<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">Hi Terry, <br><br>Is your receiver actually receiving stream ?<br>
You have to specify caps for faad to decode stream.<br>
Use gst-launch -v on your sender to get "application/x-rtp" caps used
for aac and report them as a caps filter in your receiver.<br>
gst-launch -vm udpsrc port=19790 ! "application/x-rtp, media=audio,
clock-rate=8000, encoding-name=MPEG4-GENERIC, ....." ! rtpmp4gdepay !
faad ! fakesink<br>
<br>
Also, your aac maybe stereo and amrnbenc only supports mono, so you
have to use deinterleave.<br>
<br>Aurelien<br><br>--- En date de : <b>Lun 25.8.08, Terry Leung <i><terry83@gmail.com></i></b> a écrit :<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">De: Terry Leung <terry83@gmail.com><br>Objet: [gst-devel] convert aac rtp stream to amr stream<br>À: gstreamer-devel@lists.sourceforge.net<br>Date: Lundi 25 Août 2008, 6h05<br><br><pre>Hi all,<br><br>I am trying to convert a aac rtp stream to a amr(nb) stream<br><br>Here is the pipleine i am using<br><br>generate aac stream from a 3gp file:<br>gst-launch -m filesrc location=test-aac.3gp ! qtdemux ! rtpmp4gpay !<br>udpsink host=172.20.122.9 port=19790<br><br>receive and convert the stream:<br>gst-launch -m udpsrc port=19790 ! rtpmp4gdepay ! faad ! queue !<br>amrnbenc ! rtpamrpay pt=98 ! udpsink host=172.20.122.23 port=2006<br><br>I have tried to capture those packet, I can see that there is acc rtp<br>packet generated by the
first pipleine but it dont have packet come<br>out from the second piple<br>Also, I am not sure if it is related to the udp checksum<br>The udp checksum generated by the first pipeline is wrong in wire shark<br>I also attach the capture file<br><br><br>I also have another two pipeline doing similar job but it is for g711 to amr<br>those two pipeline work gracefully<br><br>gst-launch -m filesrc location=test3.wav ! wavparse ! audioconvert !<br>audioresample !alawenc ! rtppcmapay ! udpsink host=172.20.122.9<br>port=19790<br><br>gst-launch -m udpsrc port=19790 ! rtppcmadepay ! alawdec ! queue !<br>amrnbenc ! rtpamrpay pt=98 ! udpsink host=172.20.122.23 port=2006<br><br>The difference from the first pipeline is that the udpsink of the<br>second piple change its state to playing when rtp stream arrive from<br>the udpsrc<br><br>Anyone have idea to solve
this?<br><br>Terry<br></pre><pre>-------------------------------------------------------------------------<br>This SF.Net email is sponsored by the Moblin Your Move Developer's<br>challenge<br>Build the coolest Linux based applications with Moblin SDK & win great<br>prizes<br>Grand prize is a trip for two to an Open Source event anywhere in the world<br>http://moblin-contest.org/redirect.php?banner_id=100&url=/</pre><pre>_______________________________________________<br>gstreamer-devel mailing list<br>gstreamer-devel@lists.sourceforge.net<br>https://lists.sourceforge.net/lists/listinfo/gstreamer-devel<br></pre></blockquote></td></tr></table><br>
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