>Both analog-to-digital (A/D), and its inverse digital-to-analog (D/A),
>conversion require filtering. When digital audio is "played", the
>output of the D/A converter is filtered to remove the "steps", resulting in a
>continuously varying "smoothed" analog output that can closely
>replicate the analog input. Fidelity to the original depends on the
>initial sampling frequency, i.e. how close together the "steps" are,
>and also bits per sample that determines how accurately the recorded
>amplitude represents the original amplitude and also determines the
The filtering is not to "remove the steps" or even to smooth them.
The steps don't exist. When you hear digital playback, you aren't
hearing digital, you're hearing analog. Those steps are simply
details given to the D/A converter to tell it how to generate an
analog waveform that matches what the original was. That analog
waveform will not have any steps to it. Connect a decent D/A to a
oscilloscope (sp?) and you'll notice smooth analog waveforms. No steps.
The filtering is to remove noise that the sampling frequency would
cause. For example, if you are sampling at 44.1 KHz, then any signal
over 22,050 Hz would get incorrectly digitized. The filtering on
playback is to prevent anything like that from causing interference
with the desired signal.
Filtering can't smooth steps, it would only muddy the picture so that
you didn't notice them. That's definitely not what it's for.
You can get a better worded explanation here:
It covers the filters as well as what oversampling has to do with anything.
Preserving the past for the future.
Dave Bradley President