[Live-devel] Live Audio RTSP Streaming

[Ross Finlayson]

> I am trying to send a MP2 encoded frame via RTSP (live555) from a live source, (microphone).  I am using ffmpeg to encode audio stream, and I am sending it overriding FramedSource and for the OnDemandServer I am using MPEG1or2AudioRTPSink.hh and MPEG1or2AudioStreamFramer.hh.

"MPEG1or2AudioStreamFramer" is used only when the input source is a *byte stream* (e.g., a MP3 file).  If, instead, your "FramedSource" subclass is delivering discrete MPEG audio frames (i.e., delivering one frame at a time), then you should *not* use a "MPEG1or2AudioStreamFramer".  Instead, your input source should be fed directly into a "MPEG1or2AudioRTPSink", with no intermediate 'framer' object.

Note, however, that if your "FramedSource" subclass delivers discrete frames, then you must set "fPresentationTime" correctly for each frame, before completing delivery.


> Now in VLC I get to hear maybe half a second worth of sound and then it just stops...

VLC is not our software.  You should first use "testRTSPClient" and "openRTSP" as RTSP clients, before using VLC.

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

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