[Live-devel] Noob wandering

[Jesse Hemingway]

Apologies in advance for following deluge.  I'm new to Live555, RTP and
RTSP in general; and trying to gather resources to understand how to
consume video+audio streams.

Based on the testRTSPClient example, I've gotten a stable RTSP connection
working on my target platform; in this case to a video+audio RTSP source.
 But, now I'm struggling to figure out the next step.  My custom MediaSink
classes do not receive any frame data via afterGettingFrame(), but guessing
there is significantly more logic required in these classes than shown
in testRTSPClient.

What's a 'frame', exactly, anyway - my research indicates these are RTP
specific, but a concise definition is elusive.  E.g. a frame of audio -
seems inefficient to call a function for the typical definition of 'frame'
e.g. multi-channel sample.  Without deep study, rfc4571's definition is too
generic to help.  The best FAQ I've found is
http://www.cs.columbia.edu/~hgs/rtp/faq.html, but still assumes more
understanding that I have so far (plus the color scheme makes my eyeballs
implode).

And how about these higher-level MediaSinks - do those work right out of
the box?  Seems too good to be true.  Let's say I had AAC+VP8 streams
coming in.  Would I conditionally create a MPEG4LATMAudioRTPSink (what if
it's non-LATM MPEG4?) and a VP8VideoRTPSink in continueAfterSETUP() based
on inspecting the subsession?  I suppose I will have to try this out myself
;)

I know we're encouraged to study the examples and browse the library
source, but there is a LOT to scan without having a roadmap!  A lot of
these examples seem to have more to do with serving rather than consuming
streams, can someone point out a good one for study?  In the class graph I
also see some specific MediaSink classes, e.g. MPEG4ESVideoRTPSink.  Would
it make more sense to study these implementations?  I guess I'm struggling
to understand the high-level roadmap of information.

Many thanks!
Jesse
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