What 2 Channel A/D Is More Transparent Than A Lavry Gold?

[sexxy]

MYTEK DIGITAL USA

Comparison of the sound of 8X192ADDA to other high end converters:
David Chesky of audiophile label Chesky Records have recorded this test session before acquiring two Mytek 8X192ADDA's for the label recording rack. This is a life recording of a jazz trio recorded in New York City in 2006 through a very simple signal path: mic>mic pre>custom high end tube mixer>Mytek 8X192ADDA.
David is a firm believer in high sampling rates, he also conducted recording/listening tests at 96k and decided to make the 176.4k through Mytek the new recording standard for all Chesky Records sessions.
· Trio-Mytek8X192ADDA-192k.WAV 80MB

· Trio-UniversalAudio2192-192k.WAV 80MB

· Trio-WeissADC2-192k.WAV 80MB


mytek has a verry detailed classic sound,
detailed becouse the clock and classic that could come from the original recording that none of the others AD could record
or... its colored in a classic way

Weiss uhmmm in the middle but more similar to Ua2192

ua2192 has a verry nice 3d deep bass, but for highs... that sounds blurr, could be too much harmonics or could be the clock jitter

ua2192 has akm
mytek has akm
lavry black has analogdesign


why dont you try altmann micro machines
ive read that some one sold the lavry black becouse the altman dac
The Altmann Creation ADC
The Altmann Attraction DAC


or prism ad-2
AD-2 Home Page

[pinwale]

Originally Posted by sexxy

why dont you try altmann micro machines
ive read that some one sold the lavry black becouse the altman dac
The Altmann Creation ADC
The Altmann Attraction DAC

From the website linked above "the complete electronic assembly of the Creation ADC is vibrationally optimized, as the pcb is flat-press-mounted on a specially treated spruce-board. This construction is time-consuming during manufacture, but improves sound quality, as all components are mechanically coupled to a musical sound board."

Spruce is pants. Give me my oak!

[Brad Lunde]

I think the latest DAD AX24 converters are now the ones to hear vs the Pacific. Very interesting in that they work at very high sample rates in PCM or DSD64 or DSD128. They have been winning every shootout I send them to..... [I import them so I am biased probably]

There is the practical issue of higher sample rates (>192) that will be reality soon and the neccessity to deal with them. I expect we'll see consoles with such sample rates before too much longer. I like DAD's ability to use at any sample rate up to 384 and theri modular approach toconnections, and the ability to pay only for what you need and upgrade later.

The Sonoma DSD system has a unique interface and EMM had that sort of locked up, but DAD is developing that interface now as well. I got Telarc a DAD and they are using it in DSD mode and very happy!
Brad

[lucey]

Those are the ones you mentioned at AES? Good to know, for the dark day when my PM dies.

[Brad Lunde]

Originally Posted by lucey

Those are the ones you mentioned at AES? Good to know, for the dark day when my PM dies.

Yes Brian, but lets hope that your PM never dies!
Brad

[DSD_Mastering]

Originally Posted by Brad Lunde

The Sonoma DSD system has a unique interface and EMM had that sort of locked up, but DAD is developing that interface now as well. I got Telarc a DAD and they are using it in DSD mode and very happy!Brad

Mytek has the ST-optical interface now to use with the Sonoma system.

I tried the DAD/dCS/Prism and EMM in DSD and felt the EMM was the best sounding to me. I know Telarc uses the EMM as well as the DAD.

Regards,
Bruce

[Coyoteous]

Semi-OT: I think I'm getting a feel for a consensus of hierarchy regarding converters for mastering. What about the Lynx Studio Aurora series? They say "mastering grade," but are we looking at quality below Lavry Black and Blue - which seem to be at the bottom for serious work? Thanks!

[jamsmith]

Someone enlightten me ont enlighten me a little on the asthetics vs the science on AD conversion. Now certainly DA conversion has so many factor as you are taking a finite set of snapshots of an analog signal and trying to recreated a infinitely variable signal while handling slew and distortion. And the real challenges of that are largley in the analog circuitry of the DAC,

But DA conversion is a matter of getting the instantenous voltage level of a signal. Other than the quality of the sample hold circuit, that is very simple task. What is it you are looking for in these higher quality ADCs. Are they some how skewing the sampling in a way the translates better in DA process?

