Reggie wrote: "Oh, so in digital you can only record and play back square waves?"

Well, that's essentially true. It's blown WAY up in that image, that's not a realistic view of modern digital audio.
The other thing is that the speakers and wires degrade any steps to the point where it almost couldn't be measured anymore.

It's just really there to show the difference between the two technologies.

Analog may not be a more "true" form of recording, but it does work on a pure-sine system as opposed to a sampled signal, so it is a clearer representation of the music. If you were to imagine a "perfect world" situation, where there was no signal loss or degradation, then yes, a vinyl record would more truthfully represent the actual wave. In reality, on the other hand, digital media makes much more sense to the market because it does not break down or wear out... it is just simply read with light.

So, I don't really disagree with you... those are my thoughts on the topic.

Hey Reggie -

I just LOVE articles like that. They really do help to propogate bad and just plain old DUMB information about digital. ARRRGGGHHHH. :evil:

They suffer from one of the most common problems of digital explanation I see on a daily basis - the confusion of bit depth and sample rate and their relation to amplitude and frequency respectively.

Remember, with 2 sampling points, an accurate sinewave can be recreated.

That Dumbass Article wrote:
Some sounds that have very quick transitions, such as a drum beat or a trumpet's tone, will be distorted because they change too quickly for the sample rate.

Hmmm...so frequencies are moving greater than the possible sampling bandwidth....That's where the brickwall filter comes into play at the end of band...

Besides, a note changing quickly causes digital to distort?? Huh...?

Personally I don't put a whole lot of stock into the whole "Analog sounds better than digital."

The fact is - YES, theoretically analog CAN sound better than digital. But, then you factor in all of the errors associated with analog (Wow, Flutter, Distortion, Oscillation, Noise, etc.) and you now have a severly flawed medium. Just think if all of those things existed within the digital realm. It would have NEVER been accepted on the market. EVER.

So, to me, the majority of the Analog VS Digital debate is perpetuated by those that are too snobby to realize that their precious "perfect medium" is more flawed than Britney Spears' boob job or those who only read musings of said individuals.

Don't get me wrong, I love the sound of a good record. BUT...my ear has to do so much filtering of stuff that I have to "live with" the sound of analog to get the warmth and truth of it.

Well done digital lacks all of the "harsh, sterile, cold" attributes that its opponents tout as its handicap but also lacks most of the errors and inadequacies of mediocre to even high-quality analog.

Just some thoughts -
Donning flack jacket now....

8-)

corrupted wrote: [quote=Reggie]"Oh, so in digital you can only record and play back square waves?"

Well, that's essentially true. It's blown WAY up in that image, that's not a realistic view of modern digital audio.
The other thing is that the speakers and wires degrade any steps to the point where it almost couldn't be measured anymore.

It's just really there to show the difference between the two technologies.

Analog may not be a more "true" form of recording, but it does work on a pure-sine system as opposed to a sampled signal, so it is a clearer representation of the music. If you were to imagine a "perfect world" situation, where there was no signal loss or degradation, then yes, a vinyl record would more truthfully represent the actual wave. In reality, on the other hand, digital media makes much more sense to the market because it does not break down or wear out... it is just simply read with light.

So, I don't really disagree with you... those are my thoughts on the topic.

Dan 'The Man' Lavry wrote: And for those that insist on thinking about digital as some sort of a "stepped wave", take a look at the signal that comes out of a DA: It is an analog wave. The steps are gone. They are removed by the analog anti imaging filter, and the wave is very analog.

And yeah, that part about the drum hit distorting because of the sampling rate was rediculous. I mean, seriously. Who comes up with this stuff?

The whole analog vs digital controversy has been raging since the introduction of the CD. There are still people who point to some of the original releases done on CD as classic examples of what is wrong with digital and they fail to realize that was eons ago and that things have progressed greatly since then. I know some reviewers would still like us to believe that their 1966 mono records of the Beatles sound better than a CD of the same event that has been mastered from the original tapes and sounds (IMHO) quite a bit better.

