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Does Phase Distortion/Shift Matter in Audio? (no*)

You are essentially restating what I already said.
No you didn't. You have said, but have yet to show how that impulse is band limited as you claimed. Exactly what band limited it? And what is the non-band limited version looks like if that is the case?

Remember, the nature of the signal is at it sits. Not after it has been filtered by the DAC's reconstruction filter has been applied to it.
 
I previously mentioned a sine wave near Nyquist because it is a clear example of a signal that can be created using a realizable antialiasing filter which will also be reconstructed inaccurately by the filters found in most DACs.
This has nothing to do with the discussion at hand. We are talking about the bandwidth of the excitation signal. In this case, the impulse.

What a perfect or imperfect filter in the DAC looks like is orthogonal to this topic.

Indeed, the discussion started with talking about the impulse signal being used to test speakers which would be in fully analog domain. Explain how that signal has limited bandwidth with no filtering applied to it.
 
Continuous time? How did you get from discrete time domain to continuous time? Let me guess: by having a reconstruction filter? I can't believe I am having this conversation.

One more time: we use an impulse function for one and only one reason: that it has infinite bandwidth with flat spectrum so can characterize any (frequency) transform of a black box system. The notion that it is already band limited by some unknown magic, completely takes away this value.
 
I can't believe I am having this conversation.
Likewise. This discussion is going nowhere if you refuse to read what I wrote or the references I linked. What you are disputing is proven mathematics that can be found in any digital signals text or in the relevant Wikipedia pages.
 
I have not measured, yet, a real world DAC that would not have a kind of mirror image issues, spread as well below Fs/2, when the signal frequency approaches to Fs/2. Maybe the manufacturers and designers might add some margin and decrease filter Fc a bit to get rid of the mirror images garbage. The same applies to ADC anti-aliasing filters, which are often worse.

Below is an example, which is still very good. However not technically perfect.
JCally JM6 and Samsung dongle can do that, although as per their response (third graph), JCally will heavily attenuate that 20 kHz signal and Samsung dongle will produce some images for signals slightly above 20 kHz. Here's 20 kHz @ 44.1k played through them and captured with RME Adi-2 Pro, 13 dBu @ 96k, cursor at 24 kHz:

JCally JM6
20k.jm6.png


Samsung dongle
20k.samsung.png
 
The visible images in a DAC (or converse HF aliasing in an ADC) often arises from a rather annoying bad habit of some, but not all, DAC makers, who use "half-band filters". There are advantages, every other tap of the filter except at center, is zero, and so you can make the filter longer. This, however, means you're only 6dB down at exactly FS/2. Now you can make those filters long and very, very steep, which is a common choice, but that means right around fs/2 there will be some visible crud.

There are also regularity problems (read Debauchies and Calderbank's regularity paper, which I read long ago, sorry, and haven't a copy) that can bite you with half-band filters. The conclusion is that there must be AT LEAST one zero at fs/2 in the system, but that message has not gotten to a lot of folks, apparently.

Not all DAC's have this problem. Of course, the 'longer filter' also runs more risk of having issues with pre-echo, and some with a 1dB ripple (*&(( well do have pre-echo that's easily audible, but at least most of those have aged out and gone away.

The real problem, of course, is that price rules in chip manufacture, always and forever, at 44 and 48, and that many people making 88/96k ADC/DAC's just stick with the sharp filters rolling off above 40K or so, rather than use a broad transition band, much shorter filter that starts to roll off at maybe 25k or 30k, which gets them completely out of danger at the same time eliminates any chance of audible pre-echo.
 
You are misrepresenting what I said. I did not say that the original, continuous time sine wave has infinite bandwidth. A discrete time signal is conceptually a sequence of shifted and scaled Dirac delta functions when represented in continuous time. This creates an infinite series of images above Fs/2 (assuming uniform sampling). The reconstruction filter removes these images (imperfectly in practice as an ideal filter is not realizable).
Yes, that's nicely explained (imo) in https://www.dspguide.com/ch3/2.htm , starting from paragraph:
Now we will dive into a more detailed analysis of sampling and how aliasing occurs. Our overall goal is to understand what happens to the information when a signal is converted from a continuous to a discrete form. The problem is, these are very different things; one is a continuous waveform while the other is an array of numbers. This "apples-to-oranges" comparison makes the analysis very difficult. The solution is to introduce a theoretical concept called the impulse train.
Of course it won't hurt to read that chapter from the beginning :-)

