Alsa + sblive + alsamixer = ???
Edit: I figured it all out! No need to read on unless you have an SBLive and want to use my configuration.
[code]state.Live {
control.1 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Wave Playback Volume'
value.0 80
value.1 80
control.2 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Wave Surround Playback Volume'
value.0 100
value.1 100
control.3 {
comment.access 'read write'
comment.type INTEGER
comment.count 1
comment.range '0 - 100'
iface MIXER
name 'Wave Center Playback Volume'
value 100
control.4 {
comment.access 'read write'
comment.type INTEGER
comment.count 1
comment.range '0 - 100'
iface MIXER
name 'Wave LFE Playback Volume'
value 100
control.5 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Wave Capture Volume'
value.0 0
value.1 0
control.6 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Wave Capture Switch'
value.0 true
value.1 true
control.7 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Synth Playback Volume'
value.0 100
value.1 100
control.8 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Synth Capture Volume'
value.0 0
value.1 0
control.9 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Synth Capture Switch'
value.0 false
value.1 false
control.10 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Surround Playback Volume'
value.0 0
value.1 0
control.11 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Surround Capture Volume'
value.0 0
value.1 0
control.12 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Surround Capture Switch'
value.0 true
value.1 true
control.13 {
comment.access 'read write'
comment.type INTEGER
comment.count 1
comment.range '0 - 100'
iface MIXER
name 'Center Playback Volume'
value 0
control.14 {
comment.access 'read write'
comment.type INTEGER
comment.count 1
comment.range '0 - 100'
iface MIXER
name 'LFE Playback Volume'
value 100
control.16 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Front Capture Volume'
value.0 0
value.1 0
control.17 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Front Capture Switch'
value false
control.18 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'AC97 Playback Volume'
value.0 0
value.1 0
control.19 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'AC97 Capture Volume'
value.0 100
value.1 100
control.20 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'IEC958 TTL Playback Volume'
value.0 0
value.1 0
control.21 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'IEC958 TTL Capture Volume'
value.0 0
value.1 0
control.22 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'IEC958 TTL Capture Switch'
value.0 false
value.1 false
control.23 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'IEC958 LiveDrive Playback Volume'
value.0 0
value.1 0
control.24 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'IEC958 LiveDrive Capture Volume'
value.0 0
value.1 0
control.25 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'IEC958 LiveDrive Capture Switch'
value.0 false
value.1 false
control.26 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Line LiveDrive Playback Volume'
value.0 0
value.1 0
control.27 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Line LiveDrive Capture Volume'
value.0 0
value.1 0
control.28 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Line LiveDrive Capture Switch'
value.0 false
value.1 false
control.29 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'IEC958 Coaxial Playback Volume'
value.0 0
value.1 0
control.30 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'IEC958 Coaxial Capture Volume'
value.0 0
value.1 0
control.31 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'IEC958 Coaxial Capture Switch'
value.0 false
value.1 false
control.32 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Line2 LiveDrive Playback Volume'
index 1
value.0 0
value.1 0
control.33 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Line2 LiveDrive Capture Volume'
index 1
value.0 0
value.1 0
control.34 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Line2 LiveDrive Capture Switch'
index 1
value.0 false
value.1 false
control.35 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 40'
iface MIXER
name 'Tone Control - Bass'
value.0 0
value.1 0
control.36 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 40'
iface MIXER
name 'Tone Control - Treble'
value.0 0
value.1 0
control.37 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Tone Control - Switch'
value.0 false
value.1 false
control.38 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'IEC958 Optical Raw Playback Switch'
value.0 false
value.1 false
control.39 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 100'
iface MIXER
name 'Headphone Playback Volume'
index 1
value.0 0
value.1 0
control.40 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Headphone Center Playback Switch'
index 1
value false
control.41 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Headphone LFE Playback Switch'
index 1
value false
control.42 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 32
iface PCM
device 2
name 'Captured FX8010 Outputs'
value.0 false
value.1 false
value.2 false
value.3 false
value.4 false
value.5 false
value.6 false
value.7 false
value.8 false
value.9 false
value.10 false
value.11 false
value.12 false
value.13 false
value.14 false
value.15 false
value.16 true
value.17 true
value.18 true
value.19 true
value.20 true
value.21 true
value.22 true
value.23 true
value.24 true
value.25 true
value.26 true
value.27 true
value.28 true
value.29 true
value.30 true
value.31 true
control.43 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Master Playback Switch'
value true
control.44 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 31'
iface MIXER
name 'Master Playback Volume'
value.0 31
value.1 31
control.52 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 31'
iface MIXER
name 'Front Playback Volume'
value.0 30
value.1 30
control.55 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'PC Speaker Playback Switch'
value false
control.56 {
comment.access 'read write'
comment.type INTEGER
comment.count 1
comment.range '0 - 15'
iface MIXER
name 'PC Speaker Playback Volume'
value 0
control.57 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Phone Playback Switch'
value false
control.58 {
comment.access 'read write'
comment.type INTEGER
comment.count 1
comment.range '0 - 31'
iface MIXER
name 'Phone Playback Volume'
value 0
control.59 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Mic Playback Switch'
value.0 false
value.1 false
control.60 {
comment.access 'read write'
comment.type INTEGER
comment.count 1
comment.range '0 - 31'
iface MIXER
name 'Mic Playback Volume'
value 0
control.61 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Mic Boost (+20dB)'
value false
control.62 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Line Playback Switch'
value.0 false
value.1 false
control.63 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 31'
iface MIXER
name 'Line Playback Volume'
value.0 0
value.1 0
control.64 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'CD Playback Switch'
value.0 false
value.1 false
control.65 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 31'
iface MIXER
name 'CD Playback Volume'
value.0 0
value.1 0
control.66 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Surround Mix Playback Switch'
value.0 false
value.1 false
control.67 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 31'
iface MIXER
name 'Surround Mix Playback Volume'
value.0 23
value.1 23
control.68 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Aux Playback Switch'
value.0 false
value.1 false
control.69 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 31'
iface MIXER
name 'Aux Playback Volume'
value.0 0
value.1 0
control.70 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'PCM Playback Switch'
value.0 false
value.1 false
control.71 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 31'
iface MIXER
name 'PCM Playback Volume'
value.0 23
value.1 23
control.72 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 2
comment.item.0 Mic
comment.item.1 CD
comment.item.2 Video
comment.item.3 Aux
comment.item.4 Line
comment.item.5 Mix
comment.item.6 'Mix Mono'
comment.item.7 Phone
iface MIXER
name 'Capture Source'
value.0 Mic
value.1 Mic
control.73 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Capture Switch'
value.0 true
value.1 true
control.74 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 15'
iface MIXER
name 'Capture Volume'
value.0 0
value.1 0
control.76 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name '3D Control - Switch'
value true
control.78 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 Mic1
comment.item.1 Mic2
iface MIXER
name 'Mic Select'
value Mic2
control.79 {
comment.access 'read write'
comment.type INTEGER
comment.count 1
comment.range '0 - 3'
iface MIXER
name '3D Control Sigmatel - Depth'
value 3
control.80 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Sigmatel 4-Speaker Stereo Playback Switch'
value false
control.81 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Sigmatel Surround Phase Inversion Playback '
value false
control.82 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 Input/Disabled
comment.item.1 'Front Output'
comment.item.2 'Rear Output'
comment.item.3 'Center/LFE Output'
comment.item.4 'Mixer Output'
iface MIXER
name 'Mic1 Jack'
value Input/Disabled
control.83 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 Input/Disabled
comment.item.1 'Front Output'
comment.item.2 'Rear Output'
comment.item.3 'Center/LFE Output'
comment.item.4 'Mixer Output'
iface MIXER
name 'LineIn Jack'
value Input/Disabled
control.84 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 Input/Disabled
comment.item.1 'Front Output'
comment.item.2 'Rear Output'
comment.item.3 'Center/LFE Output'
comment.item.4 'Mixer Output'
iface MIXER
name 'Front Jack'
value 'Front Output'
control.85 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 Input/Disabled
comment.item.1 'Front Output'
comment.item.2 'Rear Output'
comment.item.3 'Center/LFE Output'
comment.item.4 'Mixer Output'
iface MIXER
name 'Rear Jack'
value 'Rear Output'
control.86 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 Input/Disabled
comment.item.1 'Front Output'
comment.item.2 'Rear Output'
comment.item.3 'Center/LFE Output'
comment.item.4 'Mixer Output'
iface MIXER
name 'Center/LFE Jack'
value 'Center/LFE Output'
control.87 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 'Mic2 Jack'
comment.item.1 'Mic1 Jack'
comment.item.2 'Line In Jack'
comment.item.3 'Front Jack'
comment.item.4 'Rear Jack'
comment.item.5 'Center/LFE Jack'
comment.item.6 Mute
iface MIXER
name 'Mic Input Source'
value Mute
control.88 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 'Mic2 Jack'
comment.item.1 'Mic1 Jack'
comment.item.2 'Line In Jack'
comment.item.3 'Front Jack'
comment.item.4 'Rear Jack'
comment.item.5 'Center/LFE Jack'
comment.item.6 Mute
iface MIXER
name 'Line Input Source'
value Mute
control.89 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
comment.item.0 None
comment.item.1 'Front Jack'
comment.item.2 'Rear Jack'
iface MIXER
name 'Headphone Amp'
value None
control.90 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Exchange Center/LFE'
value false
control.91 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Headphone +3dB Boost'
value false
control.92 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'External Amplifier'
value false
control.93 {
comment.access 'read write'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.94 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 1
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.95 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 2
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.96 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 3
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.97 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 4
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.98 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 5
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.99 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 6
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.100 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 7
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.101 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 8
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.102 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 9
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.103 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 10
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.104 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 11
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.105 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 12
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.106 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 13
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.107 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 14
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.108 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 15
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.109 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 16
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.110 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 17
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.111 {
comment.access 'read write inactive'
comment.type INTEGER
comment.count 12
comment.range '0 - 15'
iface PCM
name 'EMU10K1 PCM Send Routing'
index 18
value.0 0
value.1 1
value.2 2
value.3 3
value.4 0
value.5 1
value.6 2
value.7 3
value.8 0
value.9 1
value.10 2
value.11 3
control.112 {
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It's because of alsa 1.0.16.
Check that this shows 1.0.16, for your kernel sound modules:
$ cat /proc/asound/version
Advanced Linux Sound Architecture Driver Version 1.0.16.
