Do you have any experience using reel to reel tape machines?I learned about it back in the 1980s: https://en.wikipedia.org/wiki/Tape_bias. Contrary to popular opinion, the gap of recording head is not the limiting factor, the gradient of magnetic field at the trailing edge of the recording head is. Yes, nearly-perfect azimuth is important, and setting it on a machine without a servo-adjuster could be an unpleasant chore.
The gap at playback head, yes, is a limiting factor. We replaced worn out playback heads much more often than the recording heads. I recall the laser-cut gaps being on the order of several micrometers on high-end machines back then, and https://ccrma.stanford.edu/courses/192a/Lecture7-Magnetic_recording.pdf confirms that.
I'm taking about serious machines from the 1970s. Like Ampex ATR-100. See page 32 of https://www.americanradiohistory.com/Archive-DB-Magazine/70s/DB-1976-12.pdf, and page 36 of https://www.americanradiohistory.com/Archive-DB-Magazine/70s/DB-1977-02.pdf. Note the bias frequency: 432 KHz. Note servo motors power: 1/4 HP. Note the discussion about phase coherency and testing with square waves.
ATR-100 SNR spec is not entirely clear: I've seen 68, 72, and 80 dB at 30 ips (depending on tape material and weighting?). Was this that much worse than the PCM decks of that era, such as 13-bit Sony PCM-1: http://www.thevintageknob.org/sony-PCM-1.html? In its marketing materials, Sony only dared to compare the PCM-1 with a 15 ips reel-to-reel.
Moving on to 1980s, consider Studer A820: http://www.theaudioarchive.com/TAA_Tape_Studer_A820.htm. SNR up to 77 dB (A-weighted). For a while, those monsters were still competitive with the direct PCM recorders. Yes, as I mentioned in the previous post, distortions and noise were always their weak spots. Yet were they as bad as those of the consumer-grade tape recorders? Clearly not.
What gets me is people not getting that analog SNR is not the same as digital SNR. In the analog case, as long as the noise floor at -77 .. -68 dB is masked by music, the music itself doesn't really have the amplitude resolution of just 13 or 14 bits as so many people think. It is analog: you can digitize it with 24 bits and this will be meaningful.
Similarly with timing. Yes, electronics band-limits the signal, typically to 22 KHz on high-end machines. Yet again, being an analog system, it doesn't need the 10-20 ms for the dithered PCM samples to catch up with the true signal value with a precision of ~0.2 LSB, after a signal's component "jumps": what's recorded on the tape just "jumps" too, straight to a very close approximation of the true value.
Of course, if you just plug the output of a reel-to-reel deck into DAC and push play and record buttons, you are going to record the hiss on silent intervals. Gate it out, digitally if you must. The real-life music's dynamic range rarely exceeds 45 dB anyway. IMHO, 192/24 PCM carefully captured from a high-end analog tape master can be better sounding than a CD.
Theorem. You keep using that word. I do not think it means what you think it means.
And seriously, I don't think you have a good grasp of Fourier or Shannon, and until such a time that you do, you're still going to be all over the place with irrelevancies.
More or less what's being said here:The solution for you is really simple. Playback your digital recordings (the ones that have a higher S/N ratio from the studio recording) and mix some analog noise of around -70dB which 'masks' the things you seem to worry about.
http://www.aes.org/aeshc/docs/3mtape/printthrough.pdfThere isn't an analog recording tape made that doesn't suffer from print through...
...Signal-to-print may be the constant that measures print-through potential, but the real print-through that makes you cringe is measured by the print-to-noise ratio. A tape with a poor signal-to-noise ratio will actually make you cringe less, because the noise on that tape will mask the print through you would otherwise hear. In other words if the signal-to-noise ratio is lower than the signal-to-print ratio, you would not hear the print through. However, you will definitely hear the noise.
