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

Eric Zhang nicolas.m.zhang at gmail.com
Fri Dec 19 09:10:15 CET 2008

Hi, gstreamer-devel:

    Also the gstrtpbin will be used by uridecodebin because gstrtpbin will
be used in rtspsrc which is a source element handles RTSP medias and
inherits from GstBin.

Eric Zhang

2008/12/19 Eric Zhang <nicolas.m.zhang at gmail.com>

> 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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