I can forever see the challenges or perfecting DA conversion, but after all these years, I just can see how AD conversion should anymore than a matter of more bits and higher sampling rates.

[Plush]

I think your source material is what is determining the sound of the a/d converter.

[jayfrigo]

Your post unfortunately misses some of the fundamentals of sampling theory, both A to D and D to A. It would take too long to go into all the detail, but let's just take a few basics to get the ball rolling.

ADC and DAC have many things in common. First, higher sample rate only buys you higher frequency reproduction. It IN NO WAY makes the reproduction within Nyquist any better (filter issues aside - more on that later). More bits buys you greater dynamic range and lower noise floor. It in no way makes the reproduction of the higher bits any better. Since the very best (and very expensive) analog electronics today have a practical limit of the equivalent of about 22 1/2 bits, anything below bit 22 (and change) is conveying the self noise of the electronics and contains nothing of your desired signal.

The key is that you must have a band limited signal. Nyquist and Shannon showed that you can perfectly recreate any frequency information below one half the sample rate. Fourier showed that all complex waves can be broken down into their individual sine components. Between these two concepts, anything below Nyquist can be captured, and reproduced again.

One problem you run into is how to band limit the signal. If you don't filter the Nyquist frequency and above, you are left with aliases on the ADC side, and image frequencies on the DAC. Since there is no theoretically perfect filter where all frequencies below Nyquist are perfectly passed, and all frequencies above are perfectly rejected, you run into many issues.

The three parts of a filter's response are the passband (frequencies you want to keep), the stopband (frequencies you want to get rid of) and the transition band (where the rolloff happens between pass and stop bands). You also get filter ripple to either side of the rolloff, which is deviations in the response at to either side of the rolloff region. Depending on whether you are using analog, digital, IIR, or FIR filtering, you get a host of other problems. Most is done with digital FIR filters these days.

IIR stands for Infinite Impulse response and is the typical EQ we see in music, which equates to minimum phase analog designs, and most digital plug in EQs as well. FIR is Finte Impulse Response and usually (but not necessarily) exhibits a linear phase response. These will always be digital. IIR filters have problems of phase distortion in the form of group delay, or certain frequencies arriving out of time with others. FIR filters delay all frequencies by the same amout, thereby maintaing a linear phase response, but they exhibit problematic pre-ringing. This is a bit of signal that arrives ahead of the main lobe of the filter and in practice can smear transients and skew imaging and such. The steeper the filters, the worse the problems.

Filtering is a big deal. Higher sample rates will allow you to move filter issues such as rolloff and ripple above the range of human hearing, and also allow you to use much gentler filters which will reduce ringing problems. So, analog electronics that perform well in both ADC and DAC, and filtering for both ADC and DAC are a huge part of the sound of converters. There is a bunch more that goes into why a converter sounds good or bad, but I think I've rambled long enough for now.

Originally Posted by jamsmith

Someone enlightten me ont enlighten me a little on the asthetics vs the science on AD conversion. Now certainly DA conversion has so many factor as you are taking a finite set of snapshots of an analog signal and trying to recreated a infinitely variable signal while handling slew and distortion. And the real challenges of that are largley in the analog circuitry of the DAC,

But DA conversion is a matter of getting the instantenous voltage level of a signal. Other than the quality of the sample hold circuit, that is very simple task. What is it you are looking for in these higher quality ADCs. Are they some how skewing the sampling in a way the translates better in DA process?

I can forever see the challenges or perfecting DA conversion, but after all these years, I just can see how AD conversion should anymore than a matter of more bits and higher sampling rates.

[jamsmith]

Well, theoretically it still is about hire rates. If you could sample at rate higher than than twice maximum frequency you could possibly encounter, you would solve all your problems! But of course, this not practical.

In practical application, I see your point. What I was thinking about is that filter design is much easier and far less expensive in the digital domain. For some reason I was thinking digital filters in the ADC and analog filters in the DAC. I have been out of this for a long time. It had always been practical to use digital filtering in a DAC for a studio where you have no hardware and software considerations. Not so easy to pack all of that into a CD player, though I am sure that is not much of an issue today.

