Basic technical question about digital source signals


Forgive if this is a stupid question, but the current thread about digital vs analog made me curious: if you look at an analog music signal you see (I think) summations of sine waves i.e. a signal waveform which is "smooth". I realize that there are many contributions to digital sound, but starting with the most basic, if you look at the output from a digital source e.g. on an oscilloscope, would it appear "smooth" i.e. has all the stairstepping that occurs when you convert digital to analog been smoothed out or would the signal appear jagged to some extent?

Thanks for your time.
berner99
You are equating accuracy with preference. That never ends well in the audiophile world.  People will make up all kinds of justifications why their preference is "more accurate". But the key phrase here is "make up".

Given the number of people who prefer nonoversampled DACs something seems wrong/incomplete with this theory

Given the number of people who prefer nonoversampled DACs something seems wrong/incomplete with this theory
Well, first, i’m providing a technical answer. You need not like it, but it is true.

Next, people have many opinions. Many are odd. Many will disagree. Look at politics.

Third - done right i clearly DO like it, ad anyone who listens to bitstream or DSD is listening to hugely over sampled streams by definition.

Fourth - we have no idea what was done in the anti-alias filter; that occurred in the studio. oversmapling was almost certainly employed, statistically, but we don't know. And aliasing is NEVER a good thing. It creates (maybe) tones that never existed in the first place.

I cannot personally imagine how allowing hgih frequency junk through, or having brick-wall,phase incoherent filters can possibly be preferable, but hey.
G
Not broken, a design decision to go NOS without a filter.

Its broken in the sense that such a design decision breaks the theory of digital audio. Just as a design decision not to band-limit the input breaks the theory.
such a design decision breaks the theory of digital audio.
Not really, it presumes that subsequent filters (mechanical limits in your speakers and your ear, which respond per F=MA; inherent limits in subsequent components) achieve he filtering.  There will be no aliasing after the DAC.  Yes, there could be HF noise residue; but hearing is highly attenuated above 22 kHz (if present at all) anyway.

Just as a design decision not to band-limit the input breaks the theory.

That would truly violate Nyquist's paper.Without band limiting various forms of aliasing and their effects can occur. 

Really quite different

Not really, it presumes that subsequent filters (mechanical limits in your speakers and your ear, which respond per F=MA; inherent limits in subsequent components) achieve he filtering. There will be no aliasing after the DAC. Yes, there could be HF noise residue; but hearing is highly attenuated above 22 kHz (if present at all) anyway.
Any presumptive filters won't be ones that do it by the book. HF noise residue most certainly will be present in the absence of an anti-imaging filter.

I would agree that subjectively, having images present is far, far preferable to having aliases.