Yes, this dither convergence thing might be real in the sense that it can be analysed mathematically (as I was saying earlier, you can keep zooming in until the errors fill the screen unlike analogue which just isn't analysable that way) but the size of the problem is so small that even if you boost it by 60dB without the masking of the music it still can't be heard. It's just a question of getting things in proportion - which the zoomed-in view of theoretical digital errors fails to do.
What about that quote I linked to earlier? Is it correct in what it is saying?
Sounds like modulated noise, the very worst thing imaginable in a digital system - apparently. What effect does this have on timing?
Hmm. You seem to be saying that the noise and distortion has 'infinite' resolution, therefore it's benign. But if it's modulated noise, distortion or print-through etc. then just because it has 'infinite' resolution doesn't mean it isn't producing genuine, repeatable, signal-dependent timing errors (and all the other errors) that supposedly destroy the sense of space, etc. It's just that there isn't a finite fundamental, theoretical basis to its 'resolution'.Yes, what you are saying is correct. As is what RayDunzl said, and Blumlein 88, and Amir, which all repeated what I've been saying all along: analog has more distortions and noise than digital. Should I agree with you all about "way more distortions and noise"? Fine, I agree.
But what doesn't the analog have that the 44/16 has? Such rigid informational bandwidth limitation! The analog spacial resolution goes down to micrometers on physical media, which translates to microseconds and microvolts at the output. If you calculate to what bit depth and sampling rate these resolutions correspond, and then multiply them, it could be a ginormous number of bits per second.
Some people simply prefer this tradeoff: a nearly-perfect tracking of the analog signal, unfortunately burdened with baggage of distortions and noise. My hypothesis is that this could be a rational explanation of why RTR and LP didn't yet follow the way of NTSC. The oft-cited alternative explanation - these people are stupid, and thus let themselves be fooled by unscrupulous marketers and salesmen - doesn't satisfy me.
The analog spacial resolution goes down to micrometers on physical media, which translates to microseconds and microvolts at the output. If you calculate to what bit depth and sampling rate these resolutions correspond, and then multiply them, it could be a ginormous number of bits per second.
The analog spacial resolution goes down to micrometers on physical media, which translates to microseconds and microvolts at the output. If you calculate to what bit depth and sampling rate these resolutions correspond, and then multiply them, it could be a ginormous number of bits per second.
It seems to be that MQA has a secret that no-one can critically recognise or identify. Whether it is correct or not is irrelevant in terms of current internet charges. MQA is dead on download cost vs licensing cost.
Hmm. You seem to be saying that the noise and distortion has 'infinite' resolution, therefore it's benign. But if it's modulated noise, distortion or print-through etc. then just because it has 'infinite' resolution doesn't mean it isn't producing genuine, repeatable, signal-dependent timing errors (and all the other errors) that supposedly destroy the sense of space, etc. It's just that there isn't a finite fundamental, theoretical basis to its 'resolution'.
I don't buy this idea that analogue systems have evolved to make them "perceptually transparent" despite their obvious shortcomings. Analogue systems were developed by people watching moving coil meters, playing the same specs game as manufacturers do today. In order for there to be evolution based on pure perception, there would have to be feedback from Golden-eared listeners that resulted in changes to lathe settings, coil winding, metal composition, etc. that didn't result in better conventional specs or prosaic improvements like easier maintainability. If you can show how that happened, I would be interested.
It would seem that this path is not that expensive, and doesn't require 192/24. For example, some of the inexpensive DACs from Topping measure extremely well. And whatever is missing from 44-16, is difficult for the most sensitive well trained listeners to discern, and requires critical listening under controlled conditions with top notch equipment....
(1) Make all the distortions in audio delivery chain vanishingly small. That's the path that professional community, and part of audiophile community, has taken. It is not simple, it is expensive, yet it is universal for all genres of music, and - I agree with Cosmic - easier for scientists and engineers to understand, model, and measure. 192/24 PCM and its compressed variations, in combination with seemingly unreasonably low THD and IMD at every step, relentlessly chased down by investigators like Amir, is the contemporary practical embodiment of this approach.
...
