[gst-devel] RTP sync and sender report ntp timestamps

[Eric Zhang]

Hi, gstreamer-devel:

    1. Seems you should write some codes to achieve your goal because you
mentioned you have other non-gstreamer codes to handle the audio data. So,
you can read the description and algorithms on RTP lip-synchronization and
do it yourself. "gst_rtp_bin_associate" is a good reference as well.

    2. uridecodebin will select rtspsrc and autoplug other stuffs when you
play a RTSP movie. I skimmed the source codes of uridecodebin and noticed
there is function named "gen_source_element" and it also calls
"gst_element_make_from_uri" as well.

Eric Zhang

2008/12/18 Jon Burgess <jkburges at gmail.com>

>
>>     "gst_rtp_bin_associate" in gstrtpbin.c do what you want.  But I think
>> it needs improvements. :)
>>
>>
>
> Thanks Olivier and Eric...
>
> From what I can tell, this bit of code handles synchronization between RTP
> streams within the same session, or in other words, all streams handled by
> one RTP bin will be synced.
>
> But my case is slightly different because of the fact that one stream
> (video) will be received by a gstreamer pipeline, while the other (audio)
> will be received by legacy code.
>
> So my thought of how to sync them (either when playing live, or playing
> back recorded data) would be to have gstreamer timestamp the buffers with
> something like a unix (wallclock) timestamp, have the legacy code do the
> same, and then things should stay synced.
>
> Would this work, or would timestamping like this upset gstreamer?
>
> Also, I'm using a uridecodebin - would that use a gstrtpbin internally for
> an RTSP source?
>
> Jono
>
>
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