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[PATCH 5.10] ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf()
by Alexandra Diupina 17 Nov '25

17 Nov '25

From: Takashi Iwai <tiwai(a)suse.de> commit 94c1ceb043c1a002de9649bb630c8e8347645982 upstream. snprintf() returns the would-be-filled size when the string overflows the given buffer size, hence using this value may result in the buffer overflow (although it's unrealistic). This patch replaces with a safer version, scnprintf() for papering over such a potential issue. Fixes: 29c8e4398f02 ("ASoC: SOF: Intel: hda: add extended rom status dump to error log") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20220801165420.25978-4-tiwai@suse.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Alexandra Diupina <adiupina(a)astralinux.ru> --- Backport fix for CVE-2022-50050 sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index b4cc72483137..1d879c2b81e1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -437,7 +437,7 @@ static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev) for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_ROM_STATUS + i * 0x4); - len += snprintf(msg + len, sizeof(msg) - len, " 0x%x", value); + len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); } sof_dev_dbg_or_err(sdev->dev, hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS, -- 2.30.2

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snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` ``` $pacman -Q | grep alsa alsa-card-profiles 1:1.4.9-1 alsa-lib 1.2.14-2 alsa-plugins 1:1.2.12-5 alsa-topology-conf 1.2.5.1-4 alsa-ucm-conf 1.2.14-2 alsa-utils 1.2.14-1 lib32-alsa-lib 1.2.14-2 lib32-alsa-plugins 1.2.12-1 vlc-plugin-alsa 3.0.21-31 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

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snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

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snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` I don't understand what I'm doing wrong can sm1 guess Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

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snd-loopback does not work
by GitHub issues - opened 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was opened from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 I don't understand what I'm doing wrong can sm1 guess Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

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[GIT PULL] ASoC fixes for v6.18-rc5
by Mark Brown 14 Nov '25

14 Nov '25

The following changes since commit 6146a0f1dfae5d37442a9ddcba012add260bceb0: Linux 6.18-rc4 (2025-11-02 11:28:02 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.18-rc5 for you to fetch changes up to 360b3730f8eab6c4467c6cca4cb0e30902174a63: ASoC: rsnd: fix OF node reference leak in rsnd_ssiu_probe() (2025-11-13 00:36:01 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.18 A small collection of fixes, all driver specific and none especially remarkable unless you have the hardware (many not even then). ---------------------------------------------------------------- Claudiu Beznea (1): ASoC: da7213: Use component driver suspend/resume Haotian Zhang (3): ASoC: cs4271: Fix regulator leak on probe failure ASoC: codecs: va-macro: fix resource leak in probe error path ASoC: rsnd: fix OF node reference leak in rsnd_ssiu_probe() Miaoqian Lin (1): ASoC: sdw_utils: fix device reference leak in is_sdca_endpoint_present() Niranjan H Y (2): ASoC: tas2783A: Fix issues in firmware parsing ASoC: SDCA: bug fix while parsing mipi-sdca-control-cn-list Richard Fitzgerald (1): ASoC: doc: cs35l56: Update firmware filename description for B0 silicon Shenghao Ding (1): ASoC: tas2781: fix getting the wrong device number Documentation/sound/codecs/cs35l56.rst | 9 ++--- sound/soc/codecs/cs4271.c | 10 +++-- sound/soc/codecs/da7213.c | 69 ++++++++++++++++++++++------------ sound/soc/codecs/da7213.h | 1 + sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/tas2781-i2c.c | 9 ++++- sound/soc/codecs/tas2783-sdw.c | 20 +++++++++- sound/soc/renesas/rcar/ssiu.c | 3 +- sound/soc/sdca/sdca_functions.c | 3 +- sound/soc/sdw_utils/soc_sdw_utils.c | 20 +++++++--- 10 files changed, 99 insertions(+), 47 deletions(-)

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[PATCH alsa-utils] amidi: Ignore inactive MIDI ports as default at listing
by Takashi Iwai 13 Nov '25

