Unable to un-mute PCM in alsa

Hi, recently audio stopped working. It maybe after the upgrade of alsa [2010-07-23 07:19] installed alsa-utils (1.0.23-2), but not sure. It is a SiS SI7012 Chip: C-Media Electronics CMI9739 and has been working for a while without problems.
I checked in xfce4-mixer and found PCM muted. Tried to unmute, but it just muted itself again. Tried the same in alsamixer and could not unmute. Also tried as root. I am member of audio
I have tried restarting alsa as root /etc/rc.d/alsa restart and got this
/usr/sbin/alsactl: set_control:1388: Cannot write control '2:0:0:PCM Playback Volume:0' : Operation not permitted
Also as root amixer sset PCM 90% unmute -d
amixer: Invalid command!
Tried on the IRC with a few suggestions, tried setting card options again, but still dead. Hoping someone here may have some clues.
Dan
Last edited by alleyoopster (2010-07-30 21:31:56)

I tried running alsaconf (root) and that does not change things, but I have made some progress. It seems that stopping X and running alsamixer from a tty and  un-muting PCM fixes the problem. It is still fixed going back into a desktop. After rebooting and then going into XFCE once again PCM is muted an cannot un-mute. Weird!

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    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-mono+input:iec958-stereo
    D: alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-mono+input:iec958-surround-40
    D: alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-stereo
    D: alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:analog-stereo+input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-stereo+input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-40
    D: alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40)
    D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open surround40:0
    D: alsa-util.c: Maximum hw buffer size is 170 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-40 supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-40+input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-41
    D: alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41)
    D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open surround41:0
    D: alsa-util.c: Maximum hw buffer size is 113 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-41 supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-41+input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-50
    D: alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50)
    D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open surround50:0
    D: alsa-util.c: Maximum hw buffer size is 113 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-50 supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-50+input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-51
    D: alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51)
    D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open surround51:0
    D: alsa-util.c: Maximum hw buffer size is 113 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-51 supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-51+input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-71
    D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
    D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open surround71:0
    D: alsa-util.c: Maximum hw buffer size is 85 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-71 supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:analog-surround-71+input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-stereo
    D: alsa-mixer.c: Checking for playback on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:iec958-stereo supported.
    D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:iec958-stereo+input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-surround-40
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:analog-mono
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:analog-stereo
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:iec958-stereo
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:iec958-surround-40
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:iec958:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:analog-mono
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:analog-stereo
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:iec958-stereo
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:iec958-surround-40
    D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51
    D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:analog-mono
    D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:analog-stereo
    D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:iec958-stereo
    D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:iec958-surround-40
    D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
    D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm.c: Unknown PCM a52:0
    I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:hdmi-stereo
    D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
    D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hdmi:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:hdmi-stereo supported.
    D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile output:hdmi-stereo+input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile input:analog-mono
    D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
    D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open plug:hw:0
    D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
    I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
    D: alsa-mixer.c: Looking at profile input:analog-stereo
    D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 4800 samples), period size second (to 1200 samples).
    D: alsa-mixer.c: Profile input:analog-stereo supported.
    D: alsa-mixer.c: Looking at profile input:iec958-stereo
    D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    D: alsa-mixer.c: Looking at profile input:iec958-surround-40
    D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
    D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
    I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
    I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
    I: card.c: Created 0 "alsa_card.pci-0000_00_09.0"
    D: reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio0'
    D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open surround51:0
    D: alsa-util.c: Maximum hw buffer size is 113 ms
    D: alsa-util.c: Set buffer size first (to 96000 samples), period size second (to 96000 samples).
    I: alsa-sink.c: Successfully opened device surround51:0.
    I: alsa-sink.c: Selected mapping 'Analog Surround 5.1' (analog-surround-51).
    I: alsa-sink.c: Successfully enabled mmap() mode.
    I: alsa-sink.c: Successfully enabled timer-based scheduling mode.
    I: (alsa-lib)control.c: Invalid CTL surround51:0
    I: alsa-mixer.c: Unable to attach to mixer surround51:0: No such file or directory
    I: alsa-mixer.c: Successfully attached to mixer 'hw:0'
    D: alsa-mixer.c: Probing path 'analog-output'
    D: alsa-mixer.c: Probing path 'analog-output-speaker'
    D: alsa-mixer.c: Probing path 'analog-output-speaker'
    D: alsa-mixer.c: Probe of element 'Desktop Speaker' failed.
    D: alsa-mixer.c: Probing path 'analog-output-lfe-on-mono'
    D: alsa-mixer.c: Probe of element 'Master Mono' failed.
    D: alsa-sink.c: Probed mixer paths:
    D: alsa-mixer.c: Path Set 0xee7800, direction=1, probed=yes
    D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-144, max_dB=0
    D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
    D: alsa-mixer.c: Element Speaker, direction=1, switch=1, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
    D: alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
    D: alsa-mixer.c: Path analog-output-speaker (Analog Speakers), direction=1, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-144, max_dB=0
    D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
    D: alsa-mixer.c: Element Speaker, direction=1, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
    D: alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
    D: alsa-mixer.c: Added 2 ports
    I: sink.c: Created sink 0 "alsa_output.pci-0000_00_09.0.analog-surround-51" with sample spec s16le 6ch 48000Hz and channel map front-left,front-right,rear-left,rear-right,front-center,lfe
    I: sink.c: alsa.resolution_bits = "16"
    I: sink.c: device.api = "alsa"
    I: sink.c: device.class = "sound"
    I: sink.c: alsa.class = "generic"
    I: sink.c: alsa.subclass = "generic-mix"
    I: sink.c: alsa.name = "ALC1200 Analog"
    I: sink.c: alsa.id = "ALC1200 Analog"
    I: sink.c: alsa.subdevice = "0"
    I: sink.c: alsa.subdevice_name = "subdevice #0"
    I: sink.c: alsa.device = "0"
    I: sink.c: alsa.card = "0"
    I: sink.c: alsa.card_name = "HDA NVidia"
    I: sink.c: alsa.long_card_name = "HDA NVidia at 0xf9e78000 irq 20"
    I: sink.c: alsa.driver_name = "snd_hda_intel"
    I: sink.c: device.bus_path = "pci-0000:00:09.0"
    I: sink.c: sysfs.path = "/devices/pci0000:00/0000:00:09.0/sound/card0"
    I: sink.c: device.bus = "pci"
    I: sink.c: device.vendor.id = "10de"
    I: sink.c: device.vendor.name = "nVidia Corporation"
    I: sink.c: device.product.id = "07fc"
    I: sink.c: device.product.name = "MCP73 High Definition Audio"
    I: sink.c: device.form_factor = "internal"
    I: sink.c: device.string = "surround51:0"
    I: sink.c: device.buffering.buffer_size = "65280"
    I: sink.c: device.buffering.fragment_size = "32640"
    I: sink.c: device.access_mode = "mmap+timer"
    I: sink.c: device.profile.name = "analog-surround-51"
    I: sink.c: device.profile.description = "Analog Surround 5.1"
    I: sink.c: device.description = "Internal Audio Analog Surround 5.1"
    I: sink.c: alsa.mixer_name = "Nvidia MCP73 HDMI"
    I: sink.c: alsa.components = "HDA:10ec0888,10250137,00100101 HDA:10de8001,10de0101,00100000"
    I: sink.c: module-udev-detect.discovered = "1"
    I: sink.c: device.icon_name = "audio-card-pci"
    D: core-subscribe.c: Dropped redundant event due to change event.
    I: source.c: Created source 0 "alsa_output.pci-0000_00_09.0.analog-surround-51.monitor" with sample spec s16le 6ch 48000Hz and channel map front-left,front-right,rear-left,rear-right,front-center,lfe
    I: source.c: device.description = "Monitor of Internal Audio Analog Surround 5.1"
    I: source.c: device.class = "monitor"
    I: source.c: alsa.card = "0"
    I: source.c: alsa.card_name = "HDA NVidia"
    I: source.c: alsa.long_card_name = "HDA NVidia at 0xf9e78000 irq 20"
    I: source.c: alsa.driver_name = "snd_hda_intel"
    I: source.c: device.bus_path = "pci-0000:00:09.0"
    I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:09.0/sound/card0"
    I: source.c: device.bus = "pci"
    I: source.c: device.vendor.id = "10de"
    I: source.c: device.vendor.name = "nVidia Corporation"
    I: source.c: device.product.id = "07fc"
    I: source.c: device.product.name = "MCP73 High Definition Audio"
    I: source.c: device.form_factor = "internal"
    I: source.c: device.string = "0"
    I: source.c: module-udev-detect.discovered = "1"
    I: source.c: device.icon_name = "audio-card-pci"
    I: alsa-sink.c: Using 2.0 fragments of size 32640 bytes (56.67ms), buffer size is 65280 bytes (113.33ms)
    I: alsa-sink.c: Time scheduling watermark is 20.00ms
    D: alsa-sink.c: hwbuf_unused=0
    D: alsa-sink.c: setting avail_min=4480
    D: alsa-mixer.c: Activating path analog-output-speaker
    D: alsa-mixer.c: Path analog-output-speaker (Analog Speakers), direction=1, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-144, max_dB=0
    D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
    D: alsa-mixer.c: Element Speaker, direction=1, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
    D: alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
    D: alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
    I: alsa-sink.c: Hardware volume ranges from -144.00 dB to 0.00 dB.
    I: alsa-sink.c: No particular base volume set, fixing to 0 dB
    I: alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
    I: alsa-sink.c: Using hardware mute control.
    D: alsa-util.c: snd_pcm_dump():
    D: alsa-util.c: Soft volume PCM
    D: alsa-util.c: Control: PCM Playback Volume
    D: alsa-util.c: min_dB: -51
    D: alsa-util.c: max_dB: 0
    D: alsa-util.c: resolution: 256
    D: alsa-util.c: Its setup is:
    D: alsa-util.c: stream : PLAYBACK
    D: alsa-util.c: access : MMAP_INTERLEAVED
    D: alsa-util.c: format : S16_LE
    D: alsa-util.c: subformat : STD
    D: alsa-util.c: channels : 6
    D: alsa-util.c: rate : 48000
    D: alsa-util.c: exact rate : 48000 (48000/1)
    D: alsa-util.c: msbits : 16
    D: alsa-util.c: buffer_size : 5440
    D: alsa-util.c: period_size : 2720
    D: alsa-util.c: period_time : 56666
    D: alsa-util.c: tstamp_mode : ENABLE
    D: alsa-util.c: period_step : 1
    D: alsa-util.c: avail_min : 4480
    D: alsa-util.c: period_event : 0
    D: alsa-util.c: start_threshold : -1
    D: alsa-util.c: stop_threshold : 6124895493223874560
