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Offline bensyverson

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Re: An engineer's take
« Reply #75 on: December 12, 2007, 12:31:38 AM »
:coolguy:  Thanks bensyverson for your lengthy posts - they made a lot of sense to me.  I still need to review another time or two to digest fully, though.
Hey, thanks! It's a lot of information, but hopefully people can use it as reference too. It always helps me to have pictures to visualize tricky concepts like these...

Offline DSatz

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Re: Analog to 24bit question
« Reply #76 on: December 12, 2007, 12:32:16 AM »
Focusing on the original question: The only aspect of audio quality affected by the choice of bit depth (or word length, or whatever else you want to call it--maybe "moo shu") in a properly made digital recording is dynamic range. The noise floor will be lower when more bits are used to quantize the signal, "all other things being equal" as they say. There are no inherent differences in frequency response, distortion, imaging, warmth, jitter, toenail fungus or any other aspect of audio quality beside dynamic range where the number of bits per sample is concerned (in linear PCM at a fixed sampling rate, anyway).

As a result, the original question doesn't really have anything to do with digital! The same question would occur even if digital recording had never been invented (and yes, I do realize that for certain people that is a fond wish ...).

What matters is this: If the medium that you're copying to has a distinctly (say 10+ dB) lower noise floor than the noise floor of the source material, and you use the available range fully, then the copying process will add only negligible noise. What certain people seem not to realize (or to want to admit) is this: Once you've reached that point, any meaningful further reduction in the noise level of the copy (i.e. by choosing a quieter medium to copy to) is simply not possible. Copying from an analog source can never add zero noise, nor can it add "negative noise". And once you're adding, say, only 1/2 dB of noise, you couldn't notice the difference if that could be knocked down to, say, 1/3 dB instead, even with extremely critical listening.

But that's the only real issue in the choice between a 16- and a 24-bit transfer. The noise floor of a 16-bit digital recording is WAY the frigging fark below the noise floor of the cassette recording; heck, a 16-bit recording has distinctly wider dynamic range than a 15 ips half-track Dolby "A" master tape (which I should know--having made many, many such recordings back in the day). Thus it will not matter at all whether 24 bits are used instead of 16 for this application, as long as the copy is made with reasonable care on reasonably good equipment. Nothing audible will be gained or lost as the result of 16 vs 24 bit quantization of a cassette playback--so I say, the person should feel free to use either one! If 24 bits feels more comfortable, use it. If 16 bits feels more economical, use it. Flip a coin, use Tarot cards, spin the bottle, whatever.

Let me also just say that for live concert recording, I use 24 bits whenever I can--but that's because live performances have a far wider dynamic range than cassettes.

--best regards
« Last Edit: December 12, 2007, 12:51:28 AM by DSatz »
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Offline bensyverson

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Re: Analog to 24bit question
« Reply #77 on: December 12, 2007, 01:16:37 AM »
The noise floor of a 16-bit digital recording is WAY the frigging fark below the noise floor of the cassette recording; heck, a 16-bit recording has distinctly wider dynamic range than a 15 ips half-track Dolby "A" master tape (which I should know--having made many, many such recordings back in the day).

It depends on what you accept as "dynamic range." 16 bit maxes out at 90db of theoretical DR, but the last 30db or so are pretty nasty. So if you have a tape with 60db of DR, and you really want to faithfully reproduce the quietest parts of a recording, 24bit becomes not a luxury but a necessity.

Of course, popular music doesn't really utilize that much DR -- it's mastered so that you don't have to keep fiddling with the volume when you're in the car. It might only be an issue with classical or experimental stuff...

Offline szumsteg

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Re: Analog to 24bit question
« Reply #78 on: December 12, 2007, 01:19:59 AM »
First, thanks to all of you for the good arguments on both sides. Lots of great info here, and it seems you all have taken this as far as possible which is good discussion. I knew this could be a rough discussion, and you all brought up great points. What am I personally going to do, well I will still keep taking my analog cassettes in at 16bit mainly because part of preserving the tapes I do is that I do not eq them, edit them or enhance them which many of you agreed the best part of being 24bit is. I am simply capturing what is there, for better or worse, splitting it in CDWAV and making backup flacs and making CDs for myself. Most of what I am preserving as well is simply bass, elec keyboard, vocals, electric guitar and drums, no stringed instruments or natural sound, just big rock concerts coming out of large PA's. Another part brought up is that what is being argued as "better" becomes so incremental at the end, the master taper just by being in a better spot, or using a better tape, or whatever could make 100x more difference than some of the final benefits of being at 24bit. Funny thing is many of these tapes are from 1980s and from europe, where they all used normal bias cassettes, so I bet the freq range you were speaking of was for high bias casettes. Like someone said in one of the posts, it still boils down to good equipment, good cables, a nice high performance computer or everything at the end is not consequential.

When I am doing masters from here out, its at 24bit because we can do it with the equipment today, so I am not against 24bit at all. Just for purposes of getting old items archived, ill keep doing what I was doing. Its not as if people have had any complaints as of yet

Reminds me of just how many problems are solved with tapes by the three L's of taping. Location, Location, Location...nail that and you don't need to spend hours and hours post editing.
« Last Edit: December 12, 2007, 01:26:38 AM by szumsteg »

Offline DSatz

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Re: Analog to 24bit question
« Reply #79 on: December 13, 2007, 12:14:47 AM »
bensyverson wrote:

> It depends on what you accept as "dynamic range." 16 bit maxes out at 90db of theoretical DR, but the last 30db or so are pretty nasty. So if you have a tape with 60db of DR, and you really want to faithfully reproduce the quietest parts of a recording, 24bit becomes not a luxury but a necessity.

If the term "dynamic range" is too vague for you, we can use signal-to-noise ratio instead. 16-bit linear PCM has 93 - 94 dB of it (unweighted) depending on how you like your dither, while Dolby cassettes have maybe 65 dB on a good day, and I'm being pretty generous with that number; you should write and thank me. 60 dB would be more like it. You can't tape an LP onto a cassette and not hear any tape hiss during the silences; that should tell you something important about how much dynamic range a cassette doesn't have.

Meanwhile your remark about the "last 30 dB" is something that nearly everyone on this board could find out for him- or herself within ten minutes, which I think would be very nice for all concerned if they (and you) did.

--best regards
« Last Edit: December 13, 2007, 12:54:50 AM by DSatz »
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Offline Petrus

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Re: Analog to 24bit question
« Reply #80 on: December 13, 2007, 06:26:40 AM »
Using the digital photography analogy with digital audio is sometimes usefull. There is one danger, though: in photogaphy more resolution is better because we routinelly want to enlarge the pictures or zoom in into them. In audio the corresponding thing would be making the waves larger (wider) which correspods to SLOWING DOWN the audio. In normal life this is never done. In photography it is always possible to reach the limit of the resolution by routine enlargement, in audio there is a certain limit, 20 kHz, past which human audio resolution does not reach and for this reason those frequences are not needed in normal life.

In photography better true resolution is better, in audio after certain level more resolution does not improve the signal anymore. Resolution = frequency range in this context.

Unnormal life would be slowing down audio for effects etc. unnatural manipulations.

Offline DSatz

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Re: Analog to 24bit question
« Reply #81 on: December 13, 2007, 08:55:54 AM »
Analogies with photography seem unfortunately to be misleading when it comes to the specific things that we are talking about here. To see why that is, we would need to talk about how the statistical distribution of particle size and shape is controlled during film manufacture--it would get very specific, which would turn people off. Plus I don't actually know anything about film manufacture, so let me pay respect to the very wonderful field of photography by not making analogies with it.

But that's just the thing: Analogies make people comfortable drawing conclusions about "X" by applying what they know (or imagine to be true) about "Y," without ever having looked at how "X" actually works. Thus analogies can persuade people that they "understand" things which may not be factual! This is why demagogues lurve them so much. When you pursue an analogy, how do you know when you've exceeded the limits of its validity? You don't--unless you methodically test your beliefs against the real thing. And that takes time and effort, which are the very things people try to avoid. Instead you end up comparing one concept that you have to another concept that you have, and saying, "Yeah, that feels right."

That was the whole problem with medieval thinking, and why science was such a big breakthrough for our species. That's also a big problem with so many speculative discussions on the Internet--because all discussions about things that you're not actually doing are free from the constraint of reality. It is perfectly feasible on an Internet forum to discuss the theoretical basis of digital recording and to make it as clear as day in all essentials. Unfortunately it seems equally possible for people to make interesting, intelligent guesses which are plausible, sincere, self-consistent and very persuasive--but which lack just that one tiny element of being what happens in reality. And people literally don't know which is which. The true and the false assertions weigh the same, they smell the same, so people judge based on whatever they feel like believing.

