- Alsa-devel - mailman.alsa-project.org

[PATCH v2 0/5] Add DisplayPort sound support for Fairphone 5 smartphone
by Luca Weiss 23 Apr '25

23 Apr '25

Add the necessary sound card bits and some dts additions to enable sound over DisplayPort-over-USB-C, e.g. to a connected TV or monitor. The UCM files can be found here: https://gitlab.postmarketos.org/postmarketOS/pmaports/-/tree/master/device/… This series - in spirit - depends on the series enabling DisplayPort in the first place, but can land pretty independently, especially the ASoC bits: https://lore.kernel.org/linux-arm-msm/20250312-fp5-pmic-glink-dp-v2-0-a5592… Signed-off-by: Luca Weiss <luca.weiss(a)fairphone.com> --- Changes in v2: - Revamp series based on comments on v1, doesn't have much too much in common anymore - Use sm8250 instead of sc8280xp sndcard file, so port other required changes from sc8280xp.c to sm8250.c - This also changes the sound card compatible to from qcom,qcm6490-sndcard to fairphone,fp5-sndcard - Link to v1: https://lore.kernel.org/r/20240809-fp5-dp-sound-v1-0-d7ba2c24f6b9@fairphone… --- Luca Weiss (5): ASoC: dt-bindings: qcom,sm8250: Add Fairphone 5 sound card ASoC: qcom: sm8250: set card driver name from match data ASoC: qcom: sm8250: add DisplayPort Jack support ASoC: qcom: sm8250: Add Fairphone 5 soundcard compatible arm64: dts: qcom: qcm6490-fairphone-fp5: Add DisplayPort sound support .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qcm6490-fairphone-fp5.dts | 31 ++++++++++++++++++++++ sound/soc/qcom/sm8250.c | 28 +++++++++++-------- 3 files changed, 49 insertions(+), 11 deletions(-) --- base-commit: 7f1dd4e6634d085f3c9652d4e0f1903659fb96f9 change-id: 20240809-fp5-dp-sound-b3768f3019bd Best regards, -- Luca Weiss <luca.weiss(a)fairphone.com>

4 12

[PATCH] ALSA: usb-audio: Add Pioneer DJ DJM-V10 support
by František Kučera 22 Apr '25