    D: alsa-util.c: silence_threshold: 0
    D: alsa-util.c: silence_size : 0
    D: alsa-util.c: boundary : 6124895493223874560
    D: alsa-util.c: Slave: Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0
    D: alsa-util.c: Its setup is:
    D: alsa-util.c: stream : PLAYBACK
    D: alsa-util.c: access : MMAP_INTERLEAVED
    D: alsa-util.c: format : S16_LE
    D: alsa-util.c: subformat : STD
    D: alsa-util.c: channels : 6
    D: alsa-util.c: rate : 48000
    D: alsa-util.c: exact rate : 48000 (48000/1)
    D: alsa-util.c: msbits : 16
    D: alsa-util.c: buffer_size : 5440
    D: alsa-util.c: period_size : 2720
    D: alsa-util.c: period_time : 56666
    D: alsa-util.c: tstamp_mode : ENABLE
    D: alsa-util.c: period_step : 1
    D: alsa-util.c: avail_min : 4480
    D: alsa-util.c: period_event : 0
    D: alsa-util.c: start_threshold : -1
    D: alsa-util.c: stop_threshold : 6124895493223874560
    D: alsa-util.c: silence_threshold: 0
    D: alsa-util.c: silence_size : 0
    D: alsa-util.c: boundary : 6124895493223874560
    D: alsa-util.c: appl_ptr : 0
    D: alsa-util.c: hw_ptr : 0
    D: alsa-sink.c: Read hardware volume: 0: 20% 1: 20% 2: 20% 3: 20% 4: 20% 5: 20%
    D: alsa-sink.c: Thread starting up
    D: core-util.c: SCHED_RR|SCHED_RESET_ON_FORK worked.
    I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5.
    I: alsa-sink.c: Starting playback.
    D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
    D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
    D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
    D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
    D: alsa-util.c: Managed to open front:0
    D: alsa-util.c: Maximum hw buffer size is 341 ms
    D: alsa-util.c: Set buffer size first (to 96000 samples), period size second (to 96000 samples).
    I: alsa-source.c: Successfully opened device front:0.
    I: alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo).
    I: alsa-source.c: Successfully enabled mmap() mode.
    I: alsa-source.c: Successfully enabled timer-based scheduling mode.
    I: (alsa-lib)control.c: Invalid CTL front:0
    I: alsa-mixer.c: Unable to attach to mixer front:0: No such file or directory
    I: alsa-mixer.c: Successfully attached to mixer 'hw:0'
    D: alsa-mixer.c: Probing path 'analog-input'
    D: alsa-mixer.c: Probing path 'analog-input-microphone'
    D: alsa-mixer.c: Probe of element 'Mic' failed.
    D: alsa-mixer.c: Probing path 'analog-input-linein'
    D: alsa-mixer.c: Probe of element 'Line' failed.
    D: alsa-mixer.c: Probing path 'analog-input'
    D: alsa-mixer.c: Probe of element 'Aux' failed.
    D: alsa-mixer.c: Probing path 'analog-input-video'
    D: alsa-mixer.c: Probe of element 'Video' failed.
    D: alsa-mixer.c: Probing path 'analog-input-video'
    D: alsa-mixer.c: Probe of element 'TV Tuner' failed.
    D: alsa-mixer.c: Probing path 'analog-input-radio'
    D: alsa-mixer.c: Probe of element 'FM' failed.
    D: alsa-mixer.c: Probing path 'analog-input'
    D: alsa-mixer.c: Probe of element 'Mic/Line' failed.
    D: alsa-source.c: Probed mixer paths:
    D: alsa-mixer.c: Path Set 0x7f4604004330, direction=2, probed=yes
    D: alsa-mixer.c: Path analog-input (Analog Input), direction=2, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-16.5, max_dB=30
    D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=2, required_absent=0, mask=0x403f600000000f66, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, enumeration=1, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
    D: alsa-mixer.c: Option Mic (input-microphone-1/Microphone 1) index=0, priority=20
    D: alsa-mixer.c: Option Front Mic (input-microphone-2/Microphone 2) index=1, priority=19
    D: alsa-mixer.c: Option Line (input-linein/Line-In) index=2, priority=18
    D: alsa-mixer.c: Setting input-microphone-1 (Microphone 1) priority=20
    D: alsa-mixer.c: Setting input-microphone-2 (Microphone 2) priority=19
    D: alsa-mixer.c: Setting input-linein (Line-In) priority=18
    D: alsa-mixer.c: Added 3 ports
    D: core-subscribe.c: Dropped redundant event due to change event.
    I: source.c: Created source 1 "alsa_input.pci-0000_00_09.0.analog-stereo" with sample spec s16le 2ch 48000Hz and channel map front-left,front-right
    I: source.c: alsa.resolution_bits = "16"
    I: source.c: device.api = "alsa"
    I: source.c: device.class = "sound"
    I: source.c: alsa.class = "generic"
    I: source.c: alsa.subclass = "generic-mix"
    I: source.c: alsa.name = "ALC1200 Analog"
    I: source.c: alsa.id = "ALC1200 Analog"
    I: source.c: alsa.subdevice = "0"
    I: source.c: alsa.subdevice_name = "subdevice #0"
    I: source.c: alsa.device = "0"
    I: source.c: alsa.card = "0"
    I: source.c: alsa.card_name = "HDA NVidia"
    I: source.c: alsa.long_card_name = "HDA NVidia at 0xf9e78000 irq 20"
    I: source.c: alsa.driver_name = "snd_hda_intel"
    I: source.c: device.bus_path = "pci-0000:00:09.0"
    I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:09.0/sound/card0"
    I: source.c: device.bus = "pci"
    I: source.c: device.vendor.id = "10de"
    I: source.c: device.vendor.name = "nVidia Corporation"
    I: source.c: device.product.id = "07fc"
    I: source.c: device.product.name = "MCP73 High Definition Audio"
    I: source.c: device.form_factor = "internal"
    I: source.c: device.string = "front:0"
    I: source.c: device.buffering.buffer_size = "65536"
    I: source.c: device.buffering.fragment_size = "32768"
    I: source.c: device.access_mode = "mmap+timer"
    I: source.c: device.profile.name = "analog-stereo"
    I: source.c: device.profile.description = "Analog Stereo"
    I: source.c: device.description = "Internal Audio Analog Stereo"
    I: source.c: alsa.mixer_name = "Nvidia MCP73 HDMI"
    I: source.c: alsa.components = "HDA:10ec0888,10250137,00100101 HDA:10de8001,10de0101,00100000"
    I: source.c: module-udev-detect.discovered = "1"
    I: source.c: device.icon_name = "audio-card-pci"
    I: alsa-source.c: Using 2.0 fragments of size 32768 bytes (170.67ms), buffer size is 65536 bytes (341.33ms)
    I: alsa-source.c: Time scheduling watermark is 20.00ms
    D: alsa-source.c: hwbuf_unused=0
    D: alsa-source.c: setting avail_min=15424
    D: alsa-mixer.c: Activating path analog-input
    D: alsa-mixer.c: Path analog-input (Analog Input), direction=2, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-16.5, max_dB=30
    D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=2, required_absent=0, mask=0x403f600000000f66, n_channels=2, override_map=yes
    D: alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, enumeration=1, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
    D: alsa-mixer.c: Option Mic (input-microphone-1/Microphone 1) index=0, priority=20
    D: alsa-mixer.c: Option Front Mic (input-microphone-2/Microphone 2) index=1, priority=19
    D: alsa-mixer.c: Option Line (input-linein/Line-In) index=2, priority=18
    D: alsa-mixer.c: Setting input-microphone-1 (Microphone 1) priority=20
    D: alsa-mixer.c: Setting input-microphone-2 (Microphone 2) priority=19
    D: alsa-mixer.c: Setting input-linein (Line-In) priority=18
    I: alsa-source.c: Hardware volume ranges from -16.50 dB to 30.00 dB.
    I: alsa-source.c: Fixing base volume to -30.00 dB
    I: alsa-source.c: Using hardware volume control. Hardware dB scale supported.
    I: alsa-source.c: Using hardware mute control.
    D: alsa-util.c: snd_pcm_dump():
    D: alsa-util.c: Soft volume PCM
    D: alsa-util.c: Control: PCM Playback Volume
    D: alsa-util.c: min_dB: -51
    D: alsa-util.c: max_dB: 0
    D: alsa-util.c: resolution: 256
    D: alsa-util.c: Its setup is:
    D: alsa-util.c: stream : CAPTURE
    D: alsa-util.c: access : MMAP_INTERLEAVED
    D: alsa-util.c: format : S16_LE
    D: alsa-util.c: subformat : STD
    D: alsa-util.c: channels : 2
    D: alsa-util.c: rate : 48000
    D: alsa-util.c: exact rate : 48000 (48000/1)
    D: alsa-util.c: msbits : 16
    D: alsa-util.c: buffer_size : 16384
    D: alsa-util.c: period_size : 8192
    D: alsa-util.c: period_time : 170666
    D: alsa-util.c: tstamp_mode : ENABLE
    D: alsa-util.c: period_step : 1
    D: alsa-util.c: avail_min : 15424
    D: alsa-util.c: period_event : 0
    D: alsa-util.c: start_threshold : -1
    D: alsa-util.c: stop_threshold : 4611686018427387904
    D: alsa-util.c: silence_threshold: 0
    D: alsa-util.c: silence_size : 0
    D: alsa-util.c: boundary : 4611686018427387904
    D: alsa-util.c: Slave: Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0
    D: alsa-util.c: Its setup is:
    D: alsa-util.c: stream : CAPTURE
    D: alsa-util.c: access : MMAP_INTERLEAVED
    D: alsa-util.c: format : S16_LE
    D: alsa-util.c: subformat : STD
    D: alsa-util.c: channels : 2
    D: alsa-util.c: rate : 48000
    D: alsa-util.c: exact rate : 48000 (48000/1)
    D: alsa-util.c: msbits : 16
    D: alsa-util.c: buffer_size : 16384
    D: alsa-util.c: period_size : 8192
    D: alsa-util.c: period_time : 170666
    D: alsa-util.c: tstamp_mode : ENABLE
    D: alsa-util.c: period_step : 1
    D: alsa-util.c: avail_min : 15424
    D: alsa-util.c: period_event : 0
    D: alsa-util.c: start_threshold : -1
    D: alsa-util.c: stop_threshold : 4611686018427387904
    D: alsa-util.c: silence_threshold: 0
    D: alsa-util.c: silence_size : 0
    D: alsa-util.c: boundary : 4611686018427387904
    D: alsa-util.c: appl_ptr : 0
    D: alsa-util.c: hw_ptr : 0
    D: alsa-source.c: Read hardware volume: 0: 27% 1: 27%
    D: alsa-source.c: Thread starting up
    D: core-util.c: SCHED_RR|SCHED_RESET_ON_FORK worked.
    I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5.
    I: alsa-source.c: Starting capture.
    I: module.c: Loaded "module-alsa-card" (index: #4; argument: "device_id="0" name="pci-0000_00_09.0" card_name="alsa_card.pci-0000_00_09.0" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1"").
    I: module-udev-detect.c: Card /devices/pci0000:00/0000:00:09.0/sound/card0 (alsa_card.pci-0000_00_09.0) module loaded.
    I: module-udev-detect.c: Found 1 cards.
    I: module.c: Loaded "module-udev-detect" (index: #5; argument: "").
    D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.22/modules/module-bluetooth-discover.so': success
    D: dbus-util.c: Successfully connected to D-Bus system bus 388ec2ad109f18dc6cb6313a00000021 as :1.108
    D: bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
    D: bluetooth-util.c: Bluetooth daemon is apparently not available.
    I: module.c: Loaded "module-bluetooth-discover" (index: #6; argument: "").
    D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.22/modules/module-esound-protocol-unix.so': success
    E: socket-server.c: bind(): Address already in use
    E: module.c: Failed to load module "module-esound-protocol-unix" (argument: ""): initialization failed.
    E: main.c: Module load failed.
    E: main.c: Failed to initialize daemon.
    I: module.c: Unloading "module-device-restore" (index: #0).
    I: module.c: Unloaded "module-device-restore" (index: #0).
    D: core-subscribe.c: Dropped redundant event due to remove event.
    I: module.c: Unloading "module-stream-restore" (index: #1).
    I: module.c: Unloaded "module-stream-restore" (index: #1).
    D: core-subscribe.c: Dropped redundant event due to remove event.
    I: module.c: Unloading "module-card-restore" (index: #2).
    I: module.c: Unloaded "module-card-restore" (index: #2).
    D: core-subscribe.c: Dropped redundant event due to remove event.
    I: module.c: Unloading "module-augment-properties" (index: #3).
    I: module.c: Unloaded "module-augment-properties" (index: #3).
    D: core-subscribe.c: Dropped redundant event due to remove event.
    I: module.c: Unloading "module-alsa-card" (index: #4).
    D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
    D: core-subscribe.c: Dropped redundant event due to remove event.
    D: core-subscribe.c: Dropped redundant event due to remove event.
    D: core-subscribe.c: Dropped redundant event due to remove event.
    D: core-subscribe.c: Dropped redundant event due to remove event.
    D: alsa-sink.c: Thread shutting down
    I: sink.c: Freeing sink 0 "alsa_output.pci-0000_00_09.0.analog-surround-51"
    I: source.c: Freeing source 0 "alsa_output.pci-0000_00_09.0.analog-surround-51.monitor"
    D: core-subscribe.c: Dropped redundant event due to remove event.
    D: core-subscribe.c: Dropped redundant event due to remove event.
    D: alsa-source.c: Thread shutting down
    I: source.c: Freeing source 1 "alsa_input.pci-0000_00_09.0.analog-stereo"
    I: card.c: Freed 0 "alsa_card.pci-0000_00_09.0"
    D: core-subscribe.c: Dropped redundant event due to remove event.
    I: module.c: Unloaded "module-alsa-card" (index: #4).
    D: core-subscribe.c: Dropped redundant event due to remove event.
    I: module.c: Unloading "module-udev-detect" (index: #5).
    I: module.c: Unloaded "module-udev-detect" (index: #5).
    D: core-subscribe.c: Dropped redundant event due to remove event.
    I: module.c: Unloading "module-bluetooth-discover" (index: #6).
    I: module.c: Unloaded "module-bluetooth-discover" (index: #6).
    D: core-subscribe.c: Dropped redundant event due to remove event.
    I: main.c: Daemon terminated.

