[Live-devel] Queries: regarding AAC

[Brian D'Souza]

Hi,

I have been tracking posts regarding streaming MPEG4 audio( encoded in AAC)
and in an earlier post too I was instructed to have a look at the
MPEG4GenericRTPSink and ADTSAudioFileServerMediaSubsession.

My queries:

1. I am unable to find any framer class ( as was used for
MPEG4ESVideoStreamFramer ; as well as a corresponding parser) for the same.
The issue is that I am using Live555 as an RTP packetizer for aac content
and relaying it to Darwin which does the necessary RTSP part). So I cannot
just use live555 as an RTSP server. I want to know that just creating an RTP
sink using the MPEG4GenericRTPSink ; then jusing the source
ADTSAudioFileSource(which works from file/encoder) and then invoking
startPlaying() is enough?

Wouldn't I need a framer; which decides what is in an audio frame; has
methods like continuePlaying(); getNextFrame() etc and processes it
appropriately.


2. In the MPEG4ESVideo Elementary streaming (relay to darwin)test program ,
there is a hack to read configuration information from the stream ; to
generate an SDP description; and an announce RTSP operation is done. Does
the hack need to be there for MPEG4 audio as well.. Similarily if I were to
stream h264 video content, is this hack needed?


Thanks for the help,
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