Distortions' dynamics and human perceptions-what to measure?

[Guest]

[quote author="JohnRoberts"][quote author="recnsci"]Few (uncorellated) thoughts on some issues arrised here and elswhere:

Subjective impressions:
Many people dismiss subjective impressions ("ears are lying, brain is
lying even more, so personal opinions are irrelevant"). And people
that prefer subjective impressions over measurments go to great
leanghts in fighting each other over which device sound good and
bad.
And here we have a problem, IMveryHO.

I will give you a story as example. Long ago I was very into analog
synths and their "guts" and saturated my inbox with mailing list very
similar to this forum. And everybody had opionion on various topologies
and incarnations of filters, oscilators and rest.
What I noticed is that my attention was focused on how voltage controlled
filters react to transients on voltage controll input, that is to fast envelope.
I am not refering to controll law ( linear or expo or something else). I am
reffering to the fact that different circuits react very diferently to sudden
changes of some state variables (like resistance of optoresistor).
So imagine little mind experiment. Take two filters that measure
same in THD, gain vs freq and other departments, but differ in a way
they respond to fast envelopes. And me and some guy that doesnt care
about reaction to envelope that much are listening tho those two filters.
He would say "They are same" and I would say "They are way off".
And, equaly imporant, I noticed one interesting pattern. As time went by
my capability to diferentiate various sounds got better. That is, at first
every filter and every EQ sounded same to me. But with more and more listenting I developed capabilitie to "focus" on some aspects of a sound
(like, fine details of what is 5KHz boost doing to a cymbal). This
focusing was actually concious brain activity. IMHO, ears dont get tired
at normal listening levels, but our brains have limited attention span.
I have a theory that what differs big money mixers (like Lord-Alge bros)
to us mortals is that their focusing capability remain intact during 12 hours
mixing session (and they know what to pay attention to).

Conclusion? Well, I think ears are as valid testing machines as any bench
procedure. And similar to bench testing, one pair of ears is focusing
on this aspect (because brain pays attention to that aspect) and other
pair focus on some other. One bench test will reveal CMRR other will
tell THD.

I'm not saying that everything people think they hear is real. There is
lot if imagination (especially in audiophile world) involved. People
decide do discern some info and make up nonexisting sounds.
I am however saying that ear itself is perfectly respectble tool. Its
up to us to train our brains to focus and to force our self into "unbiased"
mode.

And one final remark on this topic ( i promisse). I grow more and more
concerned on how unbiased ABX tests are (ok now I'm pulling flame
resistant suite). Especilay in psychology tests that revealed final limitations
of our hearing systems. I went numerous times to hearing tests. My
brain got lazy after 30 seconds of listening to static sinewaves. After
10 minutes I was barely able to hear. I asume that after 6 hours
of listening to that crap I would become psychotic. I guess you got my
point (and probably dont agree with it).

Now other stuff.

Testing
I think we should start new topic and resolve this issue once and
for all.
Test like THD number with barely loaded output (without notion of
exact topology) at 0 dBu sine on input are hardly important. No one
will dissagree that two circuits with same THD number could sound
raddicaly different. Also coments like "this IMD test will reveal a lot".
What da fuk is that test revealing ? differences? apples? oranges?
holly grail?
We need:
1) comprahensive set of testing procedures
2) comprahensive set of theory papers that will discuss what are
we measuring and how
3) rigid set of rules for measurment like amp loading, topology etc
so that we are sure we are comparing apples to apples
4) disscusion of how to interpret results
5) (very important IMHO) growing database of conducted tests
so that we have benchmark to look at

So pretty please, could we open discussion on this matter ?

OpAmps and discrete vs IC:

Well, back to actuall toppic. When discussing merrits of those 8 legged
monsters, we should always keep in mind context, and that is using
them in actuall audio circuits. There are few important audio tasks
where ic opamps still fail short, stuff like driving dificult loads and
providing lot of gain. And if they do that stuff gracefully they tend to be
expensive. And, opamp=clean discrete=dirty(euphonic) is simply not
true as general statement. TL0xx-driving-1K = dirty , gordon audio
preamp = clean beyond capabilities of any current ic topology.
But in a next few
years we will probably have ic opamps that will do most of tasks in
true wire with a gain aproach at reasonable prices (maybe we have them
already).

And finaly:
DO we need transparent ?

