[Live-devel] adaptive streaming

[Jason L. Ellison]

Ross,

  I'm currently trying to feed livecaster using a fifo in linux.  After a
few hours the audio latency begins to increase.  I can resolve this by
cat'ing the data from the fifo to /dev/null.  It seems data is building up
in the fifo.  I'm guessing that this is because of the way livecaster gets
the data in blocks at a set frequecy?

  Is there any thing i could modify on the generating progams config to
help this?  like the size of chunks being sent to the fifo maybe?

  Can I get the source to live caster if I buy a license?

  I'm also experimenting with testMP3Stream program.

-Jason Ellison

On Sat, 6 May 2006, Ross Finlayson wrote:

>
> >I would like to ask if it is possible to implement an adaptive
> >streaming using
> >the "testOnDemandRTSPServer".
> >
> >The server reads the client's RTCP report and if the packet loss rate
> >increases, (the server) should be able to stream the same video but using a
> >file with greater compression.
> >
> >This means that the server should have the ability of switching between
> >different versions of the same file that have been encoded at different bit-
> >rates.
>
> Yes, I think this would be possible, in principle - depending on the
> codec(s) that you plan to use.  (Some types of stream - e.g., MPEG-4
> and H.264 video - work by passing codec-specific 'configuration'
> information to the decoder before the stream starts.  You would
> probably not be able to change the bit rates of such streams
> on-the-fly.  For other codecs, however - e.g., MPEG-1 or 2 - you
> could probably get away with doing this.
>
>
> 	Ross Finlayson
> 	Live Networks, Inc. (LIVE555.COM)
> 	<http://www.live555.com/>
>
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