March 2025 - Alsa-devel - mailman.alsa-project.org

No Micrphone detected on ALC4080 - Gigabyte B850I Aorus Pro (rev 1.0)
by GitHub issues - edited 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was edited from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `amixer -c events` (unplugging and plugging mic) ``` event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 ``` `amixer -c 2 contents` ``` numid=8,iface=CARD,name='Clock Source 3 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=12,iface=CARD,name='Clock Source 4 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=15,iface=CARD,name='Clock Source 5 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=19,iface=CARD,name='Clock Source 6 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=20,iface=CARD,name='Headphone - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=5,iface=CARD,name='Mic - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=9,iface=CARD,name='Mic - Input Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=16,iface=CARD,name='Speaker - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=13,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=17,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=14,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=18,iface=MIXER,name='PCM Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=6,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=10,iface=MIXER,name='Mic Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=11,iface=MIXER,name='Mic Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=2,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=4,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Micrphone detected on ALC4080 - Gigabyte B850I Aorus Pro (rev 1.0)
by GitHub issues - edited 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was edited from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Micrphone detected on ALC4080
by GitHub issues - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was opened from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Unable to attach an LED to headphones without HW mute ALSA Control
by GitHub issues - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #527 was opened from stefdb: I have a laptop with a CS42L43 Codec which supplies the Headphone Jack. I'm trying to attach the mute state of the Headphones to the Speaker LED. However, the issue is that the CS42L43 does not possess a Hardware Mute ALSA Control which could be attached to the Speaker LED. I first tried to use the Volume Control to attach to the LED, however, that was a non-starter because the Mute is software controlled, and thus muting headphones did not touch the Volume Control. I then tried to add a new control using cset-new, and link it to the Headphones device ([0001-sof-soundwire-Attach-cs42l43-Headphone-Switch-to-spe.patch](https://gi…), and whilst that worked with the LED, the mute no longer actually muted the Headphone, since there was now a "real" control to set the mute which did not actually mute the Device. Other things I have tried include mapping the Volume control as a Switch, but that does not work, I assume, because the switch control needs to be Boolean. Is there a way to create a new control to track the LED status, whilst also ensuring that the Headphones still mute? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/527 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM2 Conditional cset in BootSequence
by Giedrius Trainavičius 18 Mar '25

18 Mar '25

Hello, I'm writing a UCM2 .conf file, depending on Device Tree options, some kcontrols may or may not exist - how do I handle this in the BootSequence, so alsactl does not error out if some controls don't exist? I know there's a possible If condition to check if a control exists, but how do I use that to conditionally append to the BootSequence? Thank you! Giedrius.

2 2

[PATCH] ALSA: hda/realtek: Support mute LED on HP Laptop 15s-du3xxx The mute LED on this HP laptop uses ALC236 and requires a quirk to function. This patch enables the existing quirk for the device.
by Dhruv Deshpande 18 Mar '25

18 Mar '25

Signed-off-by: Dhruv Deshpande <dhrv.d(a)proton.me> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a84857a3c2bf..c0b97c29bf89 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10519,6 +10519,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x881d, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x881e, "HP Laptop 15s-du3xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.48.1

2 3

[PATCH] ALSA: usb-audio: separate DJM-A9 cap lvl options
by Olivia Mackintosh 18 Mar '25

