how to balance audio data between rtp src(in the network) and decoder sink

[Soho Soho123]

Hi all,


I have created a APPSrc to implement the pipeline below:
gst-launch-0.10 filesrc location=/bin/audio_src_48k_le.wav !
decodebin2 ! audioconvert ! alsasink device="hw:0,0"
filesrc--->decodebin2 ---> audioconvert--->alsasink

the src is from file "/bin/audio_src_48_le.wav"
and the signal setting
g_signal_connect(app->src, "need-data", G_CALLBACK(start_feed), app);
 g_signal_connect(app->src, "enough-data", G_CALLBACK(stop_feed), app);
start_feed is callback function when decoder need-data.
It is workable when I read data from file.

I would like to use rtp socket as data src to replace file src.

rtpsrc--->decodebin2 ---> audioconvert--->alsasink

rtp is the code that was ready in my target system,
I just like to use decoder in gstreamer ONLY.

in start_feed function, I use gst_app_src_push_buffer(app->src,
buffer) to push data to decoder,
appsrc element emits signal to tell the application to start and stop
feeding data.

is it possible to read rtp socket when appsrc element emits need-data signal?

because rtp data is from the other src in the network. the rtp src may
not know when appsrc emits "need-data" signal,
how to get balance between rtp src and appsrc element signal?


Thanks!
Soho