Bytestream or sample jitterbuffer

[Peter Maersk-Moller]

Hi.

Every now and then I have kind of needed a sort of jitterbuffer for a
bytestream, but can't really find the right solution for GStreamer. Can
anybody suggest a solution to this?

Assuming you have a simple setup (just an example):

  gst-launch-1.0 fdsrc ! $SOMEFORMAT ! audioparse ! queue  ! audioconert !
autoaudiosink

Here is the thing. Because there is no jitterbuffer in the system, then to
avoid gap in playout, the autoaudiosink MUST receive N+1th audio frame no
later d time later it received the Nth audio frame and where d=duration of
the Nth audio frame. What would be needed would be an element that would
not forward data before it had received N amounts of data or T amount of
time data since it received the first data.

Now queue elements have a limit you can set for when it will start
forwarding data meaning something like .... you must contain at least N
amount of data before forwarding, but it is not a jitterbuffer as it can
not go below the threshold. A jitterbuffer needs to be able to do that. The
jitterbuffer element can only handle RTP data and I find it a little
excessive to have to encapsulate the audio data in RTP first and then have
the jitterbuffer queue it up and then decapsulate it again. If the queue
element was allowed to drain below its threshold, we would in effect have
ajitterbuffer working on bytes, buffers and time.

Any suggestions?

Thanks in advance.
Peter Maersk-Moller
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