---------------------------- Original Message ----------------------------
Subject: Re: [music-dsp] Antialias question (Kevin Chi)
From: "Kevin Chi" <s...@finecutbodies.com>
Date: Fri, June 1, 2018 2:50 pm
To: music-dsp@music.columbia.edu
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> Thanks Frank,
>
> I use cubic interpolation to interpolate between samples and it seems a
> bit better than linear for me. I am more worried
> about the downsampling, when the playhead is going faster than the
> original samplerate
or the output pointer ("playhead") is moving faster than the input pointer
("recordhead").� the latter is always moving at a rate of one sample
displacement per sample period.� the former, the output pointer has both
integer and fractional
parts.� the integer part of the output pointer tells you which samples you are
going to combine, and the fractional part tells you how you will be combining
them.
>� and that's
> when the high frequencies of the original material start to fold back at
> Nyquist to cause aliasing.
>
> My concern is if it would be a constant pitching up (downsample) rate
> then it would be "easy" to apply an
> antialias filter based on the resampling rate. But as the resampling
> rate can change by every step (as the speed of the
> playhead is modulating), you should apply different antialias filters
> for every step. Or at least the filter needs to use
> different coefficients every time it's applied... That's what I am
> unsure of what is the best solution to solve.
�
what you can do, but it's a little expensive, is when your output pointer
stepsize is larger than one, cut that step size in half and compute two
samples.� lowpass filter that stream of samples and pick out every other
sample, discarding the samples in between.
�
--
r b-j� � � � � � � � � � � � �r...@audioimagination.com
"Imagination is more important than knowledge."
�
�
�
�
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