From: Marco B. <gib...@gm...> - 2010-11-20 09:22:32
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Hi, a few notes: On Fri, Nov 19, 2010 at 5:08 PM, Wes Miller <wm...@sd...> wrote: > > I have written simple pipes to encode an audiotestsrc sine wave into aac then > mp4a payloadit and send it over the network to a matching udpsik, depay, > decode and pulsesink. > > The sound comes out choppy, uniformly spaced, but chopped into beeps. > Sounds a lot like Morse code but it's all E's. > > Here are the pipes: (they are on the same PC but I've also tried it between > 2 computers) > > Source > > gst-launch audiotestsrc \ > ! audio/x-raw-int, rate=44100, channels=2, endianness=1234, > width=16, depth=16 \ > ! ffenc_aac \ > ! rtpmp4apay \ > ! udpsink host=10.253.5.151 port=5002 > > and Sink > > gst-launch-0.10 -v -e udpsrc port=5002 I usually use gstrtpbin for this kind of pipelines. Btw I reas it did not help in your case. Did you build the pipeline as reported in the gstrtpbin examples? > caps=application/x-rtp,media=audio,clock-rate=90000, > encoding-name=MP4A-LATM,cpresent=0, \ > > config=NULL,payload=96,ssrc=3574762534,clock-base=2565233379,seqnum-base=29343 you don't need to explicitly set ssrc, clock-base and seqnum-base. Very likely the same receiving pipeline does not work across two separate sessions. > \ > ! gstrtpjitterbuffer \ dd you try to increase the latency value of the jitter buffer? You can find a proper one by tcpdump-ing the traffic, measuring the maximim jitter with e.g. wireshark and multiplying it by, let's say, a value between 2 and 3. > ! rtpmp4adepay \ > ! > audio/mpeg,channels=2,rate=44100,mpegversion=4,stream-format=raw,codec_data=\(buffer\)1210 > ! ffdec_aac > ! pulsesink As a trest, what if you set sync=false in pulsesink? > > NOTE: Actually sounds better without the jitter buffer. This is pretty strange, even though "better" is somewhat a subjective issue. > > I tried modifying the source pipe to do ... ffenc-aac ! ffdec_aac ! > pulsesink. Sound great. I'm pretty sure that means the network or > network elements are my problem. Probably yes, but it's unlikely in case you've a very good quality network (e.g. ethernet over wire and very few clients active on it). In any cases, you definitely need a jitter buffer. Regards, Marco > > So, anyone have suggestions for getting better sound quality across the net? > I know it's possible; other programs do it every day. > > ps. I have also coded the receiver in C, with and without gstrtpbin. Does > not help. > > Warmly, > > Wers > > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/udpsink-to-udpsrc-choppy-stacato-sine-wave-tp3050472p3050472.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Beautiful is writing same markup. Internet Explorer 9 supports > standards for HTML5, CSS3, SVG 1.1, ECMAScript5, and DOM L2 & L3. > Spend less time writing and rewriting code and more time creating great > experiences on the web. Be a part of the beta today > http://p.sf.net/sfu/msIE9-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gst...@li... > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |