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -3579,6 +3579,7 @@ enum { ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, ALC294_FIXUP_ASUS_MIC, ALC294_FIXUP_ASUS_HEADSET_MIC, + ALC294_FIXUP_ASUS_I2C_HEADSET_MIC, ALC294_FIXUP_ASUS_SPK, ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, @@ -4889,6 +4890,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_I2C_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a19020 }, /* use as headset mic */ + { } + }, + .chained = true, + .chain_id = ALC287_FIXUP_CS35L41_I2C_2 + }, [ALC294_FIXUP_ASUS_SPK] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -6730,7 +6740,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC294_FIXUP_ASUS_I2C_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), From f4b3cef55f5f96fdb4e7f9ca90b7d6213689faeb Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 19 Aug 2025 14:03:44 +0800 Subject: [PATCH 06/12] ALSA: hda/realtek: Audio disappears on HP 15-fc000 after warm boot again There was a similar bug in the past (Bug 217440), which was fixed for this laptop. The same issue is occurring again as of kernel v.6.12.2. The symptoms are very similar - initially audio works but after a warm reboot, the audio completely disappears until the computer is powered off (there is no audio output at all). The issue is also related by caused by a different change now. By bisecting different kernel versions, I found that reverting cc3d0b5dd989 in patch_realtek.c[*] restores the sound and it works fine after the reboot. [*] https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/commit/sound/pci/hda/patch_realtek.c?h=v6.12.2&id=4ed7f16070a8475c088ff423b2eb11ba15eb89b6 [ patch description reformatted by tiwai ] Fixes: cc3d0b5dd989 ("ALSA: hda/realtek: Update ALC256 depop procedure") Link: https://bugzilla.kernel.org/show_bug.cgi?id=220109 Signed-off-by: Kailang Yang Link: https://lore.kernel.org/5317ca723c82447a938414fcca85cbf5@realtek.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 6c78a286172c..0323606b3d6d 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -510,6 +510,15 @@ static void alc256_shutup(struct hda_codec *codec) hp_pin = 0x21; alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */ + + /* 3k pull low control for Headset jack. */ + /* NOTE: call this before clearing the pin, otherwise codec stalls */ + /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly + * when booting with headset plugged. So skip setting it for the codec alc257 + */ + if (spec->en_3kpull_low) + alc_update_coef_idx(codec, 0x46, 0, 3 << 12); + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); if (hp_pin_sense) { @@ -520,14 +529,6 @@ static void alc256_shutup(struct hda_codec *codec) msleep(75); - /* 3k pull low control for Headset jack. */ - /* NOTE: call this before clearing the pin, otherwise codec stalls */ - /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly - * when booting with headset plugged. So skip setting it for the codec alc257 - */ - if (spec->en_3kpull_low) - alc_update_coef_idx(codec, 0x46, 0, 3 << 12); - if (!spec->no_shutup_pins) snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); From 3f4422e7c9436abf81a00270be7e4d6d3760ec0e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Aug 2025 07:19:01 +0200 Subject: [PATCH 07/12] ALSA: hda: tas2781: Fix wrong reference of tasdevice_priv During the conversion to unify the calibration data management, the reference to tasdevice_priv was wrongly set to h->hda_priv instead of h->priv. This resulted in memory corruption and crashes eventually. Unfortunately it's a void pointer, hence the compiler couldn't know that it's wrong. