simple question to be sure :
in a digital audio file or audio stream , can frequency above nquyst exist?
or aliasing only occurs when converting analog to digital or digital to analog
aliasing nquyst question
- KVRAF
- 15272 posts since 8 Mar, 2005 from Utrecht, Holland
No. The nyquist theorem states frequencies above half the sampling rate cannot be represented. These cannot exist. Any wrong attempt to do so will result in aliasing. They get mirrored and fold back. Compare to cart wheels spinning reverse in old cowboy movies.
Aliasing can occur when not removing content above nyquist in analog-digital conversion, or simular when downsampling (reducing the sample rate) in the digital domain. Eg the wav file is 48kHz and soundcard is fixed to 44kHz and resamples too simple.or aliasing only occurs when converting analog to digital or digital to analog
See https://en.m.wikipedia.org/wiki/Nyquist ... ng_theorem
(also for me too mathematical)
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- KVRist
- Topic Starter
- 245 posts since 22 Feb, 2014
Thank you BertKoor it's clear in my mind now. ... i just have to tidy my plugins between those who have oversampling choice (or built in oversampling) and those who don't..
does somebody know a kind of vst wrapper or vst that host vst , that can oversample (the only one i know is ddmf metaplugin) ?
does somebody know a kind of vst wrapper or vst that host vst , that can oversample (the only one i know is ddmf metaplugin) ?