[Live-devel] Jitter Buffer h264 rtsp client

[Kevin Bailey]

What exactly does the presentation time entail? Is it actually the time in
which a frame should be passed off to the decoder and then to the screen?
If so how could there be many frames that have the same presentation time
(down to the miliseconds)? And if that is not the case is this just an
arbitrary time?

Also implementing a queue probably would not work in my use case. I have
set up my rtsp server to only report screen buffers when there is an actual
difference in the images on my stream (to save on bandwidth). If I
implement a queue whatever playback delay I incorporate would totally
disappear after any period of still frames.

On Thu, Jan 3, 2019 at 3:22 PM Ross Finlayson <finlayson at live555.com> wrote:

>
>
> > On Jan 4, 2019, at 12:07 PM, Kevin Bailey <kevin at advantaacomputer.com>
> wrote:
> >
> > Hi all! Can anyone point me to an example of how to implement a jitter
> buffer into an rtsp client using live555?
>
> You could probably look at the VLC code for an example.  But the idea is
> quite straightforward:  A basic jitter buffer is just a queue of incoming
> frames (or, for H.264, "NAL units”).  (Because the LIVE555 code always
> delivers frames in order, you don’t need to worry about frames arriving out
> of order - thus, it can just be a queue.)  You should mark each frame with
> its “presentation time”.  The length of the queue will depend on how much
> playback delay you want (again, based on the presentation times).
>
> You can get a lot more complex than this (i.e., varying the size of the
> buffer depending on the measured jitter, etc.).  But for a RTSP source -
> where low latency usually isn’t crucial - that’s usually not necessary.
>
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
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>
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