Hi Ross -
I am prototyping a streaming application using an MPEG-1, Layer 3 (.mp3)
audio file and an MPEG-4 video elementary stream (.m4v) file as inputs.
I am doing this to simulate our actual encoder outputs since they are
not yet available. I recorded these files from two different physical
sources on two different days. When I wrap them into an MPEG-2
transport stream, I experience synchronization issues. I seem to be
dropping a lot of audio packets, making it sound like the audio is
"jumping ahead" to catch up to the video (VLC client).
Here is a depiction of what I am doing:
|ByteStream| |MPEG4 | | |
.m4v ->|FileSource|->|Video |->| |
file |Stream | | |
|Framer | |MPEG2 | |MPEG2 | | |
|Transport |->|Transport | |Simple|
|Stream | |Stream |->|RTP |
|MPEG1or2| |From | |Framer | |Sink |
|ByteStream| |Audio | |ESSource | | |
.mp3 ->|FileSource|->|Stream |->| |
file |Framer | | |
(I didn't show the RTCP Instance associated with my RTP Sink)
Am I doing something wrong here? How does Live555 ensure synchronization?
Note that if I "disconnect" one of the inputs, the resulting transport
stream (audio or video) plays fine in my VLC client.
Thanks a ton,
Mike.
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