22 Apr '25

From: František Kučera <franta-linux(a)frantovo.cz> Pioneer DJ DJM-V10 is a high end mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Input and output is enabled by this patch: - 12 channels (6 stereo) in each direction - 44 100 / 48 000 / 96 000 Hz - S24_3LE Capture channels can be configured through alsamixer. Playback channels are fixed and controlled by hardware knobs on the mixer. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/mixer_quirks.c | 62 ++++++++++++++++++++++++++++++++++++++-- sound/usb/quirks-table.h | 51 +++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 1 + 3 files changed, 111 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a90673d43822..01091c3a83fc 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3525,10 +3525,13 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_CAP_RECOUT 0x0a #define SND_DJM_CAP_RECOUT_NOMIC 0x0e #define SND_DJM_CAP_NONE 0x0f +#define SND_DJM_CAP_FXSEND 0x10 #define SND_DJM_CAP_CH1PFADER 0x11 #define SND_DJM_CAP_CH2PFADER 0x12 #define SND_DJM_CAP_CH3PFADER 0x13 #define SND_DJM_CAP_CH4PFADER 0x14 +#define SND_DJM_CAP_EXT1SEND 0x21 +#define SND_DJM_CAP_EXT2SEND 0x22 #define SND_DJM_CAP_CH1PREFADER 0x31 #define SND_DJM_CAP_CH2PREFADER 0x32 #define SND_DJM_CAP_CH3PREFADER 0x33 @@ -3559,6 +3562,7 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_450_IDX 0x5 #define SND_DJM_A9_IDX 0x6 +#define SND_DJM_V10_IDX 0x7 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -3598,8 +3602,8 @@ static const char *snd_djm_get_label_caplevel_common(u16 wvalue) } }; -// The DJM-A9 has different capture levels than other, older models -static const char *snd_djm_get_label_caplevel_a9(u16 wvalue) +// Models like DJM-A9 or DJM-V10 have different capture levels than others +static const char *snd_djm_get_label_caplevel_high(u16 wvalue) { switch (wvalue) { case 0x0000: return "+15dB"; @@ -3627,6 +3631,7 @@ static const char *snd_djm_get_label_cap_common(u16 wvalue) case SND_DJM_CAP_RECOUT_NOMIC: return "Rec Out without Mic"; case SND_DJM_CAP_AUX: return "Aux"; case SND_DJM_CAP_NONE: return "None"; + case SND_DJM_CAP_FXSEND: return "FX SEND"; case SND_DJM_CAP_CH1PREFADER: return "Pre Fader Ch1"; case SND_DJM_CAP_CH2PREFADER: return "Pre Fader Ch2"; case SND_DJM_CAP_CH3PREFADER: return "Pre Fader Ch3"; @@ -3635,6 +3640,8 @@ static const char *snd_djm_get_label_cap_common(u16 wvalue) case SND_DJM_CAP_CH2PFADER: return "Post Fader Ch2"; case SND_DJM_CAP_CH3PFADER: return "Post Fader Ch3"; case SND_DJM_CAP_CH4PFADER: return "Post Fader Ch4"; + case SND_DJM_CAP_EXT1SEND: return "EXT1 SEND"; + case SND_DJM_CAP_EXT2SEND: return "EXT2 SEND"; default: return NULL; } }; @@ -3653,7 +3660,8 @@ static const char *snd_djm_get_label_cap_850(u16 wvalue) static const char *snd_djm_get_label_caplevel(u8 device_idx, u16 wvalue) { switch (device_idx) { - case SND_DJM_A9_IDX: return snd_djm_get_label_caplevel_a9(wvalue); + case SND_DJM_A9_IDX: return snd_djm_get_label_caplevel_high(wvalue); + case SND_DJM_V10_IDX: return snd_djm_get_label_caplevel_high(wvalue); default: return snd_djm_get_label_caplevel_common(wvalue); } }; @@ -3853,6 +3861,50 @@ static const struct snd_djm_ctl snd_djm_ctls_a9[] = { SND_DJM_CTL("Ch4 Input", a9_cap5, 2, SND_DJM_WINDEX_CAP) }; +// DJM-V10 +static const u16 snd_djm_opts_v10_cap_level[] = { + 0x0000, 0x0100, 0x0200, 0x0300, 0x0400, 0x0500 +}; +static const u16 snd_djm_opts_v10_cap1[] = { + 0x0103, + 0x0100, 0x0102, 0x0106, 0x0110, 0x0107, + 0x0108, 0x0109, 0x010a, 0x0121, 0x0122 +}; +static const u16 snd_djm_opts_v10_cap2[] = { + 0x0200, 0x0202, 0x0206, 0x0210, 0x0207, + 0x0208, 0x0209, 0x020a, 0x0221, 0x0222 +}; +static const u16 snd_djm_opts_v10_cap3[] = { + 0x0303, + 0x0300, 0x0302, 0x0306, 0x0310, 0x0307, + 0x0308, 0x0309, 0x030a, 0x0321, 0x0322 +}; +static const u16 snd_djm_opts_v10_cap4[] = { + 0x0403, + 0x0400, 0x0402, 0x0406, 0x0410, 0x0407, + 0x0408, 0x0409, 0x040a, 0x0421, 0x0422 +}; +static const u16 snd_djm_opts_v10_cap5[] = { + 0x0500, 0x0502, 0x0506, 0x0510, 0x0507, + 0x0508, 0x0509, 0x050a, 0x0521, 0x0522 +}; +static const u16 snd_djm_opts_v10_cap6[] = { + 0x0603, + 0x0600, 0x0602, 0x0606, 0x0610, 0x0607, + 0x0608, 0x0609, 0x060a, 0x0621, 0x0622 +}; + +static const struct snd_djm_ctl snd_djm_ctls_v10[] = { + SND_DJM_CTL("Master Capture Level Capture Switch", v10_cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Capture Ch1 Capture Switch", v10_cap1, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch2 Capture Switch", v10_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch3 Capture Switch", v10_cap3, 0, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch4 Capture Switch", v10_cap4, 0, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch5 Capture Switch", v10_cap5, 0, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch6 Capture Switch", v10_cap6, 0, SND_DJM_WINDEX_CAP) + // playback channels are fixed and controlled by hardware knobs on the mixer +}; + static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_250MK2_IDX] = SND_DJM_DEVICE(250mk2), [SND_DJM_750_IDX] = SND_DJM_DEVICE(750), @@ -3861,6 +3913,7 @@ static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2), [SND_DJM_450_IDX] = SND_DJM_DEVICE(450), [SND_DJM_A9_IDX] = SND_DJM_DEVICE(a9), + [SND_DJM_V10_IDX] = SND_DJM_DEVICE(v10), }; @@ -4151,6 +4204,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2b73, 0x003c): /* Pioneer DJ / AlphaTheta DJM-A9 */ err = snd_djm_controls_create(mixer, SND_DJM_A9_IDX); break; + case USB_ID(0x2b73, 0x0034): /* Pioneer DJ DJM-V10 */ + err = snd_djm_controls_create(mixer, SND_DJM_V10_IDX); + break; } return err; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8954be23325c..eafc0d73cca1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3187,6 +3187,57 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Pioneer DJ DJM-V10 + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0034), + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + QUIRK_COMPOSITE_END + } + } +}, /* * MacroSilicon MS2100/MS2106 based AV capture cards diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9112313a9dbc..4dcddab1a0e3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1872,6 +1872,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, break; case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */ case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ + case USB_ID(0x2b73, 0x0034): /* Pioneer DJM-V10 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; case USB_ID(0x08e4, 0x017f): /* Pioneer DJM-750 */ -- 2.43.0

2 1

[PATCH 0/9] ALSA: usb-audio: More fixes / improvements on PCM
by Takashi Iwai 22 Apr '25

22 Apr '25

Hi, this is a patch set for USB-audio to fix and improve the PCM stream handling. The primary target is the enhancement of the low-latency playback that was introduced recently for dealing with the small period size (that is used typically by JACK). There are a few other fixes that are relevant with it. The current patches are found in topic/usb-audio branch of sound.git tree, too. Takashi === Takashi Iwai (9): ALSA: usb-audio: Restrict rates for the shared clocks ALSA: usb-audio: Fix possible race at sync of urb completions ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback ALSA: usb-audio: Disable low-latency playback for free-wheel mode ALSA: usb-audio: Disable low-latency mode for implicit feedback sync ALSA: usb-audio: Check available frames for the next packet size ALSA: usb-audio: Add spinlock to stop_urbs() ALSA: usb-audio: Improved lowlatency playback support ALSA: usb-audio: Avoid killing in-flight URBs during draining sound/usb/card.h | 11 ++- sound/usb/endpoint.c | 230 +++++++++++++++++++++++++++++++------------ sound/usb/endpoint.h | 13 ++- sound/usb/pcm.c | 157 ++++++++++++++++++++++------- 4 files changed, 305 insertions(+), 106 deletions(-) -- 2.26.2

2 23

acp3x-alc5682-max98357: Fix path of HiFi.conf
by GitHub pull_request - opened 22 Apr '25

22 Apr '25

alsa-project/alsa-ucm-conf pull request #551 was opened from meithecatte: Closes: https://github.com/alsa-project/alsa-ucm-conf/issues/550 Fixes: 1048796e7fa9 ("Rename ucm2/AMD/acp3xalc5682m98 to ucm2/AMD/acp3x-alc5682-max98357") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/551 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/551.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Audio doesn't work on acp3xalc5682m98357 since 1.2.14
by GitHub issues - opened 22 Apr '25

