I then copied a section of the audio, to the clipboard, set the project rate to (256 x 44100 / 100) = 112896, then opened a new mono track and pasted in the audio data. I know I think the filtering should actually be done at the analogue stage of recording before it is digitized, but I only had a digital wav file to start with. So I opened the wav file in Audacity and run the Effects Low pass filter corner freq. To create the data array for teensy I need 256 samples (256 + 1 same start and end, an array of 257). If I have a wav file of frequency 441 Hz say that gives me 100 samples for 1 period. I understand teensy is playing it from a data array in memory and Audacity is playing direct from the file. What would be the difference in Audacity playing a 44.1 wave or teensy playing it.?.both at 44.1. I thought if I used wavs sampled at 44.1 that played at 44.1 on a PC, and teensy was sampling at 44.1 there would not be a problem. I also tried converting some sound samples for arbitrary sounds from wav files and find alot of aliasing. I can get it to work fine using the samples downloaded from your links. I am trying out synth waveforms using teens圓.1 and the audio adaptor.
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