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Link: https://bugzilla.suse.com/show_bug.cgi?id=1248270 Cc: Link: https://patch.msgid.link/20250820051902.4523-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 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 b91fff3fde97..e34b17f0c9b9 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -305,7 +305,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) efi_char16_t efi_name[TAS2563_CAL_VAR_NAME_MAX]; unsigned long max_size = TAS2563_CAL_DATA_SIZE; unsigned char var8[TAS2563_CAL_VAR_NAME_MAX]; - struct tasdevice_priv *p = h->hda_priv; + struct tasdevice_priv *p = h->priv; struct calidata *cd = &p->cali_data; struct cali_reg *r = &cd->cali_reg_array; unsigned int offset = 0; From f135fb24ef29335b94921077588cae445bc7f099 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 20 Aug 2025 15:22:00 +0100 Subject: [PATCH 08/12] ASoC: cs35l56: Update Firmware Addresses for CS35L63 for production silicon Production silicon for CS36L63 has some small differences compared to pre-production silicon. Update firmware addresses, which are different. No product was ever released with pre-production silicon so there is no need for the driver to include support for it. Fixes: 978858791ced ("ASoC: cs35l56: Add initial support for CS35L63 for I2C and SoundWire") Signed-off-by: Stefan Binding Link: https://patch.msgid.link/20250820142209.127575-2-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index e17c4cadd04d..f44aabde805e 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -107,8 +107,8 @@ #define CS35L56_DSP1_PMEM_5114 0x3804FE8 #define CS35L63_DSP1_FW_VER CS35L56_DSP1_FW_VER -#define CS35L63_DSP1_HALO_STATE 0x280396C -#define CS35L63_DSP1_PM_CUR_STATE 0x28042C8 +#define CS35L63_DSP1_HALO_STATE 0x2803C04 +#define CS35L63_DSP1_PM_CUR_STATE 0x2804518 #define CS35L63_PROTECTION_STATUS 0x340009C #define CS35L63_TRANSDUCER_ACTUAL_PS 0x34000F4 #define CS35L63_MAIN_RENDER_USER_MUTE 0x3400020 From 8dadc11b67d4b83deff45e4889b3b5540b9c0a7f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 20 Aug 2025 15:22:01 +0100 Subject: [PATCH 09/12] ASoC: cs35l56: Handle new algorithms IDs for CS35L63 CS35L63 uses different algorithm IDs from CS35L56. Add a new mechanism to handle different alg IDs between parts in the CS35L56 driver. Fixes: 978858791ced ("ASoC: cs35l56: Add initial support for CS35L63 for I2C and SoundWire") Signed-off-by: Richard Fitzgerald Signed-off-by: Stefan Binding Link: https://patch.msgid.link/20250820142209.127575-3-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 1 + sound/soc/codecs/cs35l56-shared.c | 29 ++++++++++++++++++++++++++--- sound/soc/codecs/cs35l56.c | 2 +- 3 files changed, 28 insertions(+), 4 deletions(-) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index f44aabde805e..7c8bbe8ad1e2 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -306,6 +306,7 @@ struct cs35l56_base { struct gpio_desc *reset_gpio; struct cs35l56_spi_payload *spi_payload_buf; const struct cs35l56_fw_reg *fw_reg; + const struct cirrus_amp_cal_controls *calibration_controls; }; static inline bool cs35l56_is_otp_register(unsigned int reg) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index ba653f6ccfae..850fcf385996 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -838,6 +838,15 @@ const struct cirrus_amp_cal_controls cs35l56_calibration_controls = { }; EXPORT_SYMBOL_NS_GPL(cs35l56_calibration_controls, "SND_SOC_CS35L56_SHARED"); +static const struct cirrus_amp_cal_controls cs35l63_calibration_controls = { + .alg_id = 0xbf210, + .mem_region = WMFW_ADSP2_YM, + .ambient = "CAL_AMBIENT", + .calr = "CAL_R", + .status = "CAL_STATUS", + .checksum = "CAL_CHECKSUM", +}; + int cs35l56_get_calibration(struct cs35l56_base *cs35l56_base) { u64 silicon_uid = 0; @@ -912,19 +921,31 @@ EXPORT_SYMBOL_NS_GPL(cs35l56_read_prot_status, "SND_SOC_CS35L56_SHARED"); void cs35l56_log_tuning(struct cs35l56_base *cs35l56_base, struct cs_dsp *cs_dsp) { __be32 pid, sid, tid; + unsigned int alg_id; int ret; + switch (cs35l56_base->type) { + case 0x54: + case 0x56: + case 0x57: + alg_id = 0x9f212; + break; + default: + alg_id = 0xbf212; + break; + } + scoped_guard(mutex, &cs_dsp->pwr_lock) { ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_PRJCT_ID", - WMFW_ADSP2_XM, 0x9f212), + WMFW_ADSP2_XM, alg_id), 0, &pid, sizeof(pid)); if (!ret) ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_CHNNL_ID", - WMFW_ADSP2_XM, 0x9f212), + WMFW_ADSP2_XM, alg_id), 0, &sid, sizeof(sid)); if (!ret) ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_SNPSHT_ID", - WMFW_ADSP2_XM, 0x9f212), + WMFW_ADSP2_XM, alg_id), 0, &tid, sizeof(tid)); } @@ -974,8 +995,10 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) case 0x35A54: case 0x35A56: case 0x35A57: + cs35l56_base->calibration_controls = &cs35l56_calibration_controls; break; case 0x35A630: + cs35l56_base->calibration_controls = &cs35l63_calibration_controls; devid = devid >> 4; break; default: diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index b1c65d8331e7..2c1edbd636ef 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -695,7 +695,7 @@ static int cs35l56_write_cal(struct cs35l56_private *cs35l56) return ret; ret = cs_amp_write_cal_coeffs(&cs35l56->dsp.cs_dsp, - &cs35l56_calibration_controls, + cs35l56->base.calibration_controls, &cs35l56->base.cal_data); wm_adsp_stop(&cs35l56->dsp); From 8d13d1bdb59d0a2c526869ee571ec51a3a887463 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 20 Aug 2025 15:22:02 +0100 Subject: [PATCH 10/12] ASoC: cs35l56: Remove SoundWire Clock Divider workaround for CS35L63 Production silicon for CS36L63 has some small differences compared to pre-production silicon. Remove soundwire clock workaround as no longer necessary. We don't want to do tricks with low-level clocking controls if we don't need to. Fixes: 978858791ced ("ASoC: cs35l56: Add initial support for CS35L63 for I2C and SoundWire") Signed-off-by: Stefan Binding Link: https://patch.msgid.link/20250820142209.127575-4-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-sdw.c | 69 ---------------------------------- sound/soc/codecs/cs35l56.h | 3 -- 2 files changed, 72 deletions(-) diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index ee14031695a1..3905c9cb188a 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -393,74 +393,6 @@ static int cs35l56_sdw_update_status(struct sdw_slave *peripheral, return 0; } -static int cs35l63_sdw_kick_divider(struct cs35l56_private *cs35l56, - struct sdw_slave *peripheral) -{ - unsigned int curr_scale_reg, next_scale_reg; - int curr_scale, next_scale, ret; - - if (!cs35l56->base.init_done) - return 0; - - if (peripheral->bus->params.curr_bank) { - curr_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B1; - next_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B0; - } else { - curr_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B0; - next_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B1; - } - - /* - * Current