22 Apr '25

alsa-project/alsa-ucm-conf issue #550 was opened from meithecatte: Since 1.2.14, pipewire doesn't detect the audio device, and trying to use `aplay` results in ``` ALSA lib pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave aplay: main:850: audio open error: No such file or directory ``` `aplay -l` lists ``` card 1: acp3xalc5682m98 [acp3xalc5682m98357], device 0: Playback rt5682-aif1-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: acp3xalc5682m98 [acp3xalc5682m98357], device 1: HiFi Playback HiFi-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `git bisect` says 0a87bd36b77e4e22397450d7ba3f9d36ce9eb5e0 is the first bad commit. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/550 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ASoC: tas2781: Add "Acoustic Tuning" kcontrol for acoustic application
by Shenghao Ding 21 Apr '25

21 Apr '25

"Acoustic Tuning" kcontrol is a bridge to the acoustic tuning application tool which can tune the chips' acoustic effect. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781.h | 13 ++++ sound/soc/codecs/tas2781-i2c.c | 138 ++++++++++++++++++++++++++++++++- 2 files changed, 149 insertions(+), 2 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index eff011444cc8..a9c236ab8cba 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -32,6 +32,8 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define TASDEV_DATA_PAYLOAD_SIZE 128 + /* PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 @@ -159,10 +161,21 @@ struct calidata { unsigned int cali_dat_sz_per_dev; }; +struct acoustic_data { + unsigned char len; + unsigned char id; + unsigned char addr; + unsigned char book; + unsigned char page; + unsigned char reg; + unsigned char data[TASDEV_DATA_PAYLOAD_SIZE]; +}; + struct tasdevice_priv { struct tasdevice tasdevice[TASDEVICE_MAX_CHANNELS]; struct tasdevice_rca rcabin; struct calidata cali_data; + struct acoustic_data acou_data; struct tasdevice_fw *fmw; struct gpio_desc *speaker_id; struct gpio_desc *reset; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b950914b7d48..0d52cfc12743 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -328,6 +328,24 @@ static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, return 0; } +static int tasdev_acoustic_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + struct acoustic_data *p = &priv->acou_data; + + if (p->id == 'r' && p->len <= bytes_ext->max) + memcpy(dst, p, p->len); + else + dev_err(priv->dev, "Not ready for get.\n"); + + return 0; +} + static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, unsigned char *dst) { @@ -700,6 +718,84 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, return calib_data_get(tas_priv, reg, &dst[1]); } +static int tasdev_acoustic_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *src = ucontrol->value.bytes.data; + int j, len, ret, value, reg; + unsigned short chn; + + if (src[0] > bytes_ext->max) { + dev_err(priv->dev, "pkg len(%u) is larger than max(%d).\n", + src[0], bytes_ext->max); + return 0; + } + + switch (src[1]) { + case 'r': + /* length of data to read */ + len = src[6]; + break; + case 'w': + /* Skip 6 bytes for package type and register address */ + len = src[0] - 6; + break; + default: + dev_err(priv->dev, "%s Wrong code %02x.\n", __func__, src[1]); + return 0; + } + + if (len < 1) { + dev_err(priv->dev, "pkg fmt invalid %02x.\n", len); + return 0; + } + + for (j = 0; j < priv->ndev; j++) + if (src[2] == priv->tasdevice[j].dev_addr) { + chn = j; + break; + } + if (j >= priv->ndev) { + dev_err(priv->dev, "no such device 0x%02x.\n", src[2]); + return 0; + } + + reg = TASDEVICE_REG(src[3], src[4], src[5]); + + guard(mutex)(&priv->codec_lock); + + if (src[1] == 'w') { + if (len > 1) + ret = tasdevice_dev_bulk_write(priv, chn, reg, + &src[6], len); + else + ret = tasdevice_dev_write(priv, chn, reg, src[6]); + } else { + struct acoustic_data *p = &priv->acou_data; + + memcpy(p, src, 6); + if (len > 1) { + ret = tasdevice_dev_bulk_read(priv, chn, reg, + p->data, len); + } else { + ret = tasdevice_dev_read(priv, chn, reg, &value); + p->data[0] = value; + } + p->len = len + 6; + } + + if (ret) { + dev_err(priv->dev, "i2c communication error.\n"); + return 0; + } + + return 1; +} + static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1043,10 +1139,11 @@ static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; - int nr_controls = 1; + struct soc_bytes_ext *ext_acoustic; + char *acoustic_name, *name; + int nr_controls = 2; int mix_index = 0; int ret; - char *name; prof_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(prof_ctrls[0]), GFP_KERNEL); @@ -1068,6 +1165,43 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) prof_ctrls[mix_index].put = tasdevice_set_profile_id; mix_index++; + ext_acoustic = devm_kzalloc(tas_priv->dev, sizeof(*ext_acoustic), + GFP_KERNEL); + if (!ext_acoustic) + return -ENOMEM; + + /* + * This kcontrol is a bridge to the acoustic tuning application + * tool which can tune the chips' acoustic effect. + */ + acoustic_name = devm_kstrdup(tas_priv->dev, "Acoustic Tuning", + GFP_KERNEL); + if (!acoustic_name) + return -ENOMEM; + /* + * package structure for PPC3 communications: + * Pkg len (1 byte) + * Pkg id (1 byte, 'r' or 'w') + * Dev id (1 byte, i2c address) + * Book id (1 byte) + * Page id (1 byte) + * Reg id (1 byte) + * switch (pkg id) { + * case 'w': + * 1 byte, length of data to read + * case 'r': + * data payload (1~128 bytes) + * } + */ + ext_acoustic->max = sizeof(tas_priv->acou_data); + prof_ctrls[mix_index].name = acoustic_name; + prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + prof_ctrls[mix_index].info = snd_soc_bytes_info_ext; + prof_ctrls[mix_index].put = tasdev_acoustic_data_put; + prof_ctrls[mix_index].get = tasdev_acoustic_data_get; + prof_ctrls[mix_index].private_value = (unsigned long)ext_acoustic; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, prof_ctrls, nr_controls < mix_index ? nr_controls : mix_index); -- 2.34.1

