I have a problem with the following pipeline:
gst-launch-1.0 -e uridecodebin caps="video/x-h264, parsed=true; audio/x-raw"
uri=<camera url> name=rtspsrc matroskamux name=mux ! filesink
location=test.mkv rtspsrc. ! identity ! mux. rtspsrc. ! audioresample !
audio/x-raw,rate=16000 ! voaacenc ! aacparse ! mux.
I am getting two streams from an rtsp soruce and placing them into a
matroska container which is saved to a file. Essentially a diamond-shaped
pipeline:
audio
rtspsrc< > matroska -> file
video
The pipeline works, creates a playable file. But there is a problem with
rtpjitterbuffer. By setting
GST_DEBUG=2,rtpjit*:5 I can see that on some cameras audio packets come out
later and later, they are actually stacking inside Jitter buffer as
indicated by lines like the following:
rtpjitterbuffer
gstrtpjitterbuffer.c:3268:gst_rtp_jitter_buffer_chain:<rtpjitterbuffer1>
Pushed packet #18896, now 31 packets, head: 0, percent -1
Over time I see that the number of packets inside rtpjitterbuffer1 keeps
rising.
On the other hand if I simply play the same source like in the following
pipeline:
gst-launch-1.0 -e uridecodebin caps="video/x-h264, parsed=true; audio/x-raw"
uri=<camera url> name=rtspsrc rtspsrc. ! avdec_h264 ! videoconvert !
autovideosink rtspsrc. ! audioresample ! audioconvert ! autoaudiosink
then all is OK, audio plays correctly with no problem/delay- Looks like the
insertion of the matroska container forces some kind of synchronization
between the two sources that is badly estimated on audio side. Note:
inserting a queue on both branches does not help
Not all cameras exhibit this problem; one some it works without any problem.
Note: this is not my real problem, I am working with a pipeline build
programmatically where I noticed that audio packet come out later and later.
The above pipeline (with the GST_DEBUG set accordingly) is the simplest way
to reproduce the problem I have found.
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