Hi, sorry for double posting, I've forgot to add a few comments in my previous email. <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 \ </blockquote><div> Are the audioresample/audioconvert necessary? What if you use pulsesrc / pulsesink and remove them? </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> <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,</blockquote><div> here clock-rate is the RTP clock, you should set it to a value according with your sample rate, check the caps negotiated from the source element on the sender side to have an hint. You can do this using gst-launch -v </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">payload=96" port=5002 \ ! rtpbin.recv_rtp_sink_1 \ rtpbin. \ ! rtpmp4adepay \ ! decodebin \ </blockquote><div> you don't need decodebin here. Just put the decoder element. </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> ! audioconvert \ ! audioresample \ ! alsasink sync=true \ </blockquote><div> Again, try replacing audioconvert/audioresample/alsasink with a plain pulsesink (if you can ;) ). Regards, Marco </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> 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. 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>