[linux-audio-dev] [ann] unmatched - a LADSPA amp tone

Steve Harris S.W.Harris at ecs.soton.ac.uk
Mon Oct 21 18:44:01 UTC 2002


On Mon, Oct 21, 2002 at 07:11:41 +0200, Tim Goetze wrote:
> Steve Harris wrote:
> 
> >On Mon, Oct 21, 2002 at 02:39:05 +0200, Tim Goetze wrote:
> >> pleased to announce the release of 'unmatched'.
> >> 
> >> 'unmatched' is a simple effort to recreate some aspects of
> >> the tone shaping of a real instrument amplifier. unlike
> >
> >Excellent. Can I ask how you did the FIR -> IIR mapping? Is it neccesary
> >or significantly better to use doubles for the coefficents?
> 
> a quick test (%s/double/float/g) shows the cpu usage doubling,
> but i'm unsure what may cause this huge performance drop.

Lets just put it down to chache issues and ignore it ;)
 
> doubles should, on average, mean we seldom hit the denormal number
> bounds, or at least less frequently than with floats. i also expect
> doubles to be beneficial to the filter stability by minimizing
> round-off error, though i may be wrong here. lastly, x87 uses 80 bits

Yes, though can manually truncate the floats, and floats will give half
the cache footprint, so play better with other software.  I dont really
have a feel for when doubles are neccesary for filters yet.

> >Have you experimented with adding a delay line and LP filter to simulate
> >reflections off the back wall of the cabinet?
> 
> actually the next thing i'd have done do to come closer to the
> original impulse would be an added IIR operating behind a delay 
> line, yes. the current plugin response does not capture the later
> parts of the original response so well (now that you mention it, 
> i think it must be those reflections). however personally i'll
> probably do more research on nonlinear effects before refining
> this method.

Yes, you can always pep things up with a nonlinear effect of somekind
before the cabinet+speaker sim. In FFT convolvers you take several
impulses at different amplitudes and shift between them.

I wouldn't worry too much aboout being very close the the impulse, the're
only for a particular input amplitude and I can't remember when they came
from so may not be fantastic.

I'm wondering if this technique can be used for reverbs too - generate a
purely "white" synthetic reverb tail, and apply an IIR the aproximate
shape of the rooms impulse to it to make it sound more real...

- Steve



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