[Live-devel] Lip synchronization problem

[David Arnold]

Ross,

My server code is based on testOnDemandRTSPServer.  In other words, I
modeled the code based on how testOnDemandRTSPServer created sessions and
subsessions, etc.  Is there anything extra that I need to do to create the
required "RTCPInstance" objects?

Dave

-----Original Message-----
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com]On Behalf Of Ross Finlayson
Sent: Friday, February 03, 2006 12:26 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] Lip synchronization problem


At 10:39 AM 2/3/2006, you wrote:
>I am experiencing a lip synchronization problem.  I am streaming audio and
>video using liveMedia and viewing the stream using VLC and I am noticing
>that the audio and video are out of sync.  The video is an MPEG4 ES and the
>implementation is derived from FramedSource and uses
>MPEG4VideoStreamDiscreteFramer.  The audio is MP3 and uses the
>MP3AudioFileServerMediaSubsession class.  Using LiveMedia and VLC, should I
>expect the audio and video to be synchronized perfectly?  I'm trying to
>determine if there is a problem in my code, LiveMedia, VLC, or if what I am
>experiencing is normal.

Dave,

The problem is most likely in your server code.  For A/V
synchronization to work, you (i) *must* use RTCP (i.e., you must have
a "RTCPInstance" for each "RTPSink"), and (ii) your source objects
(for both audio and video) *must* generate accurate presentation time
values (by setting "fPresentationTime"), which must correspond to
real, 'wall clock' time.  I.e., the presentation times generated by
your source objects can't start with 0.0; they must start with a time
generated by calling "gettimeofday()".


	Ross Finlayson
	Live Networks, Inc. (LIVE555.COM)
	<http://www.live555.com/>

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