[Linuxsampler-devel] Re: Envelopes , filtering and zipper noise

[Paul K. <pau...@ma...>]

I'm replying to things in the digest - sorry I'm out of sync with everyone 
else...!


Benno Senoner <be...@ga...> wrote:
>
> Regarding adding a let's say resonant LP filter to the sample output:
> I guess the coefficients need to be smoothed out in some way too,
> otherwise zipper noise will probably show up again.

Filter cutoff is much less sensitive to zipper nose than amplitude, and
for typical sampler patches, updating the filter coefficients every 32
or 64 samples is ok, with no interpolation.  The only time this is a 
problem is for fast, high-resonance filter "chirps", but it still sounds
acceptable, just not as good as an analog or virtual analog synth.


> What would a good tradeoff for achieving low-CPU usage zippernoise-free
> filter modulation would be ? (some kind of interpolation of precomputed
> coefficients ?)

Use a state variable filter.  Almost every other hardware and software
synth does, and you can adjust cutoff and resonance directly without 
worrying about stability, or any heavy calculations.



Steve Harris <S.W...@ec...> wrote:
> 
> BTW do we know for sure than samplers use exponential envelopes? I guess 
> we need linear ones too, but they are easy to implement. 

Decay and release should be exponential.  It's nice to offer a choice of
linear or other curves (mostly useful for drum sounds) but for most 
instruments, linear decay/release sounds bad - the fade out is too sudden.
But the ear is quite tolerant if the curves are only approximately
exponential.



> get some recordings from samples of high freq sinewaves with envelopes.
> I think Frank N. has done this allready for the S2000, Frank are you on
> the list?

It's better to use square waves, then the shape of the envelope (for 
example if there are sharp corners between each stage) is not hidden by
the shape of the waveform.




Benno Senoner <be...@ga...> wrote:
> 
> It is probably the easiest to keep track of the previous pitch value
> and then simply interpolate linearly to the new value within a let's say
> 0.5 - 1 msec time frame. I think this would smooth out things nicely,
> right ?
> 
> (Steve, others, do you agree ?)

Agree.  But a lot of the time, where the envelopes and any LFOs are
changing slowly, you may not even need to interpolate pitch and filter
cutoff, just step every 1ms.

  
> Ok but assume these nice LP filter  wah-wah effects.
> The LFO frequency in this case is up to a few Hz, but the modulation
> frequency (filter coefficient change rate) ? How low can it go so that
> zipper noise can be avoided ?

There is no simple answer to this - you will hear zipper noise on a low/mid
frequency sine wave that you won't hear on any other sound, so maybe it
is better to have a user-adjustable "quality" setting for what gets 
interpolated.  If someone is going to be playing big piano and orchestral
samples with no processing, they can switch to the lowest setting (so best
if it's not actually labelled "quality"!) or offer an unprocessed playback
option like HALion does.




Paul.

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