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In Reply to: If thats the case then why are digitial masters... posted by EdAInWestOC on September 26, 2005 at 08:39:05:
"cut using sampling rates quits a bit higher than 44.1 kHz? If 44.1 was all thats required than wouldn't it be a waste of storage space to use higher rates? "
Becuase it's harder to create a good master in redbook, but that doesn;t mean it can;t be done. It just puts greater burden on the engineer.The googled text is correct.
YOur statement:
"Since music is so dynamic (or any non-cyclical waveform for that matter) then the sampling rate is still critical to faithfully reproducing an analog waveform. Its not as simple as the theorem would indicate since little in nature is as well behaved as we would like. The project I was involved with started with such an assumption but ended up being sorely insufficient. The resulting sampling rate ended up being 8X what Nyquist Shannon would have indicated. I believe thats why the theorem states at least 2 times."Is incorrect. at least 2 times is reqwuerd to PERFECTLY reproduce the analog waveform. ANything over that is a waste.
I have no idea what you mean by well behaved. Are you saying music is not a perfect sine signal, and hence is "more complex" than if it were a perfect sine wave?
IF that is the case, you are agin pain wrong. Google fourier transforms and see!
ML
Follow Ups:
I can see why.Ed
We don't shush around here! (Siegfried)
My system
Just out of curiosity...What is your background? You throw Nyquist and sampling theorem around as if you just graduated tech school and want to play expert.
Is that the case? Or have you just had you head buried in textbooks so long that you can't see the forest for the trees? Have you ever done any work in the field?Oh, and consider this: If an amplifier was flat to 20K and dropped like a stone after that, conventional wisdom would state that the amp is severly flawed. The same could be said for a loudspeaker. (and before you jump up and down asking for references, please read Otala and virtually anything written in the late '70's through now on slew limiting). So, given that... how can anyone accept the same limitations from a source and call it high fidelity?
Does it matter who the messenger is? The question is: is the message correct? Would it help if I had a phD? Help you that is? Well, I do.I have never worked professionally in the "field", but I do dabble a lot in pro & way-out-there SET equipment. I have set up 5-6 systems of my own, in different rooms in my house, mostly using speakers I built myself.
Been doing this for about 4 years now. It's fun. and "Cheap"...ok maybe it is cheap but you sure end up spending a lot.
"If an amplifier was flat to 20K and dropped like a stone after that, conventional wisdom would state that the amp is severly flawed. The same could be said for a loudspeaker. "
Why? IF we can hear only upto 17 Khz or so, and the amp behaves gerat till 20KHZ, it should be fine.
Class D amplifiers work great within limited frequencies. What do yuo think the cutoff should be? 50KhZ?
For your hypothetical amplifier, what do you mean by dropping like a stone? What drops like a stone? Is it the slope of the curve that worries you, or that it would stop reproducing the waveform accurately at 20 KHZ, but work great till then?"Or have you just had you head buried in textbooks so long that you can't see the forest for the trees?"
I could counter with an equally cheap shot about naive beliefs in vinyl groove encoding being superior to redbook, but I won't.
I'd prefer if you just responded to the content of my posts, and ceased trying to make this personal.
ML
And also
...if you're going to defend digital recording on technical/theoretical grounds.You're taking an overly simplistic "Engineering 101" aproach to something that is technically a lot more complex than you acknowledge.
This paragraph:
Why? IF we can hear only upto 17 Khz or so, and the amp behaves gerat till 20KHZ, it should be fine.
Class D amplifiers work great within limited frequencies. What do yuo think the cutoff should be? 50KhZ?
For your hypothetical amplifier, what do you mean by dropping like a stone? What drops like a stone? Is it the slope of the curve that worries you, or that it would stop reproducing the waveform accurately at 20 KHZ, but work great till then?suggests that you don't understand how analog domain filters behave. 17kHz to 20kHz is a fraction of an octave, and any filter that had a steep rolloff across such a narrow range will have severe time domain problems, including phase effects that will reach way down into the range of audibility. Depending on the selected filter characteristic, there is likely to be SEVERE ringing, whcih will almost certainly be audible. It becomes pretty hard to keep supporting the "technical superiority of CD" argument once you understand what's really happening.
Peter
You are right, I don't understand how they behave.
But I can think it through.
That's why Ed Mitner's IDAT and then BIDAT are such great advances, and they happened 10 years ago.
ALso, I think (and I may be wrong, you seem to know more than I do, so maybe you can help) recent DAC chips seem to have solved some of these problems as well (the more recent Burr brown & Crystal chips).
What do you think?
ML
Ok, fair enough....Any time you implement an analog filter there are a number of tradeoffs that a designer has to make. The closer the filter gets to "perfect" brickwall performance in the frequencey domain, the further it gets from perfect time-domain behavior. In other words, you can't have everything; there is no free lunch. Every designer chooses based on their interpretation of what's important.
The filter at the front of an AD has to be incredibly steep, so it MUST introduce time-domain problems.
