From 92a8b5e2eff6920bf815cd6a80b088ec3fdf01a3 Mon Sep 17 00:00:00 2001 From: Yuriy Padlyak Date: Thu, 30 Apr 2026 01:09:03 +0300 Subject: [PATCH 01/28] ALSA: hda/realtek: Fix speaker silence after S3 resume on Xiaomi Mi Laptop Pro 15 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The Xiaomi Mi Laptop Pro 15 (TM1905, subsystem 1d72:1905) ships with the Realtek ALC256 codec on Intel Comet Lake PCH-LP. After S3 resume the codec sets coefficient register 0x10 to 0x0220 instead of 0x0020 — bit 9 is erroneously set, which silences the internal speaker. Bluetooth and HDMI audio are unaffected because they use different paths. This is the same mechanism fixed for Clevo NJ51CU by commit edca7cc4b0ac ("ALSA: hda/realtek: Fix quirk for Clevo NJ51CU"), but the existing ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME also reconfigures pin 0x19 as a front mic, which is wrong for this Xiaomi where pin 0x19 default is 0x411111f0 (disabled). Add a minimal fixup that only clears the stuck coef bit, and add the Xiaomi SSID to the quirk table. Verified by reading coef 0x10 with hda-verb after resume (returns 0x0220), writing 0x0020, and confirming the internal speaker resumes output. With this fixup applied the bit is cleared on every codec init, including post-resume. Signed-off-by: Yuriy Padlyak Cc: Tested-by: Yuriy Padlyak Link: https://patch.msgid.link/20260429220903.14918-1-yuriypadlyak@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index a9cd03bb73c4..9a5747c2e359 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -3390,6 +3390,19 @@ static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, } } +static void alc256_fixup_xiaomi_pro15_resume(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * On the Xiaomi Mi Laptop Pro 15 (TM1905, SSID 1d72:1905) the ALC256 + * codec sets coefficient 0x10 bit 9 to 1 after S3 resume, silencing + * the internal speaker. Bluetooth and HDMI audio are unaffected. + * Clear the bit so the speaker keeps working across suspend cycles. + */ + alc_update_coef_idx(codec, 0x10, 1<<9, 0); +} + static void alc256_decrease_headphone_amp_val(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4052,6 +4065,7 @@ enum { ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC233_FIXUP_NO_AUDIO_JACK, ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME, + ALC256_FIXUP_XIAOMI_PRO15_RESUME, ALC285_FIXUP_LEGION_Y9000X_SPEAKERS, ALC285_FIXUP_LEGION_Y9000X_AUTOMUTE, ALC287_FIXUP_LEGION_16ACHG6, @@ -6240,6 +6254,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC256_FIXUP_XIAOMI_PRO15_RESUME] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_xiaomi_pro15_resume, + }, [ALC287_FIXUP_LEGION_16ACHG6] = { .type = HDA_FIXUP_FUNC, .v.func = alc287_fixup_legion_16achg6_speakers, @@ -7774,6 +7792,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1d72, 0x1905, "Xiaomi Mi Laptop Pro 15", ALC256_FIXUP_XIAOMI_PRO15_RESUME), SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1e39, 0xca14, "MEDION NM14LNL", ALC233_FIXUP_MEDION_MTL_SPK), From 56722cfbb78d7eb41756cd78dc5192d08bd14f3d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?R=C3=A1mon=20van=20Raaij?= Date: Thu, 30 Apr 2026 21:12:24 +0200 Subject: [PATCH 02/28] ALSA: hda/realtek: Add codec SSID quirk for Lenovo Yoga Pro 9 16IMH9 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The Yoga Pro 9 16IMH9 (codec SSID 17aa:38d6) shares PCI audio device subsystem ID 17aa:3811 with the Legion S7 15IMH05. The existing SND_PCI_QUIRK entry for the Legion routes both machines to ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, which does not bind the TAS2781 smart amplifiers, resulting in near-silent built-in speakers. Add an HDA_CODEC_QUIRK entry immediately before the conflicting PCI quirk that matches the Yoga Pro 9's unique codec SSID and routes it to ALC287_FIXUP_TAS2781_I2C. Codec quirks are evaluated after PCI quirks and take precedence, leaving the Legion S7 15IMH05 entry unaffected. This follows the same pattern used to disambiguate PCI SSID 17aa:3847 (shared between Yoga Pro 7 14IMH9 and Legion 7 16ACHG6), where a HDA_CODEC_QUIRK for codec SSID 17aa:38cf resolves the conflict. Signed-off-by: Rámon van Raaij Link: https://patch.msgid.link/20260430191224.patch1-ramon@vanraaij.eu Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 9a5747c2e359..e061673d3c2e 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7647,6 +7647,10 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), HDA_CODEC_QUIRK(0x17aa, 0x3802, "DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga Pro 9 14IRP8", ALC287_FIXUP_TAS2781_I2C), + /* Yoga Pro 9 16IMH9 shares PCI SSID 17aa:3811 with Legion S7 15IMH05; + * use codec SSID to distinguish them + */ + HDA_CODEC_QUIRK(0x17aa, 0x38d6, "Lenovo Yoga Pro 9 16IMH9", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3811, "Legion S7 15IMH05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), From ad39a189bfebb3de580f390bc000f9e121c6aca3 Mon Sep 17 00:00:00 2001 From: Pauli Virtanen Date: Thu, 30 Apr 2026 23:49:50 +0300 Subject: [PATCH 03/28] ALSA: usb-audio: add min_mute quirk for Razer Nommo V2 X ID 1532:055e Razer USA, Ltd Razer Nommo V2 X is tested to have muted min playback volume. Apply quirk for that. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/work_items/5235 Signed-off-by: Pauli Virtanen Link: https://patch.msgid.link/94449577332d14d7974864903825f27e5824ddbc.1777579951.git.pav@iki.fi 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 0b4ecc2c6bcc..f77ff05dff0b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2366,6 +2366,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x152a, 0x880a, /* NeuralDSP Quad Cortex */ 0), /* Doesn't have the vendor quirk which would otherwise apply */ + DEVICE_FLG(0x1532, 0x055e, /* Razer Nommo V2 X */ + QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE), DEVICE_FLG(0x154e, 0x1002, /* Denon DCD-1500RE */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), DEVICE_FLG(0x154e, 0x1003, /* Denon DA-300USB */ From 359b626d362127451dbe8a687ac5c240f896ae2e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Fri, 1 May 2026 14:45:14 -0300 Subject: [PATCH 04/28] ALSA: pcmtest: Return -EFAULT on pattern read copy failure MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit pattern_write() reports -EFAULT when copy_from_user() fails, but pattern_read() converts copy_to_user() failures into a zero-length read. That makes a userspace buffer fault look like EOF instead of reporting the actual error. Return -EFAULT from pattern_read() when copying the pattern data to userspace fails, and update the file offset only after a successful copy. Fixes: 315a3d57c64c ("ALSA: Implement the new Virtual PCM Test Driver") Signed-off-by: Cássio Gabriel Link: https://patch.msgid.link/20260501-alsa-pcmtest-pattern-read-efault-v1-1-53e1e8c11dda@gmail.com Signed-off-by: Takashi Iwai --- sound/drivers/pcmtest.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 5bfec4c7bf71..7f93557b51ec 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -679,9 +679,9 @@ static ssize_t pattern_read(struct file *file, char __user *u_buff, size_t len, return 0; if (copy_to_user(u_buff, patt_buf->buf + *off, to_read)) - to_read = 0; - else - *off += to_read; + return -EFAULT; + + *off += to_read; return to_read; } From e366ce8b22ec68189ffea2bb8009f7b20d549b0f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig=20=28The=20Capable=20Hub=29?