18 Mar '25

Mixer quicks for the Pioneer DJM-A9 mixer was added in 5289d00 with additional capture level values added to the common DJM array of values. This breaks the existing DJM mixers however as alsa-utils relies on enumeration of the actual mixer options based on the value array which results in error when storing state. This commit just separates the A9 values into a separate array and references them in the corresponding mixer control. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/mixer_quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 79c8c1540ee2..49fb849f7d40 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3688,8 +3688,7 @@ static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) // common DJM capture level option values static const u16 snd_djm_opts_cap_level[] = { - 0x0000, 0x0100, 0x0200, 0x0300, 0x400, 0x500 }; - + 0x0000, 0x0100, 0x0200, 0x0300 }; // DJM-250MK2 static const u16 snd_djm_opts_250mk2_cap1[] = { @@ -3831,6 +3830,8 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { // DJM-A9 +static const u16 snd_djm_opts_a9_cap_level[] = { + 0x0000, 0x0100, 0x0200, 0x0300, 0x0400, 0x0500 }; static const u16 snd_djm_opts_a9_cap1[] = { 0x0107, 0x0108, 0x0109, 0x010a, 0x010e, 0x111, 0x112, 0x113, 0x114, 0x0131, 0x132, 0x133, 0x134 }; @@ -3844,7 +3845,7 @@ static const u16 snd_djm_opts_a9_cap5[] = { 0x0501, 0x0502, 0x0503, 0x0505, 0x0506, 0x0507, 0x0508, 0x0509, 0x050a, 0x050e }; static const struct snd_djm_ctl snd_djm_ctls_a9[] = { - SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Capture Level", a9_cap_level, 0, SND_DJM_WINDEX_CAPLVL), SND_DJM_CTL("Master Input", a9_cap1, 3, SND_DJM_WINDEX_CAP), SND_DJM_CTL("Ch1 Input", a9_cap2, 2, SND_DJM_WINDEX_CAP), SND_DJM_CTL("Ch2 Input", a9_cap3, 2, SND_DJM_WINDEX_CAP), -- 2.48.1

2 1

ucm2: HDA: HiFi-analog/mic: Refactor the analog mic discovery and pre…
by GitHub pull_request - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf pull request #526 was opened from ujfalusi: …sentation The current mic device creation works on certain machines and fails on others. There are several places of conflicts and setups which can only just fail, but this is mostly not an issue if the user never uses the mic, only the speaker/headset - which, to be honest is what most of us do ;) As an example: The mic selection in most codecs are via enum and it is assumed to be named 'Input Source', which is not always the case as some device uses 'Capture Source' for the control's name. There is also different sets of mics that one can select from: Exhibit A numid=6,iface=MIXER,name='Input Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Headphone Mic' : values=1 Exhibit B numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Internal Mic' ; Item #1 'Headset Mic' : values=0 Exhibit C numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Internal Mic' ; Item #1 'Headset Mic' ; Item #2 'Headphone Mic' : values=0 Exhibit D (this pushes the limits... The patch will ignore item 1) numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Internal Mic' ; Item #1 'Internal Mic 1' ; Item #2 'Mic' : values=2 Other issue is that we have this 'Headphone Mic', which turned out to be a 'Stereo Microphone in Headphone Jack', so if it is selected then the Headphone cannot work, they conflict, they use the same rings for different direction and purpose. This patch aims to make the mic discovery a bit more deterministic and pragmatic. But even if the UCM creates the use case profiles correctly, it is still up to UIs (KDE/GNOME/etc) to misunderstand how UCM presents the profiles, what they mean and most of all what 'Mic1', `Mic2', etc is. KDE presents the profiles as they are and user can selct between them to pick the right combination of output and input. GNOME goes further with simplification (and fails with it) and presents 'random' Configuration profiles for Output and Input, plus a device selection and they do work in an interesting way. GNOME also have popup for specifying the type of the plugged accessory, which does not worl at all with UCM profiles. But, this patch is meant for a small step to have clear rules based mic presentation for HDA. The expectation is that what have worked will work as it used to and what did not worked should be detected and presented correctly. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/526 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/526.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/3] ASoC: amd: acp: Fix snd_soc_acpi_mach id's duplicate symbol error
by Venkata Prasad Potturu 17 Mar '25