  • [Kernel] USB and network trouble

    Hey all, i have some trouble getting my system fully running.
    I have a built in 3com network card, it uses the 3c59x module in the kernel (got it build into the kernel). I also have a usb cd writer. When i start writing my network fails after sometime. I came from vectorlinux to arch. In VL it worked without a prob. Here check my dmesg output. The most interesting is at the end.
    chitecture supported.
    Intel machine check reporting enabled on CPU#0.
    CPU: Intel Pentium III (Coppermine) stepping 06
    Enabling fast FPU save and restore... done.
    Enabling unmasked SIMD FPU exception support... done.
    Checking 'hlt' instruction... OK.
    ACPI: setting ELCR to 0200 (from 0808)
    checking if image is initramfs... it is
    Freeing initrd memory: 567k freed
    NET: Registered protocol family 16
    PCI: PCI BIOS revision 2.10 entry at 0xfed9b, last bus=21
    PCI: Using configuration type 1
    mtrr: v2.0 (20020519)
    ACPI: Subsystem revision 20041105
    ACPI-0169: *** Error: No object was returned from [_SB_.LNKA._STA] (Node d3fdef00), AE_NOT_EXIST
    ACPI-0169: *** Error: No object was returned from [_SB_.LNKB._STA] (Node d3fdee00), AE_NOT_EXIST
    ACPI-0169: *** Error: No object was returned from [_SB_.LNKC._STA] (Node d3fded00), AE_NOT_EXIST
    ACPI-0169: *** Error: No object was returned from [_SB_.LNKD._STA] (Node d3fdec00), AE_NOT_EXIST
    ACPI-0169: *** Error: No object was returned from [_SB_.PCI0.FNC0.FDD_._STA] (Node c12f75a0), AE_NOT_EXIST
    ACPI-0169: *** Error: No object was returned from [_SB_.PCI0.FNC0.COM_._STA] (Node c12f7460), AE_NOT_EXIST
    ACPI-0169: *** Error: No object was returned from [_SB_.PCI0.FNC0.PRT_._STA] (Node c12f72e0), AE_NOT_EXIST
    ACPI-0169: *** Error: No object was returned from [_SB_.PCI0.FNC0.PRT1._STA] (Node c12f7200), AE_NOT_EXIST
    ACPI-0169: *** Error: No object was returned from [_SB_.PCI0.FNC0.PCC0._STA] (Node c12f7fe0), AE_NOT_EXIST
    ACPI: Interpreter enabled
    ACPI: Using PIC for interrupt routing
    ACPI-0169: *** Error: No object was returned from [_SB_.LNKE._PRS] (Node c12fc780), AE_NOT_EXIST
    ACPI: PCI Root Bridge [PCI0] (00:00)
    PCI: Probing PCI hardware (bus 00)
    ACPI: PCI Interrupt Routing Table [_SB_.PCI0._PRT]
    ACPI: Power Resource [PIHD] (on)
    ACPI: Power Resource [PMHD] (off)
    ACPI: Power Resource [PDOC] (off)
    ACPI: PCI Interrupt Routing Table [_SB_.PCI0.PCI1._PRT]
    ACPI: Power Resource [PFAN] (off)
    Linux Plug and Play Support v0.97 (c) Adam Belay
    pnp: PnP ACPI init
    ACPI-0169: *** Error: No object was returned from [_SB_.MEM_._CRS] (Node c12fc6a0), AE_NOT_EXIST
    pnp: PnPACPI: METHOD_NAME__CRS failure for PNP0c01
    PCI: setting IRQ 13 as level-triggered
    pnp: PnP ACPI: found 9 devices
    SCSI subsystem initialized
    Linux Kernel Card Services
    options: [pci] [cardbus] [pm]
    usbcore: registered new driver usbfs
    usbcore: registered new driver hub
    PCI: Using ACPI for IRQ routing
    ** PCI interrupts are no longer routed automatically. If this
    ** causes a device to stop working, it is probably because the
    ** driver failed to call pci_enable_device(). As a temporary
    ** workaround, the "pci=routeirq" argument restores the old
    ** behavior. If this argument makes the device work again,
    ** please email the output of "lspci" to [email protected]
    ** so I can fix the driver.
    ACPI-0169: *** Error: No object was returned from [_SB_.MEM_._CRS] (Node c12fc6a0), AE_NOT_EXIST
    apm: BIOS version 1.2 Flags 0x02 (Driver version 1.16ac)
    apm: overridden by ACPI.
    devfs: 2004-01-31 Richard Gooch ([email protected])
    devfs: boot_options: 0x1
    Initializing Cryptographic API
    Limiting direct PCI/PCI transfers.
    cpci_hotplug: CompactPCI Hot Plug Core version: 0.2
    pci_hotplug: PCI Hot Plug PCI Core version: 0.5
    acpiphp: ACPI Hot Plug PCI Controller Driver version: 0.4
    acpiphp_glue: can't get bus number, assuming 0
    Linux agpgart interface v0.100 (c) Dave Jones
    agpgart: Detected an Intel 440BX Chipset.
    agpgart: Maximum main memory to use for agp memory: 262M
    agpgart: AGP aperture is 256M @ 0xd0000000
    vesafb: framebuffer at 0xf0000000, mapped to 0xd4900000, using 6144k, total 8192k
    vesafb: mode is 1024x768x32, linelength=4096, pages=1
    vesafb: protected mode interface info at c000:8751
    vesafb: scrolling: redraw
    vesafb: Truecolor: size=8:8:8:8, shift=24:16:8:0
    Console: switching to colour frame buffer device 128x48
    fb0: VESA VGA frame buffer device
    ACPI: AC Adapter [ADP1] (on-line)
    ACPI: Battery Slot [BAT1] (battery present)
    ACPI: Power Button (FF) [PWRF]
    ACPI: Lid Switch [LID]
    ACPI: Fan [FAN] (off)
    ACPI: Video Device [VGA] (multi-head: yes rom: yes post: no)
    ACPI: Processor [CPU0] (supports C1 C2)
    ACPI: Thermal Zone [THRM] (66 C)
    serio: i8042 AUX port at 0x60,0x64 irq 12
    serio: i8042 KBD port at 0x60,0x64 irq 1
    io scheduler noop registered
    io scheduler anticipatory registered
    io scheduler deadline registered
    io scheduler cfq registered
    elevator: using anticipatory as default io scheduler
    Floppy drive(s): fd0 is 1.44M
    FDC 0 is a post-1991 82077
    RAMDISK driver initialized: 16 RAM disks of 4096K size 1024 blocksize
    loop: loaded (max 8 devices)
    ACPI: PCI interrupt 0000:00:0f.0[A]: no GSI - using IRQ 11
    3c59x: Donald Becker and others. www.scyld.com/network/vortex.html
    0000:00:0f.0: 3Com PCI 3c905C Tornado at 0xfb00. Vers LK1.1.19
    Uniform Multi-Platform E-IDE driver Revision: 7.00alpha2
    ide: Assuming 33MHz system bus speed for PIO modes; override with idebus=xx
    PIIX4: IDE controller at PCI slot 0000:00:05.1
    PIIX4: chipset revision 1
    PIIX4: not 100% native mode: will probe irqs later
    ide0: BM-DMA at 0xfff0-0xfff7, BIOS settings: hda:DMA, hdb:pio
    ide1: BM-DMA at 0xfff8-0xffff, BIOS settings: hdc:DMA, hdd:pio
    Probing IDE interface ide0...
    hda: TOSHIBA MK1516GAP, ATA DISK drive
    ide0 at 0x1f0-0x1f7,0x3f6 on irq 14
    Probing IDE interface ide1...
    hdc: TOSHIBA DVD-ROM SD-C2402, ATAPI CD/DVD-ROM drive
    ide1 at 0x170-0x177,0x376 on irq 15
    Probing IDE interface ide2...
    ide2: Wait for ready failed before probe !
    Probing IDE interface ide3...
    ide3: Wait for ready failed before probe !
    Probing IDE interface ide4...
    ide4: Wait for ready failed before probe !
    Probing IDE interface ide5...
    ide5: Wait for ready failed before probe !
    hda: max request size: 128KiB
    hda: 23579136 sectors (12072 MB), CHS=23392/16/63, UDMA(33)
    hda: cache flushes not supported
    /dev/ide/host0/bus0/target0/lun0: p1 p2 < p5 p6 p7 p8 >
    hdc: ATAPI 24X DVD-ROM drive, 128kB Cache, UDMA(33)
    Uniform CD-ROM driver Revision: 3.20
    PCI: Enabling device 0000:00:0b.0 (0000 -> 0002)
    ACPI: PCI interrupt 0000:00:0b.0[A]: no GSI
    Yenta: CardBus bridge found at 0000:00:0b.0 [1179:0001]
    Yenta: ISA IRQ mask 0x0cb8, PCI irq 0
    Socket status: 30000007
    PCI: Enabling device 0000:00:0b.1 (0000 -> 0002)
    ACPI: PCI interrupt 0000:00:0b.1[b]: no GSI
    Yenta: CardBus bridge found at 0000:00:0b.1 [1179:0001]
    Yenta: ISA IRQ mask 0x0cb8, PCI irq 0
    Socket status: 30000007
    ohci_hcd: 2004 Nov 08 USB 1.1 'Open' Host Controller (OHCI) Driver (PCI)
    USB Universal Host Controller Interface driver v2.2
    ACPI: PCI interrupt 0000:00:05.2[D]: no GSI - using IRQ 11
    uhci_hcd 0000:00:05.2: Intel Corp. 82371AB/EB/MB PIIX4 USB
    uhci_hcd 0000:00:05.2: irq 11, io base 0xff80
    uhci_hcd 0000:00:05.2: new USB bus registered, assigned bus number 1
    hub 1-0:1.0: USB hub found
    hub 1-0:1.0: 2 ports detected
    sl811: driver sl811-hcd, 06 Dec 2004
    usb 1-2: new full speed USB device using uhci_hcd and address 2
    usbaudio: device 2 audiocontrol interface 0 has 1 input and 0 output AudioStreaming interfaces
    usbaudio: valid input sample rate 8000
    usbaudio: valid input sample rate 48000
    usbaudio: valid input sample rate 44100
    usbaudio: valid input sample rate 22050
    usbaudio: valid input sample rate 11025
    usbaudio: device 2 interface 1 altsetting 1: format 0x00000010 sratelo 8000 sratehi 48000 attributes 0x01
    usbaudio: valid input sample rate 8000
    usbaudio: valid input sample rate 48000
    usbaudio: valid input sample rate 44100
    usbaudio: valid input sample rate 22050
    usbaudio: valid input sample rate 11025
