[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
--
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.
More information about the gstreamer-devel
mailing list