17 Mar '25

Move snd_soc_acpi_mach id's of all acp platforms form header file to amd-acpi-mach.c file to avoid below errors. ld.lld: error: duplicate symbol: amp_rt1019 ld.lld: error: duplicate symbol: amp_max ld.lld: error: duplicate symbol: snd_soc_acpi_amd_acp63_acp_machines ld.lld: error: duplicate symbol: snd_soc_acpi_amd_acp70_acp_machines ld.lld: error: duplicate symbol: snd_soc_acpi_amd_rmb_acp_machines ld.lld: error: duplicate symbol: snd_soc_acpi_amd_acp_machines Fixes: 6e60db74b69c ("ASoC: amd: acp: Refactor acp machine select") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202503160801.yExt0K2E-lkp@intel.com/ Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/Kconfig | 8 +++ sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-pci.c | 8 +-- sound/soc/amd/acp/amd-acpi-mach.c | 93 +++++++++++++++++++++++++++++++ sound/soc/amd/acp/amd.h | 79 ++------------------------ 5 files changed, 111 insertions(+), 79 deletions(-) create mode 100644 sound/soc/amd/acp/amd-acpi-mach.c diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 157c124570c8..b9432052c638 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -28,6 +28,9 @@ config SND_SOC_AMD_ACP_LEGACY_COMMON config SND_SOC_AMD_ACP_I2S tristate +config SND_SOC_AMD_ACPI_MACH + tristate + config SND_SOC_AMD_ACP_PCM tristate select SND_SOC_ACPI if ACPI @@ -37,6 +40,7 @@ config SND_SOC_AMD_ACP_PCI depends on X86 && PCI depends on ACPI select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH help This options enables generic PCI driver for ACP device. @@ -47,6 +51,7 @@ config SND_AMD_ASOC_RENOIR select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH depends on X86 && PCI help This option enables Renoir I2S support on AMD platform. @@ -58,6 +63,7 @@ config SND_AMD_ASOC_REMBRANDT select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH depends on AMD_NODE depends on X86 && PCI help @@ -74,6 +80,7 @@ config SND_AMD_ASOC_ACP63 select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH help This option enables Acp6.3 I2S support on AMD platform. Say Y if you want to enable AUDIO on ACP6.3 @@ -88,6 +95,7 @@ config SND_AMD_ASOC_ACP70 select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH help This option enables Acp7.0 PDM support on AMD platform. Say Y if you want to enable AUDIO on ACP7.0 diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 7c75892e678b..08220b9a3802 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -11,6 +11,7 @@ snd-acp-pdm-y := acp-pdm.o snd-acp-legacy-common-y := acp-legacy-common.o snd-acp-pci-y := acp-pci.o snd-amd-sdw-acpi-y := amd-sdw-acpi.o +snd-amd-acpi-mach-y := amd-acpi-mach.o #platform specific driver snd-acp-renoir-y := acp-renoir.o @@ -32,6 +33,7 @@ obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o obj-$(CONFIG_SND_SOC_AMD_ACP_PDM) += snd-acp-pdm.o obj-$(CONFIG_SND_SOC_AMD_ACP_LEGACY_COMMON) += snd-acp-legacy-common.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCI) += snd-acp-pci.o +obj-$(CONFIG_SND_SOC_AMD_ACPI_MACH) += snd-amd-acpi-mach.o obj-$(CONFIG_SND_AMD_ASOC_RENOIR) += snd-acp-renoir.o obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 9322379cb36f..e15904aabc06 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -137,26 +137,26 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id chip->name = "acp_asoc_renoir"; chip->rsrc = &rn_rsrc; chip->acp_hw_ops_init = acp31_hw_ops_init; - chip->machines = snd_soc_acpi_amd_acp_machines; + chip->machines = &snd_soc_acpi_amd_acp_machines; break; case 0x6f: chip->name = "acp_asoc_rembrandt"; chip->rsrc = &rmb_rsrc; chip->acp_hw_ops_init = acp6x_hw_ops_init; - chip->machines = snd_soc_acpi_amd_rmb_acp_machines; + chip->machines = &snd_soc_acpi_amd_rmb_acp_machines; break; case 0x63: chip->name = "acp_asoc_acp63"; chip->rsrc = &acp63_rsrc; chip->acp_hw_ops_init = acp63_hw_ops_init; - chip->machines = snd_soc_acpi_amd_acp63_acp_machines; + chip->machines = &snd_soc_acpi_amd_acp63_acp_machines; break; case 0x70: case 0x71: chip->name = "acp_asoc_acp70"; chip->rsrc = &acp70_rsrc; chip->acp_hw_ops_init = acp70_hw_ops_init; - chip->machines = snd_soc_acpi_amd_acp70_acp_machines; + chip->machines = &snd_soc_acpi_amd_acp70_acp_machines; break; default: dev_err(dev, "Unsupported device revision:0x%x\n", pci->revision); diff --git