Hey hey,
I want to sample the waveform of an oscillator of a hardware synth. One period
of the waveform must have 65536 points in the end. Will the following process
work:
Samplerate 48kHz, Oscillator pitch 71.0449218750000Hz, becasue:
71.0449218750000Hz / 48000Hz = 97/65536
In theory I can now create an empty table with 65536 entries and fill every
97th entry (with wraparound) with successive samples from the original audio
recording.
I see a problem: the soundcards ADC must average/interpolate the digital
samples from the analogue input arriving in between samples. I can't tune my
synth to 48000/65536Hz.
I can increase my samplerate to 96kHz and work through the sample process as
above, but will that help? Am I completely on the wrong track for creation of
an adequate copy of an analogue waveform?
If a correct process is more complicated and requires more mathematics, I will
drop the idea as such.
Ta-ta
----
Ffanci
* Internet: https://freeshell.de/~silvain
Twitter: http://twitter.com/ffanci_silvain
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