I don't know how widely disseminated this information is but one of the reasons that early CDs sounded so harsh and brittle was because the CD players of that time were designed by digital engineers who knew a heck of a lot more about digital than they new about analog and when it came time to put in the analog circuitry the went to am IC audio cookbook (written by Walter G. Jung ), took the general purpose schematic out of the book and presto instant analog audio. Now of course no one would dream of doing anything like that but back then they needed to get the CD players out and copying a circuit from a book what a heck of a lot easier than having to sit down and design one from scratch. A lot of reviewers still remember those terrible CD players and can't get past them to the present.

I love when people show digital as large steps superimposed over a sine wave. What was the sampling frequency in this example may 1000 Hz? Plus they never really explain all the rest of the scenario to their readers so the poor reader is left with the idea that they are listening to bumpy audio when it is digital and pure sine waves when it is analog.

ARG!

Another thing that many of us don't like but have learned how to live with is the necessity of a "brick wall filter" at 20kHz when recording with a 16-bit 44.1kHz recording system. It's the brick wall filter that does more to screw up the sound than PCM process alone.

Plus we all have realized that the high frequency resolution is less than it should be in a 16-bit 44.1kHz world.

A lot of people did not like the way CDs sounded when they were released from analog masters that had been optimized for vinyl release. Who could blame them? The analog master was not necessarily the analog master. The analog master was copied by the disk mastering engineer so that if any of the Stamper plates ever broke, they could create an identical Stamper plate, without having to go through the disk mastering process all over again thus improving consistency, even though it was down another generation.

Where's my wire recorder?
Ms. Remy Ann David

Analog tape and processing still has its merits, and I like the warmth, but I am SO over vinyl..... Below is an old rant that I was going to run elsewhere.....perhaps the time has come to revisit it. (Getting out the flame-retardant suit!)

I've been through enough Vinyl BS over the years to say it is nothing but a subjective JOKE continuing to be foisted on that portion of the gullible public still looking for "vintage" jollies.

Who wants to buy a format that begins wearing out the moment you play it? Quick; where can I blow more of my money on depleting resources! Gasoline, cars, food, and hey, why not! Let's go back to Vinyl!

Anyone ever notice the gradual 5 db drop at 15K that happens from the beginning of the disc to the end? Of course not, it happens so slowly you don't notice it. Kinda like cancer or hair loss. (Ever wonder why they always put the best tracks at the beginning of the disc? Ever pick up the needle and start at an inner track? Notice something missing up there?)

Few know this, but none but the largest of the REAL transcription turntable tone arms can accurately play a disc without distortion and tracking errors across most of the surface anyway. There are really only one or two "Sweet spots", (the very beginning and again about 2/3 of the way in); the rest is inaccurate.

Am I the only one that hears a CONSTANT low level morass of surface noise whenever an LP is playing? Why do the "golden ears" hear oooh-so-much BETTER stuff coming off a Vinyl record, yet ignore all the artifacts? Talk about subjective filtering!!!!

What about low-level AC hum? Ground wire or not, most of the time, there's STILL a hum present in any turntable feed, unless you've got a four or five figure $ystem with a preamp costing more than most people's car.

What about platter rumble? What about constant tics and pops, no matter how well you take care of your collection? (Remember when people stacked them? HA!!!! Little bits of sand and dirt would rub between them causing even more damage.)

What about off-center pressing, wow and flutter? What about pre and post echo from grooves cut too loud? What about cartridges skipping out of grooves when records are cut too bassy or loud? What about feedback from the cartridge (via the surface of the record) to the speakers when you turn it up too loud?

I get mad just thinking about it. Vinyl was just a stupid, lousy, lossy format for so many years, and it p*sses me off to no end whenver I hear someone blathering on about how GREAT it is. BULL!!!!

What people are hearing are old, comfortable, smoothed-over artifacts and rounded-down harmonics that remind them of their long lost youth, and their sense of accuracy goes out the window. Ever wonder why there is so much prep that goes into a 'Pre-vinyl" master tape, from EQ to read-ahead limiting to groove spacing? The compromises visited upon a completed master tape to "Fit" the audio into the grooves was/is galling. The very first thing to go is accurate transient response. ANY master tape - digital or analog - copied to vinyl is a sonic compromise, top to bottom.

