This guy is very entertaining. He explains why CD-R sounds different than pressed CDs: it's due to "clock jitter due to power supply noise". Oh.

I agree the guy writes as if he's been hit in the head by a windmill or whatever, but as far as I can ascertain, he's got a point there. Clock jitter is different for all types rewritables apparently (rather wildly) and they filter out the noise, but not perfectly. So even bit-perfect can sound different on another medium - that much appears to be grounded in fact.

Deconstructing kooks is only fun for me if it's a learning experience, and I still wonder how he got his degree(s) then- in Germany they don't take abuse of titles lightly, and they don't really have Cardboard Colleges AFAIK.

I still wonder how he got his degree(s) then- in Germany they don't take abuse of titles lightly, and they don't really have Cardboard Colleges AFAIK.

What is his degree in? Not electrical engineering, I can assure you. Probably not in any of the physical sciences. Psychology, perhaps.

Probably not in any of the physical sciences.

Think this is where he got his MS (is what Dipl Ing stands for), but cannot find a real bio:

1993: Microprocessor controlled security locking system with one-wire interface and serial network programmability, University of Wuppertal, Germany.

Yep, digital psychology, allright. The guy is obviously an outcast, and not a little bit weird judging from his writing, but I think I'll make a little detour here anyway - not *all* of his stuff is wrong, anyway - and most of it new to me anyway. Also, the baby-talk in his exposes can be annoying, I do get the point a lot faster than from a normal tech book, where they assume you're familiar with stuff from last semester, which in my case never happened - so for at least a few basics I'm grateful already - and that stuff is easily checked.

[edit] Just so you don't get worried - I will bring LOTS of salt when I go read it.

ORIGINAL: tarsier

Sorry I don't have some spectrum shots to show you, but yes there is stuff above 20kHz. I normally record at 44.1 or 48 kHz depending on final delivery format, but I was curious about the whole 96 kHz thing as well. So I did some recordings of cymbals with an Earthworks mic (flat response out to 30 kHz) into a MOTU 828 mkII at 96 kHz sample rate. There was plenty of stuff above 20 kHz that got recorded. Then I tried recording a clarinet, and there were still plenty of harmonics being recorded above 20 kHz. They were down around -85 dBFS and lower, but they were there.

So at least with that combination of mics/converters there was plenty of material being recorded above 20 kHz.

Here is a spectrum example of "Triangle" (instrument), recorded using 24/96:

http://img265.imageshack.us/img265/4561/triangle2496ci4.png


Junski

ORIGINAL: Roflcopter

This guy is very entertaining. He explains why CD-R sounds different than pressed CDs: it's due to "clock jitter due to power supply noise". Oh.

I agree the guy writes as if he's been hit in the head by a windmill or whatever, but as far as I can ascertain, he's got a point there. Clock jitter is different for all types rewritables apparently (rather wildly) and they filter out the noise, but not perfectly. So even bit-perfect can sound different on another medium - that much appears to be grounded in fact.

Regarding CDs of the same bits sounding different. Here is a link to a report where they test that assertion. To sum up: Yes, different pressings of the same material can result in different jitter in the resulting clock signal.

Now, is it audible? My interpretation of the paper is: probably not. Look at the statistical analysis of the results. There is virtually no concordance between the different CDs and subjective quality.

Finally, read the part at the end with the two mastering engineers who when tested couldn't reliably tell the difference between different CDs, even though they claimed to be able to. When they knew which CDs were different they could hear the difference. When blind tested they couldn't tell the difference at a rate about equal to chance.

ORIGINAL: bitflipper
I can, however, recommend one source that did a pretty good job of explaining the Nyquist magic theorem -- a book by Nika Aldrich called "Digital Audio Explained for the Audio Engineer". Parts of it are mathematically intimidating, but he does a good job of illustrating how the audio is reconstructed from as few as two samples per cycle -- something that is not intuitive at all!

Until you have a chance to read the book, here's the bottom line, which you'll have to take on faith for now: Nyquist says that you can encode and subsequently recreate ANY waveform EXACTLY as long as you sample it slightly over twice the highest frequency you need to record.

