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

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838250/lsus… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

New device naming breaks HDMI audio if ELD lookup fails
by GitHub issues - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-lib issue #233 was opened from berolinux: The new device naming introduced by commit 859448f01033b40feebf9a0aab467ba57a655b5b breaks HDMI audio if the ELD lookup fails. This happens e.g. with an older Sharp Aquos TV (from 2008) connected to an AMD Radeon HD 7450 graphics card with Linux kernel 5.18.1. Adding some debug statements to control/eld.c shows it fails on the check for `l == 0`, with `l` being `0`, `cinfo.count` being `24`. A hexdump of the 24 bytes in eld is `10 00 05 00 00 10 00 01 00 00 00 00 00 00 00 00 00 00 00 00 09 07 01 00`. The EDID info from the graphics card in sysfs makes sense though: ``` $ edid-decode /sys/class/drm/card0-HDMI-A-1/edid edid-decode (hex): 00 ff ff ff ff ff ff 00 4d 10 22 10 00 00 00 00 ff 12 01 03 80 52 2e 78 2a 1b be a2 55 34 b3 26 14 4a 52 af ce 00 a9 40 90 40 81 80 01 01 01 01 01 01 01 01 01 01 02 3a 80 d0 72 38 2d 40 10 2c 45 80 34 cc 31 00 00 1a 66 21 50 b0 51 00 1b 30 40 70 36 00 00 00 00 00 00 1e 00 00 00 fc 00 53 48 41 52 50 20 48 44 4d 49 0a 20 20 00 00 00 fd 00 17 4c 0f 4b 11 00 0a 20 20 20 20 20 20 01 40 02 03 2e 72 50 9f 90 20 14 05 13 04 12 03 11 02 16 07 15 06 01 23 09 07 01 83 01 00 00 e3 05 03 01 6c 03 0c 00 20 00 80 22 c0 11 11 19 19 02 3a 80 18 71 38 2d 40 58 2c 45 00 34 cc 31 00 00 1e 01 1d 80 d0 72 1c 16 20 10 2c 25 80 34 cc 31 00 00 9e 01 1d 80 18 71 1c 16 20 58 2c 25 00 34 cc 31 00 00 9e 01 1d 00 bc 52 d0 1e 20 b8 28 55 40 34 cc 31 00 00 1e 00 00 00 00 00 00 00 00 00 63 ---------------- Block 0, Base EDID: EDID Structure Version & Revision: 1.3 Vendor & Product Identification: Manufacturer: SHP Model: 4130 Model year: 2008 Basic Display Parameters & Features: Digital display Maximum image size: 82 cm x 46 cm Gamma: 2.20 DPMS levels: Off RGB color display First detailed timing is the preferred timing Color Characteristics: Red : 0.6328, 0.3330 Green: 0.2050, 0.7021 Blue : 0.1503, 0.0810 White: 0.2919, 0.3222 Established Timings I & II: IBM : 720x400 70.082 Hz 9:5 31.467 kHz 28.320 MHz DMT 0x04: 640x480 59.940 Hz 4:3 31.469 kHz 25.175 MHz DMT 0x05: 640x480 72.809 Hz 4:3 37.861 kHz 31.500 MHz DMT 0x06: 640x480 75.000 Hz 4:3 37.500 kHz 31.500 MHz DMT 0x08: 800x600 56.250 Hz 4:3 35.156 kHz 36.000 MHz DMT 0x09: 800x600 60.317 Hz 4:3 37.879 kHz 40.000 MHz DMT 0x0a: 800x600 72.188 Hz 4:3 48.077 kHz 50.000 MHz DMT 0x0b: 800x600 75.000 Hz 4:3 46.875 kHz 49.500 MHz DMT 0x10: 1024x768 60.004 Hz 4:3 48.363 kHz 65.000 MHz DMT 0x11: 1024x768 70.069 Hz 4:3 56.476 kHz 75.000 MHz DMT 0x12: 1024x768 75.029 Hz 4:3 60.023 kHz 78.750 MHz Standard Timings: DMT 0x33: 1600x1200 60.000 Hz 4:3 75.000 kHz 162.000 MHz DMT 0x2a: 1400x1050 59.978 Hz 4:3 65.317 kHz 121.750 MHz DMT 0x23: 1280x1024 60.020 Hz 5:4 63.981 kHz 108.000 MHz Detailed Timing Descriptors: DTD 1: 1920x1080 50.000 Hz 16:9 56.250 kHz 148.500 MHz (820 mm x 460 mm) Hfront 528 Hsync 44 Hback 148 Hpol P Vfront 4 Vsync 5 Vback 36 Vpol N DTD 2: 1360x768 60.015 Hz 85:48 47.712 kHz 85.500 MHz Hfront 64 Hsync 112 Hback 256 Hpol P Vfront 3 Vsync 6 Vback 18 Vpol P Display Product Name: 'SHARP HDMI' Display Range Limits: Monitor ranges (GTF): 23-76 Hz V, 