Originally Posted by chaos
Well, you *are* asking for a square wave, aren't you?
Why is this impossible? You can't produce a square signal without a bunch of higher frequency components. How do you define a Nyquist limit in this case? Plus, isn't a waveform just a collection of samples?
You can't define a Nyquist limit is this case. That's my point. Approaching the Amiga sound quality question from the perspective of sampling theory doesn't work.
Take this example:
Suppose a 500 Hz sine wave is sampled at 1000 samples per second and stored.
Is the waveform a sample of a sine wave or square wave?
It's obviously a sample of
a sine wave but will be output as a square wave and new frequencies will appear in the output that weren't in the originally sampled signal.
Now sample the 500 Hz sine wave at 10000 samples per second.
When output this will be look more like a sine wave. On the other hand, outputting the first sample at 10000 values per second will still look somewhat like a square wave.
The point is, when considering Amiga sound quality, the system should be seen as more than a simple recorder. It outputs arbitrary waveforms at arbitrary rates and volumes with a modest amount of filtering.
Seeing Paula as a mere sample player is what was responsible for some of the poor PC-based mod players that came out very early on. They were developed with the (wrong) idea that the Amiga was a sample player and so they used simple interpolation and filtering to get a 44.1KHz PC output. The sound was very different and was obviously missing something.
Proper PC mod players got better when they started to up sample instead.