[Live-devel] real time streaming

[Ross Finlayson]

At 06:47 AM 8/30/2006, you wrote:
>I am using the Live Media for sending real time video and audio based
>on the test programs (mpeg4 and wav).
>When I play the stream I get:
>1. Latency about off 600 ms till the video played - maybe because the
>program pack several video frames before sending them.

No, that doesn't happen for video.  When streaming to a media player, 
any significant latency that you see almost always occurs inside the 
receiving media player.  There is little latency in the "LIVE555 
Streaming Media" code for transmitting RTP.

>2.  I get delay between the video and the audio streams (lipsing
>synchronization).

Audio/video synchronization should work, *provided that*:
- You implement RTCP, by creating a "RTCPInstance" object for both 
the audio and video streams, and
- The presentation timestamps ("fPresentationTime") generated at the 
server end must (for both audio and video) must be accurate, and 
synchronized to 'wall clock' time (e.g., as returned by "gettimeofday()").

Because you've developed your own code, I probably can't help you 
more than this.  But Remember, You Have Complete Source Code.


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