a/sound/soc/amd/acp/amd-acpi-mach.c b/sound/soc/amd/acp/amd-acpi-mach.c new file mode 100644 index 000000000000..d95047d2ee94 --- /dev/null +++ b/sound/soc/amd/acp/amd-acpi-mach.c @@ -0,0 +1,93 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * amd-acpi-match.c - tables and support for ACP platforms + * ACPI enumeration. + * + * Copyright 2025 Advanced Micro Devices, Inc. + */ + +#include <sound/soc-acpi.h> + +struct snd_soc_acpi_codecs amp_rt1019 = { + .num_codecs = 1, + .codecs = {"10EC1019"} +}; + +struct snd_soc_acpi_codecs amp_max = { + .num_codecs = 1, + .codecs = {"MX98360A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines[] = { + { + .id = "10EC5682", + .drv_name = "acp3xalc56821019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_rt1019, + }, + { + .id = "RTL5682", + .drv_name = "acp3xalc5682sm98360", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_max, + }, + { + .id = "RTL5682", + .drv_name = "acp3xalc5682s1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_rt1019, + }, + { + .id = "AMDI1019", + .drv_name = "renoir-acp", + }, + { + .id = "ESSX8336", + .drv_name = "acp3x-es83xx", + }, + {}, +}; +EXPORT_SYMBOL_NS_GPL(snd_soc_acpi_amd_acp_machines, "SND_SOC_ACP_COMMON"); + +struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_acp_machines[] = { + { + .id = "10508825", + .drv_name = "rmb-nau8825-max", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_max, + }, + { + .id = "AMDI0007", + .drv_name = "rembrandt-acp", + }, + { + .id = "RTL5682", + .drv_name = "rmb-rt5682s-rt1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_rt1019, + }, + {}, +}; +EXPORT_SYMBOL_NS_GPL(snd_soc_acpi_amd_rmb_acp_machines, "SND_SOC_ACP_COMMON"); + +struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_acp_machines[] = { + { + .id = "AMDI0052", + .drv_name = "acp63-acp", + }, + {}, +}; +EXPORT_SYMBOL_NS_GPL(snd_soc_acpi_amd_acp63_acp_machines, "SND_SOC_ACP_COMMON"); + +struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_acp_machines[] = { + { + .id = "AMDI0029", + .drv_name = "acp70-acp", + }, + {}, +}; +EXPORT_SYMBOL_NS_GPL(snd_soc_acpi_amd_acp70_acp_machines, "SND_SOC_ACP_COMMON"); + +MODULE_DESCRIPTION("AMD ACP tables and support for ACPI enumeration"); +MODULE_LICENSE("GPL"); +MODULE_AUTHOR("Venkataprasad.potturu(a)amd.com"); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 796f8efd395c..928aa60ab386 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -277,81 +277,10 @@ struct acp_resource acp70_rsrc = { .sram_pte_offset = 0x03800000, }; -struct snd_soc_acpi_codecs amp_rt1019 = { - .num_codecs = 1, - .codecs = {"10EC1019"} -}; - -struct snd_soc_acpi_codecs amp_max = { - .num_codecs = 1, - .codecs = {"MX98360A"} -}; - -struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines[] = { - { - .id = "10EC5682", - .drv_name = "acp3xalc56821019", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_rt1019, - }, - { - .id = "RTL5682", - .drv_name = "acp3xalc5682sm98360", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_max, - }, - { - .id = "RTL5682", - .drv_name = "acp3xalc5682s1019", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_rt1019, - }, - { - .id = "AMDI1019", - .drv_name = "renoir-acp", - }, - { - .id = "ESSX8336", - .drv_name = "acp3x-es83xx", - }, - {}, -}; - -struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_acp_machines[] = { - { - .id = "10508825", - .drv_name = "rmb-nau8825-max", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_max, - }, - { - .id = "AMDI0007", - .drv_name = "rembrandt-acp", - }, - { - .id = "RTL5682", - .drv_name = "rmb-rt5682s-rt1019", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_rt1019, - }, - {}, -}; - -struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_acp_machines[] = { - { - .id = "AMDI0052", - .drv_name = "acp63-acp", - }, - {}, -}; - -struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_acp_machines[] = { - { - .id = "AMDI0029", - .drv_name = "acp70-acp", - }, - {}, -}; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_acp_machines; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_acp_machines; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_acp_machines; extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops; extern const struct snd_soc_dai_ops acp_dmic_dai_ops; -- 2.39.2