15-75 kHz H, max dotclock 170 MHz Extension blocks: 1 Checksum: 0x40 ---------------- Block 1, CTA-861 Extension Block: Revision: 3 Basic audio support Supports YCbCr 4:4:4 Supports YCbCr 4:2:2 Native detailed modes: 2 Video Data Block: VIC 31: 1920x1080 50.000 Hz 16:9 56.250 kHz 148.500 MHz (native) VIC 16: 1920x1080 60.000 Hz 16:9 67.500 kHz 148.500 MHz (native) VIC 32: 1920x1080 24.000 Hz 16:9 27.000 kHz 74.250 MHz VIC 20: 1920x1080i 50.000 Hz 16:9 28.125 kHz 74.250 MHz VIC 5: 1920x1080i 60.000 Hz 16:9 33.750 kHz 74.250 MHz VIC 19: 1280x720 50.000 Hz 16:9 37.500 kHz 74.250 MHz VIC 4: 1280x720 60.000 Hz 16:9 45.000 kHz 74.250 MHz VIC 18: 720x576 50.000 Hz 16:9 31.250 kHz 27.000 MHz VIC 3: 720x480 59.940 Hz 16:9 31.469 kHz 27.000 MHz VIC 17: 720x576 50.000 Hz 4:3 31.250 kHz 27.000 MHz VIC 2: 720x480 59.940 Hz 4:3 31.469 kHz 27.000 MHz VIC 22: 1440x576i 50.000 Hz 16:9 15.625 kHz 27.000 MHz VIC 7: 1440x480i 59.940 Hz 16:9 15.734 kHz 27.000 MHz VIC 21: 1440x576i 50.000 Hz 4:3 15.625 kHz 27.000 MHz VIC 6: 1440x480i 59.940 Hz 4:3 15.734 kHz 27.000 MHz VIC 1: 640x480 59.940 Hz 4:3 31.469 kHz 25.175 MHz Audio Data Block: Linear PCM: Max channels: 2 Supported sample rates (kHz): 48 44.1 32 Supported sample sizes (bits): 16 Speaker Allocation Data Block: FL/FR - Front Left/Right Colorimetry Data Block: xvYCC601 xvYCC709 Vendor-Specific Data Block (HDMI), OUI 00-0C-03: Source physical address: 2.0.0.0 Supports_AI Maximum TMDS clock: 170 MHz Video latency: 35 ms Audio latency: 35 ms Interlaced video latency: 51 ms Interlaced audio latency: 51 ms Detailed Timing Descriptors: DTD 3: 1920x1080 60.000 Hz 16:9 67.500 kHz 148.500 MHz (820 mm x 460 mm) Hfront 88 Hsync 44 Hback 148 Hpol P Vfront 4 Vsync 5 Vback 36 Vpol P DTD 4: 1920x1080i 50.000 Hz 16:9 28.125 kHz 74.250 MHz (820 mm x 460 mm) Hfront 528 Hsync 44 Hback 148 Hpol P Vfront 2 Vsync 5 Vback 15 Vpol P Vfront +0.5 Odd Field Vfront 2 Vsync 5 Vback 15 Vpol P Vback +0.5 Even Field DTD 5: 1920x1080i 60.000 Hz 16:9 33.750 kHz 74.250 MHz (820 mm x 460 mm) Hfront 88 Hsync 44 Hback 148 Hpol P Vfront 2 Vsync 5 Vback 15 Vpol P Vfront +0.5 Odd Field Vfront 2 Vsync 5 Vback 15 Vpol P Vback +0.5 Even Field DTD 6: 1280x720 50.000 Hz 16:9 37.500 kHz 74.250 MHz (820 mm x 460 mm) Hfront 440 Hsync 40 Hback 220 Hpol P Vfront 5 Vsync 5 Vback 20 Vpol P Checksum: 0x63 ``` Since an error code (`-EIO`) is passed on from `__snd_pcm_info_eld_fixup`, the HDMI audio output goes away altogether instead of just keeping the old name. Will send a PR to keep the audio working, but it would be nice to actually make the new naming work as well. Issue URL : https://github.com/alsa-project/alsa-lib/issues/233 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was opened from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Fixes for veyron chromebooks
by GitHub pull_request - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf pull request #164 was opened from jenneron: Add PlaybackVolume and JackControl. I didn't add JackControl for headset mic because internal mic breaks after switching to headset and back Add symlink to VEYRON-I2S, so ucm config will get probed Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/164 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/164.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [Sound-open-firmware] jack detection via input/event on linux
by Pierre-Louis Bossart 03 Jun '22