clock scale value must be different to new value. - * Modify current to guarantee this. If next still has the dummy - * value we wrote when it was current, the core code has not set - * a new scale so restore its original good value - */ - curr_scale = sdw_read_no_pm(peripheral, curr_scale_reg); - if (curr_scale < 0) { - dev_err(cs35l56->base.dev, "Failed to read current clock scale: %d\n", curr_scale); - return curr_scale; - } - - next_scale = sdw_read_no_pm(peripheral, next_scale_reg); - if (next_scale < 0) { - dev_err(cs35l56->base.dev, "Failed to read next clock scale: %d\n", next_scale); - return next_scale; - } - - if (next_scale == CS35L56_SDW_INVALID_BUS_SCALE) { - next_scale = cs35l56->old_sdw_clock_scale; - ret = sdw_write_no_pm(peripheral, next_scale_reg, next_scale); - if (ret < 0) { - dev_err(cs35l56->base.dev, "Failed to modify current clock scale: %d\n", - ret); - return ret; - } - } - - cs35l56->old_sdw_clock_scale = curr_scale; - ret = sdw_write_no_pm(peripheral, curr_scale_reg, CS35L56_SDW_INVALID_BUS_SCALE); - if (ret < 0) { - dev_err(cs35l56->base.dev, "Failed to modify current clock scale: %d\n", ret); - return ret; - } - - dev_dbg(cs35l56->base.dev, "Next bus scale: %#x\n", next_scale); - - return 0; -} - -static int cs35l56_sdw_bus_config(struct sdw_slave *peripheral, - struct sdw_bus_params *params) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); - - if ((cs35l56->base.type == 0x63) && (cs35l56->base.rev < 0xa1)) - return cs35l63_sdw_kick_divider(cs35l56, peripheral); - - return 0; -} - static int __maybe_unused cs35l56_sdw_clk_stop(struct sdw_slave *peripheral, enum sdw_clk_stop_mode mode, enum sdw_clk_stop_type type) @@ -476,7 +408,6 @@ static const struct sdw_slave_ops cs35l56_sdw_ops = { .read_prop = cs35l56_sdw_read_prop, .interrupt_callback = cs35l56_sdw_interrupt, .update_status = cs35l56_sdw_update_status, - .bus_config = cs35l56_sdw_bus_config, #ifdef DEBUG .clk_stop = cs35l56_sdw_clk_stop, #endif diff --git a/sound/soc/codecs/cs35l56.h b/sound/soc/codecs/cs35l56.h index bd77a57249d7..40a1800a4585 100644 --- a/sound/soc/codecs/cs35l56.h +++ b/sound/soc/codecs/cs35l56.h @@ -20,8 +20,6 @@ #define CS35L56_SDW_GEN_INT_MASK_1 0xc1 #define CS35L56_SDW_INT_MASK_CODEC_IRQ BIT(0) -#define CS35L56_SDW_INVALID_BUS_SCALE 0xf - #define CS35L56_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) #define CS35L56_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ | SNDRV_PCM_FMTBIT_S32_LE) @@ -52,7 +50,6 @@ struct cs35l56_private { u8 asp_slot_count; bool tdm_mode; bool sysclk_set; - u8 old_sdw_clock_scale; u8 sdw_link_num; u8 sdw_unique_id; }; From 5003a65790ed66be882d1987cc2ca86af0de3db1 Mon Sep 17 00:00:00 2001 From: Dewei Meng Date: Thu, 21 Aug 2025 09:43:17 +0800 Subject: [PATCH 11/12] ALSA: timer: fix ida_free call while not allocated In the snd_utimer_create() function, if the kasprintf() function return NULL, snd_utimer_put_id() will be called, finally use ida_free() to free the unallocated id 0. the syzkaller reported the following information: ------------[ cut here ]------------ ida_free called for id=0 which is not allocated. WARNING: CPU: 1 PID: 1286 at lib/idr.c:592 ida_free+0x1fd/0x2f0 lib/idr.c:592 Modules linked in: CPU: 1 UID: 0 PID: 1286 Comm: syz-executor164 Not tainted 6.15.8 #3 PREEMPT(lazy) Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.16.3-4.fc42 04/01/2014 RIP: 0010:ida_free+0x1fd/0x2f0 lib/idr.c:592 Code: f8 fc 41 83 fc 3e 76 69 