I would recommend deleting /etc/asound.conf, rebooting (just to be mega-certain that the soundcard setup isn't screwed up), then setting up alsamixer from scratch. I've seen a bad /etc/asound.conf do crazy things to soundcards
I use an Audigy 4, so it's fairly similar - here's my result of amixer, for 5.1 sound.
Similar Messages
-
please help me, i have not idea about that.
and my English is very pooooooooooooooor....
I follow:
https://wiki.archlinux.org/index.php/Ad … chitecture
$ pacman -S alsa-utils
$ alsamixer
F6 Select sound card
sound card
┌───── Sound Card ──────┐
│- (default) │
│0 HDA Intel HDMI │
│1 HDA Intel PCH │
│ enter device name... │
└───────────────────────┘
I hosed the HDA Intel PCH
press M turn the Master to unmute and turn it up (Item: Master [dB gain: 0.00])
press Esc
$ speaker-test -c 2
speaker-test 1.0.28
Playback device is default
Stream parameters are 48000Hz, S16_LE, 2 channels
Using 16 octaves of pink noise
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
Playback open error: -2,No such file or directory
$ speaker-test -c 8
speaker-test 1.0.28
Playback device is default
Stream parameters are 48000Hz, S16_LE, 8 channels
Using 16 octaves of pink noise
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
Playback open error: -2,No such file or directory
$ lspci -v
00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05)
Subsystem: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller
Flags: bus master, fast devsel, latency 0, IRQ 47
Memory at f7e10000 (64-bit, non-prefetchable) [size=16K]
Capabilities: <access denied>
Kernel driver in use: snd_hda_intel
Kernel modules: snd_hda_intel
Last edited by keol (2014-10-23 16:57:30)Set the default sound card
$lspci -nn | grep -i audio
00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06)
00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 05)
$vim /etc/modprobe.d/alsa-base.conf
options snd-hda-intel index=0 model=auto vid=8086 pid=8c20
options snd-hda-intel index=1 model=auto vid=8086 pid=0c0c
saved and reboot. -
I've installed the 5/14 release on an Acer C720 Chromebook according tom directions found at https://wiki.archlinux.org/index.php/Ac … Chromebook and https://wiki.archlinux.org/index.php/Beginners%27_Guide. All seems well except that I can't find ALSA.
According to https://wiki.archlinux.org/index.php/Ad … chitecture ALSA is included in the kernel, yet I get 'command not found' from bash to all of the following:
alsa
amixer
alsamixer
alsaconf
alsactl
Where can I get this package?
Last edited by porphyry5 (2014-06-09 17:54:07)https://wiki.archlinux.org/index.php/FA … ge_is_X.3F
$ pkgfile amixer
extra/alsa-utils
Please remember to mark the thread as solved https://bbs.archlinux.org/viewtopic.php?id=130309 -
[SOLVED] ALSA and intel soundcard - not detected, /dev/snd is empty
Hi, there was some topics mounths ago about this but I still can't figure out how to get my sound to work.
The main problem is that my soundcard is not detected and /dev/snd seems a little empty. I can't find the controlc0,1 files either....
Here is some useful info.
root@red red]# ls -l /sys/module/snd/holders/
total 0
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_hda_codec -> ../../snd_hda_codec
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_hda_intel -> ../../snd_hda_intel
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_hwdep -> ../../snd_hwdep
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_mixer_oss -> ../../snd_mixer_oss
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_pcm -> ../../snd_pcm
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_pcm_oss -> ../../snd_pcm_oss
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_seq -> ../../snd_seq
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_seq_device -> ../../snd_seq_device
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_seq_oss -> ../../snd_seq_oss
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_timer -> ../../snd_timer
[root@red red]# lsmod | grep '^snd' | column -t
snd_seq_dummy 1439 0
snd_seq_oss 28928 0
snd_seq_midi_event 5412 1 snd_seq_oss
snd_seq 50530 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 5233 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_hda_intel 22514 0
snd_pcm_oss 39096 0
snd_hda_codec 68863 1 snd_hda_intel
snd_hwdep 6150 1 snd_hda_codec
snd_mixer_oss 16956 1 snd_pcm_oss
snd_pcm 70924 3 snd_hda_intel,snd_pcm_oss,snd_hda_codec
snd_timer 19684 2 snd_seq,snd_pcm
snd 57209 10 snd_seq_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm_oss,snd_hda_codec,snd_hwdep,snd_mixer_oss,snd_pcm,snd_timer
snd_page_alloc 7161 2 snd_hda_intel,snd_pcm
[root@red red]# ls -l /dev/snd/
total 0
crw-rw---- 1 root audio 116, 3 Jun 15 17:00 seq
crw-rw---- 1 root audio 116, 2 Jun 15 17:00 timer
Souncard : Don't know the exact name, the "intel hd" one with an ICH7 bridge
Mobo : intel d945gclf (little atom board, not the newer ones)
Notice that line : snd_hda_intel 22514 0
Seems to me that the module is loaded, am I right?
EDIT: Sorry, maybe I am in the wrong section...
Last edited by Lrouge (2010-06-18 18:33:54)Looks pretty complete to me. (OK, /dev/snd does look sparse.) What do you see when you run alsamixer and press the F5 key?
$ sudo su -
# pacman -Sy alsa-utils
# alsamixer
Press F5 and check for:
* Master, Speaker, Front, Line, PCM volume down (if any)
* Any MM (bottom of vertical bar, signifies: line is muted) -- mute / un-mute with the 'm' key
Press Esc key to exit alsamixer + save changes.
Also, right after booting, run: dmesg | grep -i codec
Look for something like "Realtek" or "ALC662" or similar.
Manually add codec driver: modprobe snd_hda_codec_realtek
ls -al /dev/snd
drwxr-xr-x 2 root root 60 May 28 19:35 by-path
crw-rw---- 1 root audio 116, 7 May 28 19:35 controlC0
crw-rw---- 1 root audio 116, 6 May 28 19:35 hwC0D2
crw-rw---- 1 root audio 116, 5 May 28 19:35 pcmC0D0c
crw-rw---- 1 root audio 116, 4 May 28 19:35 pcmC0D0p
crw-rw---- 1 root audio 116, 3 May 28 19:35 seq
crw-rw---- 1 root audio 116, 2 May 28 19:35 timer
lsmod | grep snd
snd_hda_codec_realtek 179443 1
snd_seq_dummy 1099 0
snd_seq_oss 25304 0
snd_seq_midi_event 4452 1 snd_seq_oss
snd_hda_intel 18921 0
snd_seq 42628 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 4313 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_hda_codec 56728 2 snd_hda_codec_realtek,snd_hda_intel
snd_pcm_oss 33693 0
snd_hwdep 5102 1 snd_hda_codec
snd_pcm 57479 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss
snd_mixer_oss 14810 1 snd_pcm_oss
snd_timer 16117 2 snd_seq,snd_pcm
snd 43847 11 snd_hda_codec_realtek,snd_seq_oss,snd_hda_intel,snd_seq,snd_seq_device,snd_hda_codec,snd_pcm_oss,snd_hwdep,snd_pcm,snd_mixer_oss,snd_timer
soundcore 5007 1 snd
snd_page_alloc 5809 2 snd_hda_intel,snd_pcm -
Hello dear all!
There are 2 sound cards on my PC: hda_intel (ALC888 from Realtek) and SBLive 5.1
1. I could that ALSA make software resampling (44.1 to 48 kHz) when sound playing through SBLive and do not resampling if using hda_intel. How asoundrc must be configured in this case?
2. In my OS (Kubuntu 10.04) installed not latest version of ALSA libs. How I can know exactly which version of ALSA using. I mean are there any incompatibilities between for instance 1.0.23 ver of ALSA and 1.0.24 of alsa-lib, alsa-plugins (I have choose right version of alsa-lib, alsa-plugins and install them).
3. Is it possible to use OSS and ALSA both? I could that hda_intel uses ALSA, SBLive - OSS. How exactly?
Thank you in advanceThank you for your reply and detailed config.
1. I need separate settings for each sound card.
2. "Is your SBLive a hardware-mixing model".
SBLive 5.1 supports hardware mixing streams but with current ALSA settings it does not - I don't know why.
3. "Whereas surround51 doesn't use dmix". I have 4 satellites, subwoofer.
I try these
blop and cmt was installed by Synaptic.
4./etc/asound.conf was created and then I paste to it follow text:
pcm.upmix_20to51 {
type plug
slave.pcm lowpass_21to21
slave.channels 3
ttable {
0.0 1 # left channel
1.1 1 # right channel
0.2 0.5 # mix left and right ...
1.2 0.5 # ... channel for subwoofer
pcm.lowpass_21to21 {
type ladspa
slave.pcm upmix_21to51
path "/usr/lib/ladspa"
channels 3
plugins {
0 {
id 1098 # Identity (Audio) (1098/identity_audio)
policy duplicate
input.bindings.0 "Input";
output.bindings.0 "Output";
1 {
id 1672 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa)
policy none
input.bindings.2 "Input";
output.bindings.2 "Output";
input {
controls [ 300 2 ]
pcm.upmix_21to51 {
type plug
slave.pcm surround51
slave.channels 5
ttable {
0.0 1 # front left
1.1 1 # front right
0.2 1 # rear left
1.3 1 # rear right
2.5 1 # subwoofer
hint {
show on
description "Upmix channel"
A problem is: when I try to check sound in SystemSettings/Multimedia a message displayed : "The device Upmix channel does not work", and sounds played by snd_hda_intel. So this settings does not applyed to snd_emu10k1.
Where is my mistake? I should use Creative SBLive (snd_emu10k1).
Thank you in advance -
[SOLVED] Can't get sound working.
The system has onboard sound (EliteGroup GeForce6100SM-M motherboard) which I am not interested in, instead I would like to use my PCI card (Sound Blaster Live! 5.1), but I cannot get either working. And yes, I've already confirmed I am in the "audio" group, and logged out and in since...
I get the following during boot:
[ 4.173534] emu10k1-gp: unable to grab region 0xfdf00020-0xfdf0002f
[ 5.453359] AC'97 0 access is not valid [0xffffffff], removing mixer.
