Beside Bitwig which DAW allows audiorate modulation?

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Don´t worry :)

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Audio rate modulation means in simple words, you could create a ringmodulator just with lfos, and the lfo isn‘t a low frequency oscillator anymore, it creates a sound with all harmonics up to nyquist...

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Tj Shredder wrote: Fri Oct 26, 2018 1:50 pm Audio rate modulation means in simple words, you could create a ringmodulator just with lfos, and the lfo isn‘t a low frequency oscillator anymore, it creates a sound with all harmonics up to nyquist...
No...that´s just 1% of the truth in simple words... :hihi:

Audio rate modulation means in simple words you can modulate all kind of targets at a frequency rate, which goes high up into audible range... don´t know, if there is a upper limit though

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The upper limit is half the sampling rate also called nyquist frequency...

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Nyquist frequency is the upper frequency limit at which a sample sequence can represent an analogue waveform. It doesn't specify anything about a limit at which the sample sequence can be processed.

You might be capturing and reproducing an audio waveform at 48000Hz, hence have a Nyquist frequency of 24000Hz. A modulation circuit could happen to alter the signal frequency cyclically between half and double:

* A source signal at 3000Hz would be captured accurately, processed accurately and reproduced accurately (at 1500Hz to 6000Hz).

* A source signal at 18000Hz would be captured accurately, should be processed accurately, with reproduction aliasing for frequencies above 24000Hz (i.e. 9000Hz to 36000Hz, with aliasing).

But the sample rate isn't affecting the audio rate modulation here - it's affecting the reproduction of the computed waveform.

Admittedly, as far as I'm aware, most DAWs would do the calculation at the "project sample rate", which would usually be the hardware rate. However, that's not necessarily the case. If you load a file that contains audio sampled at 48000 but your project sample rate is 192000, you can render both those examples internally with no aliasing. If you then play back the second output at 48000, you'll probably get aliasing (unless the DAW automatically low pass filters - destructive but less unpleasant...).

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