[Live-devel] RTSP and assembling audio buffer from audio frames

[Denis Z]

Hi!

I'm not really sure which side is responsible for the problem... More like
that I need your advice what to do and investigate next.

I use live555 to receive video and audio streams using RTSP client. The
problem is: when I fill audio buffer with audio-frames according to their
timestamps, I get a small gaps between frames in the buffer. That causes
unacceptable artefacts while playing. From the other side, if I simply
concatenate incoming audio-frames, I got clear audio without artefacts. The
problem is protocol-independent, I can hear the same artefacts on both TCP
and UDP protocols.

Audio is μ-law encoded.

Here (http://www.sendspace.com/file/ba7f3n) you can download small archive
(~200 kilobytes) with two samples:

"solid.wav" created via simple concatenation of incoming audio-frames. Plays
clear, no artefacts.

"assembled.wav" created using prerequisite buffer, where incoming
audio-frames were arranged according to their timestamps. Plays with
artefacts. If you'll open this file in hex-editor you will see small
null-filled gaps.

I can't use simple concatenation method because of synchronization issue.
What should I do, what solutions can fix that problem?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.live555.com/pipermail/live-devel/attachments/20100320/9bba40fd/attachment.html>