Think about loudpeaker xovers; there is lots of debate about the audible effects of high-order filters, but very few designers use steeper than 4th order xover slopes (Jeff Joseph excepted..). If you used a 4th order 17kHz filter on your AD, you're looking at only 24dB down at 34kHz; nowhere near enough. Oh, and you're also 3dB down at 17kHz, which will definitely be audible....
Vinyl, on the other hand, tends to have a much more well-behaved HF rolloff, with a response characteristic defined up to about 50kHz (I think that's about the cutter head limit). Vinyl certainly has lots of other problems (my pet peeve is wow on piano recording from not-perfectly-centered LPs..) but it is NOT clear that CD is inherently superior in many areas.
BTW, check out the link below. Quite an eye-opener. Could vinyl actually be technically superior?
Peter
I've seen this link before and while I fall squarely in the vinyl camp, that site can't be taken seriously. He's using a Rega P3 and has somehow matched waveforms to CD's. I had a P3 and I've heard and seen numerous others and not one measured at the correct speed. Also, how did he match levels for the recordings when going from his amp outs to his sound card? It's a good trick and one I wish I could do. CD's and records almost always have different levels. Perhaps he used a normalizer. At any rate, there's too many variables to form an opinion from the info given, either for or against.
"Vinyl, on the other hand, tends to have a much more well-behaved HF rolloff, with a response characteristic defined up to about 50kHz (I think that's about the cutter head limit)"Depending on the cutter there are low-pass filters inserted in the cutting amp from 30kHz to 50kHz (I've seen the first, and heard from the second). These screw around with the RIAA pre-emphasis and assure that what you hear at home, with a 'perfect RIAA' phonostage, is not quite what was intended. In addition to this, just about every cutting room has a hard treble limiter that kicks in when the music contains too much treble. 'too much' being not an awful lot as cutting coils really can't sustain a lot of treble.
If you want to name this a well-defined response characteristic ...
Now during vinyl replay distortion rises with frequency (this a law), assuring that what you get back in the 20-40kHz band exists more of distortion components from the baseband signal (0-20kHz) than anything really valid above 20kHz.
If you don't believe this, then just measure it yourself.
I was trying to be careful in that post; not to suggest that there is anything particularly wonderful about vinyl HF behavior, but simply to point out that it probably has a much simpler filter characteristic than the typical redbook CD beyond 10kHz or so. Given all of the variables, what do you think is the typical "effective order" of vinyl HF rolloff with an MC cartridge? I can't imagine that it's much more than 3rd or 4th order, and therefore a lot more benign than the CD brickwall. Hence the "well-behaved" term (as opposed to well-defined).As you point out, the RIAA curve pre-emphasis can't really extend forever, so I always (like lots of others) add a break to the EQ curve somewhere out in that 50kHz region to allow for this.
Obviously, there's lots that is imperfect about vinyl playback, but the simplistic "44.1/16 is more than you need" is just that -- an oversimplification.
Thanks for the post clarifying the ADC situation; I suspected that most real world designs used an oversampling technique, but wasn't certain.
Peter
The Sampling Theorem is very valid and in fact very beautiful (read Shannon's proof, if you need proof ;-).But all too often people saying that the ST is Valid are misunderstood, as if they were saying that 44.1k is plenty.
It isn't, even though we only (might) need a 20kHz bandwidth.
If the AA filtering is done in the analogue domain, then the end result is below the quality of a 30kHz tape machine. So a 60kHz sampling rate would have been better.
If the AA filtering is done in the digital domain, linear-phase, through oversampling and decimation, then ringing is inserted at about 20kHz. This is unnatural (as opposed to the ringing from oversampling DAC filters, which is entirely appropriate, read Shannon closely!).
There is no way around this, unless increasing sample rate again to 60kHz or so, and adding a little bit of minimum-phase low-pass filtering in the 20-30kHz band. This kills the ADC ringing.The only alternative would be to do filtering in the 15-20kHz band and live with a low-bandwidth medium.
Nice posts, Peter & Werner.
Is redbook cd, properly done, fine for the 0-20KHZ bandwidth?
I feel we have had an entertaining and nuanced discussion, but do w ehave a conclusion, or should we all just agree to disagree a little, but generally disagree?
ML
My conclusions are1) the sampling theorem is pretty neat
2) if back in the late seventies the aim was to launch a digital
distribution format with qualities exceeding those of an average
decent record player or an average decent consumer tape deck,
then they succeeded with 44.1k.3) if back in the late seventies the aim was to launch a digital
distribution format with qualities matching those of an average
decent master recorder then they failed, and 60kHz would have been
much better.But 2) is a stupid decision, from an audiophile point of view.
It is of course more than good enough from a marketing POV. And 3) would have stretched technology, requiring a much steeper investment and later time to market. Don't forget that back then several formats were competing for the position of follow-up to the LP and cassette.
In the end, back in the late seventies the only real aim was to assure making a lot of money in the 80s and 90s.
University of Tulsa, OK. I case anybody give a flying.Ed
We don't shush around here! (Siegfried)
My system
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