= Date: Thu, 30 Apr 2026 17:45:24 +0200 Subject: [PATCH 05/28] ASoC: codecs: ab8500: Remove suspicious code MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit anc_configure() passed values from drvdata->anc_fir_values[], drvdata->anc_iir_values[] and drvdata->sid_fir_values[] as register offset to snd_soc_component_read(). The content of these arrays are user controllable via the component controls "ANC FIR Coefficients", "ANC IIR Coefficients" and "Sidetone FIR Coefficients" which I assume are supposed to hold register values, not register offsets. Without a datasheet for that component and given that before commit a201aef1a88b ("ASoC: codecs: ab8500: Fix casting of private data") the arrays overlapped with driver control structures and thus didn't work properly since 2012, drop that functionality and let someone repair it who has an actual need for it. With the core functionally removed several code parts become essentially unused and are removed, too. Reported-by: Sashiko (gemini/gemini-3.1-pro-preview) Link: https://sashiko.dev/#/patchset/20260428192255.2294705-2-u.kleine-koenig%40baylibre.com Fixes: 679d7abdc754 ("ASoC: codecs: Add AB8500 codec-driver") Signed-off-by: Uwe Kleine-König (The Capable Hub) Link: https://patch.msgid.link/20260430154524.338912-2-u.kleine-koenig@baylibre.com Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 304 +------------------------------- 1 file changed, 3 insertions(+), 301 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 8ab2e60f80b4..6e8ef9cd1b31 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -60,19 +60,6 @@ low before proceeding with the configuration sequence */ #define AB8500_ANC_SM_DELAY 2000 -#define AB8500_FILTER_CONTROL(xname, xcount, xmin, xmax) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .info = filter_control_info, \ - .get = filter_control_get, .put = filter_control_put, \ - .private_value = (unsigned long)&(struct filter_control) \ - {.count = xcount, .min = xmin, .max = xmax} } - -struct filter_control { - long min, max; - unsigned int count; - long value[128]; -}; - /* Sidetone states */ static const char * const enum_sid_state[] = { "Unconfigured", @@ -85,45 +72,13 @@ enum sid_state { SID_FIR_CONFIGURED = 2, }; -static const char * const enum_anc_state[] = { - "Unconfigured", - "Apply FIR and IIR", - "FIR and IIR are configured", - "Apply FIR", - "FIR is configured", - "Apply IIR", - "IIR is configured" -}; -enum anc_state { - ANC_UNCONFIGURED = 0, - ANC_APPLY_FIR_IIR = 1, - ANC_FIR_IIR_CONFIGURED = 2, - ANC_APPLY_FIR = 3, - ANC_FIR_CONFIGURED = 4, - ANC_APPLY_IIR = 5, - ANC_IIR_CONFIGURED = 6 -}; - -/* Analog microphones */ -enum amic_idx { - AMIC_IDX_1A, - AMIC_IDX_1B, - AMIC_IDX_2 -}; - /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { struct regmap *regmap; struct mutex ctrl_lock; /* Sidetone */ - long *sid_fir_values; enum sid_state sid_status; - - /* ANC */ - long *anc_fir_values; - long *anc_iir_values; - enum anc_state anc_status; }; static inline const char *amic_micbias_str(enum amic_micbias micbias) @@ -1024,89 +979,6 @@ static const struct snd_soc_dapm_route ab8500_dapm_routes_mic2_vamicx[] = { {"MIC2 V-AMICx Enable", NULL, "V-AMIC2"}, }; -/* ANC FIR-coefficients configuration sequence */ -static void anc_fir(struct snd_soc_component *component, - unsigned int bnk, unsigned int par, unsigned int val) -{ - if (par == 0 && bnk == 0) - snd_soc_component_update_bits(component, AB8500_ANCCONF1, - BIT(AB8500_ANCCONF1_ANCFIRUPDATE), - BIT(AB8500_ANCCONF1_ANCFIRUPDATE)); - - snd_soc_component_write(component, AB8500_ANCCONF5, val >> 8 & 0xff); - snd_soc_component_write(component, AB8500_ANCCONF6, val & 0xff); - - if (par == AB8500_ANC_FIR_COEFFS - 1 && bnk == 1) - snd_soc_component_update_bits(component, AB8500_ANCCONF1, - BIT(AB8500_ANCCONF1_ANCFIRUPDATE), 0); -} - -/* ANC IIR-coefficients configuration sequence */ -static void anc_iir(struct snd_soc_component *component, unsigned int bnk, - unsigned int par, unsigned int val) -{ - if (par == 0) { - if (bnk == 0) { - snd_soc_component_update_bits(component, AB8500_ANCCONF1, - BIT(AB8500_ANCCONF1_ANCIIRINIT), - BIT(AB8500_ANCCONF1_ANCIIRINIT)); - usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY*2); - snd_soc_component_update_bits(component, AB8500_ANCCONF1, - BIT(AB8500_ANCCONF1_ANCIIRINIT), 0); - usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY*2); - } else { - snd_soc_component_update_bits(component, AB8500_ANCCONF1, - BIT(AB8500_ANCCONF1_ANCIIRUPDATE), - BIT(AB8500_ANCCONF1_ANCIIRUPDATE)); - } - } else if (par > 3) { - snd_soc_component_write(component, AB8500_ANCCONF7, 0); - snd_soc_component_write(component, AB8500_ANCCONF8, val >> 16 & 0xff); - } - - snd_soc_component_write(component, AB8500_ANCCONF7, val >> 8 & 0xff); - snd_soc_component_write(component, AB8500_ANCCONF8, val & 0xff); - - if (par == AB8500_ANC_IIR_COEFFS - 1 && bnk == 1) - snd_soc_component_update_bits(component, AB8500_ANCCONF1, - BIT(AB8500_ANCCONF1_ANCIIRUPDATE), 0); -} - -/* ANC IIR-/FIR-coefficients configuration sequence */ -static void anc_configure(struct snd_soc_component *component, - bool apply_fir, bool apply_iir) -{ - struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(component->dev); - unsigned int bnk, par, val; - - dev_dbg(component->dev, "%s: Enter.\n", __func__); - - if (apply_fir) - snd_soc_component_update_bits(component, AB8500_ANCCONF1, - BIT(AB8500_ANCCONF1_ENANC), 0); - - snd_soc_component_update_bits(component, AB8500_ANCCONF1, - BIT(AB8500_ANCCONF1_ENANC), BIT(AB8500_ANCCONF1_ENANC)); - - if (apply_fir) - for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++) - for (par = 0; par < AB8500_ANC_FIR_COEFFS; par++) { - val = snd_soc_component_read(component, - drvdata->anc_fir_values[par]); - anc_fir(component, bnk, par, val); - } - - if (apply_iir) - for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++) - for (par = 0; par < AB8500_ANC_IIR_COEFFS; par++) { - val = snd_soc_component_read(component, - drvdata->anc_iir_values[par]); - anc_iir(component, bnk, par, val); - } - - dev_dbg(component->dev, "%s: Exit.\n", __func__); -} - /* * Control-events */ @@ -1130,7 +1002,7 @@ static int sid_status_control_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(component->dev); - unsigned int param, sidconf, val; + unsigned int param, sidconf; int status = 1; dev_dbg(component->dev, "%s: Enter\n", __func__); @@ -1159,9 +1031,8 @@ static int sid_status_control_put(struct snd_kcontrol *kcontrol, snd_soc_component_write(component, AB8500_SIDFIRADR, 0); for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) { - val = snd_soc_component_read(component, drvdata->sid_fir_values[param]); - snd_soc_component_write(component, AB8500_SIDFIRCOEF1, val >> 8 & 0xff); - snd_soc_component_write(component, AB8500_SIDFIRCOEF2, val & 0xff); + snd_soc_component_write(component, AB8500_SIDFIRCOEF1, 0); + snd_soc_component_write(component, AB8500_SIDFIRCOEF2, 0); } snd_soc_component_update_bits(component, AB8500_SIDFIRADR, @@ -1180,136 +1051,6 @@ out: return status; } -static int anc_status_control_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(component->dev); - - mutex_lock(&drvdata->ctrl_lock); - ucontrol->value.enumerated.item[0] = drvdata->anc_status; - mutex_unlock(&drvdata->ctrl_lock); - - return 0; -} - -static int anc_status_control_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_context *dapm = snd_soc_component_to_dapm(component); - struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(component->dev); - struct device *dev = component->dev; - bool apply_fir, apply_iir; - unsigned int req; - int status; - - dev_dbg(dev, "%s: Enter.\n", __func__); - - mutex_lock(&drvdata->ctrl_lock); - - req = ucontrol->value.enumerated.item[0]; - if (req >= ARRAY_SIZE(enum_anc_state)) { - status = -EINVAL; - goto cleanup; - } - if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && - req != ANC_APPLY_IIR) { - dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", - __func__, enum_anc_state[req]); - status = -EINVAL; - goto cleanup; - } - apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR; - apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR; - - status = snd_soc_dapm_force_enable_pin(dapm, "ANC Configure Input"); - if (status < 0) { - dev_err(dev, - "%s: ERROR: Failed to enable power (status = %d)!\n", - __func__, status); - goto cleanup; - } - snd_soc_dapm_sync(dapm); - - anc_configure(component, apply_fir, apply_iir); - - if (apply_fir) { - if (drvdata->anc_status == ANC_IIR_CONFIGURED) - drvdata->anc_status = ANC_FIR_IIR_CONFIGURED; - else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED) - drvdata->anc_status = ANC_FIR_CONFIGURED; - } - if (apply_iir) { - if (drvdata->anc_status == ANC_FIR_CONFIGURED) - drvdata->anc_status = ANC_FIR_IIR_CONFIGURED; - else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED) - drvdata->anc_status = ANC_IIR_CONFIGURED; - } - - status = snd_soc_dapm_disable_pin(dapm, "ANC Configure Input"); - snd_soc_dapm_sync(dapm); - -cleanup: - mutex_unlock(&drvdata->ctrl_lock); - - if (status < 0) - dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n", - __func__, status); - - dev_dbg(dev, "%s: Exit.\n", __func__); - - return (status < 0) ? status : 1; -} - -static int filter_control_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct filter_control *fc = - (struct filter_control *)kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = fc->count; - uinfo->value.integer.min = fc->min; - uinfo->value.integer.max = fc->max; - - return 0; -} - -static int filter_control_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct ab8500_codec_drvdata *drvdata = snd_soc_component_get_drvdata(component); - struct filter_control *fc = - (struct filter_control *)kcontrol->private_value; - unsigned int i; - - mutex_lock(&drvdata->ctrl_lock); - for (i = 0; i < fc->count; i++) - ucontrol->value.integer.value[i] = fc->value[i]; - mutex_unlock(&drvdata->ctrl_lock); - - return 0; -} - -static int filter_control_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct ab8500_codec_drvdata *drvdata = snd_soc_component_get_drvdata(component); - struct filter_control *fc = - (struct filter_control *)kcontrol->private_value; - unsigned int i; - - mutex_lock(&drvdata->ctrl_lock); - for (i = 0; i < fc->count; i++) - fc->value[i] = ucontrol->value.integer.value[i]; - mutex_unlock(&drvdata->ctrl_lock); - - return 0; -} - /* * Controls - Non-DAPM ASoC */ @@ -1597,7 +1338,6 @@ static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast, static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_sidstate, enum_sid_state); /* ANC */ -static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_ancstate, enum_anc_state); static struct snd_kcontrol_new ab8500_ctrls[] = { /* Charge pump */ @@ -1873,8 +1613,6 @@ static struct snd_kcontrol_new ab8500_ctrls[] = { AB8500_FIFOCONF6_BFIFOSAMPLE_MAX, 0), /* ANC */ - SOC_ENUM_EXT("ANC Status", soc_enum_ancstate, - anc_status_control_get, anc_status_control_put), SOC_SINGLE_XR_SX("ANC Warp Delay Shift", AB8500_ANCCONF2, 1, AB8500_ANCCONF2_SHIFT, AB8500_ANCCONF2_MIN, AB8500_ANCCONF2_MAX, 0), @@ -1895,21 +1633,6 @@ static struct snd_kcontrol_new ab8500_ctrls[] = { AB8500_SIDFIRADR, AB8500_SIDFIRADR_FIRSIDSET, 0), }; -static struct snd_kcontrol_new ab8500_filter_controls[] = { - AB8500_FILTER_CONTROL("ANC FIR Coefficients", AB8500_ANC_FIR_COEFFS, - AB8500_ANC_FIR_COEFF_MIN, AB8500_ANC_FIR_COEFF_MAX), - AB8500_FILTER_CONTROL("ANC IIR Coefficients", AB8500_ANC_IIR_COEFFS, - AB8500_ANC_IIR_COEFF_MIN, AB8500_ANC_IIR_COEFF_MAX), - AB8500_FILTER_CONTROL("Sidetone FIR Coefficients", - AB8500_SID_FIR_COEFFS, AB8500_SID_FIR_COEFF_MIN, - AB8500_SID_FIR_COEFF_MAX) -}; -enum ab8500_filter { - AB8500_FILTER_ANC_FIR = 0, - AB8500_FILTER_ANC_IIR = 1, - AB8500_FILTER_SID_FIR = 2, -}; - /* * Extended interface for codec-driver */ @@ -2454,7 +2177,6 @@ static int ab8500_codec_probe(struct snd_soc_component *component) struct device_node *np = dev->of_node; struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); struct ab8500_codec_platform_data codec_pdata; - struct filter_control *fc; int status; dev_dbg(dev, "%s: Enter.\n", __func__); @@ -2486,25 +2208,6 @@ static int ab8500_codec_probe(struct snd_soc_component *component) snd_soc_component_write(component, AB8500_SHORTCIRCONF, BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); - /* Add filter controls */ - status = snd_soc_add_component_controls(component, ab8500_filter_controls, - ARRAY_SIZE(ab8500_filter_controls)); - if (status < 0) { - dev_err(dev, - "%s: failed to add ab8500 filter controls (%d).\n", - __func__, status); - return status; - } - fc = (struct filter_control *) - ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value; - drvdata->anc_fir_values = (long *)fc->value; - fc = (struct filter_control *) - ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value; - drvdata->anc_iir_values = (long *)fc->value; - fc = (struct filter_control *) - ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value; - drvdata->sid_fir_values = (long *)fc->value; - snd_soc_dapm_disable_pin(dapm, "ANC Configure Input"); mutex_init(&drvdata->ctrl_lock); @@ -2538,7 +2241,6 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) if (!drvdata) return -ENOMEM; drvdata->sid_status = SID_UNCONFIGURED; - drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, From ec0611868f2fcf29e4c2bebdc6702d3e1f272fec Mon Sep 17 00:00:00 2001 From: Troy Mitchell Date: Wed, 29 Apr 2026 17:00:50 +0800 Subject: [PATCH 06/28] ASoC: spacemit: fix RX DMA params not set when TX is running When TX is already running (SSCR_SSE is set), the hw_params callback returns early before setting up DMA parameters for the RX stream. This prevents the capture path from configuring its DMA data properly. Move the SSCR_SSE check after DMA parameter setup and format constraints, so both TX and RX streams get their DMA configuration regardless of whether the hardware is already enabled. The early return now only skips the register writes that would disrupt an active stream. Fixes: fce217449075 ("ASoC: spacemit: add i2s support for K1 SoC") Signed-off-by: Troy Mitchell Link: https://patch.msgid.link/20260429-k1-i2s-fix-v2-1-8d67835aaddc@linux.spacemit.com Signed-off-by: Mark Brown --- sound/soc/spacemit/k1_i2s.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/spacemit/k1_i2s.c b/sound/soc/spacemit/k1_i2s.c index 43481f387c44..5420ca2aefbd 100644 --- a/sound/soc/spacemit/k1_i2s.c +++ b/sound/soc/spacemit/k1_i2s.c @@ -148,10 +148,6 @@ static int spacemit_i2s_hw_params(struct snd_pcm_substream *substream, u32 val; int ret; - val = readl(i2s->base + SSCR); - if (val & SSCR_SSE) - return 0; - dma_data = &i2s->playback_dma_data; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -199,6 +195,9 @@ static int spacemit_i2s_hw_params(struct snd_pcm_substream *substream, } val = readl(i2s->base + SSCR); + if (val & SSCR_SSE) + return 0; + val &= ~SSCR_DW_32BYTE; val |= data_width; writel(val, i2s->base + SSCR); From c64e77490b7e5d9dec738850f18878edb07e0f13 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 29 Apr 2026 11:53:15 +0100 Subject: [PATCH 07/28] ASoC: cs35l56: Fix hibernate write in runtime resume error path The error path of cs35l56_runtime_resume_common() should only write the hibernation sequence if can_hibernate is true. Something has already gone badly wrong if we ever reach the error path. But triggering hibernate on hardware that does not support it is likely to make the situation unrecoverable without a full reboot because there might not be any hardware signal to exit hibernate. Fixes: a47cf4dac7dc ("ASoC: cs35l56: Change hibernate sequence to use allow auto hibernate") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260429105315.2438298-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-shared.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 033e56d5e9db..ea724101cdd1 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -851,9 +851,11 @@ out_sync: err: regcache_cache_only(cs35l56_base->regmap, true); - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_seq, - ARRAY_SIZE(cs35l56_hibernate_seq)); + if (cs35l56_base->can_hibernate) { + regmap_multi_reg_write_bypassed(cs35l56_base->regmap, + cs35l56_hibernate_seq, + ARRAY_SIZE(cs35l56_hibernate_seq)); + } return ret; } From 0e60d96616640ffcf51b81a87c71e30d92385a93 Mon Sep 17 00:00:00 2001 From: Bob Song Date: Thu, 30 Apr 2026 09:49:20 +0800 Subject: [PATCH 08/28] ASoC: amd: yc: Add DMI quirk for MSI Bravo 15 C7VE The laptop requires a quirk ID to enable its internal microphone. Add it to the DMI quirk table. Reported-by: gannovera Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218402 Signed-off-by: Bob Song Link: https://patch.msgid.link/20260430014920.141276-1-songxiebing@kylinos.cn 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 c5cf45881416..dde1a144e4d4 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -479,6 +479,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 B7ED"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."), + DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 C7VE"), + } + }, { .driver_data = &acp6x_card, .matches = { From 0f9bfb84b3f0fe1406b2555fc11b45283ea21644 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 30 Apr 2026 11:11:34 +0100 Subject: [PATCH 09/28] ASoC: cs35l56: Fix out-of-bounds in dev_err() in cs35l56_read_onchip_spkid() Remove the incorrect use of onchip_spkid_gpios[i] in the dev_err() after regmap_read() of CS35L56_GPIO_STATUS1 returns an error. This dev_err() was incorrectly copy-pasted from one inside the for-loop, where i was valid. The read of CS35L56_GPIO_STATUS1 isn't for a specific GPIO register, so the use of onchip_spkid_gpios[i] in the error message is both irrelevant and out-of-bounds here. Fixes: 4d1e3e2c404d ("ASoC: cs35l56: Support for reading speaker ID from on-chip GPIOs") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260430101134.2655938-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-shared.