2 3

[PATCH v2 00/14] Implement acp_common_hw_ops support for all acp platforms
by Venkata Prasad Potturu 17 Mar '25

17 Mar '25

This patch series is to implement acp_common_hw_ops support and refactor interrupt handler registration and interrupt handling. Venkata Prasad Potturu (14): ASoC: amd: acp: Remove redundant acp70 chip->name ASoC: amd: acp: Implement acp_common_hw_ops support for acp platforms ASoC: amd: acp: Refactor dmic-codec platform device creation ASoC: amd: acp: Refactor acp platform device creation ASoC: amd: acp: Refactor acp machine select ASoC: amd: acp: Add new interrupt handle callbacks in acp_common_hw_ops ASoC: amd: acp: Remove redundant acp_dev_data structure ASoC: amd: acp: Move spin_lock and list initialization to acp-pci driver ASoC: amd: acp: Remove white line ASoC: amd: acp: Refactor acp70 platform resource structure ASoC: amd: acp: Refactor acp63 platform resource structure ASoC: amd: acp: Refactor rembrant platform resource structure ASoC: amd: acp: Refactor renoir platform resource structure ASoC: amd: acp: Fix for enabling DMIC on acp platforms via _DSD entry sound/soc/amd/acp/acp-i2s.c | 185 +++++++++--------- sound/soc/amd/acp/acp-legacy-common.c | 254 ++++++++++++++++++------ sound/soc/amd/acp/acp-pci.c | 210 ++++++++++++-------- sound/soc/amd/acp/acp-pdm.c | 53 +++-- sound/soc/amd/acp/acp-platform.c | 145 ++++---------- sound/soc/amd/acp/acp-rembrandt.c | 111 +++-------- sound/soc/amd/acp/acp-renoir.c | 110 ++--------- sound/soc/amd/acp/acp63.c | 94 +++------ sound/soc/amd/acp/acp70.c | 84 +++----- sound/soc/amd/acp/amd.h | 270 +++++++++++++++++++++----- sound/soc/amd/acp/chip_offset_byte.h | 12 +- 11 files changed, 792 insertions(+), 736 deletions(-) -- 2.39.2

4 17

sof-soundwire: rt1318: add playback control switch
by GitHub pull_request - opened 17 Mar '25

