On 3 Mar 01, 6:28PM, David W. Tamkin wrote:
> Maybe I understand part of it now: in an actual DAC+ADC passage, analog out-
> put has to come out of the DAC within the limits of its ability to generate
> the detail in the voltage changes, and the ADC has to read it within the lim-
> itations of its ability to sense the voltage readings; plus, as I said be-
> fore, there is potential lossiness in the analog travel between them. Those
> are three weak points not present in a sampling rate converter.
>
> Nonetheless, resampling has to go from discrete samples to a representation
> of a continuous waveform and then back to discrete samples, true?
that's basically correct. the biggest difference is that whenever you're
dealing with an adc or dac, you are dealing not just with the sound, but
also with physical properties of electricity, like latent voltages &c. if
you try looking at a square wave with an oscilloscope, you'll see that an
analog electrical signal does funny things of its own to a sound, unrelated
to whether there's a dac involved. the perfect square wave you see in your
audio program is more like a bumpy peak that decreases in height until it
falls off a bumpy cliff. this kind of thing is the biggest problem with
analog audio. electrical wires just can't quite represent the sound. so if
you're running through a dac-adc, you're introducing all sorts of
electrical & magnetic funny business. meanwhile, the adc has its own
physical imperfections that show up as quantization & the other familiar
digital noises.
(of course some of these things came up in the original recording process,
but each analog pass through electrical wires changes things even more.)
none of these things come up in a sample rate conversion, since it's all a
mathematical transformation. the only similarity is that the sound is being
converted.
peter
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