Vinyl always was not much more than dragging a boulder down a tar curb, hoping for the best. Early digital got a bad name with all the pre-equalized for vinyl tapes being used for so many early CD reissues. That stuff is long long gone, and I for one have no desire to go back.

Don't get me wrong: I have an old standalone LP player with a single mono speaker there I play oldies but goodies from time to time for the sheer nostalgia of it, and I still transfer many rare LPs & 78's for clients (when there's no other version available). It's fun for what it is.

For restoration work, we spend hours removing artifacts from unopened, unplayed vinyl pressings that have never been touched before, in some cases. The artifacts present in using even the best TT, preamp and never-before-played Vinyl recordings are immediately audible on any mid-to-high-end playback system of today. It's ludicrous, and it amazes me how subjective the so-called purists are who can't hear this stuff.

You couldn't make me go back to vinyl with a gun at my head. And I make no apologies whatsoever for my belief. It's nothing but nostalgia getting in the way of the facts.

Would I rather listen to a good analog reel to reel vs. Vinyl? Absolutely. A good analog reel to reel vs. today's digital? Maybe; depends on who set it up.

Whew.....that was harsh, but I guess it was the coffee talking when I wrote it, about a year or so ago! :twisted:

Dan 'The Man' Lavry wrote: And for those that insist on thinking about digital as some sort of a "stepped wave", take a look at the signal that comes out of a DA: It is an analog wave. The steps are gone. They are removed by the analog anti imaging filter, and the wave is very analog.

Playing devil's advocate...
The reason you see an analog signal is because of interpolation and averaging. As opposed to drawing the wave as it's created, the digital system samples a ton of points, and then "connects them with a line", creating an apparently smooth sine wave.

I love digital, don't think I'm saying I'd rather have an analog setup...
All I'm saying is, if you don't record the actual sound wave, you can't really say that you have a more true representation of the sound.

It's becoming a pointless argument as digital technology improves anyway... it's like digital photography. Sure, lower end digital cameras can't get the resolution of a chemical photo, but some of the higher end digitals blow everything else away as far as clarity and resolution. It just takes time and people will all come around.

corrupted "devil's advocate" sampling theory:

The reason you see an analog signal is because of interpolation and averaging. As opposed to drawing the wave as it's created, the digital system samples a ton of points, and then "connects them with a line", creating an apparently smooth sine wave. [IOW-linear straightline interpolation is used to configure the soundwaves]

Nyquist sampling theory:

A sampled waveform contains ALL the information without any distortions, when the sampling rate exceeds twice the highest frequency contained by the sampled waveform (to capture both the positive and negative peak of the wave). [sinc functions and their sums are used to accurately configure the soundwaves within a given bandwidth. Record a 20kHz sine wave--or a complex signal bandwidth limited to 20kHz--at 44100 samples(sincs) per second, and you will get the same waveform back out when converted from digital to analog.]

I love digital, don't think I'm saying I'd rather have an analog setup...
All I'm saying is, if you don't record the actual sound wave, you can't really say that you have a more true representation of the sound.

Yeah, I don't know. What is the definition of recording the actual sound wave? Aligning X number of iron oxide molecules to represent the wave while adding in some extra random noise and distortions not present in the original soundwave? Or using numbers and math functions to reconstruct a smooth wave, given a set bandwidth? Is there an analog format that is free of any bandwidth, distortion, or dynamic limitations? Hmmm.....

And I'd be curious to hear your further thoughts on how digital photography correlates to digital audio.

A little knowledge can be very dangerous they say.

Anyway, the article does not even come close in describing digital audio -- the description given is on a five-year-old level. The description of analog is at about the same level so you might say it adds up, but it surely does not! The article pictures analog as perfect and digital as the end of the world. We all know both has both shortcomings and pluses.

