I am using MP2 audio encoding for which the compressed framesize is
supposed to be 576 bytes (sampling rate is 32 KHz, single channel).
However, occasionally fMaxSize in deliverFrame() is less than 576.
It is 240 or so. When that happens, I write only 240 bytes to fTo,
and assign fFrameSize to
nal Message-
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson
Sent: Thursday, December 17, 2009 5:19 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] Audio drift with live source
>In my impleme
In my implementation of deliverFrame() function in the classes for
the sources, I read uncompressed audio and video from ring buffers
(which are filled by another thread), compress them, and then fill
the buffers accordingly before calling
FramedSource::afterGetting(this). I also set fPresentat
HI,
I am receiving uncompressed audio and video from a live Directshow source,
encoding them with ffmpeg and streaming them out using a RTSP server based on
classes derived from Live OnDemandServerMediaSubsession. Then I play the feed
on a remote machine using VLC player. The problem is that th