03 Jun '22

[Adding alsa-devel, Mark and Takashi to this thread] On 6/3/22 08:33, Jaroslav Kysela wrote: > On 03. 06. 22 13:17, hede wrote: >> Am 03.06.2022 11:34, schrieb Jaroslav Kysela: >>> PA/PW use only ALSA controls at the moment - thus you should define >>> JackControl as defined in use-case.h. But it seems that the driver >>> does not create or publish the ALSA jack control (I see only >>> "iface=CARD,name='HDMI/DP,pcm=2 Jack'" jack controls for HDMI). >>> >>> Jaroslav >> >> Ah, thank you Jaroslav. That's the hint I was searching for atm. >> JackDev vs. JackControl. I had a vague guess on this but now you made it >> certain. >> >> Maybe I should take a look into the driver directly. > > It seems that many ASoC drivers do not create kctls - the argument pins > is zero in the snd_soc_card_jack_new() call. The SND_JACK_HEADSET should > be remapped to kctl in sound/soc/intel/boards/sof_rt5682.c, too. Did you mean something like the patch below Jaroslav? If yes, most of the Chromebook machine drivers are missing this remapping. That would also explain why the jack detection didn't work on my GLK test device, despite a UCM file that looked good... diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5d67a2c87a1d4..1db17ebd26f63 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -85,6 +85,17 @@ struct sof_card_private { bool idisp_codec; }; +static struct snd_soc_jack_pin sof_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int sof_rt5682_quirk_cb(const struct dmi_system_id *id) { sof_rt5682_quirk = (unsigned long)id->driver_data; @@ -295,11 +306,13 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->sof_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, + sof_jack_pins, + ARRAY_SIZE(sof_jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; @@ -481,7 +494,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) } static const struct snd_kcontrol_new sof_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Left Spk"), SOC_DAPM_PIN_SWITCH("Right Spk"), @@ -489,7 +502,7 @@ static const struct snd_kcontrol_new sof_controls[] = { }; static const struct snd_soc_dapm_widget sof_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("Left Spk", NULL), SND_SOC_DAPM_SPK("Right Spk", NULL), @@ -501,8 +514,8 @@ static const struct snd_soc_dapm_widget dmic_widgets[] = { static const struct snd_soc_dapm_route sof_map[] = { /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, + { "Headphone", NULL, "HPOL" }, + { "Headphone", NULL, "HPOR" }, /* other jacks */ { "IN1P", NULL, "Headset Mic" },

3 3

[PATCH] ASoC: mt6797-mt6351: Fix refcount leak in mt6797_mt6351_dev_probe
by Miaoqian Lin 03 Jun '22

03 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: f0ab0bf250da ("ASoC: add mt6797-mt6351 driver and config option") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- changes in v2: - update Fixes tag v1 link: https://lore.kernel.org/r/20220603083417.9011-1-linmq006@gmail.com/ --- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index 496f32bcfb5e..d2f6213a6bfc 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -217,7 +217,8 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -230,6 +231,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } -- 2.25.1