Transparency (for whatever that means) seems to be goal of audio for
quite some decades (whole fukkin century). Well, I have very personal
opinion that differs. It's not that I like "euphonic" here and there, its
that I need it most of the time. Two very important elements in audio
chain are badly lacking in performance after all these years, mic at front
end and speaker at back end. In every aspect, freq response, thd, transient response, you name it. Even more, whole concept of recording
is doomed from begining. Realism would mean that soundfield in recording
environment is reproduced in listening environment. Harsh physical
reality of transducers, rooms and formats tells us "Nope honey, no can do"
We are always creating arteficiall image and even if by direct divine
intervention "perfect" reproduction chain apeared tommorow, 90% of
music genres today rely on "arteficiall" interventions in recording.

While I agree that "colour" should be added when its needed, and that
we should have transparent block in chain on other occasions, problem
as I see it is not in making that transparent segment. Shit, I think
5532 is transparent for me (and most of others) in most ocassions.
Problem is creating boxes that give more on output than you provide
on input, cus I need those. And if box has 0.1% thd it doesnt mean its
good sounding. It will probably sound like shit.
So, issue is box that is making "good" 0.1% thd on some sources
(and if some box is doing good on every source in every context
I'm willing to donate kidney for schematic, layout and BOM).

If you are still reading this that means you have too much time on your
hands. Go to studio and record, or go solder something.

cheerz
urosh[/quote]

The golden ear vs. meter reader debate is probably older than some readers here. The argument is constantly corrupted by straw men (misleading characterizations) about each other's position. The GE will argue all MRs are over reliant on tests that don't reveal all flaws, while MRs argue GEs are constantly reinventing the wheel and too invested in things only they can hear that seem to disappear under scrutiny. There are surely examples of both with more than a little truth, but IMO that is not the norm for both groups.

I have a slightly different take on the GE/MR issue. If you can reliably hear some audible phenomenon that is real and not a perceptual distortion, that phenomenon will have a physical basis that can be measured. Once that is measured, it can be managed in designs. Further as perceptual distortions are understood, they too can be integrated into designs and/or practices (like loudness eq decades ago) .

I won't waste bandwidth with a full list of straw men, but the "same THD as" is a classic. Low order harmonic distortion will have a dramatically different perceived sound quality than high order crossover distortion. Trying to equate the two is disingenuous.

I believe we all have the same goal (good sound), and can probably learn from each other. If someone identifies a truly new phenomenon they'll get their name in the technical journals, but please don't ignore the several decades (or more) of work that has gone before us.

JR

PS: I too find $20 an AES paper expensive, but as a long time member I read many papers when they were fresh. I concede these days I don't follow it as closely.[/quote]

John; if you advocate several decades (or more) of work that gone before us, and was able to read all that papers, can you shed please some light?

 

[Guest]

[quote author="Svart"]It's a debate that will never end. Audiophools tend to forget that the audio they listen to through their 20K$ stereo was likely run through tons and tons of opamp ICs.

IC's like every other parts chosen, have their place in the design world. I've been doing listening studies with IC's in my console and find that for high audio frequencies I like the 5532 and bipolar caps coupling and for low frequencies I like the 2134 and no cap coupling. Each has their sound and their place in the world. As for outboard units, for insert gear a 5532 would be fine, but for main output gear before final mixdown, I find that the discrete stuff wins out.

Legos man legos.[/quote]

Svart;

I always thought that all attention must be paid on mic pres and power amps, because there are parts in the chain that have most power gain so introduce more distortions than the rest of devices that may contain hundred of ICs. Later, when I bought a TOA console as a foundation for my project I realized that I was wrong: bufers, summing amps, EQs, matters as well. Even when crossover distortions are nearly unmeasurable they impact a lot. Especially, when there are more than one device in the chain that adds them.

Speaking of audiofools, there are couple of poles about them, who prefer single ended A class amps, and who prefer compressors+noise gates on each and every channel. Both believe they are right and think of people on other pole as if they are totally deaf and/or ignorant. It may be true in some kind, but not completely...

Did you hear about NLP? Richard Bandler and John Grinder?

 

[Guest]

[quote author="bcarso"]for example:

http://en.wikipedia.org/wiki/Haas_effect
[/quote]

Thank you. It is about the fact that delay in the range 10-40 ms means more for binaural localization than levels of signals.

