Hi, <div class="gmail_quote">On Mon, Oct 18, 2010 at 9:03 PM, Wes Miller <<a href="mailto:wmiller@sdr.com">wmiller@sdr.com</a>> wrote: <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> Marco et al, Ok, new pipelines with rtpbin. These work though I am not sure they are completely correct. SENDER <><><><><><><><><><><><><><><><> gst-launch-0.10 -v gstrtpbin name=rtpbin \ alsasrc do-timestamp=true \ ! queue \ ! audioresample \ ! audioconvert \ <div class="im"> ! dmaienc_aac outputBufferSize=10000000 outputformat=2 \ </div> ! rtpmp4apay \ ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 \ ! udpsink port=5002 host=$1 ts-offset=0 name=artpsink \ rtpbin.send_rtcp_src_1 \ ! udpsink port=5003 host=$1 sync=false async=false name=artcpsink \ udpsrc port=5007 name=artpsrc \ ! rtpbin.recv_rtcp_sink_1 RECEIVER <><><><><><><><><><><><><><><> gst-launch-0.10 -v gstrtpbin name=rtpbin latency=2000 \ udpsrc caps="application/x-rtp,media=audio,clock-rate=44100,encoding-name=MP4A-LATM,payload=96" port=5002 \ ! rtpbin.recv_rtp_sink_1 \ rtpbin. \ ! rtpmp4adepay \ ! decodebin \ ! audioconvert \ ! audioresample \ ! alsasink sync=true \ udpsrc port=5002 \ ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 \ ! udpsink port=5003 host=$1 sync=false async=false I am still getting playback that is too fast and choppy. Sounds like the receiver pipe is just playing each packet too fast. I don't think I'm loosing any of the music, it just plays each packet too fast and has a dead spot in between them. Imagine the Moody Blues sung by the Chipmunks with hiccups. </blockquote><div> Ok now I understand what's the issue. If it appears to be between elves or ogres either you're reading the Lord of Rings or the sample rate differs between encoder and decoder. You may try and explicitly set the RTP clock in the receiving pipeline with something lik: <pre class="programlisting"> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)MPEG4" </pre>Even though these should already be the default values. Another (more likely) possibility is that the encoder you're using is working at a default sample rate different from the one of the decoder. You should thus checker whether any compile/run time options make it possible for you to use the same values. Regards </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> So, first, do I haave the ports mapped correctly? Am I missing any connections between the rtpbin in's and out's? Secondly, is there a retimesynch trick I can use? Replacing alsasink with fakesink -v tells me that the packets are timestamped and that the durations match the time between timestamps. I really intend to replace the mp3 player I am using with a microphone. The Leopardboard only has linelevel input so I can't use a mic right now. If passing speech, would I be more likely to hear the speed=up and gaps? Will speex through an rtpxxxpayloader still cause me to need 4k of Caps? Thanks, Wes -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/Choppy-Audio-over-UDP-tp2997741p3000742.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/Choppy-Audio-over-UDP-tp2997741p3000742.html</a> <div><div></div><div class="h5">Sent from the GStreamer-devel mailing list archive at Nabble.com. ------------------------------------------------------------------------------ Download new Adobe(R) Flash(R) Builder(TM) 4 The new Adobe(R) Flex(R) 4 and Flash(R) Builder(TM) 4 (formerly Flex(R) Builder(TM)) enable the development of rich applications that run across multiple browsers and platforms. Download your free trials today! <a href="http://p.sf.net/sfu/adobe-dev2dev" target="_blank">http://p.sf.net/sfu/adobe-dev2dev</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> </div></div></blockquote></div>