[Solved]Alsa funkiness (sound breaking)

(Note: I originally thought this was a steam issue, but turns out it's not only steam that can break my sound (see the update on the bottom))
I get an issue with Steam where I get a short (less than 1s) sound on repeat/loop (usually it's some part of the message notification sound, or a trailer that I was watching) sounding a bit like a broken record, randomly occurring after keeping steam open for long periods of time in the background. Only fix I've found so far is to restart steam.
I have the Enlightenmen 18 Window Manager.
Notable errors:
Fontconfig error: "/etc/fonts/conf.d/10-scale-bitmap-fonts.conf", line 70: non-double matrix element
Fontconfig warning: "/etc/fonts/conf.d/10-scale-bitmap-fonts.conf", line 78: saw unknown, expected number
[0706/170950:ERROR:object_proxy.cc(239)] Failed to call method: org.freedesktop.DBus.Error.ServiceUnknown: The name org.freedesktop.NetworkManager was not provided by any .service files
[0706/170950:WARNING:proxy_service.cc(958)] PAC support disabled because there is no system implementation
PulseAudio connect failed (used only for Mic Volume Control) with error: Access denied
** (steam:7341): WARNING **: Could not initialize NMClient /org/freedesktop/NetworkManager: The name org.freedesktop.NetworkManager was not provided by any .service files
(steam:7341): LIBDBUSMENU-GLIB-WARNING **: Trying to remove a child that doesn't believe we're it's parent.
[0706/171536:ERROR:reference_audio_renderer.cc(46)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnCreated(media::AudioOutputController*)
[0706/171536:ERROR:reference_audio_renderer.cc(50)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnPlaying(media::AudioOutputController*)
[0706/171657:ERROR:reference_audio_renderer.cc(54)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnPaused(media::AudioOutputController*)
AL lib: pulseaudio.c:612: Context did not connect: Access denied
I don't have any reliable method to reproduce the issue, and it does not seem to output an error of it's own in the terminal output when it occurs. I have noticed that this is more likely to occur when steam has been open for prolonged periods. I am not using pulseaudio, just Alsa. I also noticed when I was running a Wine game (not through steam) that this issue occurred, and it seemed to contaminate the sound outputs for whine too (I got like a "double" sound for every sound effect the game played). If the sound was completely broken on steam and making a loop like I explained at the start of this post and then I close steam while the Wine game is running, the sound from steam will remain until I shut down the game too (i.e. seems that this isn't just breakage internally in steam, but in alsa itself)
The one thing I have tried to solve this was install lib32-libpulse, this did not seem to fix the errors nor the problem. I imagine installing pulseaudio could fix this problem, but pulseaudio downright hates my sound card, so I would rather not use it.
I also have two other issues too that are less severe.
1: Mouse cursor sometimes gets locked in steam instance (Right clicking will show the steam context menu, pressing "stop" in that context menu will solve the issue. This sometimes happens when I use "Ctrl + C"(copy) or "Ctrl + V"(paste) when I'm typing in the discussions)
2: Fullscreen videos will not go fullscreen (they will only fill the steam window)
Update:
I found a bulletproof way to make a similar issue occur.
If I open facebook in firefox, and get a message notification while I am playing a movie in Cmplayer (my media player of choice) the sound in CMplayer breaks (but firefox's sound keeps working normally)
Update2:
I was rather desperate to get this fixed so I installed pulseaudio as much as I despise it, the quality of my audio is damaged by it, but this problem was as I suspected fixed by installing it. I am guessing pulseaudio is set up partially to prevent issues like this one, but I would relaly prefer not to have it. This does not rid me of all the errors however (reference_audio_renderer errors persist) but it does create a new error for steam:
[0706/215704:ERROR:alsa_output.cc(684)] Failed querying delay: Input/output error
Last edited by rabcor (2014-07-11 16:28:34)