    usbaudio: device 2 interface 1 altsetting 2: format 0x80000010 sratelo 8000 sratehi 48000 attributes 0x01
    usbaudio: registered dsp 14,3
    usbaudio: constructing mixer for Terminal 2 type 0x0101
    usbaudio: registered mixer 14,0
    usb_audio_parsecontrol: usb_audio_state at d3cb6560
    usbcore: registered new driver audio
    drivers/usb/class/audio.c: v1.0.0:USB Audio Class driver
    Initializing USB Mass Storage driver...
    usbcore: registered new driver usb-storage
    USB Mass Storage support registered.
    usbcore: registered new driver hiddev
    usbcore: registered new driver usbhid
    drivers/usb/input/hid-core.c: v2.0:USB HID core driver
    mice: PS/2 mouse device common for all mice
    input: AT Translated Set 2 keyboard on isa0060/serio0
    input: ImExPS/2 Generic Explorer Mouse on isa0060/serio1
    Advanced Linux Sound Architecture Driver Version 1.0.6 (Sun Aug 15 07:17:53 2004 UTC).
    ACPI: PCI interrupt 0000:00:0c.0[A]: no GSI - using IRQ 11
    es1968: clocking to 48000
    unable to register OSS PCM device 0:0
    unable to register OSS mixer device 0:0
    ALSA device list:
    #0: ESS ES1978 (Maestro 2E) at 0xfc00, irq 11
    NET: Registered protocol family 2
    IP: routing cache hash table of 2048 buckets, 16Kbytes
    TCP: Hash tables configured (established 32768 bind 65536)
    NET: Registered protocol family 1
    NET: Registered protocol family 10
    IPv6 over IPv4 tunneling driver
    NET: Registered protocol family 17
    ACPI wakeup devices:
    USB VIY0 VIY1 MODM LAN LAN2 LID
    ACPI: (supports S0 S1 S3 S4 S4bios S5)
    ReiserFS: hda1: found reiserfs format "3.6" with standard journal
    ReiserFS: hda1: using ordered data mode
    ReiserFS: hda1: journal params: device hda1, size 8192, journal first block 18, max trans len 1024, max batch 900, max commit age 30, max trans age 30
    ReiserFS: hda1: checking transaction log (hda1)
    ReiserFS: hda1: Using r5 hash to sort names
    VFS: Mounted root (reiserfs filesystem) readonly.
    Mounted devfs on /dev
    Freeing unused kernel memory: 192k freed
    Adding 40120k swap on /dev/discs/disc0/part5. Priority:-1 extents:1
    Adding 128480k swap on /dev/discs/disc0/part7. Priority:-2 extents:1
    Adding 128480k swap on /dev/discs/disc0/part8. Priority:-3 extents:1
    Real Time Clock Driver v1.12
    drivers/usb/serial/usb-serial.c: USB Serial support registered for Generic
    usbcore: registered new driver usbserial_generic
    usbcore: registered new driver usbserial
    drivers/usb/serial/usb-serial.c: USB Serial Driver core v2.0
    Disabled Privacy Extensions on device c04cd0e0(lo)
    eth0: no IPv6 routers present
    ISO 9660 Extensions: Microsoft Joliet Level 3
    ISOFS: changing to secondary root
    usb 1-1: new full speed USB device using uhci_hcd and address 3
    scsi0 : SCSI emulation for USB Mass Storage devices
    usb-storage: device found at 3
    usb-storage: waiting for device to settle before scanning
    Vendor: LITE-ON Model: LTR-48246K Rev: SKS7
    Type: CD-ROM ANSI SCSI revision: 00
    sr0: scsi3-mmc drive: 207x/48x writer cd/rw xa/form2 cdda tray
    Attached scsi CD-ROM sr0 at scsi0, channel 0, id 0, lun 0
    Attached scsi generic sg0 at scsi0, channel 0, id 0, lun 0, type 5
    usb-storage: device scan complete
    cdrom: This disc doesn't have any tracks I recognize!
    cdrom: This disc doesn't have any tracks I recognize!
    scsi: unknown opcode 0x01
    ISO 9660 Extensions: Microsoft Joliet Level 3
    ISOFS: changing to secondary root
    NETDEV WATCHDOG: eth0: transmit timed out
    eth0: transmit timed out, tx_status 00 status e601.
    diagnostics: net 0cf6 media 8880 dma 0000003a fifo 8000
    eth0: Interrupt posted but not delivered -- IRQ blocked by another device?
    Flags; bus-master 1, dirty 1328(0) current 1328(0)
    Transmit list 00000000 vs. d3c97200.
    0: @d3c97200 length 8000049c status 0c01049c
    1: @d3c972a0 length 8000049c status 0c01049c
    2: @d3c97340 length 8000004d status 0c01004d
    3: @d3c973e0 length 8000004d status 0c01004d
    4: @d3c97480 length 80000049 status 0c010049
    5: @d3c97520 length 80000049 status 0c010049
    6: @d3c975c0 length 8000002a status 0001002a
    7: @d3c97660 length 80000049 status 0c010049
    8: @d3c97700 length 8000002a status 0001002a
    9: @d3c977a0 length 8000002a status 0001002a
    10: @d3c97840 length 8000004d status 0c01004d
    11: @d3c978e0 length 8000002a status 0001002a
    12: @d3c97980 length 8000002a status 0001002a
    13: @d3c97a20 length 8000002a status 0001002a
    14: @d3c97ac0 length 8000002a status 8001002a
    15: @d3c97b60 length 8000002a status 8001002a
    eth0: Resetting the Tx ring pointer.
    NETDEV WATCHDOG: eth0: transmit timed out
    eth0: transmit timed out, tx_status 00 status e601.
    diagnostics: net 0cf6 media 8880 dma 0000003a fifo 0000
    eth0: Interrupt posted but not delivered -- IRQ blocked by another device?
    Flags; bus-master 1, dirty 1344(0) current 1344(0)
    Transmit list 00000000 vs. d3c97200.
    0: @d3c97200 length 8000002a status 0001002a
    1: @d3c972a0 length 8000002a status 0001002a
    2: @d3c97340 length 8000002a status 0001002a
    3: @d3c973e0 length 8000002a status 0001002a
    4: @d3c97480 length 80000036 status 00010036
    5: @d3c97520 length 80000036 status 00010036
    6: @d3c975c0 length 80000036 status 00010036
    7: @d3c97660 length 80000036 status 00010036
    8: @d3c97700 length 80000036 status 00010036
    9: @d3c977a0 length 80000036 status 00010036
    10: @d3c97840 length 80000036 status 00010036
    11: @d3c978e0 length 80000047 status 0c010047
    12: @d3c97980 length 80000082 status 00010082
    13: @d3c97a20 length 8000004e status 0001004e
    14: @d3c97ac0 length 8000004e status 8001004e
    15: @d3c97b60 length 8000004e status 8001004e
    eth0: Resetting the Tx ring pointer.
    NETDEV WATCHDOG: eth0: transmit timed out
    eth0: transmit timed out, tx_status 00 status e601.
    diagnostics: net 0cf6 media 8880 dma 0000003a fifo 0000
    eth0: Interrupt posted but not delivered -- IRQ blocked by another device?
    Flags; bus-master 1, dirty 1360(0) current 1360(0)
    Transmit list 00000000 vs. d3c97200.
    0: @d3c97200 length 80000080 status 00010080
    1: @d3c972a0 length 8000004a status 0001004a
    2: @d3c97340 length 80000055 status 0c010055
    3: @d3c973e0 length 8000004a status 0001004a
    4: @d3c97480 length 8000002a status 0001002a
    5: @d3c97520 length 8000002a status 0001002a
    6: @d3c975c0 length 8000002a status 0001002a
    7: @d3c97660 length 80000055 status 0c010055
    8: @d3c97700 length 8000002a status 0001002a
    9: @d3c977a0 length 8000006c status 0001006c
    10: @d3c97840 length 80000036 status 00010036
    11: @d3c978e0 length 80000036 status 00010036
    12: @d3c97980 length 80000049 status 0c010049
    13: @d3c97a20 length 80000051 status 0c010051
    14: @d3c97ac0 length 80000049 status 8c010049
    15: @d3c97b60 length 80000051 status 8c010051
    eth0: Resetting the Tx ring pointer.
    NETDEV WATCHDOG: eth0: transmit timed out
    eth0: transmit timed out, tx_status 00 status e601.
    diagnostics: net 0cf6 media 8880 dma 0000003a fifo 0000
    eth0: Interrupt posted but not delivered -- IRQ blocked by another device?
    Flags; bus-master 1, dirty 1376(0) current 1376(0)
    Transmit list 00000000 vs. d3c97200.
    0: @d3c97200 length 80000049 status 0c010049
    1: @d3c972a0 length 80000051 status 0c010051
    2: @d3c97340 length 800000ae status 000100ae
    3: @d3c973e0 length 800000ae status 000100ae
    4: @d3c97480 length 80000042 status 00010042
    5: @d3c97520 length 8000006b status 0001006b
    6: @d3c975c0 length 8000006b status 0001006b
    7: @d3c97660 length 80000080 status 00010080
    8: @d3c97700 length 8000004e status 0001004e
    9: @d3c977a0 length 80000080 status 00010080
    10: @d3c97840 length 8000004e status 0001004e
    11: @d3c978e0 length 80000080 status 00010080
    12: @d3c97980 length 80000080 status 00010080
    13: @d3c97a20 length 8000006b status 0001006b
    14: @d3c97ac0 length 80000051 status 8c010051
    15: @d3c97b60 length 8000004a status 8001004a
    eth0: Resetting the Tx ring pointer.
    eth0: no IPv6 routers present
    When my network is down, the onlything to restore is is rebooting my computer. Restarting the network daemon does not work. Currently i don't have hotplug running, but with hotplug i encounter the same problem.
    I hope anybody can help me.
    Greetinx Atze