Careful... That is what the Nyquist theory says. But it only holds true for infinite resolution sampling. In other words, the bit depth would have to be infinite to get the original waveform exactly. With digital audio, the devil is in the details so I'm afraid I have to be a stickler on this point. But 24 bits is more than enough.

Aldrich's book is excellent. Everyone should read it if they are doing digital audio. It has a few minor errors that I would quibble with, and I wish it went into greater depth on the D/A side of things because that's where the magic really happens (to me, anyway. The sync function is the whole reason it works, and I don't think the book even mentions it) but overall it's a truly terrific book.

Careful... That is what the Nyquist theory says. But it only holds true for infinite resolution sampling. In other words, the bit depth would have to be infinite to get the original waveform exactly.

I have a problem with this statement. As UnderTow so succinctly stated, it takes only two data points to fully describe a circle. Perhaps I am missing the point, and might understand it better if you could explain the term "inifnite resolution sampling". Does that mean an infinitely high sample rate? (otherwise known as "analog"?:)

Edit: Sorry, after I typed that I realized you were talking about bit depth, not sample rate. But I still have a problem with the statement.

As a programmer, I have four choices when declaring an integer variable. Namely, do I want to use an 8-, 16-, 32- or 64-bit value (actually 8 choices if you count signed/unsigned)? Which of these I choose depends on two criteria: a) what is the largest value I will need to store, and b) will the variable be used as an argument to a function that expects a specific datatype?

Note that the choice has nothing to do with precision. As long as a given value is <= the highest value the variable can store, it will always have exactly the same precision regardless of bit depth.

Whether the data represents a snapshot of an analog waveform or the number of angels currently occupying the head of a pin makes no difference. In audio, it means that bit depth determines the dynamic range, not the precision of numeric values.

tarzier, I am not flaming you. I appreciate your thoughtful contributions to this and other threads. But I suspect (and I won't take it personally if you prove me wrong) you've bought in to the notion that there are exceptions to Nyquist when there are none.

bitflipper: I think tarsier has issue with the word "EXACTLY". He is right that it isn't 100% exact unless you have infinite resolution aka infinite bit depth but then rightly goes on to say that 24 bits is more than enough [for all practical purouses]. (Bit in brackets added by me). It is all about covering the human dynamic and frequency range. We don't need "EXACT" copies of the original because we humans can't tell the difference anyway.

I think it is usefull to repeat that sampling rate does not equal resolution. It equals bandwidth (assuming you don't want aliasing). Bit depth is what amounts to resolution.

I'd also like to repeat what has been said by many people: Only in audio do we go to such ridiculous over implementation of sampling theory. In fields where accurate results are MUCH more important like RADAR, Sonar (no pun intended), medical equipment, telecommunications, satelite positioning etc etc, engineers stick to the Nyquist theorem. It is only because audio is so subjective (aka we can NOT trust our hearing and our sound perception) that things have gotten so ridiculously out of hand. There is a certain amount of machismo and self-delusion in this field that unscrupulous manufacturers take full advanatage of.

UnderTow

Well said. Last night I googled the topic and found that every audio-related forum on the net has at least one long thread on this subject. Most of them had a surprisingly contentious tone! Surprising, because a purely technical point should be easy to resolve. Especially surprising because any principle supported by something as black-and-white as a mathematical proof should not engender debate at all.

Especially surprising because any principle supported by something as black-and-white as a mathematical proof should not engender debate at all.

Well spoken. Maybe that's why it's good that this thread recapitulates everything regarding the subject, and I get a chance to familiarize myself with the bogus arguments - once bitten, twice shy etc.

Therefore I don't think it's a waste of time or effort, not in the least.

BTW according to Einstein, if you travel long enough in one direction, you should end up where you started again, so essentially you only need one vector to describe a circle.

This was ofcourse topped by Richard Feynmann who pointed out you could theoritically explain the whole universe as one ubiquitous particle moving both forward and backward in time, meaning you could do all of reality with one pixel.

And all based on pretty solid math, Ricky could do his 'rithmatic awright.

[Thx for that link Tarsier, will devour it in small bites]