[gst-devel] audio problem with gstreamer and live555

[Alessio]

Hi all

sorry for my bad english

     since the gstreamer rtspsrc module in windows doesn't support rtscp 
tcp interleaved mode, I implemented a rtsp plugin based on live555.

I based my work on how both vlc and mplayer live555 based modules are made.

I've a src plugin that manages the network connection and reads the 
frames one by one, and a demuxer plugin that receives that frames and 
creates a src pad for each subsession, based on the standard behaviour 
of gstreamer demuxer.

In the application I set up a basic pipeline based like that:

gst-launch livertsp uri=rtsp://127.0.0.1:8554/bighic320x240.avi ! 
livedemuxer name=demux  demux.audio_00 ! queue ! decodebin  ! 
audioconvert ! audioresample ! autoaudiosink   demux.video_00 ! queue ! 
decodebin ! ffmpegcolorspace ! videoscale ! autovideosink

where livertsp and livedemuxer are the two plugins I wrote.

but when I try to play a stream i see the video but i didn't hear the audio.

The behaviour in the same both on a C application and trough gst-launch, 
both in windows and in linux.

Sometimes (randomly) I hear some little and distubed frame of audio, for 
less than a second.

I checked trough a probe plugin I wrote what is received by the audio 
sink, and it to receive correctly all the frames, it is exactly the same 
that is received when I play the same file in local, with the same 
pipeline using filesrc and avidemuxer in place of my plugins, both the 
data and the timestamp and duration of the frames are correct.

I also tried to save the audio output using filesink in place of the 
audiosink and the result is the same for the rtsp and the file case, so 
the audio is received by the sink but isn't played.

In the attached file there is the output of the pipeline launched with 
gst-launch, where everything seems allright to me.

Anybody can help me?

Thanks in advance, Alessio
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