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Topic: AES 2009 Audio Myths Workshop (Read 171289 times) previous topic - next topic
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AES 2009 Audio Myths Workshop

Reply #125 –
0.002% is a -94 dB. It is impressive but, as compared to state of the art, not IMPRESSIVE.

Using a 1 kHz stimulus is an accepted means of doing this sort of measurement. The fact that they went to the trouble to describe how they did the measurement puts them above average in this department.

If you want to get a better idea of how equipment will sound with more realistic stimulus, you can do swept versions of the test - plotting distortion vs. frequency and/or amplitude. (Using DSP it is now possible to do these tests with normal program material as the test stimulus. The live sound guys routinely do this to assess acoustic performance of their systems during shows - cool stuff.)

For working equipment, the resulting graphs consume printed space, are boring and not many people know how to read them.

AES 2009 Audio Myths Workshop

Reply #126 –
Quote from: dwoz on
Thus, while the idealized console/DAW itself is "perfect", there is NO perfect idealized model of one of the primary, key components, the A/D converter.


A/D and D/A converter chips are among the most perfected of all audio components within the scope of their natural use and purpose. Check the spec sheets for TI' s PCM4220 and PCM4222 chips
and the spec sheets for the previous generation of hotties by AKM and Cirrus.

We've had about 5 years of experience with the latter two families of chipt, and they turuned out to be every bit as good as they were claimed to be.

Furthermore, most if not all digital consoles can be operated entirely in the digital domain.  Of course you've heard of the Neumann TLM103D, right?

AES 2009 Audio Myths Workshop

Reply #127 –
Quote from: Arnold B. Krueger on
Did you read it before the grass was cut?

The trimmings can be found here, btw.  Feel free to quote anything there that is actually on-topic if you wish as there are some things there that were ok.  Unfortunately the stuff that was ok also contained stuff that wasn't (and still isn't!).  Common sense should dictate (hint: ad-hominem is not ok).

AES 2009 Audio Myths Workshop

Reply #128 –


Arnold, do you really assert that the "niche" aspect of the market is relevant?  Is this not about objectivity? 

If anything the market for pro audio gear is far far larger than the market represented by fools that will spend $3000 on a damn cable!

I am confused.  On the one hand, I have people posting about the theoretical perfection of digital systems, and on the other, I have you posting that it's all subjective, subject to hearing acuity...and you seem to present yourself as a qualified arbiter of "my" hearing acuity.

One reason that I keep beating this horse, is that the listener market cares about exactly one thing:  moving a single L/R signal through some boxes, and out into air.  (maybe 5.1 signal...),  but the audio production market is concerned with the creation and manipulation of many stages of intermediate products and artifacts, where for the most part, any issues with non-linearity will be additive.  This represents SUCH a fundamental difference that I claim we CANNOT conflate the two! 

It's like claiming that all vodka is just vodka.  If you think that audiophiles will want to draw-and-quarter you for your statements, wait until you try telling a Grey Goose imbiber that "Smirnoff will do".

AES 2009 Audio Myths Workshop

Reply #129 –
Quote from: dwoz on
So, basically "a digital console or workstation" is not "ALL digital consoles or workstations", but only the specific ones that only do summing, gain changes, and...panning.  Every OTHER digital console is not included in your term?


Most good digital consoles and DAWs have features that include nonlinear operations. The usual "your gun, your bullet, your feet" warning should apply.

IOW giving a DAW some nonlinear processing features doesn't take it out of our consideration if you turn the nonlinearities on and off at will.

However, complaining about what happens when you intentionally turn the nonlinearities on sort of qualifies you for a seat in the corner with a pointed hat, no?

AES 2009 Audio Myths Workshop

Reply #130 –
Quote from: Arnold B. Krueger on
IOW giving a DAW some nonlinear processing features doesn't take it out of our consideration if you turn the nonlinearities on and off at will.

However, complaining about what happens when you intentionally turn the nonlinearities on sort of qualifies you for a seat in the corner with a pointed hat, no?



Well...perhaps.  I'm just wracking my brain to come up with a mix of ANY popular music song that has experienced ANY critical and/or sales acclaim, that just uses gain and pan.

