Hi All,
I have a plugin which is actually a network source + an asf demux. So playback pipeline I have will look like the following
mysrc name = src src.srcvideo ! queue ! decodebin ! xvimagesink src.srcaudio ! queue ! faad ! osssink
It is guaranteed that video is h.264 and audio is aac. With this my streaming starts fine and works well. But I have a problem in the following scenario.
If my client cannot process at the bitrate at which the server is streaming, the server reduces the video resolution. So at this point I get a new asf header with new resolution. My pluign parses the header. But after I pass the new video frame to the vide decoder(with new sps/pps) the video decoder stops working. Is there anyway I can change the encoding parameters without unlinking and linking the components again?
Also after my new header, the timestamps start coming from 0 again. Hence I flush and send a NEWSEGMENT event for both audio and video pieplines. But after this I am noticing a 1 second lag in the video. (This is a when I tried a new header without change in resolution).
Somebody please help me how to tackle these issues.
Thanks
KK
------------------------------------------------------------------------------
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel