[gst-devel] udpsink to udpsrc choppy - stacato sine wave

Wes Miller wmiller at sdr.com
Fri Nov 19 16:08:01 CET 2010


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 
             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 
\
           ! gstrtpjitterbuffer \
           ! rtpmp4adepay \
           !
audio/mpeg,channels=2,rate=44100,mpegversion=4,stream-format=raw,codec_data=\(buffer\)1210
           ! ffdec_aac 
           ! pulsesink

NOTE:  Actually sounds better without the jitter buffer.

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.

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


      
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