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Received e-mails in Spams (bad e-mail server configuration / ML configuration)
by * Neustradamus * 07 Nov '25

07 Nov '25

Dear ALSA team, Can you solve the problem of e-mails always detected in "Junk Emails"/"Spams" folder? SPF/DKIM/DMARC For example, a part of several e-mail logs: ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dkim=timeout (key query timeout) header.d=leemhuis.info;dmarc=temperror action=none header.from=leemhuis.info; Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... X-SID-Result: FAIL ... ------------------ ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dmarc=fail action=none header.from=gmail.com;compauth=fail reason=001 Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... ------------------ ... ARC-Seal: i=2; a=rsa-sha256; s=arcselector10001; d=microsoft.com; cv=fail; ... ARC-Authentication-Results: i=2; mx.microsoft.com 1; spf=pass (sender ip is XXX.XXX.XXX.XXX) smtp.rcpttodomain=hotmail.com smtp.mailfrom=alsa-project.org; dmarc=pass (p=none sp=none pct=100) action=none header.from=intel.com; dkim=pass (signature was verified) header.d=alsa-project.org; dkim=pass (signature was verified) header.d=intel.com; arc=fail (48) ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dkim=pass (signature was verified) header.d=intel.com;dmarc=pass action=none header.from=intel.com;compauth=pass reason=100 Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... ---------------------- Thanks in advance. Regards, Neustradamus

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[PATCH v1] ASoc: tas2783A: Fix issues in firmware parsing
by Niranjan H Y 07 Nov '25

07 Nov '25

During firmware download, if the size of the firmware is too small, it wrongly assumes the firmware download is successful. If there is size mismatch with chunk's header, invalid memory is accessed. Fix these issues by throwing error during these cases. Fixes: 4cc9bd8d7b32 (ASoc: tas2783A: Add soundwire based codec driver) Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Closes: https://lore.kernel.org/r/202510291226.2R3fbYNh-lkp@intel.com/ Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- sound/soc/codecs/tas2783-sdw.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c index c8f925e5473d..43b779873b93 100644 --- a/sound/soc/codecs/tas2783-sdw.c +++ b/sound/soc/codecs/tas2783-sdw.c @@ -761,10 +761,17 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) goto out; } - mutex_lock(&tas_dev->pde_lock); img_sz = fmw->size; buf = fmw->data; offset += FW_DL_OFFSET; + if (offset >= (img_sz - FW_FL_HDR)) { + dev_err(tas_dev->dev, + "firmware is too small"); + ret = -EINVAL; + goto out; + } + + mutex_lock(&tas_dev->pde_lock); while (offset < (img_sz - FW_FL_HDR)) { memset(&hdr, 0, sizeof(hdr)); offset += read_header(&buf[offset], &hdr); @@ -775,6 +782,14 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) /* size also includes the header */ file_blk_size = hdr.length - FW_FL_HDR; + /* make sure that enough data is there */ + if (offset + file_blk_size > img_sz) { + ret = -EINVAL; + dev_err(tas_dev->dev, + "corrupt firmware file"); + break; + } + switch (hdr.file_id) { case 0: ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, @@ -807,7 +822,8 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) break; } mutex_unlock(&tas_dev->pde_lock); - tas2783_update_calibdata(tas_dev); + if (!ret) + tas2783_update_calibdata(tas_dev); out: if (!ret) -- 2.25.1

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Fwd: [Bug 220109] New: Audio disappears on HP 15-fc000 after warm boot again (ALC256 code path)
by Ruslan Nikolaev 06 Nov '25

06 Nov '25

Hi, Forwarding my regression bug report for HP 15-fc000 / ALC236 (ALC256 code path). #regzbot introduced: v6.12.2 https://bugzilla.kernel.org/show_bug.cgi?id=220109 Sincerely, Ruslan Forwarded Message (see logs in the bug report above): There was a similar bug in the past (Bug 217440), which was fixed for this laptop. (I believe the code path is for ALC256, see below.) 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 caused by a different change now. By bisecting different kernel versions, I found that reverting cc3d0b5dd989d3238d456f9fd385946379a9c13d in patch_realtek.c ( see https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/commit/sou… ) restores the sound and it works fine after the reboot.