    It looks like the problem is with ACPI or you have a hardware IRQ conflict..
    check for a bios update for your Motherboard and check the bios settings.
    It might also help if you moved network card to another pci-slot.

  • [SOLVED] espeak won't work in crontab

    Espeak works fine from my command line.
    But I used to use espeak in my crontab to shout at me to take my pills etc etc. However, recently espeak has been generating errors when run by cronie
    Subject: Cron <robin@bunyip> sudo espeak 'robin take drugs'
    Content-Type: text/plain; charset=UTF-8
    Auto-Submitted: auto-generated
    Precedence: bulk
    X-Cron-Env: <LANG=en_GB.utf8>
    X-Cron-Env: <SHELL=/bin/sh>
    X-Cron-Env: <HOME=/home/robin>
    X-Cron-Env: <PATH=/usr/bin:/bin>
    X-Cron-Env: <LOGNAME=robin>
    X-Cron-Env: <USER=robin>
    Date: Thu, 9 Oct 2014 22:20:02 +0100 (BST)
    Status: R
    ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
    ALSA lib pcm_route.c:947:(find_matching_chmap) Found no matching channel map
    ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection r
    efused
    ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection r
    efused
    ALSA lib pcm_direct.c:1703:(snd1_pcm_direct_parse_open_conf) slave is not define
    d
    what do I need to do to allow cronie to shout at me?
    Edit: I added the sudo in the contab to try and eliminate permissions, but the same kind of errors are present for espeak hello as for sudo espeak hello.
    Edit: Seems that cronie can use espeak when vlc is not running.
    Last edited by replabrobin (2014-10-11 10:31:40)

    Spider.007 wrote:Is your user in the 'audio' group? Maybe vlc is somehow claiming the entire device to prevent sounds from distracting you from a movie? FWIW; it works fine here; even with vlc running
    Yes the user is in the audio group. I assume that vlc is somehow grabbing the device, but even when I changed to use alsa directly in the VLC audio settings this problem persisted.

  • [solved] espeak error - no sound

    hi archfriends,
    i installed espeak and mbrola and tried,
    #: espeak "hello"
    ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave
    ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
    ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused
    ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused
    ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
    connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory)
    attempt to connect to server failed
    Expression 'parameters->channelCount <= maxChans' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1513
    Expression 'ValidateParameters( outputParameters, hostApi, StreamDirection_Out )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1845
    Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838
    wave_open_sound > Pa_OpenStream : err=-9997 (Invalid sample rate)
    Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838
    Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838
    wave_open_sound > Pa_OpenStream : err=-9997 (Invalid sample rate)
    Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838
    Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838
    wave_open_sound > Pa_OpenStream : err=-9997 (Invalid sample rate)
    Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838
    Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838
    wave_open_sound > Pa_OpenStream : err=-9997 (Invalid sample rate)
    Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838
    there is no sound cant figure out whats wrong
    Last edited by xabit (2014-11-19 04:31:10)

    im glad that you wrote me moonswan,
    lspci -v|grep -i snd
    Kernel driver in use: snd_hda_intel
    Kernel modules: snd_hda_intel
    Kernel driver in use: snd_hda_intel
    Kernel modules: snd_hda_intel
    Kernel driver in use: snd_hda_intel
    Kernel modules: snd_hda_intel
    im on alsa alone, vlc and other prog. are working fine.
    my system is fresh installed no fancy stuff installed, just wine for playing games.
    wine sound works fine, but not together with non-wine application.
    if i play games on wine i get on vlc:
    Audio output failed:
    The audio device "sysdefault:CARD=PCH" could not be used:
    Device or resource busy.
    i dont know if that has something to do with my problem with espeak
    Last edited by xabit (2014-11-11 23:37:44)

