issue with the rtp streaming playback pipeline

[ashokm]

Hi,
I am developing an app to play the local ogg file and also the rtp streaming
playback.
I am using the adder, volume, and alsasink in the one thread, and using
audiobins for each local playback and rtp streaming playback.  Whenever
local/rtp streaming playback request comes, am connecting audiobin to adder
and setting pipeline play.

Below is my pipeline.

*main*: adder ! volume ! alsasink
*local-playback audiobin:* uridecodebin ! audioonvert ! volume --> connected
to adder on request of local playback
*rtp streaming playback audiobin:* appsrc ! rtpjitterbuffer ! rtpL16depay !
audioonvert ! volume

here using the *gst_app_src_push_buffer* to insert the buffers into the
pipeline.
And for appsrc, setting "do-timestamp" to true,  "format", GST_FORMAT_TIME
and "block" to false.
For rtpjitterbuffer, latency is set to 700ms, and "mode" is set to synced.


Local playback works fine without any issues.
But rtp streaming playback is having issues - missing playback of first few
secs of audio.
Also after localplayback, response playback is not giving any audio.

I have tried setting the alsasink sync property to false, but this is giving
issues in local playback(clipping at the end of audio file).

Any ideas on this?


-ashok





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