13 Nov '25

When listing the devices, currently we show all MIDI ports including inactive ones. But those inactive ports are rarely useful, and it'd be more convenient to filter them out. This patch introduces the filtering of inactive ports at listing devices via amidi -l option. When user needs to scan all MIDI ports including inactive ports, pass the new option -x in addition. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- amidi/amidi.1 | 5 +++++ amidi/amidi.c | 11 ++++++++++- 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/amidi/amidi.1 b/amidi/amidi.1 index 5bc24ba3ad54..d3d321f16558 100644 --- a/amidi/amidi.1 +++ b/amidi/amidi.1 @@ -52,6 +52,11 @@ Prints the current version. .I \-l, \-\-list\-devices Prints a list of all hardware MIDI ports. +.TP +.I \-x, \-\-list\-inactive +Use together with \fI\-l\fP option. +Print all MIDI ports including inactive ports. + .TP .I \-L, \-\-list\-rawmidis Prints all RawMIDI definitions. diff --git a/amidi/amidi.c b/amidi/amidi.c index 75fb8c0a42a3..9a2aaf7205f0 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -57,6 +57,7 @@ static int stop; static int sysex_interval; static snd_rawmidi_t *input, **inputp; static snd_rawmidi_t *output, **outputp; +static int list_all; static void error(const char *format, ...) { @@ -76,6 +77,7 @@ static void usage(void) "-h, --help this help\n" "-V, --version print current version\n" "-l, --list-devices list all hardware ports\n" + "-x, --list-inactive list inactive ports, too\n" "-L, --list-rawmidis list all RawMIDI definitions\n" "-p, --port=name select port by name\n" "-s, --send=file send the contents of a (.syx) file\n" @@ -151,6 +153,9 @@ static void list_device(snd_ctl_t *ctl, int card, int device) card, device, sub, snd_strerror(err)); return; } + if (!list_all && + (snd_rawmidi_info_get_flags(info) & SNDRV_RAWMIDI_INFO_STREAM_INACTIVE)) + continue; name = snd_rawmidi_info_get_name(info); sub_name = snd_rawmidi_info_get_subdevice_name(info); if (sub == 0 && sub_name[0] == '\0') { @@ -471,11 +476,12 @@ static void add_send_hex_data(const char *str) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlLp:s:r:S::dt:aci:T:"; + static const char short_options[] = "hVlxLp:s:r:S::dt:aci:T:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, {"list-devices", 0, NULL, 'l'}, + {"list-inactive", 0, NULL, 'x'}, {"list-rawmidis", 0, NULL, 'L'}, {"port", 1, NULL, 'p'}, {"send", 1, NULL, 's'}, @@ -508,6 +514,9 @@ int main(int argc, char *argv[]) case 'l': do_device_list = 1; break; + case 'x': + list_all = 1; + break; case 'L': do_rawmidi_list = 1; break; -- 2.51.1

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[BUG] Realtek ALC236 Internal Mic not working on HP 255 G10 (Subsystem ID 103c:8b2f)
by Pape Samba Mbodj Sarr 13 Nov '25

13 Nov '25

Bonjour, J'ai un problème avec le microphone interne (intégré) qui n'est pas détecté par le système (affiché comme "non branché" / "not available") sur mon ordinateur portable HP. J'ai épuisé toutes les options de configuration utilisateur (modèle ALSA, patchs manuels). Ceci semble être un "quirk" (correctif) manquant pour le pilote. **Informations du système :** * **Ordinateur :** HP 255 15.6 inch G10 Notebook PC * **Codec Audio :** Realtek ALC236 * **Subsystem ID :** 103c:8b2f (Hewlett-Packard Company) * **Noyau Linux :** 6.14.0-35-generic (Version très récente) **URL du rapport alsa-info.sh (Codec Dump) :**http://alsa-project.org/db/?f=55da44f60f5deb6073c9edfe2f49c3b1cb03b53e L'entrée microphone (analog-input-mic) est marquée comme "not available" (non disponible) dans pactl. Je soupçonne que le pilote ne parvient pas à initialiser le microphone numérique (DMIC) ou confond l'entrée interne avec le jack externe. Merci d'avance de votre aide pour l'intégration de ce correctif. Cordialement, SARR Mbodj

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aplay: cannot play S24_LE wav file recorded with arecord
by GitHub issues - opened 13 Nov '25