Care to cite one?  two?  If, in the 100,000 songs or so that have been released by a major label for airplay, you can find even THREE that use only gain and pan in their mix, I will gladly concede the point.

So, I guess that I am in a lot of very good company, sitting in that corner!

AES 2009 Audio Myths Workshop

Reply #131 –
Quote from: Arnold B. Krueger on
Furthermore, most if not all digital consoles can be operated entirely in the digital domain.  Of course you've heard of the Neumann TLM103D, right?


You can do bedroom techno and industrial using just digitally created sounds.  Doesn't dilute my point.  Does that microphone make any stop in the voltage/frequency domain, or does it directly create a digital stream from sound pressure gradients?

If yes, then it's just a converter that has been repackaged, no?

AES 2009 Audio Myths Workshop

Reply #132 –
I can't find a question to answer in all this mess. If someone can't understand that, yes, non-linear distortion is impossible to correct when stacked, but it doesn't matter because there's no audible (almost no measurable) non-linear distortion in the equipment we're talking about, where can you go from there?

Cheers,
David.

AES 2009 Audio Myths Workshop

Reply #133 –
The point Arnold is making is that DAWs only introduce non-linearities when their user tells them to.

AES 2009 Audio Myths Workshop

Reply #134 –
Quote from: Notat on
0.002% is a -94 dB. It is impressive but, as compared to state of the art, not IMPRESSIVE.

Using a 1 kHz stimulus is an accepted means of doing this sort of measurement. The fact that they went to the trouble to describe how they did the measurement puts them above average in this department.

If you want to get a better idea of how equipment will sound with more realistic stimulus, you can do swept versions of the test - plotting distortion vs. frequency and/or amplitude. (Using DSP it is now possible to do these tests with normal program material as the test stimulus. The live sound guys routinely do this to assess acoustic performance of their systems during shows - cool stuff.)

For working equipment, the resulting graphs consume printed space, are boring and not many people know how to read them.



Does that spec, as defined and referenced, tell me ANYTHING about how that equipment is going to handle the crack of a snare?

No.  The most I can infer from that spec as defined, is that when I hit that snare drum, the card will PASS SIGNAL.  It makes zero representation of what that signal will actually look like.

AES 2009 Audio Myths Workshop

Reply #135 –
Quote from: dwoz on
Thus, while the idealized console/DAW itself is "perfect", there is NO perfect idealized model of one of the primary, key components, the A/D converter. 

So, I ask...is this important to consider when you make your statement?

If you put imperfect signals into a digital console, you'll get a linear combination of imperfect signals coming out. The console can't possibly do any better than this. Once convinced of that, no, it doesn't need to be considered further. Converter performance can be taken as a separate problem.

AES 2009 Audio Myths Workshop

Reply #136 –
Quote from: dwoz on
One reason that I keep beating this horse, is that the listener market cares about exactly one thing:  moving a single L/R signal through some boxes, and out into air.  (maybe 5.1 signal...),  but the audio production market is concerned with the creation and manipulation of many stages of intermediate products and artifacts, where for the most part, any issues with non-linearity will be additive.


I don't understand what you're after, which kind of devices? Especially in the pro audio field, an all digital data path is nothing extra-ordinary. And a digital data path means zero percent signal alteration unless you do processing, and even that error will be a logical (e.g. rounding error) and not a physical property. The only analog elements, that introduce non-linearities, are those that you especially introduce into the audio path for their specific sound signature (as tube compressors). But who in hell would rant about the possible stacking effect by such a device, if you have brought it into the chain especially for its distorting properties?

Quote from: dwoz on
Does that spec, as defined and referenced, tell me ANYTHING about how that equipment is going to handle the crack of a snare?


Of course, what do you think isn't covered by a common ADC spec with regard to what you want?

AES 2009 Audio Myths Workshop

Reply #137 –
Quote from: dwoz on
Arnold, do you really assert that the "niche" aspect of the market is relevant?  Is this not about objectivity?


Common sense says that the presence of a few excepetions doesn not necessarily invalidate the rule.


