Ask an Audio geek

[IATIA]

Here we go.  My argument is that one cannot exactly digitize analog signal.  It is not that one cannot digitize an acceptable representation of an analog signal.

I also stated that the best one could do is to digitize 50% of the actual signal. Actually, I said that 50% was the best that could be done, but unachievable.

Here we sample a 1KHz signal at 1KHz - Flat line

---

---

Here we sample a 1KHz signal at 2KHz - Still a flat line

---

---

Here we sample a 1KHz signal at 4KHz - now we have a sawtooth wave

---

---

Here we sample a 1KHz signal at 8KHz - Now it is starting to resemble a sine wave

---

---

Here we sample a 1KHz signal at 16KHz - We now have a ‘reasonable’ facsimile of a sine wave

---

---

Now the sample point.  Do we sample on the rising edge, the falling edge or take an average reading?
None will get the actual peak

---

---

Here is a 4KHz and 20KHz sine wave sampled at 64KHz

---

---

We can keep zooming in and, just like a Mandelbrot Bug, keep finding more artifacts. The upper limit for audio frequency at STP is around 3GHz.  At that point, the molecules are too far apart to transfer the sound.

This ‘magical’ math that compensates for the information between the samples is pure guess work.  Maybe there is a pulse or a dip or it just follows the apparent path.

I have shown that a 1KHz signal needs a 16KHz sampling rate which means a 20KHz signal will need a 320KHz sampling rate and still does not cover anything in the 40KHz harmonics.

If we have an infinitesimally narrow sampling point at an infinitesimally high rate, we can still only check every other point.

So, I state it again.  One can get a reasonably accurate and acceptable digital representation of an analog signal that one would have difficulty at best to prove it was not an analog signal, but it IS NOT AN EXACT COPY of the analog signal and can never be.

(And there are significantly better ways to sample an analog signal then what is being used in CD/DVD technology today.)

[Alex K]

Concerning the magical math you say is guesswork, and the results not looking like a sine wave. Have you read anything I've written?

I don't think anyone claims that any technique on earth can make a *perfect* recording of a signal, be it analog or digital. I've already mentioned that for digital it needs to be limited to frequencies below half the sampling rate and that near the sampling rate there are problems because one cannot perfectly cut off a signal at some frequency. That the signal built from the digital needs to be sent through a second filter to properly cut frequencies again. That one can arbitrarily reduce these effects by oversampling. None of this is guesswork. A lot of people worked hard to understand these things, and they are understood.
You need to argue what it is you think analog can do better and why that difference is relevant.

[IATIA]

Concerning the magical math you say is guesswork, and the results not looking like a sine wave. Have you read anything I've written?

Yes I have.  It still is guess work on the part of the program. You do realize that music is not 100% sine waves do you not?  What if it really is a sawtooth wave?  Percussion instruments.  There is no way to know what is in between the samples.  There can be a high probability.  I am not dissing the math.  All I am saying is that digital cannot make an exact copy of analog.  I have seen many of your posts and I know you are quite smart and quite intelligent.  I do not understand why I am being fought on this.

My post has enough information to show my claim to be true and it only scratches the surface.  But I am done with this thread.  Fighting two gods is too much.

[vorlon13]

Can my ears differentiate between a reproduced 16 khz waveform sampled at 44.1 khz and one at 240 khz ??

Can anyone ??

[Alex K]

@IATIA
Digital can, in principle, exactly reproduce bandwidth limited analog. Do you disagree?

[Alex K]

Can my ears differentiate between a reproduced 16 khz waveform sampled at 44.1 khz and one at 240 khz ??

Can anyone ??

Not if it is properly done, I'm willing to bet. If you botch it by using insufficient filters etc., of course. But you also have to check other things like whether hatd transients are reproduced nicely. If you have a filter at the output that keeps ringing after a jump, that sounds nasty.

[Jackalope]

Earlier I was looking for advice for some audio gear on a limited budget - I ended up buying a Pioneer VSX-830-K A/V receiver (on sale) with a pair of Klipsch 5 1/2" bookshelf speakers (not on sale). I plan on adding some floor speakers and a subwoofer soon. I was pretty amazed at the features available for the money compared to the last time I bought audio gear 20 years ago that I lost in the divorce.

[Kosh]

Earlier I was looking for advice for some audio gear on a limited budget - I ended up buying a Pioneer VSX-830-K A/V receiver (on sale) with a pair of Klipsch 5 1/2" bookshelf speakers (not on sale).  I plan on adding some floor speakers and a subwoofer soon.   I was pretty amazed at the features available for the money compared to the last time I bought audio gear 20 years ago that I lost in the divorce.

I was slow in replying to your original question so I'm not sure you even saw my suggestion.  My response wasn't very close to your original question in the thread. 
  
A lot of people think you need a matched speaker set in a 5.1/7.1 surround setup, but I don't necessarily buy into that.  Most receivers have the ability to do room calibration with automatically sets the volume on the speakers.  I personally have a major hodge podge of speakers in my 7.1 system (Klipsch Chorus II fronts, Vandersteen 1b sides, Polk Audio RTiA3  rears, Vandersteen sub and center channel).  From my perspective it all blends together after room calibration and their volumes are set. 
 
My original suggestion still stands with a speaker and sub.  For floor standing speakers I would go with the ELAC Debut F5 or F6, and an HSU research STF-2 subwoofer.  I think the options in your price range will be limited if you want to keep it all Klipsch.

[AFTT47]

I do not understand why I am being fought on this.

My post has enough information to show my claim to be true and it only scratches the surface.  But I am done with this thread.  Fighting two gods is too much.

It's very simple actually: Because you are wrong. Worse still, you're doing exactly what a creationist does: Arguing against facts. You seem to have a mental block regarding the gaps between samples. If there are enough samples for the frequency you are sampling, it doesn't matter. The wave can be reconstructed at a fidelity that is superior to any analog technique. Or do you think all the measurements made which prove the concept works as advertised are "magic" too?

The error you are making is dissing something because you don't understand it - in spite of evidence to the contrary and even facts. If you operate that way, you're going to have a pretty messed-up picture of reality because no matter how smart you are, there are going to be things you don't understand. In those cases, defer to experts who do understand it or even better, to objective measurements and established facts. Both are available when it comes to the fidelity of digital audio. You're just plugging your ears and going, "nah, nah, nah."

[Alex K]

There are two very interesting theoretical questions to me which connected with this issue: how large is the error caused by finite quantization resolution at various frequencies after sampling, and how does the fact that we only have a finite number of samples (or equivalently, not a perfectly cut-off frequency range) affect the fidelity of the recreation near half the sampling frequency. Both resulting errors can be made systematically and arbitrarily small in a desired range of course.