[Live-devel] How is the best way to realize it ?

[Raph]

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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