Quote
If anything the market for pro audio gear is far far larger than the market represented by fools that will spend $3000 on a damn cable!


At some time in the recent past the market for pro audio gear and the market for high end audio were about the same in customers and dollars. AFAIK the revolution in portable digital audio has juggled everything up w/r/t high end and general consumer audio. For example, the market for products that turn an iPod into a stationary player is said to now be about the same size as the market for all other mainstream home audio compoennts.  The actual market for $3,000 cables has always been miniscule AFAIK.

Quote
I am confused.  On the one hand, I have people posting about the theoretical perfection of digital systems, and on the other, I have you posting that it's all subjective, subject to hearing acuity...and you seem to present yourself as a qualified arbiter of "my" hearing acuity.


If you turn off all the EFX a digital console should be very, very perfect and ideal and they seem to actually be that way.  What I'm  posting about subjective and hearing acuity is that the *requirements* for audio performance have to be related to hearing ability.  The digital perfection that is possiblw with a digital console is therefore overkill.

As far as meing being the arbiter of your hearing ability goes, I actually know nothing about your personal hearing ability, except that it is very likely no better than the best that has ever been reliably observed for any hunam being.

Quote
One reason that I keep beating this horse, is that the listener market cares about exactly one thing:  moving a single L/R signal through some boxes, and out into air.  (maybe 5.1 signal...),


Very much a 5.1 signal, since such a high percentage of all stationary and even much mobile listening is actually HT or A/V.

Quote
but the audio production market is concerned with the creation and manipulation of many stages of intermediate products and artifacts, where for the most part, any issues with non-linearity will be additive.


While Ethan and the rest of us are willing to debunk stacking, we're not going after additive distortion. As I watched Ethan's video, I saw a very orthodox presentation about additive distortion.


Quote
This represents SUCH a fundamental difference that I claim we CANNOT conflate the two!


In theory we can't conflate them, but Ethan provided a well-documented orthodox argument that I gathered and presented the evidence for back in 2002 at my now-departed www.pcabx.com web site.  The bottom line is that most good equipment performs so well that any reasonble cascading of audio gear is still unlikely to cause audible problems.

As more and more audio gear gets digial inputs and outputs, we just keep more and more of the processing solidly in the digital domain where there is inherently zero added distortion and zero added noise, unless adding them is exactly what we want to do.
 

AES 2009 Audio Myths Workshop

Reply #139 –
Quote from: dwoz on
Quote from: Arnold B. Krueger on
IOW giving a DAW some nonlinear processing features doesn't take it out of our consideration if you turn the nonlinearities on and off at will.

However, complaining about what happens when you intentionally turn the nonlinearities on sort of qualifies you for a seat in the corner with a pointed hat, no?



Well...perhaps.  I'm just wracking my brain to come up with a mix of ANY popular music song that has experienced ANY critical and/or sales acclaim, that just uses gain and pan.


Show me a public archive detailed logs of all processing and equipment settings from rehearsal through tracking, mic to pressed CD,  for a representative selection of popular songs and there is a possibility of a logical conversation. AFAIK nobody is even keeping those logs and if they existed, they would be highly proprietary.  So, if we try to generalize about them, we have a strong possibility of talking out the backs of our necks.

As far as compression and limiting go, I commented on the practical non-reversibility of them here in maybe the last week.

HA Thread about reversing dynamics procssing

This is all very all OT as compared to Ethan's 2009 AES audio myths presentation.

AES 2009 Audio Myths Workshop

Reply #140 –
Quote from: googlebot on
Of course, what do you think isn't covered by a common ADC spec with regard to what you want?


Well...last time I looked at a snare drum hit hard, it looked NOTHING like a 1kHz sine waveform.  It had a very fast-rising transient.  It had a vast multitude of enharmonic and inharmonic partials, at least two and maybe 4 fundamental frequencies, a whole lot of high frequency content well past 20k (which we will ignore because arnold said so).

A lot of these characteristics, could be and are damaged by the analog front end stages of that card...maybe once it gets into bits it's perfect, but damn.

I think you've told me all I need to know.  I think I may now understand the context that you are discussing this in.  It doesn't happen to match my personal context, but I think I understand the mapping a bit better!