e8 70 b2 f8 (...) RSP: 0018:ffffc900007f79c8 EFLAGS: 00010282 RAX: 0000000000000000 RBX: 1ffff920000fef3b RCX: ffffffff872176a5 RDX: ffff88800369d200 RSI: 0000000000000000 RDI: ffff88800369d200 RBP: 0000000000000000 R08: ffffffff87ba60a5 R09: 0000000000000000 R10: 0000000000000001 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000002 R14: 0000000000000000 R15: 0000000000000000 FS: 00007f6f1abc1740(0000) GS:ffff8880d76a0000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007f6f1ad7a784 CR3: 000000007a6e2000 CR4: 00000000000006f0 Call Trace: snd_utimer_put_id sound/core/timer.c:2043 [inline] [snd_timer] snd_utimer_create+0x59b/0x6a0 sound/core/timer.c:2184 [snd_timer] snd_utimer_ioctl_create sound/core/timer.c:2202 [inline] [snd_timer] __snd_timer_user_ioctl.isra.0+0x724/0x1340 sound/core/timer.c:2287 [snd_timer] snd_timer_user_ioctl+0x75/0xc0 sound/core/timer.c:2298 [snd_timer] vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:907 [inline] __se_sys_ioctl fs/ioctl.c:893 [inline] __x64_sys_ioctl+0x198/0x200 fs/ioctl.c:893 do_syscall_x64 arch/x86/entry/syscall_64.c:63 [inline] do_syscall_64+0x7b/0x160 arch/x86/entry/syscall_64.c:94 entry_SYSCALL_64_after_hwframe+0x76/0x7e [...] The utimer->id should be set properly before the kasprintf() function, ensures the snd_utimer_put_id() function will free the allocated id. Fixes: 37745918e0e75 ("ALSA: timer: Introduce virtual userspace-driven timers") Signed-off-by: Dewei Meng Link: https://patch.msgid.link/20250821014317.40786-1-mengdewei@cqsoftware.com.cn Signed-off-by: Takashi Iwai --- sound/core/timer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 3ce12264eed8..d9fff5c87613 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -2139,14 +2139,14 @@ static int snd_utimer_create(struct snd_timer_uinfo *utimer_info, goto err_take_id; } + utimer->id = utimer_id; + utimer->name = kasprintf(GFP_KERNEL, "snd-utimer%d", utimer_id); if (!utimer->name) { err = -ENOMEM; goto err_get_name; } - utimer->id = utimer_id; - tid.dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION; tid.dev_class = SNDRV_TIMER_CLASS_GLOBAL; tid.card = -1; From 8410fe81093ff231e964891e215b624dabb734b0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Aug 2025 17:08:34 +0200 Subject: [PATCH 12/12] ALSA: usb-audio: Use correct sub-type for UAC3 feature unit validation The entry of the validators table for UAC3 feature unit is defined with a wrong sub-type UAC_FEATURE (= 0x06) while it should have been UAC3_FEATURE (= 0x07). This patch corrects the entry value. Fixes: 57f8770620e9 ("ALSA: usb-audio: More validations of descriptor units") Link: https://patch.msgid.link/20250821150835.8894-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/validate.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/validate.c b/sound/usb/validate.c index 4f4e8e87a14c..a0d55b77c994 100644 --- a/sound/usb/validate.c +++ b/sound/usb/validate.c @@ -285,7 +285,7 @@ static const struct usb_desc_validator audio_validators[] = { /* UAC_VERSION_3, UAC3_EXTENDED_TERMINAL: not implemented yet */ FUNC(UAC_VERSION_3, UAC3_MIXER_UNIT, validate_mixer_unit), FUNC(UAC_VERSION_3, UAC3_SELECTOR_UNIT, validate_selector_unit), - FUNC(UAC_VERSION_3, UAC_FEATURE_UNIT, validate_uac3_feature_unit), + FUNC(UAC_VERSION_3, UAC3_FEATURE_UNIT, validate_uac3_feature_unit), /* UAC_VERSION_3, UAC3_EFFECT_UNIT: not implemented yet */ FUNC(UAC_VERSION_3, UAC3_PROCESSING_UNIT, validate_processing_unit), FUNC(UAC_VERSION_3, UAC3_EXTENSION_UNIT, validate_processing_unit),