[gst-devel] add latency to audio in gstrtpbin
thiagoss
thiagossantos at gmail.com
Thu Sep 4 19:11:51 CEST 2008
On Thu, Sep 4, 2008 at 1:54 PM, Wim Taymans <wim.taymans at gmail.com> wrote:
> On Thu, 2008-09-04 at 11:34 -0400, Tristan Matthews wrote:
> > Thanks for the feedback Eric. I had seen the RFC before but that book
> > looks pretty useful as well.
> > My concern isn't that the AV will get out of sync in
> > gstreamer/gstrtpbin, but rather that it will be out of sync immediately
> > at the capture stage, i.e. if i have video input from a camera with more
> > latency than a separate audio source. An example scenario is given here:
> > http://chris.pirillo.com/2007/07/11/audio-video-capture/
> > where you have video of someone clapping and you have to adjust the
> > delay to make the sound and image of the clap line up.
> > I would need to be able to manually adjust the latency to ensure the
> > video and audio are sync'd. Should I just put the audio through a
> > Ladspa-delay or is their a better solution in gstreamer?
> >
>
> This situation can not occur in gstreamer when you have a well written
> source element that generates correct timestamps and reports its latency
> correctly. You could write an element that applies an offset to all
> timestamps on the buffers.
>
> Wim
>
AFAIK, This element already exists and is called GstShift from gentrans
plugins
>
>
> > -T
> >
> > Eric Zhang wrote:
> > > 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 <mailto: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 <mailto:tristan at sat.qc.ca>
> > > web: http://www.music.mcgill.ca/~tmatthews<http://www.music.mcgill.ca/%7Etmatthews>
> > > <http://www.music.mcgill.ca/%7Etmatthews>
> > >
> > >
> > >
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--
Thiago Sousa Santos
Embedded Systems and Pervasive Computing Lab (Embedded)
Center of Electrical Engineering and Informatics (CEEI)
Federal University of Campina Grande (UFCG)
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