Somewhere on the net Dan Lavry has written quite readable stuff better describing how digital actually works, using a mathematical concept known as sinelets. Even I could sort of grasp what he was writing so I guess you would not need any higher education in math to at least follow along. If I find the link I will post it here.

Gunnar

Aha!
One more "devils advocate" point about the "Nyquist sampling theory"...
If you sample an unpredictable sine wave at twice the frequency of the highest possible rate for the sine wave... you certainly don't have enough data to properly emulate the exact nature of the wave.

Think about it this way:
A complete cycle is one full sweep in the negative direction and one in the positive, and then back to zero. Now imagine one snippet of time containing one cycle of a 20kHz wave. If I were to give you 2 single points on a graph and tell you to plot the sine wave I was thinking of... you'd be hard pressed to do so. Especially when taking into consideration the complex nature of a sound wave that possibly contained.

In my mind (which is not saying much, teehee), there is no way to reproduce a 20kHz signal by sampling at 44.1kHz. Do I think it's necessary to have 20kHz recorded precisely? Well, no.

So... I'm not really arguing for anything, I just find that interesting.

Yea, you could debate it in your own head forever... interesting topic as usual, but I could stress myself out thinking about it. :lol:

ghellquist wrote: A little knowledge can be very dangerous they say.

Anyway, the article does not even come close in describing digital audio -- the description given is on a five-year-old level. The description of analog is at about the same level so you might say it adds up, but it surely does not! The article pictures analog as perfect and digital as the end of the world. We all know both has both shortcomings and pluses.

Somewhere on the net Dan Lavry has written quite readable stuff better describing how digital actually works, using a mathematical concept known as sinelets. Even I could sort of grasp what he was writing so I guess you would not need any higher education in math to at least follow along. If I find the link I will post it here.

Gunnar

Very cool, I will look for Mr. Lavry's info... I'm always up for learnin' more about this stuff!

Dude, the stuff you are talking about, with this unpredictable sine wave, would be filtered out if it is outside of the bandwidth upper limit(half the samplerate). The lower frequency stuff that can make up an odd-shaped soundwave is no problem. The sinc functions can accurately reproduce a smooth sine wave (or whatever shape as long as it is bandwidth limited to fit Nyquist) of any speed under half the sample rate. If you want to get out of range of the slight high-freq rolloff that happens with 44.1K filters, then by all means use a higher sample rate. But don't expect your ears to hear frequencies they can't hear, don't expect your speakers to be able to reproduce frequencies they can't reproduce, and don't expect tape or whatever to faithfully capture all this unpredictable transient stuff you are talking about.

Here are the Lavry links:
Link removed
Link removed

Granted, I may be off topic, but I think you missed my point.

I'm not "taking analog's side" by saying that, I'm just curious as to how one would go about reproducing any wave with merely 2 points, given that you don't have a zero crossing to key the sample from, and the frequency is not held to a clock, of sorts.

Again, I'm not saying that this matters... I've just heard people telling me that it's "true" without giving an explanation or proof, so whenever the opportunity arises, well, I try to find an answer. I'm not at all unhappy with the quality of digital, I'm just trying to search for the meaning of life with a side of fries.

I know that most people can't hear much near the 20kHz range... and in fact most people over 25 usually can't hear past ~15kHz. I also don't agree with Nyquist's theory, and I don't believe that digital sampling at 44.1kHz will give you a true wave at 15kHz. So, yea, 20kHz doesn't matter, but if your 15kHz+ was slightly wrong, that might be a downfall.

All I'm saying is that the theory behind analog recording is more true to capturing a wave than is the theory of digital recording.
No worries, I guess it's not worth discussing anyway.

JoeH wrote: Rant rant rant rant rant rant rant rant....

Wow, that's a pretty good rant you got going there! :shock:
8)

corrupted wrote: Granted, I may be off topic, but I think you missed my point.

I'm not "taking analog's side" by saying that, I'm just curious as to how one would go about reproducing any wave with merely 2 points, given that you don't have a zero crossing to key the sample from, and the frequency is not held to a clock, of sorts.