So if I drop the other side of the argument and pose this: With the level of technology to inexpensively model expensive filters using software, why do we not have A/D/A that surpasses our ability to hear the difference?
We have CD players that upsample the 16/44.1, process it and output 24/192kHz. What perplexes me is not theory, but the fact that it sounds like the technology exists to make the argument go away. Are these arguments real or is this like Eric Johnson claiming Duracells sound better than Eveready in his stop boxes? (And the question would also be that even if Eric can tell, would the difference be enough that he would spend another 20 or 30 thousand dollars for that difference)


(I guess it IS time for me to dust of the digital audio theory books.)

[Dan Lavry]

Originally Posted by jamsmith

Well, theoretically it still is about hire rates. If you could sample at rate higher than than twice maximum frequency you could possibly encounter, you would solve all your problems! But of course, this not practical.

In practical application, I see your point. What I was thinking about is that filter design is much easier and far less expensive in the digital domain. For some reason I was thinking digital filters in the ADC and analog filters in the DAC. I have been out of this for a long time. It had always been practical to use digital filtering in a DAC for a studio where you have no hardware and software considerations. Not so easy to pack all of that into a CD player, though I am sure that is not much of an issue today.

So if I drop the other side of the argument and pose this: With the level of technology to inexpensively model expensive filters using software, why do we not have A/D/A that surpasses our ability to hear the difference?
We have CD players that upsample the 16/44.1, process it and output 24/192kHz. What perplexes me is not theory, but the fact that it sounds like the technology exists to make the argument go away. Are these arguments real or is this like Eric Johnson claiming Duracells sound better than Eveready in his stop boxes? (And the question would also be that even if Eric can tell, would the difference be enough that he would spend another 20 or 30 thousand dollars for that difference)


(I guess it IS time for me to dust of the digital audio theory books.)

Hi

In my experience, an AD is a tougher task the DA. I will not elaborate. It is sufficient to say that AD and DA design is a highly complicated matter, based on dozens of years of evolutionary, as well as revolutionary knowledge. There is no single book that can tell you all you need to know.

AD and DA have different features for different applications. My medical AD's were required to be as identical to each other as possible, because the difference between channels was the most important criteria. The design was only 16 bits, and the speed was not "outrageous" but the application called for microvolt DC offset and nearly impossible precision of absolute gain. When I designed an AD for weight scale, precision was the driving requirement, while the speed was very slow. Every application brings it's own requirements, and audio has it's own difficulties. And that level of familiarity, know-how and experience is far beyond some basic principles that often forgo real world issues. I once learned about the basic principles of the Carno and Diesel engines, but I am far from qualified as someone that knows about automobile design.

More bits? Why? My Gold AD yields 127dB dynamic range (around 121 real bits un-weighted). There are very few cases when you could even take advantage of such performance, such as a solo singer or instrumental with a very high efficiency mic, loud enough so that you do not need much micpre gain.
Most signals coming into the AD are already too noisy, and the bottleneck is almost always the mic noise and the micpre noise. The more gain you set on the micpre, the more signal, but also the more noise. My AD10 yields over 117dB un-weighted (120dB un-weighted), and that is enough bits for most cases. You need more? The gold AD will give you more. I doubt it that you need more bits. You need good conversion bits.

And aside from micpre noise limitations, how many bits (which translate to dynamic range) does the ear need? Not to mention that it is not only about more bits. It is about how good those bits are.

Higher sample rate? Why? Sample rate is a function of the application. You need higher speed for digital video. You can live with very slow speed for the weight scale... Audio speed depends on how high the ear hears, and at 96KHz we are already in overkill territory. Going faster does take it's toll on accuracy. Sampling rate needs to be fast enough, but not too fast. Another view: It can be slow, but not too slow to impact the audio. In other words, it needs to be set at an optimal sample rate point.

Dan Lavry
Lavry Engineering, Inc.

[jamsmith]

I have no experience in AD or DA design, so I am not directly challenging your expertise. However, I did study and work with electronics for about 30 years and I have read much that doesn't quite jibe with how I am reading your position. I may not the expertise on digital electronic, but I have load of experience with analog electronics in general and audio in particular.