3 5

Speaker can't mute when headphone was plugged
by Kailang 20 Apr '25

20 Apr '25

Hi Takashi, Speaker can't mute when headphone was plugged for HP platform. Add attach patch will solve issue. BR, Kailang

2 1

ucm2: MediaTek: mt8395-evk: Add support for SOF
by GitHub pull_request - opened 17 Apr '25

17 Apr '25

alsa-project/alsa-ucm-conf pull request #549 was opened from julien-massot: Similar to #513, add support for MT8395 with SOF enabled. We use different playback and capture PCM. Also, split the mixers that are not yet available upstream. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No audio on Lenovo Yoga Slim 7 14ILL10
by GitHub issues - edited 16 Apr '25

16 Apr '25

alsa-project/alsa-ucm-conf issue #548 was edited from miguel1117: Only "Dummy Output" is available as an output device on the Lenovo Yoga Slim 7 14ILL10. I'm not absolutely sure if this is the right place to report this so apologies if it's not. [ALSA information](https://alsa-project.org/db/?f=71342ec47d52681c81335d75d2947d8… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/548 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No audio on Lenovo Yoga Slim 7 14ILL10
by GitHub issues - edited 16 Apr '25

16 Apr '25

alsa-project/alsa-ucm-conf issue #548 was edited from miguel1117: Only "Dummy Output" is available as an output device on the Lenovo Yoga Slim 7 14ILL1. I'm not absolutely sure if this is the right place to report this so apologies if it's not. [ALSA information](https://alsa-project.org/db/?f=71342ec47d52681c81335d75d2947d8… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/548 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0