[gst-devel] GStreamer audio streaming on Windows

Callewaert Sven S.Callewaert at TELEVIC.com
Thu Dec 9 09:33:53 CET 2010


The problem of the distortion on the audio stream appears to be linked to the rtpL16pay payloader. When I use the rtpmp4apay payloader the problem disappears.

From: Callewaert Sven
Sent: dinsdag 7 december 2010 9:53
To: 'Discussion of the development of GStreamer'
Subject: RE: [gst-devel] GStreamer audio streaming on Windows

Replacing the autoaudiosink element with directsoundsink did the trick. Now I hear a continuous beep but there is a lot of distortion on it. When I stream the microphone by replacing audiotestsrc with directsoundsrc on the sender side, I also hear a lot of distortion. The distortion however only occurs when the microphone captures moderate to loud noises. There is no distortion with quiet noises. Could it be a signed/unsigned issue? Can I force the sender the use signed=false in some way?

From: sumit kumar [mailto:sumit.git at gmail.com]
Sent: dinsdag 7 december 2010 8:18
To: Discussion of the development of GStreamer
Subject: Re: [gst-devel] GStreamer audio streaming on Windows

Hi,

Try with directsoundsink.. else try setting sync=false for autoaudiosink

-Sumit
On Tue, Dec 7, 2010 at 3:55 AM, Callewaert Sven <S.Callewaert at televic.com<mailto:S.Callewaert at televic.com>> wrote:
Hi,

I'm experimenting a bit with GStreamer (ossbuild 0.10.7) on Windows, but I can't seem to make audio streaming between two computers work. All I hear at the receiver side is a short beep followed by silence.

This is the sender pipeline:
gst-launch -v audiotestsrc  ! audioconvert ! rtpL16pay ! udpsink host=224.0.0.7 auto-multicast=true port=4444

This is the receiver pipeline:
gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444 caps="application/x-rtp, media=(string)audio, channels=(int)1, clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay ! audioconvert ! queue ! autoaudiosink

I've already tried different queue settings and codecs. Same thing when I try to stream an audio file, all I hear is about 1 second of it. What could be the problem?


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