 

[JohnRoberts]

[quote author="Wavebourn"]

John; if you advocate several decades (or more) of work that gone before us, and was able to read all that papers, can you shed please some light?[/quote]

I will be glad to participate but can't easily do a brain dump of decades of accumulated tidbits. This could almost be a full time forum, let alone a single thread. I don't have the time or energy to provide a comprehensive answer right now but will suggest some broad areas of possible interest.

The process of human audition consists of one relatively linear part (ear), attached to a self programming computer (brain) that is anything but linear. While individuals differ there are generalities that can be applied with good results.

I already gave one pretty common such compensation for perceptual distortion with loudness contour applied to compensate for Fletcher-Munson perceived loudness curves. This perceived frequency response vs. loudness effect is rich with application ranging from shady hi-fi salesmen trying to steer us to to their preferred selection by playing it louder, to dynamic noise reductions and reduced bandwidth digital encoding schemes.

I think I also referred to time dependent mechanisms. Probably based on mechanical issues with our inner ear's AGC, and brain processing short cuts (lots of data gets ignored search "perceptual masking").

In fact the basis for stereo playback is a parlor trick to make us believe stereo is an accurate recreation of some more complex alternate sound space. Our brain is often receptive to and complicit in this subterfuge. As we become more sophisticated, or perhaps familiar with real acoustic music and performance spaces, we are less easily fooled.

Hopefully others will chime in with more suggestions. Another prolific researcher that comes to mind is Diana Deutsch http://psy.ucsd.edu/~ddeutsch/. That might keep you busy for a little while, but is only one aspect of this.

I repeat, there is a large body of knowledge out there, I am not the gatekeeper. For me much of this is filed away among my mental cobwebs and not neatly accessible, but like Ohms law it all is considered on some intuitive level during design. I can explain after I do something why I did it, but to list all constraints before hand is formidable.

JR

 

[JohnRoberts]

[quote author="Wavebourn"][quote author="bcarso"]for example:

http://en.wikipedia.org/wiki/Haas_effect
[/quote]

Thank you. It is about the fact that delay in the range 10-40 ms means more for binaural localization than levels of signals.[/quote]

Don't underestimate the significance of Haas, Madsen, et al. While indeed the basis of the mechanism is caveman stuff to recognize what direction the saber tooth was coming from, this was applied to consumer delay/surround sound ('70s to modern use in film surround). For example simply delaying an impulse a modest amount of mSec will shift some early reflections outside of the fusion region making them audible to us. Of course too much delay is counter productive and we hear a repeat.

That reminds me.. perhaps also search "Pinnae transforms ". This relates to the comb filtering that goes on in our outer ear that provides vertical directional cues (slight oversimplification).

JR

 

[Svart]

Absolutely right Wavebourn, the whole device is restricted by the weakest link in the audio chain. It would be like using some neve preamps then going through a njm4580 buffer... Your sound will be hindered by that last crap opamp i don't care what the datasheets say.

You do want your fastest, lowest distortion gain first and then you can settle for less drastic stages for buffering and such but again the weakest link rule applies.

 

[Guest]

[quote author="JohnRoberts"][quote author="Wavebourn"][quote author="bcarso"]for example:

http://en.wikipedia.org/wiki/Haas_effect
[/quote]

Thank you. It is about the fact that delay in the range 10-40 ms means more for binaural localization than levels of signals.[/quote]

Don't underestimate the significance of Haas, Madsen, et al. While indeed the basis of the mechanism is caveman stuff to recognize what direction the saber tooth was coming from, this was applied to consumer delay/surround sound ('70s to modern use in film surround). For example simply delaying an impulse a modest amount of mSec will shift some early reflections outside of the fusion region making them audible to us. Of course too much delay is counter productive and we hear a repeat.

That reminds me.. perhaps also search "Pinnae transforms ". This relates to the comb filtering that goes on in our outer ear that provides vertical directional cues (slight oversimplification).

JR[/quote]

It reminds me like in early 70'th I was impressed by comb-filter sound on one Deep Purple record. I did not know yet it was a magnetic tape effect and tried to reproduce it using complex set of rejector filters. I tried mechanically tuned LC filters, T-bridges, etc..., but nothing satisfied. Later I heard similar effect (not exactly the same, but satisfactory similar), called "Phaser". It gave me a clue, and overnight I made a device, precisionally summing direct signal with phase shifted by all pass filters (I knew already from my calculations that the closest teeth of the comb may be obtained with 2 in power number of filters). Later, in early 80'th, I've implemented "rotating space" effect, passing left master channel from console through a delay line from Ronald flanger. I expected horizontal localization oscillations, but to my surprise some listeners subconscially rotating heads rotated them vertically as well, and I heard also vertical localization change! It corellates well with what you said about vertical directional cues.