Again, I'm not saying that this matters... I've just heard people telling me that it's "true" without giving an explanation or proof, so whenever the opportunity arises, well, I try to find an answer. I'm not at all unhappy with the quality of digital, I'm just trying to search for the meaning of life with a side of fries.

I know that most people can't hear much near the 20kHz range... and in fact most people over 25 usually can't hear past ~15kHz. I also don't agree with Nyquist's theory, and I don't believe that digital sampling at 44.1kHz will give you a true wave at 15kHz. So, yea, 20kHz doesn't matter, but if your 15kHz+ was slightly wrong, that might be a downfall.

All I'm saying is that the theory behind analog recording is more true to capturing a wave than is the theory of digital recording.
No worries, I guess it's not worth discussing anyway.

I think it is totally worth discussing. And I'm not saying that I understand all the math and magic behind it, but here is a test you can try sometime: Get an analog tone generator, put some tones through an A-D at 44.1k/24, screw the phase around by however many degrees you want (in order to try to screw up the sample rate), feed it back out a D-A, put an O-scope on it, and see what you get.
You seem to be forgetting that each sample represents a sinc function, not just a point along a stair-step of points (as you may see it in your recording software, depending on what it is). Give the creators of digital audio a little credit. I mean, it would probably sound like crap if it was just stairstep plots used to reproduce the frequencies we hear, unless it was oversampled like 100 times. But that would be a waste when you can use sinc functions instead of line graphs. Besides, wouldn't we destroy our amps/speakers if we were putting 15K square waves and sawtooth waves through them all the time? :?

Well, it wouldn't exactly be square or saw waves, it would just be random noise at that frequency. Not exactly random, but not exactly correct either. I'm not implying that it's a stairstep type of plot, but more of an averaged interpolation based on points. So, theoretically, the wave would be close to correct, but not exact.
About putting a scope on it... that's implying that what you record would be pure and constant at a frequency. Any slight change could throw it out for just a nanosecond until it caught back up with the "groove" of the wave. Now if you're constantly changing amplitude as well, it would be hard to track that all of the time.

An expiriment of thought: If you sampled at 40kHz, and you were sampling a 20kHz signal, what if every sample was taken at a zero crossing? It would be silent... Is that why the extra 4.1kHz is there?
What if every sample was taken with no reference to zero because of other frequencies of the sound wave? How would the A/D converter distinguish these from other frequencies? Or, does it just track each frequency individually... I don't think that's the case because it would take a ton of processing.

It's also possible that there is some magic in these A/D converters that I just don't see.

Hmmm, now I'm going to have to do a ton of poking around the net looking for test results and stuff... I like sparking the neurons now and then!

Chew on this for a second:

...when the sampling rate exceeds twice the highest frequency contained by the sampled waveform...

All the other stuff that doesn't fit into zero crossings and all-- as long as you bandwidth limit (filter) the signal, they are accounted for still by sinc functions or the sum of several sinc functions. I guess. :? Like I said, I don't totally follow all the complex math, so this is getting into the area that I will refer to as "magic." I do know it comes out the other side just fine. 5pm on a Friday is not my finest thinking hour.

I spent about the first ten years of my career as a DSP engineer working, doing both research and implementation. There is a lot of bad information on the net about sampling. Dan Lavry's papers are pretty good. Another good resouce is Rick Lyons, Understanding Digital Signal Processing. If sampling theory was wrong, they all our our cell phones would be paper weights, DirectTV would be bankrupt, and satellites would be falling out of the sky.

mpd wrote: I spent about the first ten years of my career as a DSP engineer working, doing both research and implementation. There is a lot of bad information on the net about sampling. Dan Lavry's papers are pretty good. Another good resouce is Rick Lyons, Understanding Digital Signal Processing. If sampling theory was wrong, they all our our cell phones would be paper weights, DirectTV would be bankrupt, and satellites would be falling out of the sky.

Nika Aldrich's book is also good. Although I do find that it simplifies some things, it is helpful to the layperson (and even some non-laypersons.)

