Well, bandlimited to a bandwidth < fs/2 (but the distinction isn't
useful for audio), and given perfect reconstruction circuitry. But as
far as I can gather, Theo's concern is "what happens when, as is
inevitable in practice, the reconstruction circuitry is imperfect?"
And that is an interesting question, academically speaking, if there
really isn't a body of theory to cover reconstructive imperfection -
if only to be certain that all the improvements made to DAC technology
in the last three decades have actually been _improvements_.

But I think all of this is probably covered by existing research
anyway. Minimising phase disruption in digital filters is well
understood; LSB errors and resistor chain nonlinearities are fairly
obvious, sources of relatively predictable badness, and can be
assessed in the same way as nonlinearity in general; clock jitter is
easy to simulate... and then there are analogue reconstruction
filters. Unless I've missed any, I don't think there's anything else
to look at, unless he wants to disprove or augment Nyquist-Shannon.
Which would be an achievement, true, but... I would humbly submit that
there might be more fruitful avenues towards seeing "Verelst theorem"
in the indices of 22nd-century audio textbooks.

Still, I understand great white whales all too well; and if Theo
_needs_ to harpoon this one, we should perhaps not stand in his way.


On 05/06/2015, Stefan Stenzel <stefan.sten...@waldorfmusic.de> wrote:
> Theo,
>
> Any continuous function bandlimited to frequencies < fs/2 is completely
> determined by its samples.
> That’s the essence of the sampling theorem, which answers all your
> questions.
>
> Stefan
>
>
>> On 03 Jun 2015, at 22:47 , Theo Verelst <theo...@theover.org> wrote:
>>
>> Hi,
>>
>> Playing with analog and digital processing, I came to the conclusion I'd
>> like to contemplate about certain digital signal processing
>> considerations, I'm sure have been in the minds of pioneering people quite
>> a while ago, concerning let's say how accurate theoretically and
>> practically all kinds of basic DSP subjects really are.
>>
>> For instance, I care about what happens with a perfect sine wave getting
>> either digitized or mathematically and with an accurate computer program
>> put into a sequence of "signal samples". When a close to perfect sample
>> (in the sense of a list of signal samples) gets played over a Digital to
>> Analog Converter, how perfect is the analog signal getting out of there?
>> And if it isn't all perfect, where are the errors?
>>
>> As a very crude thinking example, suppose a square wave oscillator like in
>> a synthesizer or an electronic circuit test generator is creating a near
>> perfect square wave, and it is also digitized or an attempt is made in
>> software to somehow turn the two voltages of the square wave into samples.
>>
>> Maybe a more reasonable idea is to take into account what a DAC will do
>> with the signal represented in the samples that are taken as music,
>> speech, a musical instrument's tones, or sound effects. For instance, what
>> does the digital reconstruction window and the build in "oversampling"
>> make of a exponential curve (like the part of an envelope could easily be)
>> with it's given (usually FIR) filter length.
>>
>> In that context, you could wonder what happens if we shift a given
>> exponential signal (or signal component) by "half" a sample ? Add to the
>> consideration that a function a*exp(b*x+c) defines a unique function for
>> each a,b and c.
>>
>> Anyone here think and/or work on these kinds of subjects, I'd like to
>> hear. (I think it's an interesting subject, so I'm serious about it)
>>
>> T. Verelst
>>
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