Here's what happens when I try setting up ALSA, along with the troubleshooting output as requested on the Arch Wiki's ALSA page:
$ alsamixer
cannot open mixer: No such file or directory
$ alsaconf
alsaconf: command not found
$ amixer
amixer: Mixer attach default error: No such file or directory
$ lsmod | grep '^snd' | column -t
snd_emu10k1 137365 0
snd_util_mem 2360 1 snd_emu10k1
snd_ac97_codec 114391 1 snd_emu10k1
snd_hwdep 6389 1 snd_emu10k1
snd_rawmidi 19458 1 snd_emu10k1
snd_seq_device 5300 2 snd_rawmidi,snd_emu10k1
snd_pcm 74780 2 snd_ac97_codec,snd_emu10k1
snd_page_alloc 7153 2 snd_pcm,snd_emu10k1
snd_timer 19544 2 snd_pcm,snd_emu10k1
snd 59214 7 snd_timer,snd_pcm,snd_seq_device,snd_rawmidi,snd_hwdep,snd_ac97_codec,snd_emu10k1
$ ls -l /dev/snd
total 0
crw-rw---- 1 root audio 116, 1 Mar 26 21:40 seq
crw-rw----+ 1 root audio 116, 33 Mar 26 21:40 timer
$ aplay -l
aplay: device_list:252: no soundcards found...
I have tried manually loading the appropriate modules, adding the ALSA Wiki's "Setting up modprobe" block o' text to my modprobe.conf, and reinstalling alsa-utils/alsa-oss/linux. I've tried every bit of troubleshooting I can find, at least the ones that don't already assume I can start alsamixer in the first place. Anything obvious I've missed?
Thanks!
EDIT: Disregard, moved the card to my other PCI slot and it works fine now.
Last edited by Phieta (2012-03-28 05:44:21)Please mark this thread as solved by prepending [SOLVED] to the title in the first post.
-
Alsa sequencer init for awe/sblive (awesfx incl.)
Here are init scripts and pkgbuild for awesfx. I did this, then noticed someone had already posted a build - sorry I don't mean to step on your toes! Since I already went to the work of creating the init-scripts, and being the latest version with some changes here they are.
/etc/rc.d/alsa-sequencer:
#!/bin/bash
. /etc/rc.conf
. /etc/rc.d/functions
. /etc/alsa/aseq.conf
case "$1" in
start)
stat_busy "Loading ALSA Sequencer"
# SFXLoad requires /dev/sequencer to already exist, but devfs
# creates it when the sequencer device itself is actually created.
# Catch22! Work-around this by removing nodes if they exist
# (sending improper data to these exhibits VERY bad behaviour),
# then create the nodes in the proper locations.
if [ -e /dev/sequencer ]; then
rm -f /dev/sequencer
fi
if [ -e /dev/sound/sequencer ]; then
rm -f /dev/sound/sequencer
fi
mknod -m 666 /dev/sound/sequencer c 14 1
ln -s /dev/sound/sequencer /dev/sequencer
ln -s /proc/asound/oss/sndstat /dev/sndstat
# Last we load the soundbank into the hardware, passing all arguments...
$aseq_load $aseq_opt $aseq_target
stat_done
stop)
stat_busy "Stopping ALSA Sequencer"
# Do nothing actually...
stat_done
restart)
# The nodes already exist, so all we need is a rerun
$aseq_load $aseq_opt $aseq_target
echo "usage: $0 {start|stop|restart}"
esac
exit 0
/etc/alsa/aseq.conf:
##. /etc/alsa/aseq.conf
# Configuration file for initializing the sequencer on Awe32/64
# and Emux audio devices.
# Specify the loader to use:
# sfxload for AWE32/64 cards
# OR
# asfxload for Emux devices
aseq_load=""
# Pass additional arguments here.
#Example:
#-c 35 -r 50 -V 75
aseq_opt=""
# Specify the soundfont/soundbank to load into the device's sequencer.
# NOTE: If target is a symlink, symlink must *not* be relative.
# Example:
# /usr/share/sounds/sf2/default.sf2
aseq_target=""
and as long as I went to the trouble of pasting the rest of it....
PKGBUILD:
# Contributor: My Name <[email protected]>
pkgname=awesfx
pkgver=0.5.0c
pkgrel=1
pkgdesc="Sequencer support files for Awe32/64 and EMUx (SBLive!) audio cards."
license=GPL
url="http://www.alsa-project.org/~iwai/awedrv.html#Utils"
depends=('alsa-lib')
source=(http://www.alsa-project.org/~iwai/$pkgname-$pkgver.tar.gz)
md5sums=('b4f9192dc6357014215a3d06a99d40f2')
build() {
cd $startdir/src/$pkgname-$pkgver
./configure --prefix=/usr --disable-debug
make || return 1
make prefix=$startdir/pkg/usr install
mkdir -p $startdir/pkg/etc/alsa
mkdir -p $startdir/pkg/etc/rc.d
cp ../../aseq.conf $startdir/pkg/etc/alsa
cp ../../alsa-sequencer $startdir/pkg/etc/rc.d
Yes, I remove my email here, but it's in the full tarball that is now in incoming if anyone wants the binaries etc.:shock: Just realized, it doesn't work properly with udev. If you use udev and patch it please submit the changes to me, so I can include them.
I don't use udev, because I have problems with it (i.e. kdm won't even load) so apologize for not seeing this sooner.
I say again
UDEV does not work
DEVFS works!!! -
Script for alsa, alsamixer dmix and modules
First, excuse me for my bad english
Hi
I've wrote my first script in bash and this is for archlinux.
This one install all necessary packages, check with hwd what are the sound cards modules that have to be probe, write the /etc/rc.d/alsamixer module and install dmix
It also modify /etc/rc.conf to had the modules and the deamon and add a user to the audio group.
The script works fine on my computer but i'd like to know if it really works well.
install_alsa.shHi,
I used this script on pretty much computers now, so it has already saved me pretty much time Thanks again!
May I give a few suggestions:
In the begin you check if the user is root with $USER, maybe it is better to use whoami instead. Because whoami will also see that you are root when you are root in su
Maybe clean up the output a bit, you probably know that you can pipe the output of a program to /dev/null, for instance instead of
amixer set Master 70 unmute
you could use
echo -n 'Setting Master volume to 70...'
if amixer set Master 70 unmute > /dev/null
then
echo 'done'
else
echo 'failed'
exit 1
fi
And maybe use the same layout of output pacman uses.
Also something goes wrong with the /etc/rc.d/aslamixer script. It keeps al the n signs in it instead of new lines. I think it is better to have a seperated file alsamixer and distribute you script in a tarball, so you can easiely copy the file to /etc/rc.d. But of course this is just my opinion.
You often add something to the rc.conf file, because this happens often and is nearly the same everytime you maybe could use a function where you send the parameters, file (in this case rc.conf) variable_name (here it is DAEMONS or MODULES) and of course the value. (for instance alsamixer)
On the end of the script you ask width users you want to add to the audio group. You can check if the user is already in the audio group (cat /etc/groups | grep audio etc) and don't show that user because he's already in it and maybe filter out groups as ftp, svn, etc
This is just a very rough idee but what about that every configuration script that's (going to be) written for arch uses the same file and output layout. For instance that every configuration script has:
1: A standart name, for instance arch-config-audio, arch-config-grub etc
2: As I already mention the same output to the users
3: Two scipts, one called configure, this script would do (if needed) some hardware detection and ask the user questions. The second called install, this would do the time consuming installation and would never ask a question to the user. This way one can do all the configuration of the 'install scripts' directly after each other and than install everything, where the user can leave the computer.
4: And of course add those script to the one of the ftp servers so you can install them with pacman
Don't take the last idee too serious, I often have a lot of stupid idees
Cheers,
David -
[CLOSED] snd-hda-Intel model mono/stereo and alsamixer issues
DROPPED: I just installed the Windows disc that came with the laptop and the official drivers, while they work better (a 6 channel wav plays fine in the speakers, linux only plays 2 of them) the most important issues like mono output only in the headphones jack is repeated. It seems the hardware problem. I don't know how I break it, but I did.
When I installed Arch linux about this laptop about two years ago everything was working OK. "aplay /usr/share/sounds/alsa/Front_Right.wav" (and left) played in the corresponding speakers and headphones. "alsamixer" used to have two volume control bars per control, for left and right.
I can't pinpoint the moment but sometime in the last few months everything is playing as mono. Now alsamixer shows only one volume bar per control and "Master" can only to be (un)muted as mono. I can still mute the "Speaker" control individually using '<' and '>'. The "Headphone" volume control is ignored, neither volume nor mute work. And the strangest thing: when I mute the left Speaker it also mutes the headphones.
I never did any alsa configuration and so I'm assuming the model auto detection is broken. I tried a few model= options without luck. The problem is the laptop is brandless (pseudo-olivetti 750). Perhaps with some luck someone can help me identify which model should I use.
Physical Description:
The laptop has to on-board speakers and two jacks on the front, one labeled with a Microhone (slash) two things I cannot recognize and the other labeled SPDIF OUT (slash) Headphones. (I tried the only model described as 2-jack in the wiki guide without luck).
# lspci -v | grep -A11 Audio
00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02)
Subsystem: Uniwill Computer Corp Device 907e
Flags: bus master, fast devsel, latency 0, IRQ 27
Memory at f0000000 (64-bit, non-prefetchable) [size=16K]
Capabilities: [50] Power Management version 2
Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+
Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00
Capabilities: [100] Virtual Channel
Capabilities: [130] Root Complex Link
Kernel driver in use: HDA Intel
Kernel modules: snd-hda-intel
# aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem]
Subdevices: 1/1
Subdevice #0: subdevice #0
# aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=Intel
HDA Intel, ALC883 Analog
Default Audio Device
front:CARD=Intel,DEV=0
HDA Intel, ALC883 Analog
Front speakers
surround40:CARD=Intel,DEV=0
HDA Intel, ALC883 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=Intel,DEV=0
HDA Intel, ALC883 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Intel,DEV=0
HDA Intel, ALC883 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Intel,DEV=0
HDA Intel, ALC883 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Intel,DEV=0
HDA Intel, ALC883 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=Intel,DEV=0
HDA Intel, ALC883 Digital
IEC958 (S/PDIF) Digital Audio Output
# aplay -D default /usr/share/sounds/alsa/Front_Left.wav
Playing WAVE '/usr/share/sounds/alsa/Front_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
#aplay -D front /usr/share/sounds/alsa/Front_Left.wav
aplay: main:608: audio open error: Device or resource busy
# lsof /dev/dsp
# (nothing)
# amixer
Simple mixer control 'Master',0
Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum
Playback channels: Mono
Limits: Playback 0 - 31
Mono: Playback 31 [100%] [0.00dB] [on]
Simple mixer control 'Headphone',0
Capabilities: pvolume pswitch penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 31 [100%] [0.00dB] [on]
Front Right: Playback 31 [100%] [0.00dB] [on]
Simple mixer control 'Speaker',0
Capabilities: pvolume pswitch penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 31 [100%] [0.00dB] [on]
Front Right: Playback 31 [100%] [0.00dB] [on]
Simple mixer control 'PCM',0
Capabilities: pvolume penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 255
Mono:
Front Left: Playback 255 [100%] [0.00dB]
Front Right: Playback 255 [100%] [0.00dB]
Simple mixer control 'Mic',0
Capabilities: pvolume pswitch penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 0 [0%] [-34.50dB] [off]
Front Right: Playback 0 [0%] [-34.50dB] [off]
Simple mixer control 'Mic Boost',0
Capabilities: volume penum
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: 0 - 3
Front Left: 0 [0%]
Front Right: 0 [0%]
Simple mixer control 'IEC958',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [off]
Simple mixer control 'IEC958 Default PCM',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
Simple mixer control 'Beep',0
Capabilities: pvolume pswitch penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 0 [0%] [-34.50dB] [off]
Front Right: Playback 0 [0%] [-34.50dB] [off]
Simple mixer control 'Capture',0
Capabilities: cvolume cswitch penum
Capture channels: Front Left - Front Right
Limits: Capture 0 - 31
Front Left: Capture 16 [52%] [12.00dB] [off]
Front Right: Capture 16 [52%] [12.00dB] [off]
Simple mixer control 'Caller ID',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [off]
Simple mixer control 'Off-hook',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [off]
Thanks for any help, I tried most of the models listed as ALC883 (and some others) without luck, is there any other option that could cause this issues?
