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After switching my system over from ALSA to Pulseaudio I have noticed weird sound distortions whenever a skype event sounds until the skype event is done playing. This only seems to happen with skype because I can be listening to music and have a flash video going, or knotify4 can play a sound while amarok is playing and the sound doesn't distort. I have also noticed this distortion during a test call on skype, I don't know if this effects just audio output or if the capture is also affected. Has anyone else noticed this?
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Yes sound distortion does occur with skype. The events you note are not comparable. All the other events produce sound by waiting for the entire sound to be in memory and will pause to reload if not. Skype has multiple ways to distort audio and, in my experience, uses them all.
Skype is not a good voip service but is an excellent distributed directory service. The reason that I use skype is not for the audio quality, as that is extremely poor, but for the easy integration into the larger telephone network.
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Yea I know skype is not known for good audio quality, but need it to communicate with customers at my job, skype is not my first choice... The switch from ALSA to Pulse has made the sound quality go down considerably, I now have a crackling noise where I didn't when I was using ALSA. I have not noticed this on any of my other machines running pulseaudio for its sound server and using skype which is why I thought to ask.
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From my experience with Skype, it seems to prefer buffer_size 16384 to stop crackling. See the ALSA config in my sig.
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All I can say is that I've noticed this problem too but only with the internal soundcard. Events like rigging, file transfer and whatnot most of the times have distortion but calls over an usb headset always have crystal clear sound (if the connection is fast enough and direct).
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All I can say is that I've noticed this problem too but only with the internal soundcard. Events like rigging, file transfer and whatnot most of the times have distortion but calls over an usb headset always have crystal clear sound (if the connection is fast enough and direct).
Yea I'm sure it wouldn't be so bad if I had a USB headset but I don't XP
@brebs: if it was a problem with the buffer_size why would this not happen before when I was using ALSA instead of when the box is running Pulseaudio?
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Well have you checked what buffer_size ALSA & pulseaudio use? It differs depending on the hardware and app, I think.
cat /proc/asound/card0/pcm0p/sub0/hw_params
Last edited by brebs (2011-08-30 17:20:52)
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/proc/asound/card0/pcm0p/sub0/hw_params
/proc/asound/card0/pcm1p/sub0/hw_params
/proc/asound/card1/pcm0p/sub0/hw_params
all report as closed does something need to be playing for this to work?
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Well my buffer_size is 16384 so I'm not sure if that is the cause
└─(/home/vendion)─> cat /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 8192
buffer_size: 16384Last edited by vendion (2011-08-31 02:54:16)
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That's a very large period_size - normally it's 256, 512, 1024, or 2048.
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I only mentioned a usb headset, it can also be a cheap usb soundcard where you can plug your "normal" headset. If you are using an integrated soundcard (desktop/laptop) you might experience a worse sound playback quality but a greatly improved mic recording quality, as with everything it depends on the quality of the hardware you have.
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That's a very large period_size - normally it's 256, 512, 1024, or 2048.
How would I go about changing this? I don't know where such a setting would be for Pulseaudio and my asound.conf is just this:
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}I can't find any way to change this in pulse settings under /etc/pulse/
Last edited by vendion (2011-09-20 12:15:49)
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