17 Mar '25

alsa-project/alsa-ucm-conf pull request #525 was opened from shumingfan: rt1318: add playback control switch Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/525 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/525.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

iecset.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 15 Mar '25

15 Mar '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' -e '\\~$' <file>" to find obvious trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:1: style: .TH missing fourth argument; consider package/project name and version (e.g., "groff 1.23.0") an.tmac:<stdin>:24: misuse, warning: .BI is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. an.tmac:<stdin>:28: misuse, warning: .BI is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. an.tmac:<stdin>:33: misuse, warning: .BI is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.17-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 34-3 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.41-4 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

DRAFT [for alsa-lib-1.2.14]: USB-Audio: GoXLR: enable detection of beta firmware (25 channels)
by GitHub pull_request - opened 15 Mar '25

15 Mar '25

alsa-project/alsa-ucm-conf pull request #524 was opened from perexg: 'Syntax 8' is required (alsa-lib 1.2.14+) Fixes: 17f9b4f ("USB-Audio: GoXLR - fix the channel detection for mini, cleanups") Link: https://github.com/alsa-project/alsa-ucm-conf/issues/444 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/524 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/524.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

GoXLR: Add 'Broadcast Stream Mix 2' to Capture if channels are available
by GitHub pull_request - opened 15 Mar '25

15 Mar '25

alsa-project/alsa-ucm-conf pull request #523 was opened from FrostyCoolSlug: Fixes: #444 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/523 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/523.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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pyalsa.alsamixer.Element.get_volume_dB returns error status instead of actual value
by GitHub issues - opened 15 Mar '25

15 Mar '25

alsa-project/alsa-python issue #16 was opened from tespent: I am writing a status collector for my PC, and I noticed that `pyalsa.alsamixer.Element.get_volume_dB` always returns 0 in my test, which lead to invalid result for the collected status. After digging into code I notice that the following code returns `res` instead of `val` as the result: https://github.com/alsa-project/alsa-python/blob/f8f9260282eb9c97f53e4689e0… I am able to workaround this issue by using `elem.ask_volume_dB(elem.get_volume())` but this looks silly. Btw, I'm willing to contribute and subscribed to alsa-devel mailing list but I didn't figure out exactly how. Can someone tell me? Issue URL : https://github.com/alsa-project/alsa-python/issues/16 Repository URL: https://github.com/alsa-project/alsa-python

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ucm2: Add support for Lenovo Yoga 7x and T14s
by GitHub pull_request - opened 14 Mar '25

14 Mar '25

alsa-project/alsa-ucm-conf pull request #522 was opened from Srinivas-Kandagatla: This patch series adds support for Lenovo T14s and Yoga Slim 7x based on Qualcomm x1e80100 SoC. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/522 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/522.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: GoXLR - fix the channel detection for mini, cleanups
by GitHub pull_request - opened 14 Mar '25

14 Mar '25

alsa-project/alsa-ucm-conf pull request #521 was opened from perexg: Note that detection for 25 channels requires commit [1] (alsa-lib 1.2.14+). [1] https://github.com/alsa-project/alsa-lib/commit/7fbd47ce797939862025ef42486… BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/444 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/521 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/521.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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FFADO Save / Load for M-Audio
by Nick 13 Mar '25

13 Mar '25

Hello there, Is Save / Load implemented for FFADO - Mixer at this time? This is pertaining to M-Audio ProFire (610 and 2626). thank you! Nick