1 0

[PATCH 0/6] Fixup Cirrus SX control usage
by Charles Keepax 03 Jun '22

03 Jun '22

Mostly the usage of the SX controls seems to match the lowest gain value + number of gain levels expected. The one notable exception there being cs53l30 as David noted. However, there are a couple of other places where the minimum value/TLVs are slightly incorrectly specified. There are two I couldn't quite work out yet, any input on these greatly appreciated: SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), 0.5dB step 0x40 - -96dB 0x00 - 0dB This one is weird, the datasheet says 0x40 is the minimum, which matches, but the datasheet also says 0.5dB steps to 0x00 being 0dB which doesn't match with 0x40 being the minimum, or 0xC0 being the number of levels. Some clarification will needed from the hardware guys. SOC_SINGLE_SX_TLV("Beep Volume", CS42L56_BEEP_FREQ_OFFTIME, 0, 0x07, 0x23, beep_tlv), 2dB step 0x07 - -50dB 0x00 - 0dB 0x06 - +6dB The minimum checks out at 0x07, but rest of this is weird. The values in the datasheet don't make sense, based on starting at -50dB with a 2dB step, 0x06 should equate to 12dB, over 31 volume levels, not 6dB over 35 volume levels as stated. The control specifies 0x23(35) levels, which takes us up to a final value of 0x0A at 20dB. Again some clarification from the hardware guys might be needed here. Finally it is worth noting I don't actually have hardware for most of these devices, so mostly this is just build tested. Thanks, Charles Charles Keepax (6): ASoC: cs42l52: Fix TLV scales for mixer controls ASoC: cs35l36: Update digital volume TLV ASoC: cs53l30: Correct number of volume levels on SX controls ASoC: cs42l52: Correct TLV for Bypass Volume ASoC: cs42l56: Correct typo in minimum level for SX volume controls ASoC: cs42l51: Correct minimum value for SX volume control sound/soc/codecs/cs35l36.c | 3 ++- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 8 +++++--- sound/soc/codecs/cs42l56.c | 4 ++-- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 5 files changed, 18 insertions(+), 15 deletions(-) -- 2.30.2

2 7

[PATCH v2] ASoC: Intel: cirrus-common: fix incorrect channel mapping
by Brent Lu 03 Jun '22

03 Jun '22

From: xliu <xiang.liu(a)cirrus.com> The default mapping of ASPRX1 (DAC source) is slot 0. Change the slot mapping of right amplifiers (WR and TR) to slot 1 to receive right channel data. Also update the ACPI instance ID mapping according to HW configuration. Signed-off-by: xliu <xiang.liu(a)cirrus.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 40 +++++++++++++++++++--- 1 file changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..f4192df962d6 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -54,22 +54,29 @@ static struct snd_soc_dai_link_component cs35l41_components[] = { }, }; +/* + * Mapping between ACPI instance id and speaker position. + * + * Four speakers: + * 0: Tweeter left, 1: Woofer left + * 2: Tweeter right, 3: Woofer right + */ static struct snd_soc_codec_conf cs35l41_codec_conf[] = { { .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "WL", + .name_prefix = "TL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WR", + .name_prefix = "WL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TL", + .name_prefix = "TR", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "TR", + .name_prefix = "WR", }, }; @@ -101,6 +108,21 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) return ret; } +/* + * Channel map: + * + * TL/WL: ASPRX1 on slot 0, ASPRX2 on slot 1 (default) + * TR/WR: ASPRX1 on slot 1, ASPRX2 on slot 0 + */ +static const struct { + unsigned int rx[2]; +} cs35l41_channel_map[] = { + {.rx = {0, 1}}, /* TL */ + {.rx = {0, 1}}, /* WL */ + {.rx = {1, 0}}, /* TR */ + {.rx = {1, 0}}, /* WR */ +}; + static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -134,6 +156,16 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* setup channel map */ + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(cs35l41_channel_map[i].rx), + (unsigned int *)cs35l41_channel_map[i].rx); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } } return 0; -- 2.25.1

4 3