Last edited by dresb (2011-02-07 16:29:50)Thanks a lot for the hint! codec#0 was the modem while this is from codec#1:
# cat /proc/asound/card0/codec#1
Codec: Realtek ALC883
Address: 1
Function Id: 0x1
Vendor Id: 0x10ec0883
Subsystem Id: 0x1584907e
Revision Id: 0x100002
No Modem Function Group found
Default PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Default Amp-In caps: N/A
Default Amp-Out caps: N/A
GPIO: io=2, o=0, i=0, unsolicited=1, wake=0
IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
Node 0x02 [Audio Output] wcaps 0x11: Stereo
Converter: stream=5, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x03 [Audio Output] wcaps 0x11: Stereo
Device: name="ALC883 Analog", type="Audio", device=0
Converter: stream=5, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x04 [Audio Output] wcaps 0x11: Stereo
Converter: stream=0, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x05 [Audio Output] wcaps 0x11: Stereo
Converter: stream=0, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital
Control: name="IEC958 Playback Con Mask", index=0, device=0
Control: name="IEC958 Playback Pro Mask", index=0, device=0
Control: name="IEC958 Playback Default", index=0, device=0
Control: name="IEC958 Playback Switch", index=0, device=0
Control: name="IEC958 Default PCM Playback Switch", index=0, device=0
Device: name="ALC883 Digital", type="SPDIF", device=1
Converter: stream=0, channel=0
Digital:
Digital category: 0x0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0x1e]: 16 20 24 32
formats [0x1]: PCM
Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In
Device: name="ALC883 Analog", type="Audio", device=0
Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1
Amp-In vals: [0x90 0x90]
Converter: stream=0, channel=0
SDI-Select: 0
PCM:
rates [0x160]: 44100 48000 96000
bits [0x6]: 16 20
formats [0x1]: PCM
Connection: 1
0x23
Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In
Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1
Amp-In vals: [0x00 0x00]
Converter: stream=0, channel=0
SDI-Select: 0
PCM:
rates [0x160]: 44100 48000 96000
bits [0x6]: 16 20
formats [0x1]: PCM
Connection: 1
0x22
Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital
Converter: stream=0, channel=0
SDI-Select: 0
Digital:
Digital category: 0x0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0x1e]: 16 20 24 32
formats [0x1]: PCM
Unsolicited: tag=00, enabled=0
Connection: 1
0x1f
Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In
Control: name="Mic Playback Volume", index=0, device=0
Control: name="Mic Playback Switch", index=0, device=0
Control: name="Beep Playback Volume", index=0, device=0
Control: name="Beep Playback Switch", index=0, device=0
Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1
Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80]
Connection: 10
0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17
Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out
Control: name="Headphone Playback Volume", index=0, device=0
Control: name="Headphone Playback Switch", index=0, device=0
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x00 0x00]
Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0
Amp-Out vals: [0x1f 0x1f]
Connection: 2
0x02 0x0b
Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out
Control: name="Speaker Playback Volume", index=0, device=0
Control: name="Speaker Playback Switch", index=0, device=0
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x00 0x00]
Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0
Amp-Out vals: [0x1f 0x1f]
Connection: 2
0x03 0x0b
Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x00 0x00]
Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0
Amp-Out vals: [0x00 0x00]
Connection: 2
0x04 0x0b
Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x00 0x00]
Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0
Amp-Out vals: [0x00 0x00]
Connection: 2
0x05 0x0b
Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0000003e: IN OUT HP Detect Trigger
Pin Default 0x0121101f: [Jack] HP Out at Ext Rear
Conn = 1/8, Color = Black
DefAssociation = 0x1, Sequence = 0xf
Pin-ctls: 0xc0: OUT HP
Unsolicited: tag=04, enabled=1
Connection: 5
0x0c 0x0d* 0x0e 0x0f 0x26
Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0000003e: IN OUT HP Detect Trigger
Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI
Conn = ATAPI, Color = Unknown
DefAssociation = 0x1, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x00:
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c 0x0d* 0x0e 0x0f 0x26
Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000003e: IN OUT HP Detect Trigger
Pin Default 0x411111f0: [N/A] Speaker at Ext Rear
Conn = 1/8, Color = Black
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x20: IN
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c 0x0d 0x0e* 0x0f 0x26
Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000003e: IN OUT HP Detect Trigger
Pin Default 0x411111f0: [N/A] Speaker at Ext Rear
Conn = 1/8, Color = Black
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x20: IN
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c 0x0d 0x0e 0x0f* 0x26
Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Control: name="Mic Boost", index=0, device=0
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000173e: IN OUT HP Detect Trigger
Vref caps: HIZ 50 GRD 80
Pin Default 0x01a19830: [Jack] Mic at Ext Rear
Conn = 1/8, Color = Pink
DefAssociation = 0x3, Sequence = 0x0
Pin-ctls: 0x24: IN VREF_80
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c* 0x0d 0x0e 0x0f 0x26
Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000173e: IN OUT HP Detect Trigger
Vref caps: HIZ 50 GRD 80
Pin Default 0x411111f0: [N/A] Speaker at Ext Rear
Conn = 1/8, Color = Black
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x20: IN VREF_HIZ
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c* 0x0d 0x0e 0x0f 0x26
Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000173e: IN OUT HP Detect Trigger
Vref caps: HIZ 50 GRD 80
Pin Default 0x411111f0: [N/A] Speaker at Ext Rear
Conn = 1/8, Color = Black
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x20: IN VREF_HIZ
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c* 0x0d 0x0e 0x0f 0x26
Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000173e: IN OUT HP Detect Trigger
Vref caps: HIZ 50 GRD 80
Pin Default 0x411111f0: [N/A] Speaker at Ext Rear
Conn = 1/8, Color = Black
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x20: IN VREF_HIZ
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c* 0x0d 0x0e 0x0f 0x26
Node 0x1c [Pin Complex] wcaps 0x400001: Stereo
Pincap 0x00000020: IN
Pin Default 0x411111f0: [N/A] Speaker at Ext Rear
Conn = 1/8, Color = Black
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x00:
Node 0x1d [Pin Complex] wcaps 0x400000: Mono
Pincap 0x00000020: IN
Pin Default 0x9983013f: [Fixed] Line In at Int ATAPI
Conn = ATAPI, Color = Unknown
DefAssociation = 0x3, Sequence = 0xf
Misc = NO_PRESENCE
Pin-ctls: 0x00:
Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital
Pincap 0x00000010: OUT
Pin Default 0x01451120: [Jack] SPDIF Out at Ext Rear
Conn = Optical, Color = Black
DefAssociation = 0x2, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x00:
Connection: 1
0x06
Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital
Pincap 0x00000020: IN
Pin Default 0x411111f0: [N/A] Speaker at Ext Rear
Conn = 1/8, Color = Black
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x00:
Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono
Processing caps: benign=0, ncoeff=17
Processing Coefficient: 0x4000
Coefficient Index: 0x0b
Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80]
Amp-Out caps: N/A
Amp-Out vals: [0x00 0x00]
Connection: 11
0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b
Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80]
Amp-Out caps: N/A
Amp-Out vals: [0x00 0x00]
Connection: 11
0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b
Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x25 [Audio Output] wcaps 0x11: Stereo
Converter: stream=0, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x00 0x00]
Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0
Amp-Out vals: [0x00 0x00]
Connection: 2
0x25 0x0b
I assume the combination Vendor Id: 0x10ec0883 + Subsystem Id: 0x1584907e is what I should be looking for right? Google gave me three resuts from alsa project's DB: http://www.google.com/search?q=0x10ec08 … e&filter=0 resulting in two more models to try:
model=laptop (I'm almost certain I tried this already)
model=lenovo
I also didn't knew about alsa-info.sh so I downloaded, ran it and got this http://www.alsa-project.org/db/?f=9f7e7 … bf5c3ad9ea including:
!!DMI Information
Manufacturer: Standard
Product Name: L50IIX
L50IIX seems to be a board made by Uniwill and used by various companies which adds even more possible models:
model=uniwill
model=uniwill-m31
model=fujitsu
model=lifebook
model=fujitsu-pi2515
model=fujitsu-xa3530
It seems I have a lot of testing to do later, I will report back the results and hopefully will provide a solution for any lost soul who comes across this post in the future. Thank you. -
[SOLVED] Asus 900 : ALSA plays sound, but microphone doesn't work
Hi,
This week I installed Arch Linux (kernel 3.1.8-1-ARCH) on my Asus 900 with HDA Intel Realtek ALC662 rev1 sound card. The wiki and google and existing forum posts have been good for most issues, but I'm stumped on ALSA configuration. So here's my first forum post for advice.
I configured sound OK following advice in the wiki and twiddling settings with alsamixer, through both the internal speaker and through the headphones, but I can't get internal or external microphone to work. I'm testing with:
arecord -d 3 junk.wav
then when I play back via:
aplay junk.wav
there is only soft static (with audible soft clicks at the start and end).
