Hi,
Thanks for the guidance and I tried it. Now I can see the packets being
transmitted in testRTSPClient and also in VLC. VLC shows packets being
decoded for both audio and video. But still somehow video is running
smooth but audio just comes and goes. Its like something is choking
audio. It comes for a second and then goes back. What area should I see
for possible issue. I tried changingthe estimate bandwidth for video and
currently set it to 4500 and for audio set it to 200 but still same
issue. I checked time stamps but they seems fine in testRTSPClient.
Thanks,
Caduceus
On 2013-10-26 13:43, Ross Finlayson wrote:
I am using AC3 codec to encode audio and then stream it using
AC3AudioStreamFramer and AC3AudioRTPSink. Am i doing anything wrong
?
Yes. You should *not* be using an "AC3AudioStreamFramer". That's
intended only for streaming AC3 audio that comes in a byte stream -
e.g., from a file. Because your AC3 audio input comes from an encoder
- one frame at a time - you should feed it directly into a
"AC3AudioRTPSink". (Also, don't forget to set accurate presentation
times ("fPresentationTime") for each frame.)
Also, you should test your stream first using "testRTSPClient", before
trying VLC (or any other media player).
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/ [1]
Links:
------
[1] http://www.live555.com/
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