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Split the GoXLR configuration into Full Device and Mini
by GitHub pull_request - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-ucm-conf pull request #520 was opened from FrostyCoolSlug: This commit splits the GoXLR and GoXLR Mini into separate configurations. The GoXLR Mini will always (at this point) have 21 channels defined, where as the full sized GoXLR will have either 23 or 25 depending on Firmware (23 is FAR more common at this point, as the firmware with 25 channels is still in beta). Separating out the two devices should at least partially solve the recent issue with Pipewire. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/520 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/520.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Need to edit Steinberg configuration files
by GitHub issues - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-ucm-conf issue #519 was opened from AdrianusWest: I faced a problem with disappearance of HiFi profile in Steinberg UR22C audio card settings. Attempting to fix it myself did not lead to anything. I addressed the issue to pipewire developers, who pointed out that the existing configuration file needs to be corrected: "The problem here is that previously, the channel splitting was done with ALSA dshare plugin which as it turns out does not require the exact channel numbers to match. As this introduces additional latency, in 1.4.0 the splitting is now done on Pipewire side. But of course there was a nasty surprise (and affects multiple devices), so probably there will be 1.4.1 with a workaround for wrong channel numbers in profiles." The link to the thread is attached: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4609 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/519 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Dynamic PCM switching in ALSA with multi room audio sync support
by GitHub issues - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-lib issue #443 was opened from Asikbct: **Challenges:** The PCM device is already opened by applications, making it difficult to switch to a new output seamlessly. Since most Music Service SDKs initialize the PCM device at startup, re-initializing it would require users to manually reopen their streaming app and re-select the device, negatively impacting the user experience. **Requirement:** User can switch between different audio output options (Line Out, Optical, USB Out, Bluetooth, etc.) at any time. The transition to a new output should be seamless, whether the user is currently playing a song from any music sources. Playback should resume on the newly selected output without requiring the user to manually reopen the content provider app and restart playback from the beginning. **Problems:** 1. **Reopening the PCM** Most music services open and configure the PCM device during initialization. Changing the PCM device (e.g., switching from Line Out to Optical Out) requires reopening it, which forces music services to reinitialize. This may require the end user to relaunch the respective app, reselect the device, and restart playback, impacting the user experience. 2. **Multiroom Sync** Many music services that support multiroom synchronization rely on Linux ALSA APIs like snd_pcm_delay() to estimate the kernel buffer fill level and synchronize audio playback across devices. Any new approach to meet the requirements, should not disrupt this synchronization. I've tried virtual loopback, PulseAudio, and a customized dswitch plugin. While all these methods enable dynamic PCM switching, I couldn't achieve stable long-term latency on the same PCM. Any guidance or suggestions on achieving stable long-term latency along with dynamic PCM switching would be greatly appreciated. Issue URL : https://github.com/alsa-project/alsa-lib/issues/443 Repository URL: https://github.com/alsa-project/alsa-lib

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[GIT PULL] ASoC fixes for v6.14-rc6
by Mark Brown 13 Mar '25

13 Mar '25

The following changes since commit 7eb172143d5508b4da468ed59ee857c6e5e01da6: Linux 6.14-rc5 (2025-03-02 11:48:20 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc6 for you to fetch changes up to 658fb7fe8e7f4014ea17a4da0e0c1d9bc319fa35: ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS (2025-03-12 16:12:24 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 The bulk of this is driver specific fixes, mostly unremarkable. There's also one core fix from Charles, fixing up confusion around the limiting of maximum control values. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: codecs: wsa884x: report temps to hwmon in millidegree of Celsius Arnd Bergmann (1): ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS Bard Liao (2): ASoC: rt1320: set wake_capable = 0 explicitly ASoC: rt722-sdca: add missing readable registers Charles Keepax (2): ASoC: ops: Consistently treat platform_max as control value ASoC: cs42l43: Fix maximum ADC Volume Christophe JAILLET (1): ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() Maciej Strozek (1): ASoC: cs42l43: Add jack delay debounce after suspend Peter Ujfalusi (1): ASoC: Intel: sof_sdw: Fix unlikely uninitialized variable use in create_sdw_dailinks() Thomas Mizrahi (1): ASoC: amd: yc: Support mic on another Lenovo ThinkPad E16 Gen 2 model Thorsten Blum (1): ASoC: tegra: Fix ADX S24_LE audio format include/sound/soc.h | 5 ++++- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs42l43-jack.c | 13 ++++++++++--- sound/soc/codecs/cs42l43.c | 17 +++++++++++++++-- sound/soc/codecs/cs42l43.h | 3 +++ sound/soc/codecs/rt1320-sdw.c | 3 +++ sound/soc/codecs/rt722-sdca-sdw.c | 4 ++++ sound/soc/codecs/wm0010.c | 13 +++++++++++-- sound/soc/codecs/wsa884x.c | 4 ++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/soc-ops.c | 15 +++++++-------- sound/soc/tegra/tegra210_adx.c | 4 ++-- 12 files changed, 69 insertions(+), 21 deletions(-)