(Other files like /usr/share/sounds/alsa/Front_Center.wav play fine.)
alsamixer recognizes and reports my HDA Intel card, Realtek ALC662 rev1.
https://wiki.archlinux.org/index.php/Alsa has several suggestions of options for the snd-hda-intel module.
Different pages suggest different places, which was confusing until I discovered that they can go in any file in /etc/modprobe.d so I'm putting them in /etc/modprobe.d/snd-hda-intel.conf .
I added
options snd-hda-intel enable_msi=1
with no luck.
I have tried various model=XXX parameters (based on many suggestions found googling for Asus 900, alsa, ALC662, etc problems):
options snd-hda-intel model=auto
options snd-hda-intel model=asus
options snd-hda-intel model=asus-laptop
options snd-hda-intel model=laptop
options snd-hda-intel model=ref
options snd-hda-intel model=asus-mode1
options snd-hda-intel model=asus-mode2
options snd-hda-intel model=asus-mode8
and I do
rmomd snd-hda-intel && modprobe snd-hda-intel
after each modification, which makes a little click in the speakers as the module is restarted.
I don't know what the differences between asus-mode1, asus-mode2 etc are supposed to be - I've failed to find clear documentation about them, but AFAIK they seem to be the only valid model choices, unlike stuff like laptop, asus, etc.
I find asus-mod1 etc at https://github.com/torvalds/linux/blob/ … _realtek.c and various other webpages.
I found they vary the items presented by alsamixer but none of the 8 asus-modeN values helped me get the mic working. (The only obvious change was asus-mode8 made my headphone stop working, unlike asus-mod1 through asus-mod7.)
I looked at /var/log/kernel.log when doing the snd-hda-intel module reload for various values of model= and they all look similar to this:
snd_hda_intel 0000:00:1b.0: PCI INT A disabled
snd_hda_intel 0000:00:1b.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16
snd_hda_intel 0000:00:1b.0: irq 43 for MSI/MSI-X
snd_hda_intel 0000:00:1b.0: setting latency timer to 64
hda_code: ALC662 rev1: BIOS auto-probing
input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input24
input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input25
input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input26
(Note the emphasized line about BIOS auto-probing only comes up when not using model= or when trying various suggested model values that don't seem to be actually defined for ALC662, e.g. stuff like model=asus, etc.)
Here's a list of alsamixer settings I used for various models. (Note I also tried muting and raising various higher boost levels, and when there's an Internal Mic or F-Mic item I tried various combinations. The following are just what seem the most plausible settings to me.)
model=asus
model=asus-auto
model=auto
alsamixer presents 14 items (boosts are paired) :
Master 100 (dB gain: 0.00)
Headphone 00
Speaker 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
Mic Boost 22 (dB gain: 10.00, 10.00)
Mic Boost 22 (dB gain: 10.00, 10.00)
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Auto-Mute Mode Enabled
Digital 25 (dB gain: 0.50, 0.50)
Internal Mic 56 (dB gain: 0.00, 0.00)
Internal Mic Boost 22 (dB gain: 10.00, 10.00)
Internal Mic Boost 22 (dB gain: 10.00, 10.00)
model=asus-mode1
alsamixer [All] presents 8 items:
Master 100 (dB gain: 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
speaker & headphones work, neither mic works
model=asus-mode2
alsamixer [All] presents 11 items:
Master 100 (dB gain: 0.00)
Headphone 00
Speaker 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
F-Mic 56 (dB gain: 0.00, 0.00)
model=asus-mode3
alsamixer [All] presents 9 items:
Master 100 (dB gain: 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
F-Mic 56 (dB gain: 0.00, 0.00)
model=asus-mode4
alsamixer presents 9 items
Master 100 (dB gain: 0.00, 0.00)
Headphone 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
model=asus-mode5
alsamixer presents same 9 items as asus-mode4
model=asus-mode6
alsamixer [All] presents 9 items same as asus-mode3
model=asus-mode7
alsamixer [All] presents 12 items:
Master 100 (dB gain: 0.00, 0.00)
Headphone1 00
Headphone2 00
Speaker 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
IntMic 56 (dB gain: 0.50, 0.50)
model=asus-mode8
alsamixer [All] presents 11 items:
Master 100 (dB gain: 0.00, 0.00)
Headphone1 00
Headphone2 00
Speaker 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
No headphone output for asus-mode8!?
I'm not interested in doing anything fancy (e.g. mixing multiple sources or whatever), I just want basic mic functionality.
Finally here's some info requested from the archlinux.org/index.php/Alsa:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
$ lsmod|grep snd
snd_hda_intel 19325 0
snd_hda_codec_realtek 211044 1
snd_hda_codec 69829 2 snd_hda_intel,snd_hda_codec_realtek
snd_hwdep 4942 1 snd_hda_codec
snd_pcm 60207 2 snd_hda_intel,snd_hda_codec
snd_timer 15438 1 snd_pcm
snd 43817 6 snd_hda_intel,snd_hda_codec_realtek,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
soundcore 5018 1 snd
snd_page_alloc 5869 2 snd_hda_intel,snd_pcm
$ ls -l /dev/snd
total 0
drwxr-xr-x 2 root root 60 Jan 11 20:43 by-path
crw-rw---T 1 root audio 116, 5 Jan 11 20:43 controlC0
crw-rw---T 1 root audio 116, 4 Jan 11 20:43 hwC0D0
crw-rw---T 1 root audio 116, 3 Jan 11 21:01 pcmC0D0c
crw-rw---T 1 root audio 116, 2 Jan 11 21:01 pcmC0D0p
crw-rw---T 1 root audio 116, 1 Jan 11 15:23 seq
crw-rw---T 1 root audio 116, 33 Jan 11 15:23 timer
Has anyone got sound recording working on an Asus 900 with HDA Intel Realtek ALC662 rev1 ...?
Thanks for any help! (And for feedback whether this post gave too much info or not enough, or other suggestions for posting such requests. Sorry if this was too long, but it seemed better to err on the side of too much info...)
Last edited by goulo (2012-01-13 18:30:13)Hello,
I don't have your soundcard but maybe the following information is helpful, though:
Install the linux-docs package, then you have some information about your soundcard drivers in "/usr/src/linux-*/Documentation/sound/alsa/". Exspecially "HD-Audio.txt" and "HD-Audio-Models.txt" might be important. In the Models file it is stated that just the "asus-mode*" are legitimate model parameters for your soundcard and in the HD-Autio.txt file it is more or less said that trial and error the modules is the way to go to find the "correct module". Further there is a section
Capture Problems
~~~~~~~~~~~~~~~~
The capture problems are often because of missing setups of mixers.
Thus, before submitting a bug report, make sure that you set up the
mixer correctly. For example, both "Capture Volume" and "Capture
Switch" have to be set properly in addition to the right "Capture
Source" or "Input Source" selection. Some devices have "Mic Boost"
volume or switch.
When the PCM device is opened via "default" PCM (without pulse-audio
plugin), you'll likely have "Digital Capture Volume" control as well.
This is provided for the extra gain/attenuation of the signal in
software, especially for the inputs without the hardware volume
control such as digital microphones. Unless really needed, this
should be set to exactly 50%, corresponding to 0dB -- neither extra
gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM,
this control will have no influence, though.
It's known that some codecs / devices have fairly bad analog circuits,
and the recorded sound contains a certain DC-offset. This is no bug
of the driver.
So check your mixer, here and here are nice tutorials.
Greetings
matse
Last edited by matse (2012-01-12 16:30:09) -
Blue Yeti USB microphone not working in ALSA/PulseAudio
Hi all,
I recently bought a Blue Yeti USB microphone. It is supposed to require no drivers for Windows and OS X, and so far, it works well under Windows, but I cannot seem to get it to function on Arch Linux. As fair warning, although I am relatively experienced with Linux, I don't know much about the internals of ALSA/PA --- up until now, it has just "always worked", so this is my first real foray into the area. I've tried for many days, with no success, to get this thing working.
So, let's start off first with my configuration.
I have a few audio devices:
An on-board audio solution that I don't typically use. For testing, I have connected an old, crappy pair of headphones, but normally I keep this disabled. It has one capture device. (Analog stereo [or 5.1/7.1/etc] duplex)
An external DAC that is connected via USB and is output-only. So, no capture devices on that. (Digital/IEC958 output)
The Blue Yeti itself, which is connected via USB and supports output (it has an internal DAC) as well as input. So, one capture device here. (Digital stereo output and analog stereo input???)
The HDMI-out stuff for nvidia, which I never use
Here is the output of "arecord -l":
**** List of CAPTURE Hardware Devices ****
card 1: Microphone [Yeti Stereo Microphone], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 2: ALC892 Alt Analog [ALC892 Alt Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
Here is the the output of "arecord -L":
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default
Default ALSA Output (currently PulseAudio Sound Server)
sysdefault:CARD=Microphone
Yeti Stereo Microphone, USB Audio
Default Audio Device
front:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
Front speakers
surround40:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
4.0 Surround output to Front and Rear speakers
surround41:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
IEC958 (S/PDIF) Digital Audio Output
sysdefault:CARD=PCH
HDA Intel PCH, ALC892 Analog
Default Audio Device
front:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
Front speakers
surround40:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
Here is the output of "aplay -l" (for completeness):
**** List of PLAYBACK Hardware Devices ****
card 0: Audio [USB2.0 High-Speed True HD Audio], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: Microphone [Yeti Stereo Microphone], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3]
Subdevices: 1/1
Subdevice #0: subdevice #0
If the output of "aplay -L" is needed, let me know, but it's not all that different from arecord -L.