AES 2009 Audio Myths Workshop

Reply #141 –
Quote from: Canar on
Quote from: dwoz on
Thus, while the idealized console/DAW itself is "perfect", there is NO perfect idealized model of one of the primary, key components, the A/D converter.
There are several. The big question is: which one? http://en.wikipedia.org/wiki/Analog-to-dig...#ADC_structures


Every contemporary  hiigh performance audio converter chip that I know of is Sigma-Delta.  Several variations on the basic theme exist.

AES 2009 Audio Myths Workshop

Reply #142 –
Quote from: dwoz on
I think I may now understand the context that you are discussing this in.  It doesn't happen to match my personal context, but I think I understand the mapping a bit better!

Hopefully your personal context doesn't stray too far away from TOS #8, otherwise this forum requires that you keep it squarely to yourself.

...and yes, there are things in the Audio Myths Workshop video that do not fulfill TOS #8.

AES 2009 Audio Myths Workshop

Reply #143 –
Quote from: dwoz on
Quote from: googlebot on
Of course, what do you think isn't covered by a common ADC spec with regard to what you want?


Well...last time I looked at a snare drum hit hard, it looked NOTHING like a 1kHz sine waveform.  It had a very fast-rising transient.  It had a vast multitude of enharmonic and inharmonic partials, at least two and maybe 4 fundamental frequencies, a whole lot of high frequency content well past 20k


That question was resolved about 2 centuries back by a guy named Fourier.

Every band-limited wave  (don't care how high the band limit is, just that it must be finite) can be analyzed and accurately characterized as a collection of sine and cosine waves. 

This isn't just an oddity of math becuase it is now routine to convert audio into one of these collections of sines and cosines, and then convert the collection of sines and cosines back into a regular wave as part of normal digital audio processing.

The reconstructed wave looks right, subtracts well from the origional leaving essentially nothing (you can make the essentially nothing as small as you want by simply using longer data words), and it sounds exactly right to everybody who bothers to do a good sensitive listening test.

BTW, the process is completely linear - intelligent but simple addition & subtration of the sines and cosines. I've done this at clock rates up to 10 MHz, so don't rag on me about ignoring the high frequency components.




AES 2009 Audio Myths Workshop

Reply #144 –
Quote from: dwoz on
Quote from: googlebot on
Of course, what do you think isn't covered by a common ADC spec with regard to what you want?


Well...last time I looked at a snare drum hit hard, it looked NOTHING like a 1kHz sine waveform.  It had a very fast-rising transient.  It had a vast multitude of enharmonic and inharmonic partials, at least two and maybe 4 fundamental frequencies, a whole lot of high frequency content well past 20k (which we will ignore because arnold said so).

A lot of these characteristics, could be and are damaged by the analog front end stages of that card...maybe once it gets into bits it's perfect, but damn.


  • How can you be sure what that specific snare's waveform should look like?
  • If you are sure, how do you know that your mic, its placement and the room's characteristics, is delivering what you expect, but the ADC's "analog front end stage" isn't?
  • Regarding high frequencies. ABX the >20 kHz record against version processed with a high quality low-pass. You would be the first to be able to tell a difference. Don't get anyone wrong, I don't know anybody here, who would promote mixing at 44.1 kHz. As you say, high resolutions eases filtering constraints, and producing at nothing higher than the delivery format is not worth the hassle.


I know that capturing extreme transients with inherently band passed systems can be tricky without experience. But a sensibly chosen input gain combined with a half-way decent dynamic range in your ADC is usually all that's needed. But that's nothing specific to digital audio.

AES 2009 Audio Myths Workshop

Reply #145 –
Quote from: dwoz on
Also, I have found, FINALLY, a couple references that actually use the words "linear distortion".  The references seem to be specific to loudspeakers,


It turns out that in the eyes of most orthodox audio authorities, loudspeakers are just about the only remaining kind of audio components where nonlinear distoriton is even interesting. So people who work with loudspeakers tend to be more rigorous about these kinds of definitions because in their work they still deal with a lot of audible distortion of both kinds.

Quote
and seem to anecdotally mention level-dependency as an factor, but not as a definitional term.