  • Audacity crashing at startup

    Hello,
    I am trying to use audacity.  Can someone help me? I keep getting this error:
    ryanvade@ryan-linuux-desktop:~$ audacity -v
    ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave
    ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM bs2b
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM bs2b
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM bs2b
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM bs2b
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
    ALSA lib pcm_dmix.c:961:(snd_pcm_dmix_open) The dmix plugin supports only playback stream
    ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
    ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
    ALSA lib pcm_mmap.c:427:(snd_pcm_mmap) malloc failed: Cannot allocate memory
    Expression 'ret' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1809
    audacity: pcm_null.c:143: snd_pcm_null_drop: Assertion `null->state != SND_PCM_STATE_OPEN' failed.
    Aborted (core dumped)
    I am wondering if the issue is my ~/.asoundrc..
    # dmix - plug:dmix supports 1-8 channels, and does use dmix!
    # Whereas surround51 doesn't use dmix
    # http://bbs.archlinux.org/viewtopic.php?pid=745946#p745946
    # cat /proc/asound/card0/pcm0p/sub0/hw_params
    # Output to hw:0,0 to keep at 44.1k rather than dmix's 48k
    # 44.1k stops dmix from working, though.
    # From https://bugs.launchpad.net/debian/+source/sdl-mixer1.2/+bug/66483
    # Not needed.
    #defaults.pcm.dmix_max_periods -1
    #defaults.pcm.rate_converter "samplerate_best"
    # See /usr/share/alsa/pcm/dmix.conf
    #defaults.dmix.period_time 0
    #defaults.dmix.periods 4
    #defaults.pcm.surround51.device "0"
    # From https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1853
    # Posted at http://bbs.archlinux.org/viewtopic.php?id=95582
    # Is a dmix that actually works!
    pcm.dmixed {
    type asym
    playback.pcm {
    # See plugin:dmix at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
    type dmix
    ipc_key 5678293
    ipc_perm 0660
    ipc_gid audio
    #rate 48000 # 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 {
    channels 6
    pcm {
    format S16_LE
    rate 48000
    type hw
    card 1
    device 0
    subdevice 0
    # Play with this value, if you get errors "unable to set buffer size" or "underrun occured"
    # 4320 is effective minimum with hda-intel, but flash in firefox needs at least 5000.
    buffer_size 5000
    period_time 0
    #period_size 512
    #periods 2
    capture.pcm {
    # Dummy, but present to stop wine from moaning: ALSA lib pcm_asym.c:106:(_snd_pcm_asym_open) capture slave is not defined
    type null
    # Playing
    #pcm.!default {
    # type asym
    # playback.pcm "upmix_20to51_resample"
    # Check that e.g. Thief2 still works, if default is redefined.
    pcm.!default {
    type plug
    # Always output to all 6 channels, so the dmixer actually works if e.g. 6-channel is attempted to be started, while 2-channel is playing.
    slave.pcm "dmixed"
    pcm.!surround20 {
    type plug
    slave.pcm "dmixed"
    pcm.!surround40 {
    type plug
    slave.pcm "dmixed"
    pcm.!surround51 {
    type plug
    slave.pcm "dmixed"
    # If get error "Slave PCM not usable", then need to use plug:
    # If get error "Cannot find rate converter", then install libsamplerate and alsa-plugins
    # Lunar Linux: lin ladspa-bs2b
    # listplugins
    # analyseplugin bs2b
    pcm.bs2b {
    type ladspa
    path "/usr/lib/ladspa"
    plugins {
    0 {
    id 4221 # Bauer stereophonic-to-binaural (4221/bs2b)
    input {
    controls [ 700 6 ]
    # http://bbs.archlinux.org/viewtopic.php?id=95582
    slave.pcm "plug:dmixed"
    # speaker-test -D headphones -c 2 -t wav
    # audacious uses less CPU when running bs2b as a listed plugin, probably because of samplerate_best
    # Posted at http://bbs.archlinux.org/viewtopic.php?pid=626573#p626573
    pcm.headphones {
    type rate
    slave {
    pcm "plug:bs2b"
    rate 48000
    # Choices: samplerate_best samplerate_medium samplerate samplerate_order samplerate_linear
    converter "samplerate_best"
    hint {
    show on
    description "Headphones"
    pcm.ch51dup {
    slave.pcm "dmixed"
    slave.channels 6
    type route
    # Front and rear
    ttable.0.0 0.5
    ttable.1.1 0.5
    ttable.2.2 0.5
    ttable.3.3 0.5
    # 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
    # http://alsa.opensrc.org/SurroundSound
    # http://alsa.opensrc.org/index.php/Low-pass_filter_for_subwoofer_channel_(HOWTO)
    # Lunar: lin ladspa tap-plugins swh-plugins cmt-plugins libsamplerate
    # Fedora: yum install ladspa ladspa-blop-plugins ladspa-caps-plugins ladspa-cmt-plugins ladspa-swh-plugins ladspa-tap-plugins libsamplerate
    # Arch Linux: pacman -S ladspa blop swh-plugins libsamplerate tap-plugins cmt
    # For id 1672 - 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa), install blop-plugins
    # speaker-test -D upmix_20to51 -c 2 -t wav
    # Debugging: speaker-test -D plug:lowpass_21to21 -c 3 -t wav
    # listplugins
    # analyseplugin cmt
    # http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html
    # http://forums.gentoo.org/viewtopic-p-4528619.html#4528619
    pcm.lowpass_21to21 {
    type ladspa
    slave.pcm upmix_21to51
    # Set the path to ladspa, to fix this error:
    # Playback open error: -2,No such file or directory
    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 1052 # High-pass filter
    # policy none
    # input.bindings.0 "Input";
    # output.bindings.0 "Output";
    # input {
    # controls [ 300 ]
    #2 {
    # id 1052 # High-pass filter
    # policy none
    # input.bindings.1 "Input";
    # output.bindings.1 "Output";
    # input {
    # controls [ 300 ]
    #3 {
    # id 1051 # Low-pass filter.
    # policy none
    # input.bindings.2 "Input";
    # output.bindings.2 "Output";
    # input {
    # controls [ 300 ]
    # From http://alsa.opensrc.org/index.php/Low-pass_filter_for_subwoofer_channel_(HOWTO)
    # Can be used instead of 1-3 above.
    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 ]
    # speaker-test -D upmix_20to51 -c 2 -t wav
    # In audacious: upmix_20to51
    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
    # slave.rate 48000 makes CPU utilization 20% instead of 3%
    # Can't hear the difference with Audigy4 anyway.
    # slave.rate 44100 is 3%, so that proves audacious outputs 44100
    #slave.rate 48000
    #converter "samplerate"
    #slave.rate_converter "samplerate_best"
    # In audacious: upmix_20to51_resample
    # aplay -D upmix_20to51_resample ~/alsa/samplerate-test/udial.wav
    pcm.upmix_20to51_resample {
    type rate
    slave {
    pcm upmix_20to51
    #format S32_LE
    # Audigy4 upmixes to 48000 itself, and seems to use low-quality linear interpolation
    rate 48000
    # Choices: samplerate_best samplerate_medium samplerate samplerate_order samplerate_linear
    # 8% CPU with samplerate_medium - good choice
    converter "samplerate_medium"
    #converter "samplerate_linear"
    hint {
    show on
    description "20to51"
    # Debugging: speaker-test -D upmix_21to51 -c 3 -t wav
    pcm.upmix_21to51 {
    type plug
    # For ice1724:
    #slave.pcm surround51-ice
    # For Audigy:
    slave.pcm "dmixed"
    # http://bbs.archlinux.org/viewtopic.php?pid=745946#p745946
    #slave.pcm dmixed:6
    # For P5K ADI:
    #slave.pcm surround51-adi
    # Trying to pipe through Pulse Audio, to stop the clicks between songs.
    # Can't get Pulse Audio to work like this.
    #slave.pcm pulse
    # Don't need to specify the number of channels.
    slave.channels 6
    ttable {
    0.0 1 # front left
    1.1 1 # front right
    0.2 1 # rear left
    1.3 1 # rear right
    # Front left/right to center.
    # Imbalanced because is to the left of the monitor!
    # Would normally be 0.5 each.
    0.4 0.5
    1.4 0.5
    # Subwoofer, more powerful to compensate for bass-removal from other speakers.
    2.5 2
    # Channels are wrong way around in doom! This fixes them.
    # http://www.linuxforen.de/forums/archive/index.php/t-206470.html
    # http://forums.seriouszone.com/showthread.php?t=49869&page=10
    # http://forums.gentoo.org/viewtopic-p-4173170.html#4173170
    # For Audigy 4
    # Weird, doom3 has crappy sound if I add an alsa rate converter.
    # Posted at http://ubuntuforums.org/showthread.php?t=1304228
    pcm.doom-surround51 {
    slave.pcm "dmixed"
    slave.channels 6
    type route
    ttable.0.0 1
    ttable.1.1 1
    ttable.2.4 1
    ttable.3.5 1
    ttable.4.2 1
    ttable.5.3 1
    pcm.doom3-8930g {
    type plug
    slave.pcm {
    type dmix
    ipc_key 1093 # Must be unique
    ipc_key_add_uid false
    ipc_perm 0660
    slave {
    pcm "hw:0,0"
    rate 44100
    channels 2
    period_time 0
    period_size 1024
    buffer_time 0
    # Doom 3 wants buffer_size 8192
    # In ~/.doom3/base/autoexec.cfg
    # And ~/.quake4/q4base/autoexec.cfg
    # seta s_alsa_pcm "doom3-8930g"
    buffer_size 8192

    When you say "Audio Units Manager" are you talking about Logic's AU Manager, or some third party application? Is it the Granted Software one?
    If so, my versions of Waves 5 work well with them both. All that the Granted AU Manager does is move the component files (ie the Waveshell) between two directories to enable or disable the plugins, it won't affect Waves crashing at all.
    Without knowing what you did when things stopped working it's difficult to say, but it may well be a Waves installation, copy protection or authorisation issue, they can be notoriously odd behaving sometimes.
    Incidentally, why do you need two Waveshells? I think Logic only supports one of Waves' Waveshells at a time, at least it did in Logic 5, not sure about later ones.
    Not sure what else to suggest, really...

  • [Solved] No sound in either mplayer or vlc

    I am completely at a loss with this problem, when I play a video file on mplayer, let's say, I just type "mplayer <file>", things seem to set to their proper devices without any errors...
    Opening audio decoder: [mp3lib] MPEG layer-2, layer3
    AUDIO: 48000 Hz, 2 ch, s16le, 128.0 kbit/8.33% (ratio: 16000->192000)
    Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer3)
    ==================================================
    AO: [oss] 48000Hz 2ch s16le (2 bytes per sample)
    Starting playback...
    It makes even less sense since I'm using OSS with MPD and NCMPCPP runs freakin beautifully when it comes to music.  But trying to play even music with mplayer -- no sound, even though it appears loaded, I checked mute levels, using alsa, downloaded mplayer-w32codecs, tried it with alsa-plugins installed, used "-ao alsa," to try alsa then all others, and it chose oss again.  Most other posts I run across on google are like 2-3 yrs old, and no one here seems to be quite experiencing my same problems (which seems to be like every post I've been making recently! *grumble*)
    Last edited by Joshmotron (2009-03-13 13:09:09)