Where digital audio is concerned, that pretty much already happened a long time ago for some people. I was an analog engineer for years while digital was still in the laboratory stages. Some of the earliest digital recording systems didn't use dither, and consequently had severe defects in their performance at low signal levels. The lower you went in level, the higher went the distortion because of fewer and fewer bits being used. It got uglier and uglier as you descended, and at the very bottom there was an absolute chasm--reverberation tails simply disappeared. As sounds went bye-bye over the cliff, you could hear this odd punctuated noise that was kind of like cheesecloth gently ripping and then fading out. This was usually called "granular noise" (although it's technically a form of distortion), and most people could identify it reliably after one hearing, if it was pointed out to them.

As a result of complaints over this problem (and due in large part to the work of two AES stalwarts, Stan Lipshitz and John Vanderkooy), within two to three years most audio manufacturers saw the light and started using dither appropriately. This subtracted a couple of dB from the spec sheet signal to noise ratio of the recording system, but it cured once and for all (notice that I didn't say "covered up" or "concealed"--it fundamentally cured) those problems with "digital deafness," granular noise and the ratty sound in the lower 30 (I would say 40) dB of the dynamic range. It became very clear that they had never been an inherent part of digital audio, but were an artifact of bad design in particular systems.

And yet you see people today, a quarter century later, arguing as if all that had never happened. They not only didn't learn anything--in many cases I would say that they consciously chose not to let the information in, for whatever reason. And while it's one thing to think whatever you want to think about whatever you want to think it about--that's just being human and ornery, and it can keep you alive--it's another thing to spread ignorance and to devise more and more clever ways of covering up the fact that it is ignorance.

I want to be persuasive, not combative, but sometimes I feel like telling people to shut up and do their homework before spreading further misinformation. On the other hand, at least misinformation in the audio field only bruises some bits and some ears, rather than, say, breaking up a marriage where a child is involved (which some friends of ours are going through right now) or starting a whole war on false pretenses (which my country did and is still doing). So that helps me to stay calm about the audio issues, though I have to say I'm pretty upset about the other stuff.

--best regards
« Last Edit: December 13, 2007, 09:02:19 AM by DSatz »
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Offline Gutbucket

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Re: Analog to 24bit question
« Reply #82 on: December 13, 2007, 10:29:46 AM »
^^^
Well that's a rational, informed and well articulated perspective.  At the same time I believe dmccabe when he says he hears a difference in the tape transfers (I haven't had a chance to find some MOTB recordings and listen myself, though I plan to), and those differences must be attributable to something.  Perhaps it's the fingerprint of the particular A/D circuitry that sounds different and not limitations of the file formats themselves.

...Meanwhile your remark about the "last 30 dB" is something that nearly everyone on this board could find out for him- or herself within ten minutes, which I think would be very nice for all concerned if they (and you) did...

How do you suggest we do this, crank up the volume and seriously listen to the quietest sections of our recordings? or is there some other method?

I'll bring this in from another thread since it pertains here, it's come up several times in various threads and we never really reached a satisfactory answer to this question:

I don't see the point of doing 24 bit on this recorder [a discussion on the Edirol R-09].  The specs of the chip inside say the max SNR is 92dB.  That means, the noise floor can be no lower than 16 bits (-96dB).

I verified this.  I put a 1k resistor load on each channel of a miniplug, plugged into line in, and recorded in 24 bit.  If this was a perfect recorder, there should be zero bits of noise.  In fact, the lower 9 bits were noise.  So, go ahead and record in 24 bit, but you'll just be recording 8 bits of extra noise.

...

Considering that dithering a signal works by adding a form of low level noise before truncating the bottom 8 bits (essentially allowing us to hear details beneath the noise floor that dissolve into the dither noise instead of into quantization artifacts), would the noise you measured in the bottom 9 bits of the R-09's 24bit recorded signal act as a form of dither allowing better than 16 bit performance?

Until this question is answered conclusively, I choose record in 24/48 on the R-09 because I'd rather err on the side of potential quality, recognizing that quality difference may be illusional, and also because the additional storage space required is not a big problem for me.