Here is my /etc/asound.conf:
defaults.pcm.rate_converter "samplerate_best"
pcm.!default {
type pulse
fallback "sysdefault"
hint {
show on
description "Default ALSA Output (currently PulseAudio Sound Server)"
ctl.!default {
type pulse
fallback "sysdefault"
# vim:set ft=alsaconf:
I don't remember where the bulk of that came from; maybe it was installed like that or maybe I put it in there. Here is another useful command showing non-commented-out/non-blank lines in the pulseaudio conf:
$ egrep -v '^;|^#|^$' /etc/pulse/*.conf
/etc/pulse/client.conf:autospawn=no
/etc/pulse/daemon.conf:exit-idle-time=0
/etc/pulse/daemon.conf:default-sample-rate = 192000
So, there's not much I've done there. Here's the contents of /proc/asound/cards:
0 [Audio ]: USB-Audio - USB2.0 High-Speed True HD Audio
CMEDIA USB2.0 High-Speed True HD Audio at usb-0000:00:1d.0-1.7, high speed
1 [Microphone ]: USB-Audio - Yeti Stereo Microphone
Blue Microphones Yeti Stereo Microphone at usb-0000:00:1d.0-1.8, full speed
2 [PCH ]: HDA-Intel - HDA Intel PCH
HDA Intel PCH at 0xf7530000 irq 52
3 [NVidia ]: HDA-Intel - HDA NVidia
HDA NVidia at 0xf7080000 irq 17
And here's the contents of /proc/asound/modules:
0 snd_usb_audio
1 snd_usb_audio
2 snd_hda_intel
3 snd_hda_intel
Now, here is the output of "pactl list sources" (the Blue Yeti is last in the list):
Source #3
State: RUNNING
Name: alsa_input.pci-0000_00_1b.0.analog-stereo
Description: Built-in Audio Analog Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 192000Hz
Channel Map: front-left,front-right
Owner Module: 7
Mute: yes
Volume: front-left: 17972 / 27% / -33.71 dB, front-right: 17972 / 27% / -33.71 dB
balance 0.00
Base Volume: 6554 / 10% / -60.00 dB
Monitor of Sink: n/a
Latency: 610 usec, configured 100000 usec
Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
Properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC892 Analog"
alsa.id = "ALC892 Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "2"
alsa.card_name = "HDA Intel PCH"
alsa.long_card_name = "HDA Intel PCH at 0xf7530000 irq 52"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card2"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.id = "1e20"
device.product.name = "7 Series/C210 Series Chipset Family High Definition Audio Controller"
device.form_factor = "internal"
device.string = "front:2"
device.buffering.buffer_size = "76800"
device.buffering.fragment_size = "19200"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Built-in Audio Analog Stereo"
alsa.mixer_name = "Realtek ALC892"
alsa.components = "HDA:10ec0892,10438436,00100302"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
Ports:
analog-input-front-mic: Front Microphone (priority: 8500, not available)
analog-input-rear-mic: Rear Microphone (priority: 8200, available)
analog-input-linein: Line In (priority: 8100, not available)
Active Port: analog-input-rear-mic
Formats:
pcm
Source #11
State: RUNNING
Name: alsa_output.pci-0000_00_1b.0.analog-stereo.monitor
Description: Monitor of Built-in Audio Analog Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 192000Hz
Channel Map: front-left,front-right
Owner Module: 7
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
Base Volume: 65536 / 100% / 0.00 dB
Monitor of Sink: alsa_output.pci-0000_00_1b.0.analog-stereo
Latency: 0 usec, configured 100000 usec
Flags: DECIBEL_VOLUME LATENCY
Properties:
device.description = "Monitor of Built-in Audio Analog Stereo"
device.class = "monitor"
alsa.card = "2"
alsa.card_name = "HDA Intel PCH"
alsa.long_card_name = "HDA Intel PCH at 0xf7530000 irq 52"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card2"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.id = "1e20"
device.product.name = "7 Series/C210 Series Chipset Family High Definition Audio Controller"
device.form_factor = "internal"
device.string = "2"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
Formats:
pcm
Source #15
State: RUNNING
Name: alsa_output.usb-CMEDIA_USB2.0_High-Speed_True_HD_Audio-00-Audio.iec958-stereo.monitor
Description: Monitor of USB2.0 High-Speed True HD Audio Digital Stereo (IEC958)
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 192000Hz
Channel Map: front-left,front-right
Owner Module: 8
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
Base Volume: 65536 / 100% / 0.00 dB
Monitor of Sink: alsa_output.usb-CMEDIA_USB2.0_High-Speed_True_HD_Audio-00-Audio.iec958-stereo
Latency: 0 usec, configured 100000 usec
Flags: DECIBEL_VOLUME LATENCY
Properties:
device.description = "Monitor of USB2.0 High-Speed True HD Audio Digital Stereo (IEC958)"
device.class = "monitor"
alsa.card = "0"
alsa.card_name = "USB2.0 High-Speed True HD Audio"
alsa.long_card_name = "CMEDIA USB2.0 High-Speed True HD Audio at usb-0000:00:1d.0-1.7, high speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:1d.0-usb-0:1.7:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:1d.0/usb6/6-1/6-1.7/6-1.7:1.0/sound/card0"
udev.id = "usb-CMEDIA_USB2.0_High-Speed_True_HD_Audio-00-Audio"
device.bus = "usb"
device.vendor.id = "0d8c"
device.vendor.name = "C-Media Electronics, Inc."
device.product.id = "0319"
device.product.name = "USB2.0 High-Speed True HD Audio"
device.serial = "CMEDIA_USB2.0_High-Speed_True_HD_Audio"
device.string = "0"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
Formats:
pcm
Source #19
State: RUNNING
Name: alsa_output.usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone.iec958-stereo.monitor
Description: Monitor of Integrated Rate Matching Hub Digital Stereo (IEC958)
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 48000Hz
Channel Map: front-left,front-right
Owner Module: 9
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
Base Volume: 65536 / 100% / 0.00 dB
Monitor of Sink: alsa_output.usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone.iec958-stereo
Latency: 0 usec, configured 100000 usec
Flags: DECIBEL_VOLUME LATENCY
Properties:
device.description = "Monitor of Integrated Rate Matching Hub Digital Stereo (IEC958)"
device.class = "monitor"
alsa.card = "1"
alsa.card_name = "Yeti Stereo Microphone"
alsa.long_card_name = "Blue Microphones Yeti Stereo Microphone at usb-0000:00:1d.0-1.8, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:1d.0-usb-0:1.8:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:1d.0/usb6/6-1/6-1.8/6-1.8:1.0/sound/card1"
udev.id = "usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone"
device.bus = "usb"
device.vendor.id = "b58e"
device.vendor.name = "Intel Corp."
device.product.id = "9e84"
device.product.name = "Integrated Rate Matching Hub"
device.serial = "Blue_Microphones_Yeti_Stereo_Microphone_REV8"
device.form_factor = "microphone"
device.string = "1"
module-udev-detect.discovered = "1"
device.icon_name = "audio-input-microphone-usb"
Formats:
pcm
Source #23
State: RUNNING
Name: alsa_input.usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone.analog-stereo
Description: Integrated Rate Matching Hub Analog Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 48000Hz
Channel Map: front-left,front-right
Owner Module: 9
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
Base Volume: 17775 / 27% / -34.00 dB
Monitor of Sink: n/a
Latency: 949811526 usec, configured 100000 usec
Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
Properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "Yeti Stereo Microphone"
alsa.long_card_name = "Blue Microphones Yeti Stereo Microphone at usb-0000:00:1d.0-1.8, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:1d.0-usb-0:1.8:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:1d.0/usb6/6-1/6-1.8/6-1.8:1.0/sound/card1"
udev.id = "usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone"
device.bus = "usb"
device.vendor.id = "b58e"
device.vendor.name = "Intel Corp."
device.product.id = "9e84"
device.product.name = "Integrated Rate Matching Hub"
device.serial = "Blue_Microphones_Yeti_Stereo_Microphone_REV8"
device.form_factor = "microphone"
device.string = "front:1"
device.buffering.buffer_size = "19200"
device.buffering.fragment_size = "4800"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Integrated Rate Matching Hub Analog Stereo"
alsa.mixer_name = "USB Mixer"
alsa.components = "USBb58e:9e84"
module-udev-detect.discovered = "1"
device.icon_name = "audio-input-microphone-usb"
Ports:
analog-input-mic: Microphone (priority: 8700)
Active Port: analog-input-mic
Formats:
pcm
Honestly, I think that's about all of the configuration info I can give. Here are some screenshots of pavucontrol, though, in case you are not yet sated: http://imgur.com/a/ngyLl
Now, on to the debugging stuff. First: everything except Blue Yeti recording works. The external DAC works, the on-board audio (both input and output) work, using the DAC in the Blue Yeti to play music/monitor the microphone works (so it works as an output). The only thing that doesn't work is actual recording from the Blue Yeti itself.
When I try to record from the Blue Yeti with parecord -v, I get very strange behavior. Essentially, the "latency" of the microphone measured is exactly the same as the time it has been recording. That is, if it has been recording for 5 sec, then the latency is 5000000 usec. Also, if I try to record something in Audacity, I get no result except for a message box that says "Latency Correction has caused the recorded audio to be hidden before zero. Audacity has brought it back to start at zero." Also, pavucontrol doesn't show an input volume bar for the Blue Yeti, as in the above screenshots, whereas it normally will for microphones that are getting input. The Yeti is not muted, either, so it should be picking up ambient fan noise and such.
Does anyone have any ideas? Is there some configuration error I've made? Typically, this sort of stuff simply works out of the box for me, so I am somewhat stuck on what to do. Could it be a driver-level bug?I'm not sure what you mean by "mixed with audio in Alsamixer". Can you clarify?
With that said, my problem did mysteriously fix itself after some update (presumably to alsa). Unfortunately, I noticed it in passing one day while I was changing the volume of something in pavucontrol, so I don't remember the exact update that caused it to work. But yes, my current setup does work now: I can listen to audio through my DAC and use the Yeti for recording. However, I have found that use of the Yeti crashes Google Hangouts, although it works perfectly in every other program I've tried.
As documentation, here is my current working setup (it is very similar to the above): in PulseAudio, the DAC is set to be a Digital Stereo (IEC958) output. The Yeti (called an 'Integrated Rate Matching Hub' in pavucontrol) is set to be an Analog Stereo Input. In the input devices section, I have the Yeti's volume set to around 58% (-14.31 dB), which is past the base volume a fair ways but produces decent levels for recording. The DAC's volume is set to 100% (0dB gain, the DAC takes care of the level). In alsamixer, I have the mic level for the Yeti set to 40.
Edit: This is a long-time-later edit, but let me say that when you have multiple USB audio devices, what USB hub and in what order they are connected seems to matter a great deal. If you can't get your Blue Yeti to work, try moving it to a different set of USB ports (controlled by a different controller) on your motherboard. This magically fixes the issue for me.
Last edited by rwiggins (2014-12-14 00:26:39) -
Sound problems, ALSA can't find sound card at boot
Sometimes, my sound randomly drops out, and I can't get it back until I reboot. Even when I reboot, 1/20 times, it will not be back.
When I watch stuff on youtube, sometimes the sound just drops out. Nothing gets muted. My audio player can't produce sound.