Ok say you are sampling at 44.1 at 20K, this is barely more than 2 samples per period. You would not be able to tell if the sample was sine wave or a triangle or a sawtooth. Now I am not trying to be obtuse. I understand that no one is going hear the difference directly. But we do know that frequencies above the range of hearing can through modulation affect the sounds we are hearing. And we also know that modulation using different wave shapes creates different timbres.

So three questions:

So are you saying with absolute certainty that higher sampling rates do not affect sound quality?

How fast is too fast for accurating sampling voltages?

Considering that 24bit/96kHz ADC chips can be bought from companies like Analog Devices for 5 bucks, what justifies the high price of converters on the pro-audio market?

[arf]

Originally Posted by jamsmith

Considering that 24bit/96kHz ADC chips can be bought from companies like Analog Devices for 5 bucks, what justifies the high price of converters on the pro-audio market?

As to sine, vs. square vs. triangle waves at 20k, your ear can't hear the difference between them. As to beat frequencies caused by overtones above the audio band, musical instruments don't produce significant levels above 40kHz, so with good filtering, 96k is more than enough sampling. As to the quote above, for someone who "did study and work with electronics for about 30 years." I can't believe this is a serious question. The analog stage, board layout, pico second clocking, custom filtering are a few of the design hurdles that affect the cost of great converters.

[Sui_City]

Originally Posted by sexxy

#1. seem you are confusign dynamic range with noise floor
"my AD has 127 dynamic but mic-pres have more noise"

http://audio.rightmark.org
i would love to see lavry gold,blue,black rmaa rightmark loopback tests


#2. seem you are confusign high-sample rate with hearing freq. range
but agree on the optimal sample rate..
why?
ear cannot hear 96k tones, but if you record & playback at a faster speed, will record&playback more details in all ranges
can you hear those additional details if they are in the human hearing range ? yes

low samplerate:
1234567890
high sample rate:
12345678901234567890123456789012345678901234

[top]almost 4 times more information

more details

sometimes its good, sometimes its bad
example ?
digital film vs. standard video:::::::::
original film goes at 24frames per.second,
video goes at 60interlaced = 30 frames
new digital film goes at 24p progressive
why step back?
becouse it looks way cooler

#3. digital has 144dB aprox. of dynamic range
no AD/DA can yet match digital
why work at 24-bits ? if your AD/DA only does 127dB ?
its a dumb question...

for example its not the same to::
A) take a digital photo at 640480
vs.
B) take a digital photo at 20MegaPixels, and use hi-quality bi-cubic algorithms to rescale down to 640x480
vs.
C) take a digital photo at 160x100 and up-scale to 640x480

same as::
why use 3ccd ? if just 1ccd its enough?
dumb question...
same answare, its not the same results

similar to dacs clarity mic-pre
thumbsup

holy shit is all i can say.

[jayfrigo]

Originally Posted by sexxy

ear cannot hear 96k tones, but if you record & playback at a faster speed, will record&playback more details in all ranges
can you hear those additional details if they are in the human hearing range ? yes

First, nobody knows this stuff better than Dan Lavry, so it's silly to try to argue it with him.

Second, you display a common and understandable, yet no less completely erroneous understanding of how sampling works. You need no more than just over 2 points to unambiguously represent a sine wave. Nyquist and Shannon have had this proven for more than half a century, and nobody has been awarded a Nobel prize for proving them wrong yet.

For a given set of points, assuming a band-limited signal (everything Nyquist and above (1/2 sample rate+)filtered out), there will only be one sine wave that can fit a set of points. What about complex waveforms? Over a century and a half of Fourier to the rescue. He showed that all complex waves can be broken down into their sine wave components, musicially speaking a fundamental and harmonics, which means if it works for sines, it works for complex waves.

Other times people take pieces of a sampling system out of context, like when they worry about stairsteps in the DAC output. If you were omit the anti-alias filter, then sure, that's a problem. However, for a complete system, including filtering pre and post, and dither etc., the system works just fine, albeit with some noise added to the signal.

