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<DIV><FONT face="MS UI Gothic" size=2>Hi,</FONT></DIV>
<DIV><FONT face="MS UI Gothic" size=2></FONT> </DIV>
<DIV><FONT face="MS UI Gothic" size=2>Try the following commandline please, may
you heard the sound.</FONT></DIV>
<DIV><FONT face="MS UI Gothic" size=2></FONT> </DIV>
<DIV>gst-launch -v rtspsrc location = "rtsp://192.168.1.10/testStream.mp3" !
rtpmpadepay ! queue ! mad ! alsasink sync=false<BR></DIV>
<DIV><FONT face="MS UI Gothic" size=2>Regards.</FONT></DIV>
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style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt MS UI Gothic">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt MS UI Gothic; font-color: black"><B>From:</B>
<A title=kanzakiv@gmail.com href="mailto:kanzakiv@gmail.com">聂巍</A> </DIV>
<DIV style="FONT: 10pt MS UI Gothic"><B>To:</B> <A
title=gstreamer-devel@lists.sourceforge.net
href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</A>
</DIV>
<DIV style="FONT: 10pt MS UI Gothic"><B>Sent:</B> Saturday, April 28, 2007
2:43 PM</DIV>
<DIV style="FONT: 10pt MS UI Gothic"><B>Subject:</B> [gst-devel] gstreamer
develop---stream a mp3 file via rtsp & rtp</DIV>
<DIV><BR></DIV>I use gstreamer to stream a mp3 file via rtsp &
rtp<BR>and I only heard the sound for about
half a second and it stops at once
<BR><BR>The console displays :<BR>WARNING: Element "alsasink0"
warns:<BR>gstbaseaudiosink.c (696): gst_base_audio_sink_render ():
/pipeline0/alsasink0:<BR>Unexpected discontinuity in audio timestamps of more
than half a second (0:00:00.803061224), resyncing<BR><BR><BR>My commadline is
: gst-launch rtspsrc location = "rtsp://192.168.1.10/testStream.mp3" !
rtpmpadepay ! queue ! mad ! alsasink <BR><BR>would anybody tell me where the
problem is ?<BR>and how can i solve it ~~<BR>thank you guys in advance
:)<BR><BR>
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