Nonlinear distortion is usually level-dependent. There are a few cases where nonlinear distortion is not level dependent (half wave rectification comes quickly to mind) but generally it is level-dependent.



AES 2009 Audio Myths Workshop

Reply #146 –
Quote from: Arnold B. Krueger on
Quote from: dwoz on
Quote from: Arnold B. Krueger on
IOW giving a DAW some nonlinear processing features doesn't take it out of our consideration if you turn the nonlinearities on and off at will.

However, complaining about what happens when you intentionally turn the nonlinearities on sort of qualifies you for a seat in the corner with a pointed hat, no?



Well...perhaps.  I'm just wracking my brain to come up with a mix of ANY popular music song that has experienced ANY critical and/or sales acclaim, that just uses gain and pan.


Show me a public archive detailed logs of all processing and equipment settings from rehearsal through tracking, mic to pressed CD,  for a representative selection of popular songs and there is a possibility of a logical conversation. AFAIK nobody is even keeping those logs and if they existed, they would be highly proprietary.  So, if we try to generalize about them, we have a strong possibility of talking out the backs of our necks.



for the purposes of this discussion, I think we need not be quite so rigorous!  For example, reverb will kick it off the list, mixing into compression on the 2-buss, (but let's ignore compression applied in mastering), any major EQ, compression or gating on drums, etc. Compression on lead vocal...  All stuff that you can easily hear even on earbuds from an iPod with an mp3! 

AES 2009 Audio Myths Workshop

Reply #147 –
Quote from: Arnold B. Krueger on
Quote from: dwoz on
Quote from: googlebot on
Of course, what do you think isn't covered by a common ADC spec with regard to what you want?


Well...last time I looked at a snare drum hit hard, it looked NOTHING like a 1kHz sine waveform.  It had a very fast-rising transient.  It had a vast multitude of enharmonic and inharmonic partials, at least two and maybe 4 fundamental frequencies, a whole lot of high frequency content well past 20k


That question was resolved about 2 centuries back by a guy named Fourier.

Every band-limited wave  (don't care how high the band limit is, just that it must be finite) can be analyzed and accurately characterized as a collection of sine and cosine waves. 

This isn't just an oddity of math becuase it is now routine to convert audio into one of these collections of sines and cosines, and then convert the collection of sines and cosines back into a regular wave as part of normal digital audio processing.

The reconstructed wave looks right, subtracts well from the origional leaving essentially nothing (you can make the essentially nothing as small as you want by simply using longer data words), and it sounds exactly right to everybody who bothers to do a good sensitive listening test.

BTW, the process is completely linear - intelligent but simple addition & subtration of the sines and cosines. I've done this at clock rates up to 10 MHz, so don't rag on me about ignoring the high frequency components.


yes, quite.  but you're answering a different question.  What, in the specification of a component processing a sine at 1kHz at full range amplitude, can I extrapolate to a full-range complex signal?  Beyond faith, hope, and charity, that is?  (reference to pandora's box is very deliberate!)

AES 2009 Audio Myths Workshop

Reply #148 –

Ok, so now, by the elite and competent office of Arnold Krueger, Canar, pdg, and googlebot, I think I have arrived at a conclusion.

if:

A) all modern digital equipment is by definition highly linear;

B) highly linear equipment is transparently high fidelity;

C) converters, even cheap ones, are nearing ideal linearity;

D) pretty much all modern equipment has published specs that are testably well-below audibility for distortion, euphonic or otherwise;

E) published specs represent a median or worst-case, typically, therefore the actual performance is probably even better than the already-non-audible specs...

THEREFORE:

In assembling and setting up a music production system, the notion of "choice" and "preference" are essentially irrelevant, sonically speaking, so basically ANY SYSTEM that I can assemble out of ANY current-vintage gear, that isn't demonstrably broken, will be higher fidelity than I will ever need, or in fact, ever perceive?

AES 2009 Audio Myths Workshop

Reply #149 –
Quote from: dwoz on
D) pretty much all modern equipment has published specs that are testably well-below audibility for distortion, euphonic or otherwise;
This is the falsest part of your logic.