As for additional detail, the smaller little wiggles represent high frequency information only. Any wiggles absent are components that were above Nyquist. That's what you lose with a lower sample rate, high frequency inforamtion above Nyquist. I know it's odd to think about at first, but you truly, absolutely, do not get any more detail in the representation of the waveform components below the Nyquist frequency by increasing the sample rate. That's just not how it works. People confuse early learning and conceptual aids like "connect the dots" as the actual way it works, but it's more complex than that. It is mathematically proven, and it is proven in practical application.

[bob katz]

Dear Screaming Michael:

You need to learn a bit more about PCM audio and converter theory. The "need more points to describe a wave" is an often-repeated fallacy. Once you understand that the filtering in the converter COMPLETELY restores the original wave, up to the bandlimit of the system, then you begin to understand how PCM audio works.

That said, I highly recommend that you read chapters 5, 17 and 20 in my new book before trying to spout incorrect theory and reach the wrong conclusions!


Best wishes,


Bob

Originally Posted by jamsmith

I have no experience in AD or DA design, so I am not directly challenging your expertise. However, I did study and work with electronics for about 30 years and I have read much that doesn't quite jibe with how I am reading your position. I may not the expertise on digital electronic, but I have load of experience with analog electronics in general and audio in particular.

Ok say you are sampling at 44.1 at 20K, this is barely more than 2 samples per period. You would not be able to tell if the sample was sine wave or a triangle or a sawtooth. Now I am not trying to be obtuse. I understand that no one is going hear the difference directly. But we do know that frequencies above the range of hearing can through modulation affect the sounds we are hearing. And we also know that modulation using different wave shapes creates different timbres.

So three questions:

So are you saying with absolute certainty that higher sampling rates do not affect sound quality?

How fast is too fast for accurating sampling voltages?

Considering that 24bit/96kHz ADC chips can be bought from companies like Analog Devices for 5 bucks, what justifies the high price of converters on the pro-audio market?

[LNerell]

Originally Posted by jamsmith

Ok say you are sampling at 44.1 at 20K, this is barely more than 2 samples per period. You would not be able to tell if the sample was sine wave or a triangle or a sawtooth.

This arugment has a major flaw, that being if 20 khz is the fundamental frequency of your waveform you are not going to be able to tell what type of waveform it is as the overtones will be higher then the human hearing range. You could try this with a analog ocillator and would get the same results, they will all sound like sine waves provided you can hear that high to begin with.

[Kees de Visser]

Originally Posted by jamsmith

So are you saying with absolute certainty that higher sampling rates do not affect sound quality?

It's impossible to prove that no-one can hear differences between standard (44.1kHz, aka 1FS) rates and higher. It's up to the claimants to prove they can hear a difference. Proof would require proper double blind testing.
There are recent AES articles about such listening tests, with different outcomes. The first one (like most others) fails to find any audible difference and the second one might have found some evidence. Interesting reading, highly recommended.
1) Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback (full article AES members only or $$), some details on BAS Experiment Explanation page - Oct 2007
2) Which of the two digital audio systems best matches the quality of the analog system?

Considering that 24bit/96kHz ADC chips can be bought from companies like Analog Devices for 5 bucks, what justifies the high price of converters on the pro-audio market?

My guess: R&D and marketing budget in combination with relatively low production quantities. Quite some budget is going into R&D of details that might or might not be audible. The big question is (and will remain) what is good enough ?

[MAzevedo]

Originally Posted by Kees de Visser

My guess: R&D and marketing budget in combination with relatively low production quantities. Quite some budget is going into R&D of details that might or might not be audible. The big question is (and will remain) what is good enough ?

Well, you can get a $5 AD chip. The AD chip isn't the hard part though, it's the analog front end you need to put with it. Ultra-clean power supplies, discreet input stages, discreet alias filters, super-low-jitter clocks, all that stuff that needs to go with the $5 chip gets as expensive at boutique mic pres. Nobody is charging $10,000 for the AD chip, you're paying that for all the other stuff in the box.

[jayfrigo]

Higher sampling rates can move filter issues away from the range of human hearing (though many still use unnecessarily steep, ringing filters to improve arbitrary written specs at the cost of sound quality), and do open up more possibilities for noise shaping. This may account for people's preferencs for them. There are plenty of other reasons regarding converter design that may also account for it, from analog stages to the chipsets to the built in filters in the chips that can account for a difference in sound not related to more sample points or ultrasonic capture.