While I also find Dan Lavry's papers excellent, I find that on occassional subjective issues, he is quite opinionated and can only present the side of the argument which better suits his opinion.

He's still 10x smarter than I'll be any given day of the week, but I just take issue with a few of his more controversial ideas.

J.

NOW they're getting somewhere.... :wink:

Link removed

For maximum clarity of sound, each needle is used for one or two sides only. However, if you prefer an even more exciting and characterful sound, you can use the same needle many times. This subtly 'shapes' the groove and blends additional high frequencies into the sound. Once you have experienced the sound of a shellac disc that has been lovingly modified in this way, you can never go back to the stark clarity of digital media again.

corrupted wrote: An expiriment of thought: If you sampled at 40kHz, and you were sampling a 20kHz signal, what if every sample was taken at a zero crossing? It would be silent... Is that why the extra 4.1kHz is there?

It's also possible that there is some magic in these A/D converters that I just don't see.

No there is no magic, only physics. Remember e=mc squared ? It is said that only a few persons in the world really understands the relativity theory. That part coming out of it, e=mc2, is sort difficult to grasp. And yet if you see an atomic bomb there is no need to understand the mathematics, you can stand back and simply see the fireworks. (Let us hope none of us ever will see that).

Same with sampling. You could either learn the math from the ground and understand it from there, or you could stand back and see it work. The sinc method described by Dan (the papers are referenced above) is a simplified part of the theory when it is presented in the paper, but it should allow you to understand the e=mc2 part. And even if you do not believe that part, there is proof. Do not expect people to not having tried to spoof the theory a million times, and still it stands. You can test the proof yourself, simply download the rightmark audio analyzer and run it on your own card. That program tests if our DA/AD actually can pass a 20kHz signal when running at 44.1kHz (and most not only can but does a good job at it).
http://audio.rightmark.org/index_new.shtml

As for the exactly 20kHz and 40kHz test, theory says it will not work. According to the Nyquist sampling theorem the highest frequency you can sample is LESS than half the sampling frequency. 20/40 is exactly half, not less than. In real world applications you addionally have to go beyond the simple theory and look at the real world. In the real world there is no total brickwall filters. A total brickwall filter would pass everything below a frequence and pass everything above. Real world filters always has a gradual slope, hence the filters create a need to go to 44.1 in order to sample 20kHz.

You may not believe in e=mc2 but it is there still. You might not believe in sampling theory, but it does work. Both are at the same practical things that can be seen in the real world, and can be described with mathematical tools. Theory and practice goes hand in hand.

But, do not be depressed. Even Einstein had problems believing. The last half of his life he tried hard to disprove quantum physics, allegedly saying that "God does not play dices". We (knowing better) knows that exact theory to be the the very fundament that solid state electronics is built on. Without quantum physics, no transistors. Einstein was wrong, and so am I. Only I am wrong on such a more mundane level.

Gunnar

... double post

Reggie wrote: Besides all that, since when is vinyl a more accurate format?

Instead of trying to intellectually hash this out with theory arguments (easy to do on an www forum), you really need to experience (hear) the differences (not so easy to do over the www). Since I can't play you an example, I can only relay an anecdotal account....

For Straight Ahead Records, I engineer direct to two track projects. This stereo mix is recorded to four formats simultaneously: DSD, 24/96 PCM, 16/44 PCM, and 1/4"-30ips tape. The 1/4" is then transfered to 45rpm 200gram vinyl. Long story short; over the past two years of demonstrating our recordings, there has not been one single person who has preferred the digital sources over the analog tape or vinyl transfer. Without reservation, the analog sources more accurately reproduce the musical performances.

The caveat is, of course, that vinyl wears out, scratches, and requires very expensive playback equipment to realize this higher accuracy. So one could argue that for the common consumer, digital may be more accurate.

Zilla wrote:
The caveat is, of course, that vinyl wears out, scratches, and requires very expensive playback equipment to realize this higher accuracy. So one could argue that for the common consumer, digital may be more accurate.

