Thaks for this lot of informations.
The things I realized are somewhere 'bad' way to realize it, but what I
did is :
Copy WAVAdioFileSource.cpp to ReaStreamAudio.cpp my own file
Copy WAVAudioFileServerMediaSubsession.cpp to
ReaStreamFileServerMediaSubsession.cpp
with they corresponding .hh files. and recompile all of it.
Based with the testOnRTSPServer.cpp sample code I could do some tests that
worked.
Things that are doing in a very bad way is just that I just remove some
lines of code and delcare explicitely some variables like frequency,
bytespersamples, etc...
I really should have to use the object oriented implementation, but it
needs some times to learn it at all.
It's works !
But there is only one trouble to do it efficiently, which is :
If I want to stream from home my own mix with 48k/24b I need at least 250k
upload to do it in real time...
So with my dsl connexion I've only 100k which is not enough...
It's a problem of banddwidth about uploading datas. (fuck this limit).
I wait to get an fiber or an VDSL connexion to realize it from home, and I
will test all of these soft with an real optical fiber dsl connexion to
chack it !
All of my tests, like simulate an udp stream that would come from my home
directly on the server (cat rawWaveFile > /tmp/my_pipe) works very well to
get back an RSTP stream of 48k/24b audio file.
It's a real pleasure to discover Live555 open source code to test it on an
real media server. It works very well.
Best regards
Grag38
ps : For people that are interested streaming live audio streaming : look
at Reaper Software (http://www.reaper.fm and look at the ReaStram plugin
that can stream until 8 tracks in real time audio without compression. But
to do it, you really need a lot of bandwidth.
If someone needs some explanations, he can mail me about it.
Choose a "preferredFrameSize" that's large enough to fit within an
outgoing RTP packet. I suggest 1500 bytes (i.e., 250 samples), with
"playTimePerFrame" of 5208 microseconds (assuming a sampling rate of 48
kHz).
I made a mistake there. 1500 bytes is a bit too large to fit within a
packet (assuming the code's current packet size settings).
Instead, I suggest 1440 bytes (== 240 samples), with a
"playTimePerFrame" of 5000 microseconds (assuming a sampling rate of 48
kHz).
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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