I always had to have an asound.conf in the name of upmixing 2.0 to 5.1 This is the content of my asound.conf before I made the changes. I also had a problem with a game where I had crackling noises on 48khz so I had to tell alsa to force 96khz (or 44.1khz but since my card supports 96khz that's what I went with) to avoid that.
pcm.!default {
slave.pcm "surround51"
slave.channels 6
type route
# Front and rear
ttable.0.0 0.7
ttable.1.1 0.7
ttable.2.2 0.6
ttable.3.3 0.6
# Center and LFE
ttable.4.4 1
ttable.5.5 1
# Front left/right to center
ttable.0.4 0.5
ttable.1.4 0.5
# Front left/right to rear
ttable.0.2 0.5
ttable.1.3 0.5
Now it looks like this and everything seems to be working fine (except that mono sound only plays from my left speaker and I haven't defined a way to downmix 7.1 to 5.1):
pcm.dmixed {
type asym
playback.pcm {
# See plugin:dmix at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
type dmix
# Don't block other users, e.g. the Timidity midi-player daemon
# http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
ipc_key_add_uid true
ipc_key 5678293
ipc_perm 0660
ipc_gid audio
# Don't put the rate here! Otherwise it resets the rate & channels set below, as shown by: cat /proc/asound/card0/pcm0p/sub0/hw_params
slave {
# 2 for stereo, 6 for surround51, 8 for surround71
channels 6
pcm {
# mplayer chooses S32_LE, but others usually S16_LE
format S32_LE
#format S16_LE
# 44100 or 48000
# 44100 for music, 48000 is compatible with most h/w
#rate 44100
rate 96000
# http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
# Maybe helps
nonblock true
type hw
card 0
device 0
subdevice 0
# mplayer2 chooses 1024
# period_size 512 with buffer_size 16384 stops crackling in xmame
# 320 breaks flash - https://bbs.archlinux.org/viewtopic.php?id=129458
#period_size 512
period_size 1024
#period_size 512
# 4096 might make sound crackle
# mplayer2 chooses 8192. Half-Life 2 chooses 16384.
# If too large, use CONFIG_SND_HDA_PREALLOC_SIZE=2048
buffer_size 16384
capture.pcm "hw:0"
pcm.!default{
type plug
slave.pcm "upmix20_51"
pcm.!surround20 {
type plug
slave.pcm "upmix20_51"
pcm.!surround40 {
type plug
slave.pcm "dmixed"
route_policy duplicate
pcm.!surround51 {
type plug
slave.pcm "dmixed"
pcm.upmix20_51 {
slave.pcm "dmixed"
slave.channels 6
type route
# Front and rear
ttable.0.0 0.7
ttable.1.1 0.7
ttable.2.2 0.6
ttable.3.3 0.6
# Center and LFE
ttable.4.4 1
ttable.5.5 1
# Front left/right to center
ttable.0.4 0.5
ttable.1.4 0.5
# Front left/right to rear
ttable.0.2 0.5
ttable.1.3 0.5
Is there a better way for me to achieve the results of the above config? Because I'm all ears.
Last edited by rabcor (2014-07-08 11:09:23)

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    Last edited by EvanPurkhiser (2014-04-16 09:27:13)

    I'm interested in recording from the mixer over USB.
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    Last edited by NeanderMarcl (2013-06-15 11:53:01)

    chu887 wrote:
    NeanderMarcl wrote:
    Hi,
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    **** List of PLAYBACK Hardware Devices ****
    card 0: Intel [HDA Intel], device 0: STAC9228 Analog [STAC9228 Analog]
      Subdevices: 0/1
      Subdevice #0: subdevice #0
    card 0: Intel [HDA Intel], device 1: STAC9228 Digital [STAC9228 Digital]
      Subdevices: 1/1
      Subdevice #0: subdevice #0
    Headphones don't work

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    hello,
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    Sound breaks up in PE7 when I play back from the timeline. I have Vista 64 bit, Core 2 quad 4Gb RAM, GeForce 9600 GT. Scratch disks are set to E drive which are 2x1Tb drives in raid 0
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  • Alsa: reroute sound output to line in?

    Hello,
    i've been on this for a week now, not getting anywhere.
    but now i'm determined to crack the nut that is alsa.
    a familiar problem: i want to use alsa's sound output - pcm or master - internally.
    i have set up a loopback device now; it works with e.g. recordmydesktop, but the application i'm interested in seems to only chose the first capture channel of the default soundcard :-(
    (it's sndpeek).
    so, is it possible to duplicate & reroute the default card's pcm or master channel back into the default card's Line channel (Line in)?
    of course i'm open to other suggestions.
    thanks.
    current .asoundrc (creates a 2nd sound card that i cannot access with sndpeek, but works otherwise)
    pcm.!default { type asym
    playback.pcm "LoopAndReal"
    #capture.pcm "looprec"
    capture.pcm "hw:0,0"
    pcm.looprec { type hw
    card "Loopback"
    device 1
    subdevice 0
    pcm.LoopAndReal { type plug
    slave.pcm mdev
    route_policy "duplicate"
    pcm.mdev { type multi
    slaves.a.pcm pcm.MixReale
    slaves.a.channels 2
    slaves.b.pcm pcm.MixLoopback
    slaves.b.channels 2
    bindings.0.slave a
    bindings.0.channel 0
    bindings.1.slave a
    bindings.1.channel 1
    bindings.2.slave b
    bindings.2.channel 0
    bindings.3.slave b
    bindings.3.channel 1
    pcm.MixReale { type dmix
    ipc_key 1024
    slave { pcm "hw:0,0"
    rate 48000
    #rate 44100
    periods 128
    period_time 0
    period_size 1024 # must be power of 2
    buffer_size 8192
    pcm.MixLoopback { type dmix
    ipc_key 1025
    slave { pcm "hw:Loopback,0,0"
    rate 48000
    #rate 44100
    periods 128
    period_time 0
    period_size 1024 # must be power of 2
    buffer_size 8192
    - copied over from here.
    the soundcard is a basic laptop intel chip.
    $ dmesg|grep -i intel
    [ 1.037761] i915 0000:00:02.0: fb0: inteldrmfb frame buffer device
    [ 19.315080] snd_hda_intel 0000:00:1b.0: irq 44 for MSI/MSI-X
    [ 20.408741] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10
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    Last edited by ondoho (2013-11-07 19:37:46)