But that's it in a nutshell. Good analog costs far more than even most self-professed audiophiles can afford. An Avid turntable, a Lyra cartridge, SME tonearm and an EAR preamp sure will sound good, but that will also set you back roughly $40K.

You simply can't get anywhere near that level of performance with a $1000 Rega table and a shure cartridge!

Not to mention the care and maintenance.

Basically, good digital is FAR easier to achieve than good analog. I would also venture to say that overall (not just on the consumer level) digital would be more accurate. It's the inaccuracies of analog which actually endear us to the technology's sound. The warm, euphonic nature of subtle even harmonic distortion and the comforting feeling of a realistic noise floor all add up to a more "user friendly" sound.

But hey, I don't know, maybe I'm nuts.

I dunno, maybe you are right about vinyl now. Maybe new technology has made it better. But consider this quote I found from David Satz (formerly of RCA):

--By the way, when I was an engineer at RCA Studios I heard the master tapes for many "revered" Red Seal LPs. Without a doubt, LPs on a good playback system can sound far, far better than the master tapes from which they were made. Vinyl LP isn't even remotely a sonically transparent medium, and the old engineers had learned how to tailor the sound of the master tapes so that the LPs would sound great.

I have no doubt that some people will prefer the effects of transferring to tape or vinyl, but does that necessarily mean it is more accurate? And are these people listening to the different formats in a blind test? Otherwise it isn't too hard to trick people into convincing themselves that what they have been told should sound better, does. I must admit, you have a unique perspective, being able to record directly to so many formats at once.

Corrupted neeeds to spend some time reading Pohlmann and Watkinson...

Reggie wrote: I have no doubt that some people will prefer the effects of transferring to tape or vinyl, but does that necessarily mean it is more accurate?

I can say that for our recordings, we have designed and aligned for high fidelity, not for effect. We are not abusing tubes, iron, or magnetic tape to produce euphonic harmonic distortion or magnetic compression. If you passed a high freq square wave through our analog chain and compared it to what comes out the digital chain, you would find that the analog was more accurate. At least on a technical level.

But I think it might be helpful if you defined your use of the word "accurate". Do you mean it passes sine waves accurately? That TH+N or IMD is a certain low value? I can't tell you how many times I have used or built equipment with outstanding technical figures only to find that the musical reproduction was dead. The whole point of music recording is reproducing music, not test tones. If you can accept that all recording mediums are, in reality, horribly inaccurate when compared to the original sonic event, then it comes down to which creates the most convincing illusion. Which illusion most accurately delivers the message of the music.

If significant listeners are preferring the vinyl then I would say the vinyl is more successful at delivering the message. That seems like better accuracy to me.

Corrupted neeeds to spend some time reading Pohlmann and Watkinson...

Now that you mention it, I probably do to... :oops:

But hey, I just found this cool page on the Rane website:
http://www.rane.com/note137.html
About the best explaination of how bit depth works as I have found. Kind of confirmed a few things that I hadn't seen in print myself, like how the bits are used to divide up the voltage range. And a lot of other crazy stuff to be found on there too.

But I think it might be helpful if you defined your use of the word "accurate"

I would say accurate being that what you have recorded matches what you have put in the recorder. I suppose the degree of accuracy can be tested with tone/noise generators and Oscopes and the like; and THD+N and IMD would be a couple ways to compare although you probably already know the outcome of that. I'm afraid I'm not technical enough to do or guide the proper testing methods myself. FWIW, an ideal 16-bit A/D converter (assuming ideal analog stages) should have no more than .003% error recording the voltage level. For 24-bit, it is 0.000012% or less. I have no idea what error% your tape or vinyl is capable of. But I'm afraid a digital system will only be able to pass a square wave accurately when you have an infinite sample rate. You aren't insinuating that you can record a square wave on a vinyl rekkid, are you? Or play it back accurately on your speakers? Or hear the difference between a clean 20kHz sine wave and a hypothetically reproduced clean 20kHz square wave?

Just digging for answers and knowledge,