[sip-comm-dev] javasound bufferlength problem on Linux


#1

I'm finding that on my Linux machine, if I don't comment out the following code:

// // 1. Changing buffer size. The default buffer size (for javasound)
// // is 125 milliseconds - 1/8 sec. On MacOS this leeds to exception and
// // no audio capture. 30 value of buffer fix the problem and is ok
// // when using some pstn gateways
// // 2. Changing to 60. When it is 30 there are some issues
// // with asterisk and nat(we don't start to send stream and so
// // asterisk rtp part doesn't notice that we are behind nat)
// Control ctl = (Control)
// dataSource.getControl("javax.media.control.BufferControl");
//
// if(ctl != null)
// {
// ((BufferControl)ctl).setBufferLength(60);//buffers in
// }

That I get an error:

Cannot open audio device for input: javax.sound.sampled.LineUnavailableException: line with format PCM_SIGNED 44100.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian not supported.
Failed to configure: com.sun.media.ProcessEngine@184be29
  IO exception: line with format PCM_SIGNED 44100.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian not supported.

And the media service won't start.

I'm using ubuntu gutsy gibbon with Sun JDK 1.6.

Ken

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