
VTgreen81
Active member
regebro said:Pulse modulation is a way of making D/A conversion. Essentially, what you do is generate high frequency pulses, loads of them if you want a high value, few of them if you want a low value. Then you slap a filter on it to remove the high frequency contents, and you got yourself an audio signal.![]()
This much I understood, but how do the pulses correct the errors between the actual analog wave and the representation supplied by the LSB (least significant bit)? I know there's averaging involved but where to get the numbers to average?