[linux-audio-dev] Paper on dynamic range compression

[Dan Mills]

--- Steve Harris <S.W.Harris at ecs.soton.ac.uk> wrote:

> > But that puts potentially expensive gain
> > calculations into the fast sr code, also I was
rather planning
> > on using the impulse used for the upsampler to
> >provide the band limiting for free.
> 
> the gain calculations are relativly cheap, its
> converting those gain
> levels back into a gain coefficient on the output
> that's expensive (from
> what I remember).

But all that can still be done at 44.1/48, it is only
the final sample multiplication that needs its inputs
band limited and thus potentially needs the upsampler.
 

> > BEAST was the project that had the SSE 2* up/down
> > sampler code that seems to be reasonably quick. 
> 
> But what interpolation function? If you something
> obvious and cheap you
> may as well not bother, as you wont get accurate
> peak interpolation.

The usual half band impulse convolution based
approach. 

> I dont think a streaming resampler is more
> challenging than a correct one
> that runs on buffers. You just have to preserve
> state with every call,
> rather than at the buffer boundaries.

But that breaks the ability to use the vectorisation
capability of modern compilers and  besides the call
and return overhead will end up costing as much as the
convolution!

We seem to be discussing two separate problems here,
aliasing in the gain cell and real 'audio' peak
discovery, they are peripherally related byt are not
the same thing. 

Regards Dan (Who really now has to go and do some work).

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