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index ea724101cdd1..795e2764d67e 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -1730,8 +1730,7 @@ int cs35l56_read_onchip_spkid(struct cs35l56_base *cs35l56_base) ret = regmap_read(regmap, CS35L56_GPIO_STATUS1, &val); if (ret) { - dev_err(cs35l56_base->dev, "GPIO%d status read failed: %d\n", - cs35l56_base->onchip_spkid_gpios[i] + 1, ret); + dev_err(cs35l56_base->dev, "GPIO status read failed: %d\n", ret); return ret; } From d63c219b7ff39f897da10c160a2edef76320f16c Mon Sep 17 00:00:00 2001 From: Tommaso Soncin Date: Wed, 29 Apr 2026 18:08:57 +0200 Subject: [PATCH 10/28] ASoC: amd: yc: Add HP OMEN Gaming Laptop 16-ap0xxx product line in quirk table Add a DMI quirk for the HP OMEN Gaming Laptop 16-ap0xxx line fixing the issue where the internal microphone was not detected. Cc: stable@vger.kernel.org Signed-off-by: Tommaso Soncin Link: https://patch.msgid.link/20260429160858.538986-1-soncintommaso@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index dde1a144e4d4..7a637d6b5576 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -59,6 +59,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "HP Laptop 15-fc0xxx"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "OMEN Gaming Laptop 16-ap0xxx"), + } + }, { .driver_data = &acp6x_card, .matches = { @@ -675,6 +682,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8EE4"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8E35"), + } + }, { .driver_data = &acp6x_card, .matches = { From 56d5a9eaf60af5c824a33a83e1468aa143627a62 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Thu, 30 Apr 2026 20:10:43 +0800 Subject: [PATCH 11/28] ASoC: sdw_utils: avoid the SDCA companion function not supported failure Treat the companion amp as generic AMP until full support for companion amp is added. Signed-off-by: Derek Fang Reviewed-by: Charles Keepax Signed-off-by: Bard Liao Link: https://patch.msgid.link/20260430121043.552241-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sdw_utils/soc_sdw_utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 1637cc3f3d59..849ae876d7a4 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1608,6 +1608,7 @@ int asoc_sdw_get_dai_type(u32 type) switch (type) { case SDCA_FUNCTION_TYPE_SMART_AMP: case SDCA_FUNCTION_TYPE_SIMPLE_AMP: + case SDCA_FUNCTION_TYPE_COMPANION_AMP: return SOC_SDW_DAI_TYPE_AMP; case SDCA_FUNCTION_TYPE_SMART_MIC: case SDCA_FUNCTION_TYPE_SIMPLE_MIC: From e8446a4a574d19f0fb39c06af15dbc5165079474 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Tue, 28 Apr 2026 00:07:08 -0300 Subject: [PATCH 12/28] ASoC: fsl_xcvr: Fix event generation for cached controls MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ALSA controls should return 1 from a put callback when the control value changes. fsl_xcvr_capds_put() and fsl_xcvr_tx_cs_put() both update cached control data but always return 0, so ALSA suppresses change notifications for the Capabilities Data Structure and playback IEC958 channel status controls. Compare the old and new cached values before copying the new data, and return whether the control value changed. Fixes: 28564486866f ("ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver") Signed-off-by: Cássio Gabriel Link: https://patch.msgid.link/20260428-asoc-fsl-xcvr-event-generation-v1-1-f21cf0812c4f@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_xcvr.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index ee16cf681488..6677d3bf36ec 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -228,10 +228,14 @@ static int fsl_xcvr_capds_put(struct snd_kcontrol *kcontrol, { struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); struct fsl_xcvr *xcvr = snd_soc_dai_get_drvdata(dai); + int changed; - memcpy(xcvr->cap_ds, ucontrol->value.bytes.data, FSL_XCVR_CAPDS_SIZE); + changed = memcmp(xcvr->cap_ds, ucontrol->value.bytes.data, + sizeof(xcvr->cap_ds)) != 0; + memcpy(xcvr->cap_ds, ucontrol->value.bytes.data, + sizeof(xcvr->cap_ds)); - return 0; + return changed; } static struct snd_kcontrol_new fsl_xcvr_earc_capds_kctl = { @@ -1040,10 +1044,15 @@ static int fsl_xcvr_tx_cs_put(struct snd_kcontrol *kcontrol, { struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); struct fsl_xcvr *xcvr = snd_soc_dai_get_drvdata(dai); + int changed; - memcpy(xcvr->tx_iec958.status, ucontrol->value.iec958.status, 24); + changed = memcmp(xcvr->tx_iec958.status, + ucontrol->value.iec958.status, + sizeof(xcvr->tx_iec958.status)) != 0; + memcpy(xcvr->tx_iec958.status, ucontrol->value.iec958.status, + sizeof(xcvr->tx_iec958.status)); - return 0; + return changed; } static struct snd_kcontrol_new fsl_xcvr_rx_ctls[] = { From f80785888f7c0980a49545b87a80e3817c9ed7c6 Mon Sep 17 00:00:00 2001 From: Anton Swart Date: Sun, 3 May 2026 23:15:17 +0200 Subject: [PATCH 13/28] ALSA: usb-audio: Add quirk flags for AlphaTheta EUPHONIA The AlphaTheta EUPHONIA (VID 0x2b73, PID 0x0047) is a USB Audio Class 2 DJ mixer that requires implicit feedback for full-duplex operation. The capture endpoint (0x83 IN, interface 2) acts as the implicit feedback source for the playback endpoint (0x03 OUT, interface 1), and the device firmware does not send isochronous data on the capture endpoint unless the host is simultaneously sending data on the playback endpoint, i.e. playback must be started first. Without QUIRK_FLAG_PLAYBACK_FIRST the kernel waits for capture URBs before submitting playback URBs, creating a deadlock: the device waits for playback data and the host waits for capture data. Without QUIRK_FLAG_GENERIC_IMPLICIT_FB the kernel does not detect the implicit feedback relationship between the two interfaces. The same flag combination is already used for the Behringer UMC202HD, UMC204HD and UMC404HD (0x1397:0x0507/0x0508/0x0509), which exhibit the identical implicit-feedback topology. Tested on Raspberry Pi 5 with kernel 6.12.75; continuous full-duplex streaming at 96 kHz / 24-bit, zero XRUNs. Signed-off-by: Anton Swart Link: https://patch.msgid.link/20260503211517.14332-1-anton.swart.jhb@gmail.com 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 f77ff05dff0b..b17bdae09bc3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2460,6 +2460,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x2b73, 0x0047, /* AlphaTheta EUPHONIA */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x2d95, 0x8011, /* VIVO USB-C HEADSET */ QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x2d95, 0x8021, /* VIVO USB-C-XE710 HEADSET */ From 0749daa8eb5ab90334aaad3b0671efd7150d43b1 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Sun, 3 May 2026 21:55:52 -0300 Subject: [PATCH 14/28] ALSA: firewire-tascam: Do not drop unread control events MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit tscm_hwdep_read_queue() copies as many queued control events as fit in the userspace buffer. When the buffer is smaller than the current contiguous queue segment, length is rounded down to the number of bytes that can be copied. However, after copying that shortened length, the code advances pull_pos to the original tail_pos, marking the whole contiguous segment as consumed. Any events between the copied portion and tail_pos are lost. Limit tail_pos to the position after the entries actually copied before updating pull_pos. When the whole segment fits, this is equivalent to the old tail_pos update; when the buffer is smaller, the remaining events stay queued for the next read. Fixes: a8c0d13267a4 ("ALSA: firewire-tascam: notify events of change of state for userspace applications") Cc: stable@vger.kernel.org Suggested-by: Takashi Sakamoto Signed-off-by: Cássio Gabriel Reviewed-by: Takashi Sakamoto Co-developed-by: Takashi Sakamoto Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260503-alsa-firewire-tascam-read-queue-v2-1-126c6efd7642@gmail.com --- sound/firewire/tascam/tascam-hwdep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index 867b4ea1096e..6270263e7bf4 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -73,6 +73,7 @@ static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf, length = rounddown(remained, sizeof(*entries)); if (length == 0) break; + tail_pos = head_pos + length / sizeof(*entries); spin_unlock_irq(&tscm->lock); if (copy_to_user(pos, &entries[head_pos], length)) From db4cd7d6c571c5ec6b52a6e80ec59ad99c4aa1d6 Mon Sep 17 00:00:00 2001 From: Rong Zhang Date: Mon, 4 May 2026 19:38:05 +0800 Subject: [PATCH 15/28] ALSA: usb-audio: Add quirk flags for JBL Pebbles JBL Pebbles is a pair of desktop speakers with UAC interface. Its Playback and Capture mixers use linear volume with val = 0/999/1 and 0/3996/4. Meanwhile, the reported sample rates are truncated to multiples of 0x100 (i.e., 44100 => 44032), resulting in noisy kmsg, as a warning message is printed each time a stream is opened. Add a quirk table entry matching VID/PID=0x05fc/0x0231 and applying linear volume and sample rate quirk flags, so that it can work properly. Also note that the volume control knob on device is an incremental encoder. It does nothing but sends KEY_VOLUMEUP and KEY_VOLUMEDOWN per rotation, controlling the UAC Playback volume mixer indirectly. Hence, the linear volume quirk flags also enable the volume control knob to function properly. Quirky device sample: usb 5-1.1: new full-speed USB device number 12 using xhci_hcd usb 5-1.1: New USB device found, idVendor=05fc, idProduct=0231, bcdDevice= 1.00 usb 5-1.1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 5-1.1: Product: JBL Pebbles usb 5-1.1: Manufacturer: Harman International Industries usb 5-1.1: SerialNumber: 1.0.0 usb-storage 5-1.1:1.0: USB Mass Storage device detected scsi host0: usb-storage 5-1.1:1.0 usb 5-1.1: Found last interface = 1 usb 5-1.1: 2:1: add audio endpoint 0x5 usb 5-1.1: Creating new data endpoint #5 usb 5-1.1: 2:1 Set sample rate 44100, clock 0 usb 5-1.1: current rate 44032 is different from the runtime rate 44100 usb 5-1.1: 3:1: add audio endpoint 0x84 usb 5-1.1: Creating new data endpoint #84 usb 5-1.1: 3:1 Set sample rate 44100, clock 0 usb 5-1.1: current rate 44032 is different from the runtime rate 44100 usb 5-1.1: [2] FU [PCM Playback Switch] ch = 1, val = 0/1/1 usb 5-1.1: Warning! Unlikely big volume step count (=999), linear volume or wrong cval->res? usb 5-1.1: [2] FU [PCM Playback Volume] ch = 2, val = 0/999/1 usb 5-1.1: [5] FU [Mic Capture Switch] ch = 1, val = 0/1/1 usb 5-1.1: Warning! Unlikely big volume step count (=999), linear volume or wrong cval->res? usb 5-1.1: [5] FU [Mic Capture Volume] ch = 2, val = 0/3996/4 input: Harman International Industries JBL Pebbles as /devices/pci0000:00/0000:00:08.3/0000:67:00.3/usb5/5-1/5-1.1/5-1.1:1.4/0003:05FC:0231.0018/input/input55 hid-generic 0003:05FC:0231.0018: input,hidraw2: USB HID v2.01 Device [Harman International Industries JBL Pebbles] on usb-0000:67:00.3-1.1/input4 Signed-off-by: Rong Zhang Link: https://patch.msgid.link/20260504-uac-jbl-pebbles-v1-1-c888d592a286@rong.moe Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b17bdae09bc3..17983d9774f8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2277,6 +2277,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x05e1, 0x0480, /* Hauppauge Woodbury */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x05fc, 0x0231, /* JBL Pebbles */ + QUIRK_FLAG_MIXER_PLAYBACK_LINEAR_VOL | QUIRK_FLAG_MIXER_CAPTURE_LINEAR_VOL | + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x0624, 0x3d3f, /* AB13X USB Audio */ QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x0644, 0x8043, /* TEAC UD-501/UD-501V2/UD-503/NT-503 */ From dc1e0172be54e742bccb28d5f14c0c395e28c098 Mon Sep 17 00:00:00 2001 From: Fernando Antunez Antonio Date: Mon, 4 May 2026 07:33:26 -0600 Subject: [PATCH 16/28] ALSA: hda/realtek: Fix mute and mic-mute LEDs for HP Envy X360 15-fh0xxx This enables the mute and mic-mute LEDs on the HP Envy X360 15-fh0xxx 2-in-1 laptops. The quirk 'ALC245_FIXUP_HP_ENVY_X360_15_FH0XXX' has been created and is now enabled for this device. This is my first patch, and I'm still getting to grips with the code, so there's probably a better way to implement this fix. I apologize for any inconvenience caused by the constant release of new versions of this patch. Signed-off-by: Fernando Antunez Antonio Link: https://patch.msgid.link/20260504-hpenvy-muteled-fix-v3-1-5567fd9b3d25@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index e061673d3c2e..a7525ed83e2b 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -4137,6 +4137,7 @@ enum { ALC245_FIXUP_CS35L41_I2C_2_MUTE_LED, ALC236_FIXUP_HP_DMIC, ALC256_FIXUP_HONOR_MRB_XXX_M1020_AUDIO, + ALC245_FIXUP_HP_ENVY_X360_15_FH0XXX, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -6693,6 +6694,12 @@ static const struct hda_fixup alc269_fixups[] = { { 0x1b, 0x90170110 }, { } } + }, + [ALC245_FIXUP_HP_ENVY_X360_15_FH0XXX] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_i2c_two, + .chained = true, + .chain_id = ALC245_FIXUP_HP_X360_MUTE_LEDS } }; @@ -7115,7 +7122,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8be6, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8be7, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8be8, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy x360 2-in-1 Laptop 15-fh0xxx", ALC245_FIXUP_HP_ENVY_X360_15_FH0XXX), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre x360 2-in-1 Laptop 14-eu0xxx", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX), From f3c57c9c2a49a21d784b7c04a2c883bffc070659 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Mon, 4 May 2026 11:08:45 -0300 Subject: [PATCH 17/28] ALSA: usb-audio: midi2: Restart output URBs on resume MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit USB MIDI 2.0 suspend saves the endpoint running state, clears it and kills all endpoint URBs. Resume restores the running state, but only restarts input endpoints. For a running output endpoint, this leaves the endpoint marked running with an empty URB queue. Output transfer progress depends on either the rawmidi trigger path starting the queue or an output completion refilling it. After suspend there is no completion left, and output data that remains queued in the raw UMP or legacy rawmidi buffer can stay stalled until userspace happens to trigger the stream again. Restore the saved state with atomic accessors, keep input endpoints restarted as before, and restart output endpoints that were running before suspend. Clear the saved suspend state after restoring it. Fixes: ff49d1df79ae ("ALSA: usb-audio: USB MIDI 2.0 UMP support") Cc: stable@vger.kernel.org Signed-off-by: Cássio Gabriel Link: https://patch.msgid.link/20260504-usb-midi2-output-resume-v1-1-c089cc8ad3c6@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/midi2.