The only gap MQA is designed to fill is the one in Bob Stuart's wallet. So far, it has failed to do that.I'm not saying 44-16 is perfectly transparent so we're done, but whatever gap 44-16 has is quite thin, and that I'm skeptical that MQA does anything to fill that gap.
So here comes the insight. What is common between (A),(B), and (C)? Gross distortions! In (A), they are mostly odd-harmonics, as Cosmik so eloquently reminded me. In (B), they are mostly even harmonics. In (C), they are all harmonics, carefully crafted to induce, via nonlinearities of transducers, and aided by DSP algorithms, the Missing Fundamental effect (https://en.wikipedia.org/wiki/Missing_fundamental).
Another hint for me was the perplexingly convoluted designs of some of the early transistor amplifiers, which were replacing in RTR and LP gear the still-fresh-in-memory vacuum-tube ones. Back then, I was thinking: "Seriously, fourteen stages of amplification? With diodes in a local feedback loop? What other than massive distortions are you going to get out from this?". But that was precisely the point! Systems as a whole did sound good.
If you put (A) and (B) together, you'll get (C). Odd harmonics + Even harmonics = All harmonics = illusion of a deep bass + perceptually arguably benign timbral deviations. In other words, there are two paths to perceptual high fidelity:
(1) Make all the distortions in audio delivery chain vanishingly small. That's the path that professional community, and part of audiophile community, has taken. It is not simple, it is expensive, yet it is universal for all genres of music, and - I agree with Cosmic - easier for scientists and engineers to understand, model, and measure. 192/24 PCM and its compressed variations, in combination with seemingly unreasonably low THD and IMD at every step, relentlessly chased down by investigators like Amir, is the contemporary practical embodiment of this approach.
(2) Keep adding distortions until they perceptually cancel each other. Majority of consumer audio gear vendors, and another part of audiophile community, took this turn. Despite being cheaper and easier to achieve as a first approximation, it is not universal in regard to genres of music, is very hard to model and measure, and keeping perceptual improvements coming gets prohibitively expensive, especially in terms of tinkering time spent, after a certain point. Nevertheless, the boutiques owners seem happy playing new age and moderately paced urban contemporary on that gear.
The framework described above allows to explain the mystery behind the still raging, after all these decades, analog vs digital controversy. CD was one of the very first generations of products on the path (1) that mass-market consumers could buy. High-end RTR/LP media and gear were the last generations of analog products on path (2). The CD is superior in many, yet not all, perceptual aspects, to the high-end RTR/LP. And vice versa.
Due to differences in hearing systems, and other factors, some of which could be unrelated to the perceptual quality of sound, some people prefer CD to RTR/LP, while others have opposite preference. Yet others prefer CD for some genres of music, or even individual music pieces, and RTR/LP for other genres and pieces.
In my opinion, there is strong evidence indicating that 192/24 PCM, if delivered inexpensively, and yet in such a way that it won't be easily stolen, may persuade some of the folks currently pursuing path (2) to switch to path (1). MQA purports to be such a technology. That's why it is of interest to me personally. Yet it is not the only technology that can aid the (2)->(1) transition, in my opinion.
You may be able to design circuits where distortion in one part cancels distortion in another part, e.g adding them if they are 180 degree out of phase. Some op-amp designer do this AFAIK. The result is what you mention under (1). This approach can only work within a specific unit. It cannot work such that one unit (e.g. preamplifier) cancels distortions in another unit (e.g. CD-player), since for this the designed or the preamplifier must know which CD player will be used.[..]In other words, there are two paths to perceptual high fidelity:
(1) Make all the distortions in audio delivery chain vanishingly small. That's the path that professional community, and part of audiophile community, has taken. It is not simple, it is expensive, yet it is universal for all genres of music, and - I agree with Cosmic - easier for scientists and engineers to understand, model, and measure. 192/24 PCM and its compressed variations, in combination with seemingly unreasonably low THD and IMD at every step, relentlessly chased down by investigators like Amir, is the contemporary practical embodiment of this approach.
(2) Keep adding distortions until they perceptually cancel each other.