If I do
lsmod | grep snd
I'll see the some output.
When I reboot and try to play music from Quodlibet, I'll get an error like this:
Output Error
GStreamer output pipeline could not be initialized. The pipeline might be invalid, or the device may be in use. Check the player preferences.
"lsmod | grep snd" will have NO output and when I try to start alsamixer:
[xz@gestalt ~]$ alsamixer
cannot open mixer: No such file or directory
I can't figure out what's going on, and googling has just led me to various not-completely-related posts on this board... what could be the problem?
EDIT: A few more things to note--
I have alsa in my daemons list in rc.conf
I'm part of the "audio" groups.
soundcore is not blacklisted
Last edited by brokengestalt (2012-08-29 09:10:59)If I try
[xz@gestalt ~]$ lspci -nn
00:00.0 RAM memory [0500]: NVIDIA Corporation MCP67 Memory Controller [10de:0547] (rev a2)
00:01.0 ISA bridge [0601]: NVIDIA Corporation MCP67 ISA Bridge [10de:0548] (rev a2)
00:01.1 SMBus [0c05]: NVIDIA Corporation MCP67 SMBus [10de:0542] (rev a2)
00:01.2 RAM memory [0500]: NVIDIA Corporation MCP67 Memory Controller [10de:0541] (rev a2)
00:01.3 Co-processor [0b40]: NVIDIA Corporation MCP67 Co-processor [10de:0543] (rev a2)
00:02.0 USB controller [0c03]: NVIDIA Corporation MCP67 OHCI USB 1.1 Controller [10de:055e] (rev a2)
00:02.1 USB controller [0c03]: NVIDIA Corporation MCP67 EHCI USB 2.0 Controller [10de:055f] (rev a2)
00:04.0 USB controller [0c03]: NVIDIA Corporation MCP67 OHCI USB 1.1 Controller [10de:055e] (rev a2)
00:04.1 USB controller [0c03]: NVIDIA Corporation MCP67 EHCI USB 2.0 Controller [10de:055f] (rev a2)
00:06.0 IDE interface [0101]: NVIDIA Corporation MCP67 IDE Controller [10de:0560] (rev a1)
00:08.0 PCI bridge [0604]: NVIDIA Corporation MCP67 PCI Bridge [10de:0561] (rev a2)
00:09.0 IDE interface [0101]: NVIDIA Corporation MCP67 AHCI Controller [10de:0550] (rev a2)
00:0a.0 Ethernet controller [0200]: NVIDIA Corporation MCP67 Ethernet [10de:054c] (rev a2)
00:0c.0 PCI bridge [0604]: NVIDIA Corporation MCP67 PCI Express Bridge [10de:0563] (rev a2)
00:0d.0 PCI bridge [0604]: NVIDIA Corporation MCP67 PCI Express Bridge [10de:0563] (rev a2)
00:12.0 VGA compatible controller [0300]: NVIDIA Corporation C67 [GeForce 7150M / nForce 630M] [10de:0531] (rev a2)
00:18.0 Host bridge [0600]: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration [1022:1100]
00:18.1 Host bridge [0600]: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map [1022:1101]
00:18.2 Host bridge [0600]: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller [1022:1102]
00:18.3 Host bridge [0600]: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control [1022:1103]
02:05.0 FireWire (IEEE 1394) [0c00]: Ricoh Co Ltd R5C832 IEEE 1394 Controller [1180:0832] (rev 05)
02:05.1 SD Host controller [0805]: Ricoh Co Ltd R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter [1180:0822] (rev 22)
02:05.2 System peripheral [0880]: Ricoh Co Ltd R5C592 Memory Stick Bus Host Adapter [1180:0592] (rev 12)
02:05.3 System peripheral [0880]: Ricoh Co Ltd xD-Picture Card Controller [1180:0852] (rev 12)
03:00.0 Ethernet controller [0200]: Atheros Communications Inc. AR242x / AR542x Wireless Network Adapter (PCI-Express) [168c:001c] (rev 01)
Audio devices aren't even mentioned. I noticed when I tried to boot up that I get an error like "sound card not detected." Why would arch suddenly not be able to find a soundcard? It looks like it almost has nothing to do with my ALSA configuration, and has more to do with my sound card not being found at all in the first place.... -
Alsa issue on a Dell Studio 15
I'm actually kind of surprised that I've resorted to making a forum post on this topic, but really I'm just flat out of ideas.
I just got this laptop yesterday, and immediately reformatted it and installed arch. Everything's working (X, openbox, network to an extent, etc), with the exception of alsa. For some reason, aplay isn't working, or rather I don't hear any sound when I type the command "sudo aplay /usr/share/sounds/alsa/Front_Center.wav" (sudo just to make sure it wasn't some queer permissions problem). Every channel in alsamixer is unmuted, and I even kept on increasing the volume up until everything was at 100%, still to no avail. I followed the standard procedure in the arch beginners guide, so alsa-utils, alsa-oss, and alsa-lib are all installed.
I tried running alsaconf as root, set up every card that it discovered (didn't probe for legacy chips, though), which, for reference are...
*hda-intel Intel Corporation 82801I (ICH9 Family) HD Audio Controller)
*hda-intel ATI Technologies Inc R700 Audio Device [Radeon HD 4000 Series]
*legacy Probe Legacy ISA (non-PnP) chips
and said yes to it modifying any modprobe.conf files (this is all after I had tried doing it manually as described on the alsa wiki page, mind you).
So, can anyone point me in the right direction? I can't hear anything through the speakers, or headphones if I plug them into either of the 3.5mm output jacks (yeah, there's two on this machine). I'd really like to get mpd+sonata running on this thing next, but that would be kind of pointless if I can't hear anything.Thanks, that got my speaker and one of the two headphone jacks working.
For anyone else who's having this problem, just add
options snd-hda-intel model=dell-m6
to your modprobe file(s)
(just to make sure, I added it to these three)
/etc/modprobe.d/alsa-base.conf
/etc/modprobe.d/modprobe.conf
/etc/modprobe.conf
Now to get mpd, sonata, wireless, bluetooth, camera, mics working. It's going to be a fun weekend ;-) -
ALSA - Master volume no longer having an effect on headphone volume
I use ALSA and have both my 'front' (speakers) and 'headphone' audio volume set to max. Then I control the two by adjusting the master volume.
Since my last system update this worked fine however now when I change the master volume it only affects the sound coming from my speakers and my headphones are constantly at max unless I specifically change the headphone volume in 'alsamixer'. This is inconvenient since I have the master volume controlled by hotkeys.
I know I can work around this and hotkey my controls to change both the master and headphone volumes but this leaves room for things getting out of sync. Is there a way to revert back to the old method?Hi-
"Essentially" a clone and a clone are not the same. There are alot of things that don't get copied when you simply drag folders. Pathways to aliases are one of them.
Use SuperDuper! (the best IMHO) or Carbon Copy Cloner to get real bootable clones of your system and files. -
[Solved]No sound anymore (ALSA/PulseAudio)
Ok so I have no idea why I have no sound anymore, I didn't install anything...
Anyway, that's what I've tried:
-Check if mixer are all up in alsamixer
-Try to play a bit with pavucontrol
-Reboot
-Rerun pulseaudio with verbose
-Mute and unmute with physical button (I don't know why but I used to get this problem and it used to be solved just doing this)
-Googgling and searching this forum
Here are some more info
Pulseaudio logs:
└┼─$─┤▶ pulseaudio -k; pulseaudio -vvvvv
E: [pulseaudio] main.c: Failed to kill daemon: No such process
I: [pulseaudio] main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: [pulseaudio] main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: [pulseaudio] core-rtclock.c: Timer slack is set to 50 us.
D: [pulseaudio] core-util.c: RealtimeKit worked.
I: [pulseaudio] core-util.c: Successfully gained nice level -11.
I: [pulseaudio] main.c: This is PulseAudio 5.0
D: [pulseaudio] main.c: Compilation host: x86_64-unknown-linux-gnu
D: [pulseaudio] main.c: Compilation CFLAGS: -march=x86-64 -mtune=generic -O2 -pipe -fstack-protector --param=ssp-buffer-size=4 -Wall -W -Wextra -Wno-long-long -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing -Wwrite-strings -Wno-unused-parameter -ffast-math -fno-common -fdiagnostics-show-option
D: [pulseaudio] main.c: Running on host: Linux x86_64 3.14.1-1-ARCH #1 SMP PREEMPT Mon Apr 14 20:40:47 CEST 2014
D: [pulseaudio] main.c: Found 4 CPUs.
I: [pulseaudio] main.c: Page size is 4096 bytes
D: [pulseaudio] main.c: Compiled with Valgrind support: no
D: [pulseaudio] main.c: Running in valgrind mode: no
D: [pulseaudio] main.c: Running in VM: no
D: [pulseaudio] main.c: Optimized build: yes
D: [pulseaudio] main.c: FASTPATH defined, only fast path asserts disabled.
I: [pulseaudio] main.c: Machine ID is a3af3988434b49f3b8af2ab66eba685b.
I: [pulseaudio] main.c: Session ID is c1.
I: [pulseaudio] main.c: Using runtime directory /run/user/1001/pulse.
I: [pulseaudio] main.c: Using state directory /home/unkn0wn/.config/pulse.
I: [pulseaudio] main.c: Using modules directory /usr/lib/pulse-5.0/modules.
I: [pulseaudio] main.c: Running in system mode: no
I: [pulseaudio] main.c: Fresh high-resolution timers available! Bon appetit!
D: [pulseaudio] memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472
I: [pulseaudio] cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2
I: [pulseaudio] svolume_mmx.c: Initialising MMX optimized volume functions.
I: [pulseaudio] remap_mmx.c: Initialising MMX optimized remappers.
I: [pulseaudio] svolume_sse.c: Initialising SSE2 optimized volume functions.
I: [pulseaudio] remap_sse.c: Initialising SSE2 optimized remappers.
I: [pulseaudio] sconv_sse.c: Initialising SSE2 optimized conversions.
I: [pulseaudio] svolume_orc.c: Initialising ORC optimized volume functions.
D: [pulseaudio] database-tdb.c: Opened TDB database '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-device-volumes.tdb'
I: [pulseaudio] module-device-restore.c: Successfully opened database file '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-device-volumes'.
I: [pulseaudio] module.c: Loaded "module-device-restore" (index: #0; argument: "").
D: [pulseaudio] database-tdb.c: Opened TDB database '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-stream-volumes.tdb'
I: [pulseaudio] module-stream-restore.c: Successfully opened database file '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-stream-volumes'.