    so, is it possible to duplicate & reroute the default card's pcm or master channel back into the default card's Line channel (Line in)?
    If you're talking about the analog audio path the answer is maybe. But fair warning, you'll be spending many more hours (maybe) getting it to work while wallowing real deep in today's integrated audio hardware and ALSA if you want to use this method.
    the soundcard is a basic laptop intel chip.
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    Last edited by pigiron (2013-11-10 04:31:25)

  • Solving the distorted sound?

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    Make sure your battery is fully charged
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  • Tip: Sound breaking up. Check your HDMI cable.

    Having owned our Apple TV for only a week, I was very disappointed that the sound would break every 3 to 5 seconds when anything was broadcast from the Apple TV via our television set. The sound breaking occurred for both videos and audios, for both synced and streamed. When played locally on my iMac, the sound was perfect.
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    As part of my trouble-shooting process, I needed to eliminate the cabling as a cause. I therefore switched the HDMI cable even though it was only a week old. As soon as I did so, the sound problem went away. I can now listen to music or watch videos without any glitches at all. Life is wonderful
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    Thanks for the tip. I was having very poor audio problems where the audio was not synced with the video. Changed out the hdmi cable and it works great.

  • [Solved] Alsa Sound Card Numbering Not Persistant Between Boots

    Hello,
    I'm having trouble with alsa and my multiple sound cards. Sometimes when I boot, I get the following output from "aplay -l"
    **** List of PLAYBACK Hardware Devices ****
    card 0: SB [HDA ATI SB], device 0: ALC889A Analog [ALC889A Analog]
    Subdevices: 1/1
    Subdevice #0: subdevice #0
    card 0: SB [HDA ATI SB], device 1: ALC889A Digital [ALC889A Digital]
    Subdevices: 1/1
    Subdevice #0: subdevice #0
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    Subdevices: 1/1
    Subdevice #0: subdevice #0
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    Subdevices: 1/1
    Subdevice #0: subdevice #0
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    Subdevice #0: subdevice #0
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    Last edited by szim90 (2012-01-15 02:01:54)

    Thank you for your response, karol.
    I ran 'udevadm info -a -p /sys/class/sound/card1/' as was able to get the following information:
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    KERNEL=="card1"
    SUBSYSTEM=="sound"
    DRIVER==""
    ATTR{id}=="SB"
    ATTR{number}=="1"
    looking at parent device '/devices/pci0000:00/0000:00:14.2':
    KERNELS=="0000:00:14.2"
    SUBSYSTEMS=="pci"
    DRIVERS=="snd_hda_intel"
    ATTRS{vendor}=="0x1002"
    ATTRS{device}=="0x4383"
    ATTRS{subsystem_vendor}=="0x1458"
    ATTRS{subsystem_device}=="0xa102"
    ATTRS{class}=="0x040300"
    ATTRS{irq}=="16"
    ATTRS{local_cpus}=="00000000,0000000f"
    ATTRS{local_cpulist}=="0-3"
    ATTRS{numa_node}=="0"
    ATTRS{dma_mask_bits}=="64"
    ATTRS{consistent_dma_mask_bits}=="64"
    ATTRS{broken_parity_status}=="0"
    ATTRS{msi_bus}==""
    looking at parent device '/devices/pci0000:00':
    KERNELS=="pci0000:00"
    SUBSYSTEMS==""
    DRIVERS==""
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    ctl.!default { type hw card SB }
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    Regards,
    Sean

  • [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)

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