[vorbis-dev] Peer-to-peer audio codec

[Ross Levis]

http://www.allcast.com is using peer-to-peer streaming technology (with 
WMA unfortunately).  An Ogg solution would be great.  Their website may 
give you some ideas.

Ross Levis.

HALL CYRUS P wrote:

>A friend and I were recently talking about the possible
>downfall of Internet radio, and as everyone seems to these days, decided
>that a nice solution would be to set up a P2P broadcasting solution.
>However, streaming has a completely different set of requirements than
>file sharing, with bandwidth and QOS become much larger issues.  As such,
>we started to realize that a new audio compression scheme might be needed
>for such a network.
>
>As an open source developer and user, I  thought of the immediately Vorbis
>project, and its developers.  As such, I thought I'd run this idea by the
>list, and test it for feasibility.  the basic idea follows:  Most
>broadband users are on cable modems, which unfortunately limit up-stream
>bandwidth to a fraction of what many DSL users get to experience (I know
>the technical issues the cable companies always taut, however, I've always
>thought of it as a way to limit personal publishing ability).  As such, a
>P2P radio broadcast network could not use the upstream of such users,
>putting the burden of distribution on a few, and thereby defeating the
>major point of the whole thing P2P in the first place.
>
>If there was a way to break up the audio streams and re-sync them on the
>listeners side, this problem could be avoided (maybe).  For example,
>imagine a 128 kbps stream (OGG, of course).  A lucky cable modem user
>could barely support one user at such a bit-rate.  However, if the stream
>was split into left and right channels, each one a different stream, a
>listener could get one channel from one location, the other from somewhere
>else.
>
>Now splitting stereo channels doesn't seem to get one very far.  But what
>about some sort of a "progressive" audio codec?  Let me at this point
>state I have no real experience with compression, so all of this off the
>hip.  A progressive audio codec that could "build" itself up would be a
>very handy (and cool) solution to the problem.  A stream could be broken
>into multiple levels, where each level would add more definition to the
>previous.  For example, if each level took 16 Kbps of bandwidth, a user
>who had been able to find a source for both level 1 and 2 would be able to
>listen to the stream as if it was at a quality of 32 Kbps.  maybe the
>scaling wouldn't be that linear, I don't know, but I think you get the
>idea.
>
>If a P2P network could use such an audio codec to distribute its streams,
>it balance load and demand (number of higher quality
>"levels" for each stream goes down as demand goes up) and avoid the
>death of Internet radio.  Both of which are important. ;-)
>
>This idea doesn't really fit into OGG at all, I understand that.  However,
>I was hoping that maybe some of you would have a more technical base in
>the field than I, and could point me in the right direction to start
>researching such.
>
>thanks,
>Cyrus
>

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