..but that's really a cost/benefit justification and rationalization I make to myself. I'd prefer a more scientific analysis of the matter, if only for my own education.  Realistically I'll probably go on recording 24/48 on the R-09 regardless until I have the opportunity to do some comparative testing and listening, because I love understanding why, but in the end I always trust my ears.
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Offline aegert

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Re: Analog to 24bit question
« Reply #83 on: December 13, 2007, 12:44:49 PM »
I am an electrical engineer. I do Digital signal processing... I could argue text book reasoning around all this including Nyquist Criteria... But!

We have taken cassettes played them through killer analog electronics and transfered them in parallel using bro a/d's to 16 and 24 bit....


The results are clear!

The 24 bit and more importantly the higher sample rate make them sound better. Sample rate is the best determination factor for your signal to noise ratio and your transient response Period... Its my ears that tell me that..


Here is a snap shot that shows it (SR that is):




Now I respect any opinions on this matter and if any one asks me and they do every day what resolution and SR to use I say the more the better!

Now if you are only ever going to release 16 bit cd style stuff then 24/88.2 is a clear choice but if tou want to make 24 bit releases that people will burn to lets say dvd-A go 24/96...

Now all that being said the argument of 16 bit dats that is not the discussion that was originally started and is a whole other thread..

I will sit back now and watch the sparks but for real we have done qualitative tests to prove this with our ears...

Use your own to see if we are right but remember your DAC is critical in the listening part. I use a benchmark DAC-1



« Last Edit: December 13, 2007, 12:54:45 PM by aegert »
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Offline Shawn

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Re: Analog to 24bit question
« Reply #84 on: December 13, 2007, 12:51:53 PM »
Sample rate is the best determination factor for your signal to noise ratio and your transient response Period...

I'm just trying to make sure I understand what you are saying.... so are you saying that signal to noise ratio is most directly affected by the sample rate?

Offline aegert

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Re: Analog to 24bit question
« Reply #85 on: December 13, 2007, 01:48:30 PM »
Sample rate is the best determination factor for your signal to noise ratio and your transient response Period...

I'm just trying to make sure I understand what you are saying.... so are you saying that signal to noise ratio is most directly affected by the sample rate?

In the dynamic range pertaining to cassette transfer. The ability to manage the transient response of analog signals, yes sample rate has the biggest effect....

To describe this with what I hear and for me all the tech stuff falls away the higher sample rates and bit depths for that matter are more open... If you want me to define in techincal terms open forget about it LOL

But you hear it it is not hiss but air int he recording that exsists when you listen to the cassettes or LP's or reels that you here that does not trasnlate to the cd as well.. In the higher res/rate formats it starts to come back.. But there is no perfect digital copy of the analog signals just aproximations... In the end what are you doing with this stuff...

Are the cassettes adn other analog media dying... Yes

Do I want to best preserve the master in as close to original sound as I can with digital... Yes

Do I want to edit these transfers for listening? Yes


Well then for me there is only one clear choice.

I have sat countless doubters down with the dac-1 and a pair of sennheiser 595's ultrasone 750's and proven the point...

Let your ears be the judge... Without a killer dac thought you will not hear it.. As well one 'audiophile' fought me tooth and nail on this and he was listening on his shitty $20 computer speakers... I told him you got to work with me here ROFL...

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Offline bensyverson

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Re: Analog to 24bit question
« Reply #86 on: December 13, 2007, 02:20:41 PM »
I'm just trying to make sure I understand what you are saying.... so are you saying that signal to noise ratio is most directly affected by the sample rate?

If that is what he's saying, it's misleading. Changing the sample rate alone can not change your actual S/N ratio. But by choosing higher sample rates, you're oversampling that noise. The more you oversample the signal, the lower the ADC noise will seem to be. Really, the ADC noise will be the same amplitude, but you'll just have to zoom in further to see it. It's the same with dithering. Sure, it does help, but it can not give you more than 16 levels between -66db and -72db. It can make those 16 levels sound smoother, but if you decide to boost the levels, that range will still not hold up very well. How big of an issue is it for cassettes that already have overwhelming tape noise at -66db? Like I said before, it all comes down to how well you want to reproduce that noise.  :P  If you don't care about preserving exactly the hiss from the tape, 16/44.1 is more than enough.

I do think the picture aegert posted is a little unfair. Yes, if you really want to totally faithfully reproduce that tiny electrical "pop" with the best fidelity, you need an insanely high sample rate. (Care to tell us the duration of the pop?) But most sounds are smoother waves, and are thus a lot easier to digitize. If you run the same test with a real sample from music or voice, it will not be so clear cut. And if you say you can hear the difference in transient response between 96k SR and 192k, so be it, but I highly doubt it's physically possible. Your ears are just not designed for that.

