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

[Wim Taymans]

On Fri, 2010-11-19 at 07:08 -0800, Wes Miller 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.

The clock rate on the receiver looks wrong. You should copy and paste
the caps from the sender to the receiver without changing them. If you
change them, the receiver has no idea what kind of data it is dealing
with and it will not work.

Wim 

> 
> 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
> 
> 
>