[lucey]

What's the topic at this point?

Do we have people arguing there is no difference who neither hear difference nor have experience designing AD? If so, do you guys think there's a hype driven market in high end AD converters?

Bottom line, small differences in technical terms are audible. It's counter intuitive.

[jamsmith]

Ok, so I am currently a little over my head as far as the high level A/D/A converters. And I realize this is probably not the best forum for this disccusion. But what I am hearing is perplexing so I will clarify and ask one more to clear the air. Looking at this diagram:



I have a sine, a triangle, and square wave. I am sampling in this case at three points resulting in the 3 identical digital representations. I don't see how any kind of filtering will restore each wave accurately. Or is PCM sampling working in a completey different manner? (And yes, Bob's book is on its way)

But I guess the real issue I was getting at to begin with is are these differences in these different high end converters really that signficant? Is this a Steinway vs Dusendorfer argument or Steinway vs Kawai argument? (And thats still a bad analogy because I know a good pianist who love Kawai and hates Steinways)

[jamsmith]

Originally Posted by arf

for someone who "did study and work with electronics for about 30 years." I can't believe this is a serious question.

Everything is a serious question. I have seen manufacturers sell a lot stuff for WAY more than it is worth just for a minor improvement. Having seen the double blind test between products with huge cost differences that yielded no better results, having taken mixers and replaced parts that amounted to little more than maybe $10-15 per channel and greatly improve the quality, I have always been a little suspect of the industry. Hell, they want over $100 for a distortion box that is little more than a 42 cent op amp and 2 - 14 cent diodes.

Perhaps its unfair to apply that to this level, but I don't have the luxury of spending 10s of thousands of dollars to get and use all these units. For all I know an RME ADI 2 might get me 99% there. I am not trying to be Gateway out of the starting line. But I don't want to sell myself short so I will the questions that may sound obtuse to someone with a $20,000 collection of converters who doesn't believe he could possibly to do his job without them.

[dcollins]

Originally Posted by jamsmith

Everything is a serious question.

Nyquist–Shannon sampling theorem - Wikipedia, the free encyclopedia

DC

[jayfrigo]

Originally Posted by jamsmith

I have a sine, a triangle, and square wave. I am sampling in this case at three points resulting in the 3 identical digital representations. I don't see how any kind of filtering will restore each wave accurately. Or is PCM sampling working in a completey different manner? (And yes, Bob's book is on its way)

OK, first wave: sine, 1 frequency. Second wave: triangle, fundamental plus a complement of harmonics. Several frequencies. Third wave: square, fundamental plus all odd harmonics, infinitely high theoretically (though you can't transition infintely fast, so you can't actually have all harmonics, infinitely high, of course).

Think Fourier, and think sinc functions. A square wave is just a bunch of sines - a fundamental and a bunch of odd harmonics. What frequency square do you want? 1 kHz? Well, you'll get the 1 kHz fundamental, and all the odd harmoincs within the bandwidth of the sampling system, which is to say below Nyquist, or 1/2 the sample rate. You want 20 kHz? Well, you'll get all the harmonics up to Nyquist, but there won't be too many this time if you're sampling at 1X rates. Anything within the bandwidth will be preserved just fine, but the ones outside of the bandwidth won't be there. Of course you won't really hear the difference because those harmonics are so high as to be beyond human hearing anyway.

You only lose what's beyond the bandwidth. In trying to preserve al the sharp transitions with so high a fundamental, what you are really trying to do is preserve very high frequency information (harmonics) above Nyquist, and above human auditory perception. We already know that the bandwidth is limited for a digital sampling system, so we should not be surprised when the harmonics above Nyquist that would result in the sharp transitions of a very high frequency fundamental are not present after sampling. In fact we needed to filter those frequencies out with the anti-alias filter. You only lose those super-high harmonics beyond Nyquist, and beyone our ability to hear them (caveats and implimentation details in here, depending on filter issues and what sample rate you are using).

Connecting the dots is a great conceptual aid, but that's now how it works. Digital audio is just that: digital. It is numbers. I suppose it's possible you can't fully appreciate it without looking at the math. Digital audio is not a graphics file, a picture of the wave. It is a numerical representation of it. It is a set of numbers that describes the waveform so that you can recreate it later. 2+2=4. 1+1+1+1 isn't any more of a 4 just because you used more data to represent it.

