Triode wrote: 
> There are two buffers - the first one stores the raw stream (compressed)
> audio, the second the decode audio. PCM is a special case as the process
> of decoding is just unpacking samples into the second buffer so they are
> in 32bit format which is used in the output thread, but this is only a
> special case of decoding. The decode process thread runs whenever there
> is enough data in the first buffer and enough free space in the second
> one.
> 
> So if you want to stream the whole file at the start then probably make
> the first one big and keep the second one smaller (defaults at 10
> seconds).  If you want to stream and decode the whole file at the start
> then make the second one big.
> 
> The second buffer is really only needed to allow crossfade to happen and
> to separate the high priority output thread from the rest of the
> processing.  Note that it uses 8 bytes per sample (2x32bits) whereas the
> first buffer uses the native sample rate and so will use less memory for
> the same amount of audio.


Thx a lot. 

The decoding buffer would have to be roughly 6 times larger, if we talk
16/44.1 flacs. 
If you've got plenty of RAM that's manageable. ;)



::: ' Touch Toolbox and more' (http://soundcheck-audio.blogspot.com) :::
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