On 9/26/05, Robin Bowes <[EMAIL PROTECTED]> wrote:
Ben Sandee said the following on 26/09/2005 23:47:
> If it's live audio, you can only buffer data as fast as the player plays
> it (real-time). If you want two seconds of buffer then you're going to
> need to delay playback by two seconds in order to guarantee that two
> seconds of buffer can actually be used in the event of network or server
> issues. If you start playback immediately there isn't any data IN the
> buffer and you'll never catch up because it's live.
Not so. Streaming is asynchronous.
For example, if you stream at > the playback rate then the buffer will
soon fill up.
How can you stream faster than the playback rate when the playback rate is inherently the same as the source data rate? It's not like we've got a file full of data -- we have data coming in from a microphone (or soundboard) at the sample rate and being played on the other end at the same sample rate.
Ben
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