13 Nov '25

alsa-project/alsa-utils issue #308 was opened from yoshihisa-yamamoto-re: My environment is alsa-utils version : 1.2.14 (Debian trixie) I tried to record and play PCM audio as S24_LE format over alsa as follows: ``` arecord -f S24_LE out.wav aplay out.wav ``` However, the audio turned into noise and did not seem to play properly. This problem does **not** seem to occur when using alsa-utils version 1.2.9 (Ubuntu 24.04 LTS). Searching through the commits, I found the closed issue https://github.com/alsa-project/alsa-utils/pull/260. From reading the source, it appears that the header information has been rewritten, so I dumped it to check, and it looks like this: hexdump of wav created with alsa-utils version 1.2.14 : ``` 00000000 52 49 46 46 24 e2 04 00 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 00000010 10 00 00 00 01 00 01 00 40 1f 00 00 00 7d 00 00 |........@....}..| 00000020 04 00 20 00 64 61 74 61 00 e2 04 00 e6 67 8d ff |.. .data.....g..| 00000030 c8 fd 8e ff 0c 8e 90 ff c6 18 92 ff 0a 9e 93 ff |................| 00000040 ec 1d 95 ff 7e 98 96 ff d2 0d 98 ff fd 7d 99 ff |....~........}..| 00000050 10 e9 9a ff 1c 4f 9c ff 35 b0 9d ff 6b 0c 9f ff |.....O..5...k...| 00000060 cf 63 a0 ff 74 b6 a1 ff 68 04 a3 ff be 4d a4 ff |.c..t...h....M..| 00000070 86 92 a5 ff cf d2 a6 ff a9 0e a8 ff 25 46 a9 ff |............%F..| ``` hexdump of wav created with alsa-utils version 1.2.9 : ``` 00000000 52 49 46 46 24 e2 04 00 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 00000010 10 00 00 00 01 00 01 00 40 1f 00 00 00 7d 00 00 |........@....}..| 00000020 04 00 18 00 64 61 74 61 00 e2 04 00 17 3d fe ff |....data.....=..| 00000030 3e 3d fe ff 86 41 fe ff cb 47 fe ff 3b 46 fe ff |>=...A...G..;F..| 00000040 10 47 fe ff e3 4d fe ff 66 4f fe ff 8d 50 fe ff |.G...M..fO...P..| 00000050 be 4f fe ff 40 58 fe ff e7 5b fe ff 27 60 fe ff |.O..@X...[..'`..| 00000060 f9 60 fe ff ce 61 fe ff ce 63 fe ff d3 68 fe ff |.`...a...c...h..| 00000070 83 6f fe ff 84 6c fe ff 22 73 fe ff 91 6d fe ff |.o...l.."s...m..| ``` As the change in Commit 9cdcbe0e932b97fac88a6195f87647159a6c7194, the header information has changed BitsPerSample (bit_p_spl) from 24bit to 32bit (0x22-0x23). If I change this header info and treat the 1.2.14 wav as 24-bit per sample as before, the audio seems to play correctly. I don't have the correct information for S24_LE format, so I don't know if the header information is wrong or if I should modify the arecord to match this 32-bit. Issue URL : https://github.com/alsa-project/alsa-utils/issues/308 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] MAINTAINERS: update Tzung-Bi's email address
by Tzung-Bi Shih 11 Nov '25

11 Nov '25

Use kernel.org account instead. Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c3e9f3485449..dea293f403d9 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,7 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang(a)chromium.org> - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 1d73204451b1..ea7d4900ee4a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Realtek rt1015p codec devicetree bindings maintainers: - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Rt1015p is a rt1015 variant which does not support I2C and diff --git a/MAINTAINERS b/MAINTAINERS index cf0f18502372..f9749afc0b9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4906,7 +4906,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang <cychiang(a)chromium.org> -M: Tzung-Bi Shih <tzungbi(a)google.com> +M: Tzung-Bi Shih <tzungbi(a)kernel.org> R: Guenter Roeck <groeck(a)chromium.org> L: chrome-platform(a)lists.linux.dev S: Maintained -- 2.38.0.135.g90850a2211-goog

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