    Had to output to a file and open with mousepad to copy and paste, here's the verbose (didn't know this existed!)
    MPlayer SVN-r28928-4.3.3 (C) 2000-2009 MPlayer Team
    CPU vendor name: AuthenticAMD  max cpuid level: 1
    CPU: AMD Athlon(tm) 64 X2 Dual Core Processor 3800+ (Family: 15, Model: 43, Stepping: 1)
    extended cpuid-level: 24
    extended cache-info: 33587520
    Detected cache-line size is 64 bytes
    CPUflags:  MMX: 1 MMX2: 1 3DNow: 1 3DNowExt: 1 SSE: 1 SSE2: 1 SSSE3: 0
    Compiled for x86 CPU with extensions: MMX MMX2 3DNow 3DNowExt SSE SSE2 CMOV
    get_path('codecs.conf') -> '/home/josh/.mplayer/codecs.conf'
    Reading /home/josh/.mplayer/codecs.conf: Can't open '/home/josh/.mplayer/codecs.conf': No such file or directory
    Reading /etc/mplayer/codecs.conf: 137 audio & 295 video codecs
    Configuration: --prefix=/usr --confdir=/etc/mplayer --disable-gui --disable-runtime-cpudetection --enable-largefiles --enable-menu --disable-x264 --disable-mencoder
    CommandLine: '-v' 'e01.avi'
    init_freetype
    get_path('font/font.desc') -> '/home/josh/.mplayer/font/font.desc'
    font: can't open file: /home/josh/.mplayer/font/font.desc
    font: can't open file: /usr/share/mplayer/font/font.desc
    Using MMX (with tiny bit MMX2) Optimized OnScreenDisplay
    get_path('fonts') -> '/home/josh/.mplayer/fonts'
    Using nanosleep() timing
    get_path('input.conf') -> '/home/josh/.mplayer/input.conf'
    Can't open input config file /home/josh/.mplayer/input.conf: No such file or directory
    Parsing input config file /etc/mplayer/input.conf
    Input config file /etc/mplayer/input.conf parsed: 89 binds
    Setting up LIRC support...
    get_path('e01.avi.conf') -> '/home/josh/.mplayer/e01.avi.conf'
    Playing e01.avi.
    get_path('sub/') -> '/home/josh/.mplayer/sub/'
    [file] File size is 181786624 bytes
    STREAM: [file] e01.avi
    STREAM: Description: File
    STREAM: Author: Albeu
    STREAM: Comment: based on the code from ??? (probably Arpi)
    LAVF_check: AVI format
    AVI file format detected.
    list_end=0x229A
    ======= AVI Header =======
    us/frame: 41708  (fps=23.976)
    max bytes/sec: 0
    padding: 0
    MainAVIHeader.dwFlags: (272) HAS_INDEX IS_INTERLEAVED
    frames  total: 29879   initial: 0
    streams: 2
    Suggested BufferSize: 0
    Size:  512 x 384
    ==========================
    list_end=0x10F8
    ==> Found video stream: 0
    [aviheader] Video stream found, -vid 0
    ====== STREAM Header =====
    Type: vids   FCC: xvid (64697678)
    Flags: 0
    Priority: 0   Language: 0
    InitialFrames: 0
    Rate: 24000/1001 = 23.976
    Start: 0   Len: 29879
    Suggested BufferSize: 55855
    Quality 10000
    Sample size: 0
    ==========================
    Found 'bih', 40 bytes of 40
    ======= VIDEO Format ======
      biSize 40
      biWidth 512
      biHeight 384
      biPlanes 1
      biBitCount 12
      biCompression 1145656920='XVID'
      biSizeImage 1179648
    ===========================
    Regenerating keyframe table for MPEG-4 video.
    list_end=0x218E
    ==> Found audio stream: 1
    [aviheader] Audio stream found, -aid 1
    ====== STREAM Header =====
    Type: auds   FCC:  (0)
    Flags: 0
    Priority: 0   Language: 0
    InitialFrames: 1
    Rate: 48000/1152 = 41.667
    Start: 0   Len: 51924
    Suggested BufferSize: 768
    Quality -1
    Sample size: 0
    ==========================
    Found 'wf', 30 bytes of 18
    ======= WAVE Format =======
    Format Tag: 85 (0x55)
    Channels: 2
    Samplerate: 48000
    avg byte/sec: 27068
    Block align: 1152
    bits/sample: 0
    cbSize: 12
    mp3.wID=1
    mp3.fdwFlags=0x2
    mp3.nBlockSize=649
    mp3.nFramesPerBlock=1
    mp3.nCodecDelay=0
    ==========================================================================
    list_end=0x229A
    AVI: dmlh found (size=248) (total_frames=29879)
    list_end=0x22BA
    hdr=Software  size=11
    Software  : VirtualDub
    list_end=0xAC1D7D6
    Found movie at 0x280C - 0xAC1D7D6
    Reading INDEX block, 81803 chunks for 29879 frames (fpos=180475870).
    AVI index offset: 0x2808 (movi=0x280C idx0=0x4 idx1=0x18C)
    Auto-selected AVI audio ID = 1
    Auto-selected AVI video ID = 0
    AVI: Searching for audio stream (id:1)
    AVI video size=146059469 (29879) audio size=33731280 (51924)
    VIDEO:  [XVID]  512x384  12bpp  23.976 fps  937.6 kbps (114.5 kbyte/s)
    Auto-selected AVI audio ID = 1
    [V] filefmt:3  fourcc:0x44495658  size:512x384  fps:23.976  ftime:=0.0417
    Clip info:
    Software: VirtualDub
    get_path('sub/') -> '/home/josh/.mplayer/sub/'
    X11 opening display: :0.0
    vo: X11 color mask:  FFFFFF  (R:FF0000 G:FF00 B:FF)
    vo: X11 running at 1920x1200 with depth 24 and 32 bpp (":0.0" => local display)
    [x11] Detected wm supports NetWM.
    [x11] Detected wm supports FULLSCREEN state.
    [x11] Detected wm supports ABOVE state.
    [x11] Detected wm supports BELOW state.
    [x11] Current fstype setting honours FULLSCREEN ABOVE BELOW X atoms
    [VO_XV] Using Xv Adapter #0 (ATI Radeon AVIVO Video)
    [xv common] Drawing no colorkey.
    [xv common] Maximum source image dimensions: 4096x4096
    ==========================================================================
    Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
    INFO: libavcodec init OK!
    Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4)
    ==========================================================================
    ==========================================================================
    Opening audio decoder: [mp3lib] MPEG layer-2, layer-3
    dec_audio: Allocating 4608 + 65536 = 70144 bytes for output buffer.
    mp3lib: using SSE optimized decore!
    MP3lib: init layer2&3 finished, tables done
    MPEG 1.0, Layer III, 48000 Hz 128 kbit Joint-Stereo, BPF: 384
    Channels: 2, copyright: No, original: Yes, CRC: Yes, emphasis: 0
    AUDIO: 48000 Hz, 2 ch, s16le, 128.0 kbit/8.33% (ratio: 16000->192000)
    Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3)
    ==========================================================================
    Building audio filter chain for 48000Hz/2ch/s16le -> 0Hz/0ch/??...
    [libaf] Adding filter dummy
    [dummy] Was reinitialized: 48000Hz/2ch/s16le
    [dummy] Was reinitialized: 48000Hz/2ch/s16le
    Trying every known audio driver...
    ao2: 48000 Hz  2 chans  s16le
    audio_setup: using '/dev/dsp' dsp device
    audio_setup: using '/dev/mixer' mixer device
    audio_setup: using 'pcm' mixer device
    audio_setup: sample format: s16le (requested: s16le)
    audio_setup: using 2 channels (requested: 2)
    audio_setup: using 48000 Hz samplerate (requested: 48000)
    audio_setup: frags:  32/32  (2048 bytes/frag)  free:  65536
    AO: [oss] 48000Hz 2ch s16le (2 bytes per sample)
    AO: Description: OSS/ioctl audio output
    AO: Author: A'rpi
    Building audio filter chain for 48000Hz/2ch/s16le -> 48000Hz/2ch/s16le...
    [dummy] Was reinitialized: 48000Hz/2ch/s16le
    [dummy] Was reinitialized: 48000Hz/2ch/s16le
    Starting playback...
    Increasing filtered audio buffer size from 0 to 65536
    [ffmpeg] aspect_ratio: 1.333333
    VDec: vo config request - 512 x 384 (preferred colorspace: Planar YV12)
    Trying filter chain: vo
    VDec: using Planar YV12 as output csp (no 0)
    Movie-Aspect is 1.33:1 - prescaling to correct movie aspect.
    VO Config (512x384->512x384,flags=0,'MPlayer',0x32315659)
    VO: [xv] 512x384 => 512x384 Planar YV12
    VO: Description: X11/Xv
    VO: Author: Gerd Knorr <[email protected]> and others
    Xvideo image format: 0x32315659 (YV12) planar
    Xvideo image format: 0x30323449 (I420) planar
    Xvideo image format: 0x32595559 (YUY2) packed
    Xvideo image format: 0x59565955 (UYVY) packed
    using Xvideo port 131 for hw scaling
    *** [vo] Allocating (slices) mp_image_t, 512x384x12bpp YUV planar, 294912 bytes
    A:   0.0 V:   0.0 A-V:  0.024 ct:  0.000   1/  1 ??% ??% ??,?% 0 0 [J
    XXX initial  v_pts=0.000  a_pos=11856 (0.438)
    *** [vo] Allocating (slices) mp_image_t, 512x384x12bpp YUV planar, 294912 bytes
    get_path('subfont.ttf') -> '/home/josh/.mplayer/subfont.ttf'
    Unicode font: 5009 glyphs.
    get_path('subfont.ttf') -> '/home/josh/.mplayer/subfont.ttf'
    Unicode font: 5009 glyphs.
    A:   0.1 V:   0.0 A-V:  0.019 ct:  0.002   2/  2 ??% ??% ??,?% 0 0 [J
    *** [vo] Allocating (slices) mp_image_t, 512x384x12bpp YUV planar, 294912 bytes
    A:   0.1 V:   0.1 A-V: -0.001 ct:  0.002   3/  3 ??% ??% ??,?% 0 0 [J
    Uninit audio filters...
    [libaf] Removing filter dummy
    Uninit audio: mp3lib
    Uninit video: ffmpeg
    vo: uninit ...
    Exiting... (Quit)
    Last edited by Joshmotron (2009-03-11 00:14:26)

  • No sound in multiple applications with 5.1 upmix

    Hello,
    I already did some research about my problem but I did not find a solution for my problem yet. I configured my .asoundrc so that I have a permanent upmix from 2.0 to 5.1. But with the current configuration I cannot have multiple applications play sound at the same time.
    This is the latest .asoundrc i tried
    #define card
    pcm.snd_card {
    type hw
    card 0
    device 0
    ctl.snd_card {
    type hw
    card 0
    device 0
    # softwaremixing
    pcm.dmixer {
    type dmix
    ipc_key 1024
    ipc_perm 0666
    slave.pcm "snd_card"
    slave {
    period_time 0
    period_size 1024
    buffer_size 4096
    channels 6
    bindings {
    0 0
    1 1
    2 2
    3 3
    4 4
    5 5
    #dsnoop plugin
    pcm.dsnooper {
    type dsnoop
    ipc_key 2048
    ipc_perm 0666
    slave.pcm "snd_card"
    slave
    period_time 0
    period_size 1024
    buffer_size 4096
    channels 2
    bindings {
    0 0
    1 1
    #upmix
    pcm.ch51dup {
    type route
    slave.pcm dmixer
    slave.channels 6
    ttable.0.0 1
    ttable.1.1 1
    ttable.0.2 1
    ttable.1.3 1
    ttable.0.4 0.5
    ttable.1.4 0.5
    ttable.0.5 0.5
    ttable.1.5 0.5
    pcm.duplex {
    type asym
    playback.pcm "ch51dup"
    capture.pcm "dsnooper"
    pcm.!default {
    type asym
    slave.pcm "duplex"
    With this configuration Exaile gives an error on playback "Cannot open device for playback". VLC plays just stereo...
    What I want to have is this:
    Stereo should be upmixed to 5.1. But signals that already are 5.1 should not be changed. With pulseaudio on Ubuntu this was easy to do but I don't want to have pulseaudio at the moment. I had some trouble with it.
    I tried some .asoundrc from this thread but they did not work.
    Maybe someone can help me?

    For some reason it shows nothing. Sound is still not working but the output of mplayer is different now:
    [AO_ALSA] alsa-lib: conf.c:1645:(snd_config_load1) _toplevel_:41:26:Unexpected char
    [AO_ALSA] alsa-lib: conf.c:3425:(snd_config_hook_load) /home/florian/.asoundrc may be old or corrupted: consider to remove or fix it
    [AO_ALSA] alsa-lib: conf.c:3286:(snd_config_hooks_call) function snd_config_hook_load returned error: Invalid argument
    [AO_ALSA] alsa-lib: conf.c:3671:(snd_config_update_r) hooks failed, removing configuration
    [AO_ALSA] Playback open error: Invalid argument
    Failed to initialize audio driver 'alsa:device=ch51dup'
    Could not open/initialize audio device -> no sound.
    Audio: no sound
    Video: no video
    Exiting... (End of file)
    VLC shows the same error and Exaile says "konfigured audiosink 'bin1' does not work". But in .asoundrc there is no "bin1"...
    It's exactly the same .asoundrc you posted above.
    Edit:
    I noticed that nano messed the line breaks up. So longer comments started in a new line and were not commented of course. I fixed that. Now the output is:
    [AO_ALSA] alsa-lib: pcm_direct.c:936:(snd1_pcm_direct_initialize_slave) unable to set buffer size
    [AO_ALSA] alsa-lib: pcm_dmix.c:1030:(snd_pcm_dmix_open) unable to initialize slave
    [AO_ALSA] Playback open error: Invalid argument
    Failed to initialize audio driver 'alsa:device=ch51dup'
    Could not open/initialize audio device -> no sound.
    Audio: no sound
    Video: no video
    Edit2
    I set a different Buffer size in .asoundrc as the comment told.
    Now
    [AO_ALSA] Unable to get initial parameters: Invalid argument
    Failed to initialize audio driver 'alsa:device=ch51dup'
    Could not open/initialize audio device -> no sound.
    Audio: no sound
    Video: no video
    cat /proc/asound/card0/pcm0p/sub0/hw_params
    closed
    speaker-test -c 6 -D surround51 -t wav gives output on front left and front right And I read it might be a problem since kernel 2.6.34
    Last edited by McFlow (2010-09-07 12:36:59)