(one slightly annoying thing is that the description references figure 3-5 which is in the next chapter)
 
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Yes, that's nicely explained (imo) in https://www.dspguide.com/ch3/2.htm , starting from paragraph:

Of course it won't hurt to read that chapter from the beginning :-)

(one slightly annoying thing is that the description references figure 3-5 which is in the next chapter)
There is nothing whatsoever about the topic under discussion in that chapter or what you quoted from it. Nothing has been "sampled." Nothing started in analog/continuous domain. A file was created digitally and hence, bypassed both of those stages and with it, violates sampling theorem. To understand that, you need to have an understanding of impulse function by itself and what it represents. If you are confused about that and think it is band limited, there is no hope of you ever figuring out anything about signal processing.
 
@j_j, if you have the time, could you please comment on whether a discrete-time impulse (i.e. a single nonzero sample with infinite zero samples on either side), generated directly in the discrete-time domain violates the bandlimiting requirement imposed by sampling theorem? Do note that I'm specifically not asking about a practical A/D/A system, just the sampling theorem itself.

I'm pretty certain I understand the fundamentals of discrete-time signals (I've written resampling algorithms from first principles and many other DSP routines) and Amir's recent claims in this thread seem to directly contradict every digital signals text I've read and, in turn, the fundamental mathematics governing discrete time signals. If I happen to be wrong, especially the points I made in this post, could you point me in the right direction?
 
@j_j, if you have the time, could you please comment on whether a discrete-time impulse (i.e. a single nonzero sample with infinite zero samples on either side), generated directly in the discrete-time domain violates the bandlimiting requirement imposed by sampling theorem? Do note that I'm specifically not asking about a practical A/D/A system, just the sampling theorem itself.
The answer to that question is "it's indeterminant" at pi. Consider what the phase of the spectrum does at -1,0 . For testing a DAC filter it's ok, but it presents a problem in the same way that half-band filters in a DAC upsampling/ADC downsampling chain create a problem with aliasing/imaging reduction. Don't ask me to explain that, Ingrid Debauchies and Rob Calderbank are required to explain that. It's tied into the regularity theorem for wavelets and biorthogonal wavelets. I simply remember that it's resolved by having at least 1 zero at pi, which means you can never MAKE that signal from a continuous system that's being sampled, and that you'd better have at least one zero or zero pair at pi of the final (smallest) sampling rate of the actual system. Generation in the sampled domain of course allows you to make this signal, but it's an oddity at the very least.

It is, of course, something you could not ever generate with a time continuous signal put into an ADC with anything proper in the anti-aliasing chain, though, so it is NOT something that would appear in any real data. Ever. There are a number of issues with creation of signals in the "digital" domain (bear in mind that "analog" is just a time continuous, amplitude continuous analog, and that "digital" is a discrete time, discrete level analog.
 
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... Since we can't have non-casual filters in real life, the only method we'll have these samples (i.e. the Kronecker delta function) is by synthesizing them in the samples domain with a computer. ...
Why can't we? It seems that all we need to do is capture raw sampled values and then apply the filter afterward so it can "look ahead" in time.
 
Why can't we? It seems that all we need to do is capture raw sampled values and then apply the filter afterward so it can "look ahead" in time.
Thank you! Your post can't make it any clearer the point that this signal with only a single non-zero pulse (the Kronecker delta function) is entirely "artificial" in nature.
 
All digitally created files used for testing are “artificial” in nature. Such reasoning about their invalidity is pointless.
 
Do note that I'm specifically not asking about a practical A/D/A system, just the sampling theorem itself.
If even an idealized ADC cannot create that signal, then you have your answer.
 
All digitally created files used for testing are “artificial” in nature. Such reasoning about their invalidity is pointless.
That's like saying a car should go just as fast backward as forward.... Just because test signals can all be created by a computer doesn't mean they are all just as applicable to the system under test. Signals that fundamentally violate sampling theorem must be used with caution and full knowledge of what you are doing.
 
We should not forget that today's music hardly contains any real and unprocessed ADC capture. Therefore a DAC must cope with signals that an ADC would never produce.
Same thing as with intersample overs. A proper ADC never produces those but that's no sufficient reason why a DAC wouldn't need to care.
 
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