Just over two points will give you enough data to know exactly which sine wave within the bandwidth is able to pass through those points. There's only one that will fit it. That's the one that gets recreated. Anything squared off inside the box (those stairstep visual conceptual aids) gets filtered by the anti-image filter on the DAC. Remember, the squaring of a wave is added high frequency information, so when you filter the high frequency stuff that wasn't in your original band-limited waveform, you get back your original waveform. And because of Fourier, if it works for the fundamental, it works for the harmonics.

But I guess the real issue I was getting at to begin with is are these differences in these different high end converters really that signficant? Is this a Steinway vs Dusendorfer argument or Steinway vs Kawai argument? (And thats still a bad analogy because I know a good pianist who love Kawai and hates Steinways)

The digital sampling stuff is not why these converters sound different. It's analog stages, filtering, power supplies, and a myriad of other implimentation details. Once you get into the top end, like Lavry (gold series), Prism (dream series), dcs, PM's HDCD box, they are all excellent, and it comes down to taste. However, there are real differences between these and the mid level pro stuff like Apogee, Benchmark, even the Lavry Blue and Black series, and perhaps Prism's new, more affordable products like the Orpheus. And compare the top stuff to low end converters like an MBox or something, and your grandmother could probably hear it. The last few percent is where all the effort and money goes. It's a logarithmic scale. The question is, how important to you is that last little bit?

[jayfrigo]

The "bibles" for digital audio are by Watkinson and Pohlmann. Check amazon.

[jamsmith]

"I suppose it's possible you can't fully appreciate it without looking at the math. Digital audio is not a graphics file, a picture of the wave. It is a numerical representation of it. It is a set of numbers that describes the waveform so that you can recreate it later. 2+2=4. 1+1+1+1 isn't any more of a 4 just because you used more data to represent it."

Uh, actually once you are in the digital domain, I don't have any problems there. (I have been a software engineer for 12 years.) But the alegory here doesn't work. Four individual samples are not "4". They individual 4 samples and not added. The math works that way counting the number of bits in each sample where 1+1+1+1 = 0100. And while digital audio is not a "graphics file" it can be represented graphically. When you see a waveform in your editor, you are seen a graphic representation of a sequence of samples. In a sense, a graphic file is not to different as a graphic file is collection of bytes or words that describe individual pixels much as samples describe points on a wave.

As to the sine components and Nyquist, that is what I always knew to be true. It just appeared from the way I read things here (sometime you guys can be brief to the extreme) that something else was believed. I will not elaborate on that, I will accept it as my misinterpretation. I just wanted to clarify it for myself.

[rufus13]

Originally Posted by jamsmith

Well, theoretically it still is about hire rates. If you could sample at rate higher than ...<snip>

Yes it is. If a person is hiring a bit of gear for $200/day, is that gear better or worse than a $175/day item?

At the level of Lavry Gold, everything is subtly affected by everything else. There just is not much more quality to be had in converter hardware, even if you are willing to pay 10X more money. At the top of the range, everything is a cross-grade, not an upgrade. As Bob O. mentioned, small level differences may dwarf every other effect of a converter change, when both are of the best types available.

Of course, we have to keep abreast of new stuff, but we don't have to own one of each.

What a person can do, is not allow errors in installation or use creep in. That's a user/training problem.

Remember who has real power: Musicians who write the music / perform the sounds and microphone-placement-engineers. Every step after has diminishing power to positively affect the recorded music. Accept your place in the system, and do good work.

Cheers.

[Bob Olhsson]

Where I think this stuff gets confusing is that it's easy to try and apply analog audio concepts to digital audio. Even more confusing is the fact that some analog concepts such as gain staging and headroom very much apply to digital audio too. Finally there is the issue of how the analog audio is treated in a digital converter and how the analog clock is treated.

Bob's book excels as digital audio for those of us having lots of analog audio experience. Wilkinson is an invaluable reference that looks deep into the math and theory. I frankly never got that much out of Pohlmann which always seemed to me like it was oversimplified.