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Channel count in GoXLR SplitPCM configuration
by GitHub issues - edited 12 Mar '25

12 Mar '25

alsa-project/alsa-ucm-conf issue #518 was edited from pv: Similar issue as #515, and related to #444: We have a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602 that GoXLR UCM profile with SplitPCM=1 specifies always 23 channels, but the device can't be opened with exactly 23 channels. According to https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 the device may have 21, 23, or 25 depending on firmware/etc details. In Pipewire, I think we add a workaround that just uses the available channel count even if it is different from what UCM profile specifies, but logs warnings if the counts are different so these issues in the profiles can be caught. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/518 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Channel count in GoXLR SplitPCM configuration
by GitHub issues - opened 12 Mar '25

12 Mar '25

alsa-project/alsa-ucm-conf issue #518 was opened from pv: Similar issue as #515, and related to #444: We have a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 that GoXLR UCM profile with SplitPCM=1 specifies always 23 channels, but the device can't be opened with exactly 23 channels. According to https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 the device may have 21, 23, or 25 depending on firmware/etc details. In Pipewire, I think we add a workaround that just uses the available channel count even if it is different from what UCM profile specifies, but logs warnings if the counts are different so these issues in the profiles can be caught. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/518 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.14-rc6
by Mark Brown 12 Mar '25

12 Mar '25

The following changes since commit 7eb172143d5508b4da468ed59ee857c6e5e01da6: Linux 6.14-rc5 (2025-03-02 11:48:20 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc6 for you to fetch changes up to ed92bc5264c4357d4fca292c769ea9967cd3d3b6: ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() (2025-03-11 13:51:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 The bulk of this is driver specific fixes, mostly unremarkable. There's also one core fix from Charles, fixing up confusion around the limiting of maximum control values. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: codecs: wsa884x: report temps to hwmon in millidegree of Celsius Bard Liao (2): ASoC: rt1320: set wake_capable = 0 explicitly ASoC: rt722-sdca: add missing readable registers Charles Keepax (2): ASoC: ops: Consistently treat platform_max as control value ASoC: cs42l43: Fix maximum ADC Volume Christophe JAILLET (1): ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() Maciej Strozek (1): ASoC: cs42l43: Add jack delay debounce after suspend Peter Ujfalusi (1): ASoC: Intel: sof_sdw: Fix unlikely uninitialized variable use in create_sdw_dailinks() Thomas Mizrahi (1): ASoC: amd: yc: Support mic on another Lenovo ThinkPad E16 Gen 2 model Thorsten Blum (1): ASoC: tegra: Fix ADX S24_LE audio format include/sound/soc.h | 5 ++++- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs42l43-jack.c | 13 ++++++++++--- sound/soc/codecs/cs42l43.c | 17 +++++++++++++++-- sound/soc/codecs/cs42l43.h | 3 +++ sound/soc/codecs/rt1320-sdw.c | 3 +++ sound/soc/codecs/rt722-sdca-sdw.c | 4 ++++ sound/soc/codecs/wm0010.c | 13 +++++++++++-- sound/soc/codecs/wsa884x.c | 4 ++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/soc-ops.c | 15 +++++++-------- sound/soc/tegra/tegra210_adx.c | 4 ++-- 12 files changed, 69 insertions(+), 21 deletions(-)

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