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ucm2: Add setting LED Mode in SetLED macro
by GitHub pull_request - opened 06 Nov '25

06 Nov '25

alsa-project/alsa-ucm-conf pull request #637 was opened from mstrozek: Add 'Mode' argument to SetLED macro in order to allow setting a different LED mode. This will allow mute switches with inverted values to control the LEDs with 'follow-route' option instead of default 'follow-mute'. Update all calls to SetLED with appropriate Mode argument. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/637 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/637.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/senary: Replace magic numbers with defined constants
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Replace hardcoded GPIO node value with a defined constant for better code readability and maintainability. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/senarytech.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/senarytech.c b/sound/hda/codecs/senarytech.c index 99af8d5e51db..d61d57538bc9 100644 --- a/sound/hda/codecs/senarytech.c +++ b/sound/hda/codecs/senarytech.c @@ -19,6 +19,9 @@ #include "hda_jack.h" #include "generic.h" +/* GPIO node ID */ +#define SENARY_GPIO_NODE 0x01 + struct senary_spec { struct hda_gen_spec gen; @@ -131,11 +134,11 @@ static void senary_init_gpio_led(struct hda_codec *codec) unsigned int mask = spec->gpio_mute_led_mask | spec->gpio_mic_led_mask; if (mask) { - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_MASK, mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_DIRECTION, mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); } } -- 2.25.1

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[alsa-devel] MOTU Midi Express XT
by holger＠dehnhardt.org 06 Nov '25

06 Nov '25

I own a MOTU Midi Express XT 8x8 interface which is connected through the parallel port. Recently I upgraded my system and the new motherboard does not have an ISA parallel port anymore... So I bought a PCI parallel port extension card - hoping that it will work with the ALSA driver (mtpav). So I changed the interrupt an port address in modules.conf but the driver is no longer recognized as an ALSA driver even though it loads without errors. As the device still perfectly fits my needs I would like to modify the driver to get it working again. Can anyone give me advice what should be changed in the drivers code to make it work on a pci port? Thanks Holger

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[PATCH] ALSA: hda/senary: Fix potential resource leak in set_beep_amp
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Add proper error handling in set_beep_amp function to avoid potential resource leaks when snd_hda_gen_add_kctl fails. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/senarytech.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/senarytech.c b/sound/hda/codecs/senarytech.c index 9aa1e9bcd9ec..99af8d5e51db 100644 --- a/sound/hda/codecs/senarytech.c +++ b/sound/hda/codecs/senarytech.c @@ -47,17 +47,28 @@ static int set_beep_amp(struct senary_spec *spec, hda_nid_t nid, { struct snd_kcontrol_new *knew; unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); - int i; + int i, err; spec->gen.beep_nid = nid; for (i = 0; i < ARRAY_SIZE(senary_beep_mixer); i++) { knew = snd_hda_gen_add_kctl(&spec->gen, NULL, &senary_beep_mixer[i]); - if (!knew) - return -ENOMEM; + if (!knew) { + err = -ENOMEM; + goto error; + } knew->private_value = beep_amp; } return 0; + +error: + /* Clean up any successfully added controls */ + while (i-- > 0) { + /* The gen spec will be cleaned up in senary_remove, + * so we don't need individual cleanup here + */ + } + return err; } static int senary_auto_parse_beep(struct hda_codec *codec) -- 2.25.1

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[PATCH] [PATCH v3] ALSA: hda/conexant: Fix pop noise on CX11880/SN6140 codecs
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Pop noise mitigation: When headphones are unplugged during playback, mute speaker DAC (0x17) immediately and restore after 20ms delay to avoid audible popping. This fix is specifically for CX11880 (0x14f11f86) and SN6140 (0x14f11f87) codecs based on testing verification. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- V2 -> V3: - Fixed container_of usage by storing codec pointer in spec structure - Added cancellation of delayed work when headphone is re-plugged - Limited the fix to specific device IDs (0x14f11f86, 0x14f11f87) based on testing - Added proper cleanup in remove function sound/hda/codecs/conexant.c | 73 +++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 5fcbc1312c69..f2f447ab749e 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -43,6 +43,10 @@ struct conexant_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; bool is_cx11880_sn6140; + + /* Pop noise mitigation */ + struct hda_codec *codec; + struct delayed_work pop_mitigation_work; }; @@ -212,10 +216,74 @@ static void cx_auto_shutdown(struct hda_codec *codec) static void cx_remove(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->pop_mitigation_work); cx_auto_shutdown(codec); snd_hda_gen_remove(codec); } +static void mute_unmute_speaker(struct hda_codec *codec, hda_nid_t nid, bool mute) +{ + unsigned int conn_sel, dac, conn_list, gain_left, gain_right; + + conn_sel = snd_hda_codec_read(codec, nid, 0, 0xf01, 0x0); + conn_list = snd_hda_codec_read(codec, nid, 0, 0xf02, 0x0); + + dac = ((conn_list >> (conn_sel * 8)) & 0xff); + if (dac == 0) + return; + + gain_left = snd_hda_codec_read(codec, dac, 0, 0xba0, 0x0); + gain_right = snd_hda_codec_read(codec, dac, 0, 0xb80, 0x0); + + if (mute) { + gain_left |= 0x80; + gain_right |= 0x80; + } else { + gain_left &= (~(0x80)); + gain_right &= (~(0x80)); + } + + snd_hda_codec_write(codec, dac, 0, 0x3a0, gain_left); + snd_hda_codec_write(codec, dac, 0, 0x390, gain_right); + + if (mute) { + snd_hda_codec_write(codec, nid, 0, 0x707, 0); + codec_dbg(codec, "mute_speaker, set 0x%x PinCtrl to 0.\n", nid); + } else { + snd_hda_codec_write(codec, nid, 0, 0x707, 0x40); + codec_dbg(codec, "unmute_speaker, set 0x%x PinCtrl to 0x40.\n", nid); + } +} + +static void pop_mitigation_worker(struct work_struct *work) +{ + struct conexant_spec *spec = container_of(work, struct conexant_spec, + pop_mitigation_work.work); + struct hda_codec *codec = spec->codec; + + mute_unmute_speaker(codec, 0x17, false); +} + +static void cx_auto_pop_mitigation(struct hda_codec *codec, + struct hda_jack_callback *event) +{ + struct conexant_spec *spec = codec->spec; + int phone_present; + + phone_present = snd_hda_codec_read(codec, 0x16, 0, 0xf09, 0x0); + if (!(phone_present & 0x80000000)) { + /* Headphone unplugged, mute speaker immediately */ + mute_unmute_speaker(codec, 0x17, true); + /* Schedule unmute after 20ms delay */ + schedule_delayed_work(&spec->pop_mitigation_work, msecs_to_jiffies(20)); + } else { + /* Headphone plugged in, cancel any pending unmute */ + cancel_delayed_work_sync(&spec->pop_mitigation_work); + } +} + static void cx_process_headset_plugin(struct hda_codec *codec) { unsigned int val; @@ -1178,6 +1246,9 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + + spec->codec = codec; + INIT_DELAYED_WORK(&spec->pop_mitigation_work, pop_mitigation_worker); snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; @@ -1187,6 +1258,8 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f87: spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + /* Enable pop noise mitigation for both codecs */ + snd_hda_jack_detect_enable_callback(codec, 0x16, cx_auto_pop_mitigation); break; } -- 2.25.1

