[gst-devel] gstreamer develop---stream a mp3 file via rtsp & rtp

[Fu Hao]

Hi,

Try the following commandline please, may you heard the sound.

gst-launch -v rtspsrc location = "rtsp://192.168.1.10/testStream.mp3" !
rtpmpadepay ! queue ! mad ! alsasink sync=false

Regards.
  ----- Original Message ----- 
  From: 聂巍
  To: gstreamer-devel at lists.sourceforge.net
  Sent: Saturday, April 28, 2007 2:43 PM
  Subject: [gst-devel] gstreamer develop---stream a mp3 file via rtsp & rtp


  I use gstreamer to stream a mp3 file via rtsp & rtp
  and I only heard the sound for about half a second and it stops at once

  The console displays :
  WARNING: Element "alsasink0" warns:
  gstbaseaudiosink.c (696): gst_base_audio_sink_render ():
/pipeline0/alsasink0:
  Unexpected discontinuity in audio timestamps of more than half a second
(0:00:00.803061224), resyncing


  My commadline is : gst-launch rtspsrc location =
"rtsp://192.168.1.10/testStream.mp3" ! rtpmpadepay ! queue ! mad ! alsasink

  would anybody tell me where the problem is ?
  and how can i solve it ~~
  thank you guys in advance :)




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