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/midi2.c b/sound/usb/midi2.c index 3546ba926cb3..2785600d2312 100644 --- a/sound/usb/midi2.c +++ b/sound/usb/midi2.c @@ -227,7 +227,7 @@ static void kill_midi_urbs(struct snd_usb_midi2_endpoint *ep, bool suspending) if (!ep) return; if (suspending) - ep->suspended = ep->running; + atomic_set(&ep->suspended, atomic_read(&ep->running)); atomic_set(&ep->running, 0); for (i = 0; i < ep->num_urbs; i++) { if (!ep->urbs[i].urb) @@ -1188,10 +1188,11 @@ void snd_usb_midi_v2_suspend_all(struct snd_usb_audio *chip) static void resume_midi2_endpoint(struct snd_usb_midi2_endpoint *ep) { - ep->running = ep->suspended; - if (ep->direction == STR_IN) + atomic_set(&ep->running, atomic_read(&ep->suspended)); + atomic_set(&ep->suspended, 0); + + if (ep->direction == STR_IN || atomic_read(&ep->running)) submit_io_urbs(ep); - /* FIXME: does it all? */ } void snd_usb_midi_v2_resume_all(struct snd_usb_audio *chip) From 320e55722ca466a7d40dd69e1aea982cb6189006 Mon Sep 17 00:00:00 2001 From: Nicola Lunghi Date: Mon, 4 May 2026 16:45:20 +0200 Subject: [PATCH 18/28] ALSA: usb-audio: add clock quirk for Motu 1248 The Motu 1248 (and probably other older Motu AVB interfaces) take more than 2 seconds to switch clock. During the clock switching process the device return that the clock is not valid. This is similar to what already implemented for the Microbook II interface. Add the Motu 1248 usb id to the existing Motu quirk. Signed-off-by: Nicola Lunghi Link: https://patch.msgid.link/20260504144520.699522-2-nick83ola@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 842ba5b801ea..2e0c18e35281 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -208,11 +208,18 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, } /* - * MOTU MicroBook IIc - * Sample rate changes takes more than 2 seconds for this device. Clock - * validity request returns false during that period. + * Quirk for older MOTU AVB / hybrid interfaces + * + * These devices take more than 2 seconds to switch sample rate or + * clock source. During this period the clock validity request + * returns false, causing ALSA to fail prematurely. + * + * Affected models (all use vendor 0x07fd): + * - MicroBook IIc → 0x0004 + * - 1248, 624, 8A, UltraLite AVB, 8M, 16A, ... → 0x0005 */ - if (chip->usb_id == USB_ID(0x07fd, 0x0004)) { + if (chip->usb_id == USB_ID(0x07fd, 0x0004) || /* MicroBook IIc */ + chip->usb_id == USB_ID(0x07fd, 0x0005)) { /* 1248 / 624 / 8A / UltraLite AVB / ... */ count = 0; while ((!ret) && (count < 50)) { From 915f9860fe1c9f7eb6c48c299b2db64fd57ef32f Mon Sep 17 00:00:00 2001 From: James Calligeros Date: Sun, 3 May 2026 22:23:23 +1000 Subject: [PATCH 19/28] ASoC: tas2764: Deal with bogus initial temperature register value The TAS2764 datasheet specifies that the chip initialises the temperature register such that the temperature reading is 2.6 *C, ostensibly to prevent tripping the chip's protection circuitry. The chip is not capable of representing 2.6 *C however, and the register is actually initialised to 0. The ADC does not start sampling until the chip is powered up, and the last sampled temperature persists in the register during software shutdown. Therefore, any reading returning 0 is almost certain to be from before the ADC has actually started sampling, meaning that it is invalid. Return -ENODATA early if the temperature has not yet been sampled by the chip, and indicate a fault condition using HWMON_T_FAULT. Fixes: 186dfc85f9a8 ("ASoC: tas2764: expose die temp to hwmon") Signed-off-by: James Calligeros Signed-off-by: Mark Brown --- sound/soc/codecs/tas2764.c | 35 +++++++++++++++++++++++++++-------- 1 file changed, 27 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 6aab6d2b7419..55211266927d 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -684,18 +684,33 @@ static int tas2764_read_die_temp(struct tas2764_priv *tas2764, long *result) * As per datasheet, subtract 93 from raw value to get degrees * Celsius. hwmon wants millidegrees. * - * NOTE: The chip will initialise the TAS2764_TEMP register to - * 2.6 *C to avoid triggering temperature protection. Since the - * ADC is powered down during software shutdown, this value will - * persist until the chip is fully powered up (e.g. the PCM it's - * attached to is opened). The ADC will power down again when - * the chip is put back into software shutdown, with the last - * value sampled persisting in the ADC's register. + * NOTE: The TAS2764 datasheet mentions initialising TAS2764_TEMP + * such that the temperature is 2.6 *C, however the register + * is actually initialised to 0. The ADC is also powered down during + * software shutdown. The last sampled temperature will persist + * in the register while the amp is in this power state. */ + if (reg == 0) + return -ENODATA; + *result = (reg - 93) * 1000; return 0; } +static int tas2764_hwmon_is_fault(struct tas2764_priv *tas2764, long *result) +{ + int ret; + long temp; + + ret = tas2764_read_die_temp(tas2764, &temp); + if (ret == -ENODATA) { + *result = true; + return 0; + } + + return ret; +} + static umode_t tas2764_hwmon_is_visible(const void *data, enum hwmon_sensor_types type, u32 attr, int channel) @@ -705,6 +720,7 @@ static umode_t tas2764_hwmon_is_visible(const void *data, switch (attr) { case hwmon_temp_input: + case hwmon_temp_fault: return 0444; default: break; @@ -724,6 +740,9 @@ static int tas2764_hwmon_read(struct device *dev, case hwmon_temp_input: ret = tas2764_read_die_temp(tas2764, val); break; + case hwmon_temp_fault: + ret = tas2764_hwmon_is_fault(tas2764, val); + break; default: ret = -EOPNOTSUPP; break; @@ -733,7 +752,7 @@ static int tas2764_hwmon_read(struct device *dev, } static const struct hwmon_channel_info *const tas2764_hwmon_info[] = { - HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT), + HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT | HWMON_T_FAULT), NULL }; From d0771f4995d3285756bf496cf6e346df99481f83 Mon Sep 17 00:00:00 2001 From: James Calligeros Date: Sun, 3 May 2026 22:23:24 +1000 Subject: [PATCH 20/28] ASoC: tas2770: Deal with bogus initial temperature value TAS2770 initialises the temperature readout registers to 0. This value persists until the chip is fully powered up and the ADC starts sampling. The ADC then persists the last sampled temperature during software shutdown. The ADC should therefore never return 0 in normal operating conditions, so return -ENODATA and mark it as a fault condition using HWMON_T_FAULT. Fixes: ff73e2780169 ("ASoC: tas2770: expose die temp to hwmon") Signed-off-by: James Calligeros Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 50501bcbe916..dbda9f327535 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -633,10 +633,27 @@ static int tas2770_read_die_temp(struct tas2770_priv *tas2770, long *result) * value read back from its registers will be the last value sampled * before entering software shutdown. */ + if (reading == 0) + return -ENODATA; + *result = (reading - (93 * 16)) * 1000 / 16; return 0; } +static int tas2770_hwmon_is_fault(struct tas2770_priv *tas2770, long *result) +{ + int ret; + long temp; + + ret = tas2770_read_die_temp(tas2770, &temp); + if (ret == -ENODATA) { + *result = true; + return 0; + } + + return ret; +} + static umode_t tas2770_hwmon_is_visible(const void *data, enum hwmon_sensor_types type, u32 attr, int channel) @@ -646,6 +663,7 @@ static umode_t tas2770_hwmon_is_visible(const void *data, switch (attr) { case hwmon_temp_input: + case hwmon_temp_fault: return 0444; default: break; @@ -665,6 +683,9 @@ static int tas2770_hwmon_read(struct device *dev, case hwmon_temp_input: ret = tas2770_read_die_temp(tas2770, val); break; + case hwmon_temp_fault: + ret = tas2770_hwmon_is_fault(tas2770, val); + break; default: ret = -EOPNOTSUPP; break; @@ -674,7 +695,7 @@ static int tas2770_hwmon_read(struct device *dev, } static const struct hwmon_channel_info *const tas2770_hwmon_info[] = { - HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT), + HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT | HWMON_T_FAULT), NULL }; From 78538047717bdeabe8481ef611c9131e455e61df Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 5 May 2026 11:51:22 +0100 Subject: [PATCH 21/28] ASoC: wm_adsp_fw_find_test: Redirect wm_adsp_release_firmware_files() Redirect wm_adsp_release_firmware_files() to a replacement function that handles the dummy firmware created by the tests. Use the same cleanup function to cleanup in the test exit() function. Also call it on each loop in wm_adsp_fw_find_test_find_firmware_byindex() to free the created strings before reusing priv->found_fw on the next loop. wm_adsp_release_firmware_files() will pass the struct firmware* pointers to release_firmware(). But the pointers created by the tests are dummies and must not be passed to release_firmware(). The test never invokes wm_adsp_release_firmware_files() so it wasn't redirected. But the error handling in wm_adsp_request_firmware_files() calls wm_adsp_release_firmware_files(). The redirected function makes this safe. Using the same cleanup function to perform cleanup from the test exit() handler and wm_adsp_fw_find_test_find_firmware_byindex() avoids the risk of duplicate cleanup code that all needs updating if there is any change to the cleanup requirements. This problem was found by