I also want to say this about my digital image (not photo) analogy: it's fine if you don't like it -- don't use it. But it is valid. I made it very clear in my text that image detail frequency = audio waveform frequency, so I don't think that's misleading. Nor do I think it's medieval demagogy to try to make these abstract concepts visual. I'm not asking anyone to draw any conclusions based on the images, I'm just using them as illustrations to help people get to the "a ha" moment when it all clicks together how SR and bit depth are related.

All of that said, the engineer in me agrees that "the higher the better." Higher sample rates and better bit depths can only help you. The question is, when do you reach the point of diminishing returns? That's a subjective call that you'll need your ears for, not an engineer. For digitizing a cassette tape "clean," without the intention of applying effects, I don't believe I could hear a "worthy" difference between 16/44.1 and 24/96. For a violin in a studio, it's obviously another story...
« Last Edit: December 13, 2007, 02:23:56 PM by bensyverson »

Offline Petrus

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Re: Analog to 24bit question
« Reply #87 on: December 13, 2007, 02:32:27 PM »
aegert, if I understod your graph correctly that "analog impulse" is 3 microseconds long, which means it represents a half wave of a 166666 Hz signal. What is the point of testing these systems with signals that are about 8 times outside the frequency range they are meant and designed to record or what we can hear? Besides all systems filter out all frequency content above the Nyquist frequency of the sampling rate before letting the signal to the A/D converter. This is so BASIC! For our normal 44.1 and 48 kHz everything above 21 kHz is filtered out, that kind of 166.666 kHz signal would never get past the low pass filters of even a 196 kHz sample rate recorder.

Must be from a broshure of a DSD recorder... For even that this is totally irrelevant.

Misinformation at worst.
--------------------------------

There is one analogy connecting dither to pre-exposure of printing paper (or even film) in the old days of film. It is possible to get about one half f-stop's worth of extra latitude to the highlights by exposing the printing paper to a weak even light before exposure proper. The pre-exposure must be so weak, that it itself does not have an effect on the paper, but combined with the weakest highlight signals (shadows in the neg, remember) cumulativelly causes a weak exposure. A neat trick, in digital audio we add weak noise to help the weakest signals to raise above the lowest digitizing level.
« Last Edit: December 13, 2007, 03:20:03 PM by Petrus »

Offline Petrus

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Re: Analog to 24bit question
« Reply #88 on: December 13, 2007, 03:30:20 PM »
I hope the comment above not directed at me... :-)

All of that said, the engineer in me agrees that "the higher the better." Higher sample rates and better bit depths can only help you. The question is, when do you reach the point of diminishing returns? That's a subjective call that you'll need your ears for, not an engineer.

There are no valid scientific tests that I know of (or any of my AES engineer friends know of) proving that people can even hear a difference between 16/44.1 and 24/96. I am talking about a controlled double blind test of real world audio, not one where you listen to test signals and/or know what you are listening to. People tend to hear what they expect or want to hear, not what they actually do hear.

I do know about tests where they did NOT hear a difference between original analog live signal and 16/44.1, between 16/44.1 and 24/96, and where 24/96 signal was low pass filtered at 20 kHz.

That makes me a sceptic...
« Last Edit: December 13, 2007, 03:33:55 PM by Petrus »

Offline boojum

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Re: Analog to 24bit question
« Reply #89 on: December 13, 2007, 03:52:11 PM »
I think it would be good for us to agree if we are chasing the real or the theoretical here.  A lot of the theory discussed is interesting,  sort of, but in the real world, what can we hear and what can we differentiate?  The Lavry paper, if I remember correctly, argues against insanely high sampling rates as unnecessary and wasted.  With those really high sampling rate and deep bit depths we are in the realm of the medieval argument of "how many angels can dance on the head of a pin?"  Well, some say an infinite number and some say none at all: the difference between the theoretical and the real.

For me and for all my practical purposes I will stick with 24/48; maybe a 24/88.2 sometime, but I doubt it.  I wonder how many of those who can hear a difference on analog xferred at 16/44.1 and 24/96 and 24/48 can hear those same differences in a double-blind test?  I like double-blind as it assures neutrality; and I like seeing others replicate it, just as in the scientific world of facts.

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