[gst-devel] add latency to audio in gstrtpbin

[Eric Zhang]

Hi, Tristan:

    You should not adjust video/audio latency manually because RTP provides
a mechanism to accomplish this, called `lip-synchronization'. Refer to RFC
3550 or book `RTP: Video and Audio for the Internet' for more details. These
will help you a lot.

Eric Zhang

2008/9/3 Tristan Matthews <tristan at sat.qc.ca>

> Hi,
>
> If I have a pipeline using gstrtpbin (similar to the example in the
> documentation) to send audio and video, what is the best/most reliable
> way of adding latency to the audio? Would gst_event_new_latency work
> (and if so, how), or am I missing its intent:
>
> http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency
>
> Basically my concern is that if video capture is too slow, can I
> manually adjust the audio latency to match.
>
> Best,
>
> Tristan
>
> --
> Tristan Matthews
> Société des arts technologiques [SAT]
> email: tristan at sat.qc.ca
> web: http://www.music.mcgill.ca/~tmatthews<http://www.music.mcgill.ca/%7Etmatthews>
>
>
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