https://sashiko.dev. Fixes: bf2d44d07de7 ("ASoC: wm_adsp: Add kunit test for firmware file search") Closes: https://sashiko.dev/#/patchset/20260326100853.1582886-1-rf%40opensource.cirrus.com Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260505105123.3539778-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp_fw_find_test.c | 56 ++++++++++++++++++++----- 1 file changed, 46 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp_fw_find_test.c b/sound/soc/codecs/wm_adsp_fw_find_test.c index d0c7fb30a95d..516ba08eb5ba 100644 --- a/sound/soc/codecs/wm_adsp_fw_find_test.c +++ b/sound/soc/codecs/wm_adsp_fw_find_test.c @@ -45,6 +45,34 @@ struct wm_adsp_fw_find_test_params { /* Dummy struct firmware to return from wm_adsp_request_firmware_files */ static const struct firmware wm_adsp_find_test_dummy_firmware; +static void wm_adsp_fw_find_test_release_firmware_files_stub(struct wm_adsp_fw_files *fw) +{ + /* + * fw->wmfw.firmware and fw->coeff.firmware allocated by this KUnit + * test are dummies not allocated by the real request_firmware() call + * so they must not be passed to release_firmware(). + * This function replaces wm_adsp_release_firmware_files(). + */ + + if (!fw) + return; + + kfree(fw->wmfw.filename); + kfree(fw->coeff.filename); + + fw->wmfw.firmware = NULL; + fw->coeff.firmware = NULL; + fw->wmfw.filename = NULL; + fw->coeff.filename = NULL; +} + +static void wm_adsp_free_found_fw(struct kunit *test) +{ + struct wm_adsp_fw_find_test *priv = test->priv; + + wm_adsp_fw_find_test_release_firmware_files_stub(&priv->found_fw); +} + /* Simple lookup of a filename in a list of names */ static int wm_adsp_fw_find_test_firmware_request_simple_stub(const struct firmware **firmware, const char *filename, @@ -97,9 +125,14 @@ static void wm_adsp_fw_find_test_pick_file(struct kunit *test) kunit_activate_static_stub(test, wm_adsp_firmware_request, wm_adsp_fw_find_test_firmware_request_simple_stub); + kunit_activate_static_stub(test, + wm_adsp_release_firmware_files, + wm_adsp_fw_find_test_release_firmware_files_stub); ret = wm_adsp_request_firmware_files(dsp, &priv->found_fw); kunit_deactivate_static_stub(test, wm_adsp_firmware_request); + kunit_deactivate_static_stub(test, wm_adsp_release_firmware_files); + KUNIT_EXPECT_EQ_MSG(test, ret, (params->expect_wmfw || params->expect_bin) ? 0 : -ENOENT, "%s\n", priv->searched_fw_files); @@ -173,10 +206,13 @@ static void wm_adsp_fw_find_test_search_order(struct kunit *test) kunit_activate_static_stub(test, wm_adsp_firmware_request, wm_adsp_fw_find_test_firmware_request_stub); + kunit_activate_static_stub(test, + wm_adsp_release_firmware_files, + wm_adsp_fw_find_test_release_firmware_files_stub); wm_adsp_request_firmware_files(dsp, &priv->found_fw); - kunit_deactivate_static_stub(test, wm_adsp_firmware_request); + kunit_deactivate_static_stub(test, wm_adsp_release_firmware_files); KUNIT_EXPECT_STREQ(test, priv->searched_fw_files, params->expected_searches); @@ -201,6 +237,7 @@ static void wm_adsp_fw_find_test_find_firmware_byindex(struct kunit *test) dsp->cs_dsp.name = "cs1234"; dsp->part = "dsp1"; + for (dsp->fw = 0;; dsp->fw++) { fw_name = wm_adsp_get_fwf_name_by_index(dsp->fw); if (!fw_name) @@ -209,14 +246,21 @@ static void wm_adsp_fw_find_test_find_firmware_byindex(struct kunit *test) kunit_activate_static_stub(test, wm_adsp_firmware_request, wm_adsp_fw_find_test_firmware_request_stub); + kunit_activate_static_stub(test, + wm_adsp_release_firmware_files, + wm_adsp_fw_find_test_release_firmware_files_stub); wm_adsp_request_firmware_files(dsp, &priv->found_fw); + kunit_deactivate_static_stub(test, wm_adsp_firmware_request); + kunit_deactivate_static_stub(test, wm_adsp_release_firmware_files); KUNIT_EXPECT_NOT_NULL_MSG(test, strstr(priv->searched_fw_files, fw_name), "fw#%d Did not find '%s' in '%s'\n", dsp->fw, fw_name, priv->searched_fw_files); + + wm_adsp_free_found_fw(test); } } @@ -255,15 +299,7 @@ static int wm_adsp_fw_find_test_case_init(struct kunit *test) static void wm_adsp_fw_find_test_case_exit(struct kunit *test) { - struct wm_adsp_fw_find_test *priv = test->priv; - - /* - * priv->found_wmfw_firmware and priv->found_bin_firmware are - * dummies not allocated by the real request_firmware() call they - * must not be passed to release_firmware(). - */ - kfree(priv->found_fw.wmfw.filename); - kfree(priv->found_fw.coeff.filename); + wm_adsp_free_found_fw(test); } static void wm_adsp_fw_find_test_param_desc(const struct wm_adsp_fw_find_test_params *param, From af64f790969973b325efda7264d6860167623cdd Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 5 May 2026 11:51:23 +0100 Subject: [PATCH 22/28] ASoC: wm_adsp_fw_find_test: Clear searched_fw_files in find-by-index test In wm_adsp_fw_find_test_find_firmware_byindex() the content of priv->searched_fw_files must be cleared before starting the next iteration. The files searched for are appended to priv->searched_fw_files, so if it is not cleared on each iteration it will still contain the searches from the previous iteration. Fixes: bf2d44d07de7 ("ASoC: wm_adsp: Add kunit test for firmware file search") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260505105123.3539778-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp_fw_find_test.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm_adsp_fw_find_test.c b/sound/soc/codecs/wm_adsp_fw_find_test.c index 516ba08eb5ba..ae686dc4fa94 100644 --- a/sound/soc/codecs/wm_adsp_fw_find_test.c +++ b/sound/soc/codecs/wm_adsp_fw_find_test.c @@ -261,6 +261,7 @@ static void wm_adsp_fw_find_test_find_firmware_byindex(struct kunit *test) dsp->fw, fw_name, priv->searched_fw_files); wm_adsp_free_found_fw(test); + memset(priv->searched_fw_files, 0, sizeof(priv->searched_fw_files)); } } From 92429ca999db99febced82f23362a71b2ba4c1d8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Wed, 6 May 2026 00:15:48 -0300 Subject: [PATCH 23/28] ALSA: seq: Fix UMP group 16 filtering MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The sequencer UAPI defines group_filter as an unsigned int bitmap. Bit 0 filters groupless messages and bits 1-16 filter UMP groups 1-16. The internal snd_seq_client storage is only unsigned short, so bit 16 is truncated when userspace sets the filter. The same truncation affects the automatic UMP client filter used to avoid delivery to inactive groups, so events for group 16 cannot be filtered. Store the internal bitmap as unsigned int and keep both userspace-provided and automatically generated values limited to the defined UAPI bits. Fixes: d2b706077792 ("ALSA: seq: Add UMP group filter") Cc: stable@vger.kernel.org Signed-off-by: Cássio Gabriel Link: https://patch.msgid.link/20260506-alsa-seq-ump-group16-filter-v1-1-b75160bf6993@gmail.com Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/seq/seq_clientmgr.h | 5 ++++- sound/core/seq/seq_ump_client.c | 2 +- 3 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 75a7a2af9d8c..5719637575a9 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1253,7 +1253,7 @@ static int snd_seq_ioctl_set_client_info(struct snd_seq_client *client, if (client->user_pversion >= SNDRV_PROTOCOL_VERSION(1, 0, 3)) client->midi_version = client_info->midi_version; memcpy(client->event_filter, client_info->event_filter, 32); - client->group_filter = client_info->group_filter; + client->group_filter = client_info->group_filter & SND_SEQ_GROUP_FILTER_MASK; /* notify the change */ snd_seq_system_client_ev_client_change(client->number); diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index ece02c58db70..feea8bb7d987 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -14,6 +14,9 @@ /* client manager */ +#define SND_SEQ_GROUP_FILTER_MASK GENMASK(SNDRV_UMP_MAX_GROUPS, 0) +#define SND_SEQ_GROUP_FILTER_GROUPS GENMASK(SNDRV_UMP_MAX_GROUPS, 1) + struct snd_seq_user_client { struct file *file; /* file struct of client */ /* ... */ @@ -40,7 +43,7 @@ struct snd_seq_client { int number; /* client number */ unsigned int filter; /* filter flags */ DECLARE_BITMAP(event_filter, 256); - unsigned short group_filter; + unsigned int group_filter; snd_use_lock_t use_lock; int event_lost; /* ports */ diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index fdc76f23e03f..9079ccfdc866 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -369,7 +369,7 @@ static void setup_client_group_filter(struct seq_ump_client *client) cptr = snd_seq_kernel_client_get(client->seq_client); if (!cptr) return; - filter = ~(1U << 0); /* always allow groupless messages */ + filter = SND_SEQ_GROUP_FILTER_GROUPS; /* always allow groupless messages */ for (p = 0; p < SNDRV_UMP_MAX_GROUPS; p++) { if (client->ump->groups[p].active) filter &= ~(1U << (p + 1)); From 01801e20d69346e1e6cec0d908f1cea3a49e51b5 Mon Sep 17 00:00:00 2001 From: Rodrigo Faria Date: Tue, 5 May 2026 19:55:18 +0100 Subject: [PATCH 24/28] ALSA: hda/realtek: Add mute LED fixup for HP Pavilion 15-cs1xxx Add a new fixup for the mute LED on the HP Pavilion 15-cs1xxx series using the VREF on NID 0x1b. The BIOS on these models (tested up to F.32) incorrectly reports the mute LED on NID 0x18 via DMI OEM strings, which lacks VREF capabilities. This fixup overrides the LED pin to the correct NID 0x1b. Signed-off-by: Rodrigo Faria Link: https://patch.msgid.link/20260505185518.23625-1-rodrigofilipefaria@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index a7525ed83e2b..11d0ea8ed859 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -1669,6 +1669,21 @@ static void alc295_fixup_hp_mute_led_coefbit11(struct hda_codec *codec, } } +/* Override wrong pin to NID 0x1b (F.32 BIOS reports 0x18 via DMI OEM string) + * on HP pavilion 15-cs1xxx laptops + */ +static void alc295_fixup_hp_pavilion_mute_led_1b(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + alc269_fixup_hp_mute_led(codec, fix, action); + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->mute_led_nid = 0x1b; +} + static void alc233_fixup_lenovo_coef_micmute_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3870,6 +3885,7 @@ enum { ALC290_FIXUP_SUBWOOFER, ALC290_FIXUP_SUBWOOFER_HSJACK, ALC295_FIXUP_HP_MUTE_LED_COEFBIT11, + ALC295_FIXUP_HP_PAVILION_MUTE_LED_1B, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_LENOVO_XPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, @@ -5715,6 +5731,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_hp_mute_led_coefbit11, }, + [ALC295_FIXUP_HP_PAVILION_MUTE_LED_1B] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_hp_pavilion_mute_led_1b, + }, [ALC298_FIXUP_SAMSUNG_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc298_fixup_samsung_amp, @@ -6931,6 +6951,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { 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, 0x856a, "HP Pavilion 15-cs1xxx", ALC295_FIXUP_HP_PAVILION_MUTE_LED_1B), 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, 0x8603, "HP Omen 17-cb0xxx", ALC285_FIXUP_HP_MUTE_LED), From 5337213381df578058e2e41da93cbd0e4639935f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Wed, 6 May 2026 00:34:47 -0300 Subject: [PATCH 25/28] ALSA: core: Serialize deferred fasync state checks MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_fasync_helper() updates fasync->on under snd_fasync_lock, and snd_fasync_work_fn() now also evaluates fasync->on under the same lock. snd_kill_fasync() still tests the flag before taking the lock, leaving an unsynchronized read against FASYNC enable/disable updates. Move the enabled-state check into the locked section. Also clear fasync->on under snd_fasync_lock in snd_fasync_free() before unlinking the pending entry. Together with the locked sender-side check, this publishes teardown before flushing the deferred work and prevents a racing sender from requeueing the entry after free has started. Fixes: ef34a0ae7a26 ("ALSA: core: Add async signal helpers") Fixes: 8146cd333d23 ("ALSA: core: Fix potential data race at fasync handling") Cc: stable@vger.kernel.org Signed-off-by: Cássio Gabriel Link: https://patch.msgid.link/20260506-alsa-core-fasync-on-lock-v1-1-ea48c77d6ca4@gmail.com Signed-off-by: Takashi Iwai --- sound/core/misc.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/core/misc.c b/sound/core/misc.c index 5aca09edf971..833124c8e4fa 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -148,9 +148,11 @@ EXPORT_SYMBOL_GPL(snd_fasync_helper); void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) { - if (!fasync || !fasync->on) + if (!fasync) return; guard(spinlock_irqsave)(&snd_fasync_lock); + if (!fasync->on) + return; fasync->signal = signal; fasync->poll = poll; list_move(&fasync->list, &snd_fasync_list); @@ -163,8 +165,10 @@ void snd_fasync_free(struct snd_fasync *fasync) if (!fasync) return; - scoped_guard(spinlock_irq, &snd_fasync_lock) + scoped_guard(spinlock_irq, &snd_fasync_lock) { + fasync->on = 0; list_del_init(&fasync->list); + } flush_work(&snd_fasync_work); kfree(fasync); From 2bcbb163162789d3488562073dbb99d9bd71a762 Mon Sep 17 00:00:00 2001 From: Rosen Penev Date: Tue, 5 May 2026 20:18:54 -0700 Subject: [PATCH 26/28] ALSA: sparc/dbri: add missing fallthrough Fixes compiler error with probably newer compilers: sound/sparc/dbri.c:595:2: error: unannotated fall-through between switch labels [-Werror,-Wimplicit-fallthrough] 595 | case 1: | ^ sound/sparc/dbri.c:595:2: note: insert 'break;' to avoid fall-through 595 | case 1: | ^ | break; Signed-off-by: Rosen Penev Link: https://patch.msgid.link/20260506031854.780411-1-rosenp@gmail.com Signed-off-by: Takashi Iwai --- sound/sparc/dbri.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 75f82a92ff44..2f5f62079fa4 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -592,6 +592,7 @@ static __u32 reverse_bytes(__u32 b, int len) fallthrough; case 2: b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1); + fallthrough; case 1: case 0: break; From bee87cf0f1248c0f20710d7a79df41fe892d9f88 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 5 May 2026 17:11:23 +0100 Subject: [PATCH 27/28] ASoC: cs35l56: Don't use devres to unregister component Manually call snd_soc_unregister_component() from cs35l56_remove() instead of using devres cleanup. This ensures that the component is destroyed before cs35l56_remove() starts cleanup of anything the component code could be using. Devres cleanup happens after the driver remove() callback, so if snd_soc_register_component() is used, it will not be destroyed until after cs35l56_remove() has returned. But there is some cleanup that must be done in cs35l56_remove(), or wrapped in a custom devres cleanup handler to ensure correct ordering. The simplest option is to call snd_soc_unregister_component() at the start of cs35l56_remove(). Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Closes: https://sashiko.dev/#/patchset/20260501103002.2843735-1-rf%40opensource.cirrus.com Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260505161124.3621000-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 378017fcea10..e847bb32af2e 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1956,9 +1956,9 @@ int cs35l56_common_probe(struct cs35l56_private *cs35l56) goto err; } - ret = devm_snd_soc_register_component(cs35l56->base.dev, - &soc_component_dev_cs35l56, - cs35l56_dai, ARRAY_SIZE(cs35l56_dai)); + ret = snd_soc_register_component(cs35l56->base.dev, + &soc_component_dev_cs35l56, + cs35l56_dai, ARRAY_SIZE(cs35l56_dai)); if (ret < 0) { dev_err_probe(cs35l56->base.dev, ret, "Register codec failed\n"); goto err; @@ -2057,6 +2057,8 @@ EXPORT_SYMBOL_NS_GPL(cs35l56_init, "SND_SOC_CS35L56_CORE"); void cs35l56_remove(struct cs35l56_private *cs35l56) { + snd_soc_unregister_component(cs35l56->base.dev); + cs35l56->base.init_done = false; /* From fd4d83e1437d6395021b21531e187c8a67ac21b0 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 5 May 2026 17:11:24 +0100 Subject: [PATCH 28/28] ASoC: cs35l56: Destroy workqueue in probe error path The error path in cs35l56_common_probe() should call destroy_workqueue() on the workqueue that was created by cs35l56_dsp_init(). Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260505161124.3621000-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index e847bb32af2e..849d70ca23d6 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1970,6 +1970,9 @@ err: gpiod_set_value_cansleep(cs35l56->base.reset_gpio, 0); regulator_bulk_disable(ARRAY_SIZE(cs35l56->supplies), cs35l56->supplies); + if (cs35l56->dsp_wq) + destroy_workqueue(cs35l56->dsp_wq); + return ret; } EXPORT_SYMBOL_NS_GPL(cs35l56_common_probe, "SND_SOC_CS35L56_CORE");