  • [MuseScore] crash on launch

    Hello,
    Are some people here using MuseScore ?
    http://musescore.org/
    Since last update from AUR (yesterday), the soft crash on launch. In terminal, i read "core dumped".
    Are you experimenting the same problem ?
    Regards,
    Pierre

    Hello. I've got the same problem with musescore 1.3
    global share: </usr/share/mscore-1.3/>
    configured localeName <system>
    real localeName <en_US>
    load translator </usr/share/mscore-1.3/locale/mscore_en_US>
    load translator <qt_en_US> from </usr/share/qt4/translations>
    load translator <qt_en_US> failed
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline
    ALSA lib pcm_dmix.c:961:(snd_pcm_dmix_open) The dmix plugin supports only playback stream
    ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
    using PortAudio Version: PortAudio V19-devel (built Feb 24 2012 12:00:16)
    connect to midi input <Midi Through Port-0>
    connect to midi input <Audigy MPU-401 (UART)>
    connect to midi input <Audigy MPU-401 #2>
    load soundfont </usr/share/mscore-1.3/sound/TimGM6mb.sf2>
    printer DPI 92.000000(92) display PDPI 92.000000(92) DPMM 3.622047
    LibraryPath: </usr/lib/qt4/plugins>
    LibraryPath: </usr/bin>
    Can't open "/home/jeck/.local/share/data/MusE/MuseScore/cookies.txt" to read cookies
    Plugin Path </home/jeck/.local/share/data/MusE/MuseScore/plugins>
    Plugin Path </usr/share/mscore-1.3/plugins>
    Register Plugin </usr/share/mscore-1.3/plugins/notenames.js>
    Segmentation fault (core dumped)
    program segfaults on launch. According to log, crash happens after loading notename.js plugin, but I don't know what to do with that.
    BTW musescore-git doesn't compile at all. At the very end of process it quits with error
    mv: cannot stat ‘manual/plugins_manual/plugins/*’: No such file or directory
    So I'll appreciate any help because I really need this program working asap.
    Thank you.

  • [SOLVED] (ish) Trine 2 not working, ksmserver issue ?

    I have Trine 2 that I got from the Humble Indie Bundle 9, I failed at "installing" it manually, because of some selinux nonsense, so I used the aur package "trine2-hib" to help me install it.
    It was working perfectly, until after some updates it didn't work anymore.
    This all started together with the problems caused by the latest update of kde stuff and qt4 not being updated, that solved, I though this would've been solved too, well, not really.
    Whenever I run Trine 2 from a .desktop launcher, it kills my kde session and shows me this: "Could not start ksmserver. Check your installation " and only an option to logout.
    trine2-hib.desktop content :
    [Desktop Entry]
    Version=1.0
    Type=Application
    Categories=Game;
    Name=Trine 2: Complete Story
    Comment=Frozenbyte
    GenericName=Action-Puzzle Platformer
    Icon=trine2
    Exec=trine2
    Terminal=false
    If I run it from a console, with the command "trine2" the game starts loading,  and then it just closes.
    Terminal output:
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.hdmi.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM hdmi
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.hdmi.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM hdmi
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.modem.0:CARD=0'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline:CARD=0,DEV=0
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.modem.0:CARD=0'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline:CARD=0,DEV=0
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.modem.0:CARD=0'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM phoneline
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.modem.0:CARD=0'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM phoneline
    ALSA lib dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib32/alsa-lib/libasound_module_pcm_pulse.so
    ALSA lib dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib32/alsa-lib/libasound_module_pcm_pulse.so
    ALSA lib pcm_dmix.c:961:(snd_pcm_dmix_open) The dmix plugin supports only playback stream
    ALSA lib dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib32/alsa-lib/libasound_module_pcm_pulse.so
    ALSA lib dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib32/alsa-lib/libasound_module_pcm_pulse.so
    Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1959
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2634
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2758
    Pa_OpenDefaultStream failed Invalid sample rate
    If I run it from a terminal with "kdesudo trine2" it works perfectly, but using this on a regular basis is less than ideal.
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.hdmi.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM hdmi
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.hdmi.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM hdmi
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.modem.0:CARD=0'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline:CARD=0,DEV=0
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.modem.0:CARD=0'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline:CARD=0,DEV=0
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.modem.0:CARD=0'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM phoneline
    ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'cards.CA0106.pcm.modem.0:CARD=0'
    ALSA lib conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
    ALSA lib conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM phoneline
    ALSA lib dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib32/alsa-lib/libasound_module_pcm_pulse.so
    ALSA lib dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib32/alsa-lib/libasound_module_pcm_pulse.so
    ALSA lib pcm_dmix.c:961:(snd_pcm_dmix_open) The dmix plugin supports only playback stream
    ALSA lib dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib32/alsa-lib/libasound_module_pcm_pulse.so
    ALSA lib dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib32/alsa-lib/libasound_module_pcm_pulse.so
    Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1959
    Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2634
    Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2758
    Pa_OpenDefaultStream failed Invalid sample rate <<This appears only when I close the game
    I don't think that any of that alsa nonsense has anything to do with this.
    I'm putting this on Multimedia as this is actually a problem with a game, however I don't think the game itself is what's causing the problems.
    Any ideas ?
    Thanks in advance.
    Last edited by LucetLux (2014-01-08 16:59:55)

    Sorry I lost track of this thread, I hope it isn't too late.
    Answering to your points;
    1; No it doesn't happen, if I run it like that it only closes after the loading screen, but it doesn't crash kde.
    2: If I run it on windowed mode it works just fine.
    3: I'm using the catalyst proprietary drivers, most other things I've tried work fine, far better than using the opensource drivers, that's why I'm using the proprietary ones. i've only had problems with some windows applications on wine that could be related to the drivers.
    4: I don't really know what could be considered of interest, but this looks like it's related:
    kwin(22944) KWin::Client::readUserTimeMapTimestamp: User timestamp, ASN: 4294967295
    kwin(22944) KWin::Client::readUserTimeMapTimestamp: User timestamp, final: 'ID: 85983238 ;WMCLASS: "trine2_linux_32bit" : "trine2_linux_32bit" ;Caption: "OpenGL test" ' : 194644424
    kwin(22944) KWin::Workspace::allowClientActivation: Activation, compared: 'ID: 85983238 ;WMCLASS: "trine2_linux_32bit" : "trine2_linux_32bit" ;Caption: "OpenGL test" ' : 194644424 : 194639746 : true
    kwin(22944) KWin::Workspace::updateClientArea: screens: 1 desktops: 4
    kwin(22944) KWin::Workspace::updateClientArea: Done.
    kwin(22944) KWin::Workspace::allowFullClientRaising: Raising: Belongs to active application
    kwin(22944) KWin::Workspace::updateClientArea: screens: 1 desktops: 4
    kwin(22944) KWin::Workspace::updateClientArea: Done.
    X Error: BadWindow (invalid Window parameter) 3
    Major opcode: 20 (X_GetProperty)
    Resource id: 0x5200006
    X Error: BadWindow (invalid Window parameter) 3
    Major opcode: 20 (X_GetProperty)
    Resource id: 0x5200006
    X Error: BadWindow (invalid Window parameter) 3
    Major opcode: 20 (X_GetProperty)
    Resource id: 0x5200006
    X Error: BadWindow (invalid Window parameter) 3
    Major opcode: 20 (X_GetProperty)
    Resource id: 0x5200006
    X Error: BadWindow (invalid Window parameter) 3
    Major opcode: 20 (X_GetProperty)
    Resource id: 0x5200006
    X Error: BadWindow (invalid Window parameter) 3
    Major opcode: 20 (X_GetProperty)
    Resource id: 0x5200006
    X Error: BadWindow (invalid Window parameter) 3
    Major opcode: 20 (X_GetProperty)
    Resource id: 0x5200006
    kwin(22944) KWin::Client::readUserTimeMapTimestamp: User timestamp, ASN: 4294967295
    Thanks for your help, and sorry for the long delay.
    Last edited by LucetLux (2014-01-04 01:00:55)

  • Problem with jack

    Hi,
    I'm trying to run my simple python script with using SpeechRecognition library, but i get a error when i try to listen on microphone:
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
    ALSA lib pcm_route.c:947:(find_matching_chmap) Found no matching channel map
    ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused
    connect(2) call to /dev/shm/jack-0/default/jack_0 failed (err=No such file or directory)
    ls /dev/shm/
    pulse-shm-1786803448* pulse-shm-2918752564* pulse-shm-528682629*
    pulse-shm-2134045932* pulse-shm-3577176364* pulse-shm-589288251*
    pulse-shm-2735481799* pulse-shm-40907746* pulse-shm-899524684*
    pulse-shm-2827064006* pulse-shm-4260734428* sem.lastpassffsemaphore*
    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=PCH
    HDA Intel PCH, ALC663 Analog
    Default Audio Device
    front:CARD=PCH,DEV=0
    HDA Intel PCH, ALC663 Analog
    Front speakers
    surround21:CARD=PCH,DEV=0
    HDA Intel PCH, ALC663 Analog
    2.1 Surround output to Front and Subwoofer speakers
    surround40:CARD=PCH,DEV=0
    HDA Intel PCH, ALC663 Analog
    4.0 Surround output to Front and Rear speakers
    surround41:CARD=PCH,DEV=0
    HDA Intel PCH, ALC663 Analog
    4.1 Surround output to Front, Rear and Subwoofer speakers
    surround50:CARD=PCH,DEV=0
    HDA Intel PCH, ALC663 Analog
    5.0 Surround output to Front, Center and Rear speakers
    surround51:CARD=PCH,DEV=0
    HDA Intel PCH, ALC663 Analog
    5.1 Surround output to Front, Center, Rear and Subwoofer speakers
    surround71:CARD=PCH,DEV=0
    HDA Intel PCH, ALC663 Analog
    7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
    I read other threads about it, and that don't resolve my problems, maybe someone have the same problem...
    I will be very gratefull for any help.
    Last edited by neptyd (2014-09-11 09:56:00)

    neptyd wrote:I read other threads about it, and that don't resolve my problems, maybe someone have the same problem...
    I will be very gratefull for any help.
     Which threads have you read? Link them here for reference. You apparently use pulseaudio, which you do not mention. What simple python script is that? Does is take any arguments? Do you have pulseaudio-alsa installed?
    arecord -lL;
    lspci -vvnn | grep -A1 '040[1-3]'; cat /proc/asound/modules; amixer;
    lsusb #if there is a usb sound card;
    # or use the alsa-info script

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