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1 added for object /org/pulseaudio/stream_restore1
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry0
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry1
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry2
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry3
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry4
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry5
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry6
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry7
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry8
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry9
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry10
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry11
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry12
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry13
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry14
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry15
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry16
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry17
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry18
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry19
I: [pulseaudio] module.c: Loaded "module-stream-restore" (index: #1; argument: "").
D: [pulseaudio] database-tdb.c: Opened TDB database '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-card-database.tdb'
I: [pulseaudio] module-card-restore.c: Successfully opened database file '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-card-database'.
I: [pulseaudio] module.c: Loaded "module-card-restore" (index: #2; argument: "").
I: [pulseaudio] module.c: Loaded "module-augment-properties" (index: #3; argument: "").
I: [pulseaudio] module.c: Loaded "module-switch-on-port-available" (index: #4; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-udev-detect.so': success
D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes
D: [pulseaudio] module-udev-detect.c: Ignoring /devices/platform/thinkpad_acpi/sound/card29, because marked so.
I: [pulseaudio] module-udev-detect.c: Found 1 cards.
I: [pulseaudio] module.c: Loaded "module-udev-detect" (index: #5; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-jackdbus-detect.so': success
D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus session bus c9a6328a5bfaef44136d134253502922 as :1.32
D: [pulseaudio] module-jackdbus-detect.c: jackdbus isn't running.
I: [pulseaudio] module.c: Loaded "module-jackdbus-detect" (index: #6; argument: "channels=2").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-bluetooth-policy.so': success
I: [pulseaudio] module.c: Loaded "module-bluetooth-policy" (index: #7; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-bluetooth-discover.so': success
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-bluez5-discover.so': success
D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus system bus 54d6e91cb1d72b925a540cf453502907 as :1.40
I: [pulseaudio] module.c: Loaded "module-bluez5-discover" (index: #9; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-bluez4-discover.so': failure
I: [pulseaudio] module.c: Loaded "module-bluetooth-discover" (index: #8; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-esound-protocol-unix.so': success
I: [pulseaudio] module.c: Loaded "module-esound-protocol-unix" (index: #10; argument: "").
I: [pulseaudio] module.c: Loaded "module-native-protocol-unix" (index: #11; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-gconf.so': success
I: [pulseaudio] module.c: Loaded "module-gconf" (index: #12; argument: "").
I: [pulseaudio] module-default-device-restore.c: Saved default sink 'auto_null' not existent, not restoring default sink setting.
I: [pulseaudio] module-default-device-restore.c: Saved default source 'auto_null.monitor' not existent, not restoring default source setting.
I: [pulseaudio] module.c: Loaded "module-default-device-restore" (index: #13; argument: "").
I: [pulseaudio] module.c: Loaded "module-rescue-streams" (index: #14; argument: "").
D: [pulseaudio] module-always-sink.c: Autoloading null-sink as no other sinks detected.
I: [pulseaudio] module-device-restore.c: Restoring volume for sink auto_null: front-left: 99957 / 153%, front-right: 99957 / 153%
I: [pulseaudio] module-device-restore.c: Restoring mute state for sink auto_null.
I: [pulseaudio] sink.c: Created sink 0 "auto_null" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: [pulseaudio] sink.c: device.description = "Dummy Output"
I: [pulseaudio] sink.c: device.class = "abstract"
I: [pulseaudio] sink.c: device.icon_name = "audio-card"
D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
I: [pulseaudio] module-device-restore.c: Restoring mute state for source auto_null.monitor.
I: [pulseaudio] source.c: Created source 0 "auto_null.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: [pulseaudio] source.c: device.description = "Monitor of Dummy Output"
I: [pulseaudio] source.c: device.class = "monitor"
I: [pulseaudio] source.c: device.icon_name = "audio-input-microphone"
D: [null-sink] module-null-sink.c: Thread starting up
D: [pulseaudio] module-device-restore.c: Could not set format on sink auto_null
I: [pulseaudio] module.c: Loaded "module-null-sink" (index: #16; argument: "sink_name=auto_null sink_properties='device.description="Dummy Output"'").
I: [pulseaudio] module.c: Loaded "module-always-sink" (index: #15; argument: "").
I: [pulseaudio] module.c: Loaded "module-intended-roles" (index: #17; argument: "").
D: [pulseaudio] module-suspend-on-idle.c: Sink auto_null becomes idle, timeout in 5 seconds.
I: [pulseaudio] module.c: Loaded "module-suspend-on-idle" (index: #18; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-console-kit.so': success
I: [pulseaudio] module.c: Loaded "module-console-kit" (index: #19; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-systemd-login.so': success
I: [pulseaudio] client.c: Created 0 "Login Session c1"
D: [pulseaudio] module-systemd-login.c: Added new session c1
I: [pulseaudio] module.c: Loaded "module-systemd-login" (index: #20; argument: "").
I: [pulseaudio] module.c: Loaded "module-position-event-sounds" (index: #21; argument: "").
D: [pulseaudio] module-role-cork.c: Using role 'phone' as trigger role.
D: [pulseaudio] module-role-cork.c: Using roles 'music' and 'video' as cork roles.
I: [pulseaudio] module.c: Loaded "module-role-cork" (index: #22; argument: "").
I: [pulseaudio] module.c: Loaded "module-filter-heuristics" (index: #23; argument: "").
I: [pulseaudio] module.c: Loaded "module-filter-apply" (index: #24; argument: "").
D: [pulseaudio] main.c: Got org.PulseAudio1!
D: [pulseaudio] main.c: Got org.pulseaudio.Server!
I: [pulseaudio] main.c: Daemon startup complete.
E: [pulseaudio] bluez5-util.c: GetManagedObjects() failed: org.freedesktop.DBus.Error.ServiceUnknown: The name org.bluez was not provided by any .service files
I: [pulseaudio] module-suspend-on-idle.c: Sink auto_null idle f
└┼─$─┤▶ loginctl session-status c1
c1 - unkn0wn (1001)
Since: Thu 2014-04-17 21:18:58 CEST; 16min ago
Leader: 1048 (slim)
Seat: seat0; vc7
Display: :0.0
Remote: user root
Service: slim; type x11; class user
State: active
Unit: session-c1.scope
|-1048 /usr/bin/slim -nodaemon
|-2206 /usr/bin/openbox --startup /usr/lib/openbox/openbox-autostart OPENBOX
|-2214 dbus-launch --sh-syntax --exit-with-session
|-2215 /usr/bin/dbus-daemon --fork --print-pid 5 --print-address 7 --session
|-2227 tint2
|-2230 xcompmgr -CfF
|-2231 volumeicon
|-2236 conky -q
|-2238 /usr/lib/at-spi2-core/at-spi-bus-launcher
|-2252 /usr/bin/dbus-daemon --config-file=/etc/at-spi2/accessibility.conf --no...
|-2255 /usr/lib/at-spi2-core/at-spi2-registryd --use-gnome-session
|-2259 /usr/lib/gvfs/gvfsd
|-2263 /usr/lib/gvfs/gvfsd-fuse /run/user/1001/gvfs -f -o big_writes
|-2278 /usr/lib32/skype/skype
|-2320 /usr/lib/xfce4/notifyd/xfce4-notifyd
|-2322 /usr/lib/xfce4/xfconf/xfconfd
|-2376 firefox
|-2416 /usr/lib/virtualbox/VBoxXPCOMIPCD
-I see the mixer going up and down in pavucontrol if I play some music
└┼─$─┤▶ loginctl session-status c1
c1 - unkn0wn (1001)
Since: Thu 2014-04-17 21:18:58 CEST; 16min ago
Leader: 1048 (slim)
Seat: seat0; vc7
Display: :0.0
Remote: user root
Service: slim; type x11; class user
State: active
Unit: session-c1.scope
|-1048 /usr/bin/slim -nodaemon
|-2206 /usr/bin/openbox --startup /usr/lib/openbox/openbox-autostart OPENBOX
|-2214 dbus-launch --sh-syntax --exit-with-session
|-2215 /usr/bin/dbus-daemon --fork --print-pid 5 --print-address 7 --session
|-2227 tint2
|-2230 xcompmgr -CfF
|-2231 volumeicon
|-2236 conky -q
|-2238 /usr/lib/at-spi2-core/at-spi-bus-launcher
|-2252 /usr/bin/dbus-daemon --config-file=/etc/at-spi2/accessibility.conf --no...
|-2255 /usr/lib/at-spi2-core/at-spi2-registryd --use-gnome-session
|-2259 /usr/lib/gvfs/gvfsd
|-2263 /usr/lib/gvfs/gvfsd-fuse /run/user/1001/gvfs -f -o big_writes
|-2278 /usr/lib32/skype/skype
|-2320 /usr/lib/xfce4/notifyd/xfce4-notifyd
|-2322 /usr/lib/xfce4/xfconf/xfconfd
|-2376 firefox
|-2416 /usr/lib/virtualbox/VBoxXPCOMIPCD
└┼─$─┤▶ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: CX20590 Analog [CX20590 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2]
Subdevices: 1/1
Subdevice #0: subdevice #0
└┼─$─┤▶ aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default
Default ALSA Output (currently PulseAudio Sound Server)
sysdefault:CARD=PCH
HDA Intel PCH, CX20590 Analog
Default Audio Device
front:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
Front speakers
surround40:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
hdmi:CARD=PCH,DEV=0
HDA Intel PCH, HDMI 0
HDMI Audio Output
hdmi:CARD=PCH,DEV=1
HDA Intel PCH, HDMI 1
HDMI Audio Output
hdmi:CARD=PCH,DEV=2
HDA Intel PCH, HDMI 2
HDMI Audio Output
Don't really know what else to provide...
About my setup it's a thinkpad t420 not plugged in any hdmi screen or simething like that
Thank you
Last edited by Truc (2014-04-19 13:15:28)└┼─$─┤▶ fuser -v /dev/snd/*
USER PID ACCESS COMMAND
/dev/snd/controlC0: unkn0wn 2231 F.... volumeicon
└┼─$─┤▶ speaker-test -c 2 -t wav -D plughw:0,0
speaker-test 1.0.27.2
Playback device is plughw:0,0
Stream parameters are 48000Hz, S16_LE, 2 channels
WAV file(s)
Playback open error: -16,Device or resource busy
It says it's busy but I don't see anything that could use it. I quit skype before trying
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