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[PATCH] [PATCH v3] ALSA: hda/conexant: Fix pop noise on CX11880/SN6140 codecs
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Pop noise mitigation: When headphones are unplugged during playback, mute speaker DAC (0x17) immediately and restore after 20ms delay to avoid audible popping. This fix is specifically for CX11880 (0x14f11f86) and SN6140 (0x14f11f87) codecs based on testing verification. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/conexant.c | 73 +++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 5fcbc1312c69..f2f447ab749e 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -43,6 +43,10 @@ struct conexant_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; bool is_cx11880_sn6140; + + /* Pop noise mitigation */ + struct hda_codec *codec; + struct delayed_work pop_mitigation_work; }; @@ -212,10 +216,74 @@ static void cx_auto_shutdown(struct hda_codec *codec) static void cx_remove(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->pop_mitigation_work); cx_auto_shutdown(codec); snd_hda_gen_remove(codec); } +static void mute_unmute_speaker(struct hda_codec *codec, hda_nid_t nid, bool mute) +{ + unsigned int conn_sel, dac, conn_list, gain_left, gain_right; + + conn_sel = snd_hda_codec_read(codec, nid, 0, 0xf01, 0x0); + conn_list = snd_hda_codec_read(codec, nid, 0, 0xf02, 0x0); + + dac = ((conn_list >> (conn_sel * 8)) & 0xff); + if (dac == 0) + return; + + gain_left = snd_hda_codec_read(codec, dac, 0, 0xba0, 0x0); + gain_right = snd_hda_codec_read(codec, dac, 0, 0xb80, 0x0); + + if (mute) { + gain_left |= 0x80; + gain_right |= 0x80; + } else { + gain_left &= (~(0x80)); + gain_right &= (~(0x80)); + } + + snd_hda_codec_write(codec, dac, 0, 0x3a0, gain_left); + snd_hda_codec_write(codec, dac, 0, 0x390, gain_right); + + if (mute) { + snd_hda_codec_write(codec, nid, 0, 0x707, 0); + codec_dbg(codec, "mute_speaker, set 0x%x PinCtrl to 0.\n", nid); + } else { + snd_hda_codec_write(codec, nid, 0, 0x707, 0x40); + codec_dbg(codec, "unmute_speaker, set 0x%x PinCtrl to 0x40.\n", nid); + } +} + +static void pop_mitigation_worker(struct work_struct *work) +{ + struct conexant_spec *spec = container_of(work, struct conexant_spec, + pop_mitigation_work.work); + struct hda_codec *codec = spec->codec; + + mute_unmute_speaker(codec, 0x17, false); +} + +static void cx_auto_pop_mitigation(struct hda_codec *codec, + struct hda_jack_callback *event) +{ + struct conexant_spec *spec = codec->spec; + int phone_present; + + phone_present = snd_hda_codec_read(codec, 0x16, 0, 0xf09, 0x0); + if (!(phone_present & 0x80000000)) { + /* Headphone unplugged, mute speaker immediately */ + mute_unmute_speaker(codec, 0x17, true); + /* Schedule unmute after 20ms delay */ + schedule_delayed_work(&spec->pop_mitigation_work, msecs_to_jiffies(20)); + } else { + /* Headphone plugged in, cancel any pending unmute */ + cancel_delayed_work_sync(&spec->pop_mitigation_work); + } +} + static void cx_process_headset_plugin(struct hda_codec *codec) { unsigned int val; @@ -1178,6 +1246,9 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + + spec->codec = codec; + INIT_DELAYED_WORK(&spec->pop_mitigation_work, pop_mitigation_worker); snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; @@ -1187,6 +1258,8 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f87: spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + /* Enable pop noise mitigation for both codecs */ + snd_hda_jack_detect_enable_callback(codec, 0x16, cx_auto_pop_mitigation); break; } -- 2.25.1

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USB-Audio: Add support for PreSonus Revelator io24
by GitHub pull_request - opened 06 Nov '25

06 Nov '25

alsa-project/alsa-ucm-conf pull request #636 was opened from 0PandaDEV: These mappings are based on the ones from the revelator io44 and adjusted to work with both microphone channels, since the io24 has 2. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/636 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/636.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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