<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 12 (filtered medium)"><style></style></head><body lang=NL-BE link=blue vlink=purple><div class=WordSection1>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.<o:p></o:p><o:p> </o:p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>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<o:p></o:p></div></div><o:p> </o:p>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?<o:p></o:p><o:p> </o:p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>From: sumit kumar [mailto:sumit.git@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<o:p></o:p></div><o:p> </o:p>Hi,<o:p></o:p><div><o:p> </o:p></div><div>Try with directsoundsink.. else try setting sync=false for autoaudiosink<o:p></o:p></div><div><o:p> </o:p></div><div>-Sumit<o:p></o:p><div>On Tue, Dec 7, 2010 at 3:55 AM, Callewaert Sven <<a href="mailto:S.Callewaert@televic.com">S.Callewaert@televic.com</a>> wrote:<o:p></o:p><div><div>Hi,<o:p></o:p> <o:p></o:p>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.<o:p></o:p> <o:p></o:p>This is the sender pipeline:<o:p></o:p>gst-launch -v audiotestsrc  ! audioconvert ! rtpL16pay ! udpsink host=224.0.0.7 auto-multicast=true port=4444<o:p></o:p> <o:p></o:p>This is the receiver pipeline:<o:p></o:p>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<o:p></o:p> <o:p></o:p>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?<o:p></o:p> <o:p></o:p></div></div> ------------------------------------------------------------------------------ What happens now with your Lotus Notes apps - do you make another costly upgrade, or settle for being marooned without product support? Time to move off Lotus Notes and onto the cloud with Force.com, apps are easier to build, use, and manage than apps on traditional platforms. Sign up for the Lotus Notes Migration Kit to learn more. <a href="http://p.sf.net/sfu/salesforce-d2d" target="_blank">http://p.sf.net/sfu/salesforce-d2d</a> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><o:p></o:p></div><o:p> </o:p></div></div></body></html>