 

[Kev]

[quote author="Wavebourn"]I always thought that all attention must be paid on mic pres and power amps, because there are parts in the chain that have most power gain so introduce more distortions than ...[/quote]
apart from the voltage or current addition/gain
thay are also where the electro mechanicals take place

they will always be something that is the last Analog before the digital and then the point at which digital become analog so we can hear
:shock:
unless we get a USB fitted to our brain

good attempt at a discussion thread and I'll take part as much as I can offer something
I with John and believe that if you can identify it consistently then we should be able to develop a measurement procedure

I don't think that THD and TIM describe enough of what we need to know. It is a deeper subject than that.

Yes stereo is just a parlor trick and so to is Dolby ProLogic ... the stereo matrix trick.

 

[JohnRoberts]

[quote author="JohnRoberts"]


It reminds me like in early 70'th I was impressed by comb-filter sound on one Deep Purple record. I did not know yet it was a magnetic tape effect and tried to reproduce it using complex set of rejector filters. I tried mechanically tuned LC filters, T-bridges, etc..., but nothing satisfied. Later I heard similar effect (not exactly the same, but satisfactory similar), called "Phaser". It gave me a clue, and overnight I made a device, precisionally summing direct signal with phase shifted by all pass filters (I knew already from my calculations that the closest teeth of the comb may be obtained with 2 in power number of filters). Later, in early 80'th, I've implemented "rotating space" effect, passing left master channel from console through a delay line from Ronald flanger. I expected horizontal localization oscillations, but to my surprise some listeners subconscially rotating heads rotated them vertically as well, and I heard also vertical localization change! It corellates well with what you said about vertical directional cues.

IIRC the effect you refer to by Deep Purple was accomplished by "Tape Flanging" or aka "Reel Flanging". This involved combing two identical tracks with one slightly delayed varying around leading or lagging in time. The variable delay was accomplished by dragging a hand on the outer edge of the tape reel on the second tape machine (thus the name “reel flanging”). This was later reduced to a studio black box "Flanger" using analog delay chips. A similar but not as rich effect "Phasor" (SP?) was generated for guitar pedal effects using an all pass phase shift instead of true delay. Both approaches created comb filter responses but the number and spacing of comb tines varied.

JR

PS: FWIW I designed the Loft 440 and 450 series of studio delay-line/flangers that were used in '70s/80's. For a really obscure reference the flanging on Heart's Barracuda track was accomplished with a $39 delay/flanger kit I sold in mid '70s.

 

[Guest]

I agree with you Kev that an audio art is the same like other arts which purpose is to fool perceptions in order to create designed imaginations.

I'm moving my posting from the discrete opamp topic, it wass off-topic there...


[quote author="JohnRoberts"]

There has been a great deal of study and knowledge documented about human perception. I took advantage of some of the time dependent perception characteristics while designing tape noise reductions and dynamics processors ('70s-'80s) to help conceal the gain manipulations.

[/quote]

Is it possible to find some of them?

PS: In a perhaps not surprising data point, customers preferred an older generation noise reduction wrapped around the analog delay chips that exhibited significant amounts of low order harmonic distortion. Since the customer is always right and it was after all an effect intended to alter the sound, adding distortion in this case was perfectly consistent with the products design goal (to make stuff sound good). IMO The design goal of a reference playback system is to reveal what's there, not change it. YMMV

I agree with you about "reveal ... not change". The question is, what "not change". It is impossible "not change" at all, something always will be changed. The question is, what. People tend to pay attentions to some "side effects" of "less changed" sound such as high level of lower order additional harmonics, while main reasons why it seems to be "not changed" slips through fingers...
Like all strings are aligned perfectly horisontally, but the piano still sounds out of tune... :roll:

[quote author="Kit"]

Speaking of audio we have to measure them against subjective perceived levels

You said it yourself......"subjective" is the keyword here.

My ears might favour a different opamp than yours.

Data and specs is the only common language we got on this matter.[/quote]

Sure. My point was, distortions' change means more than some absolute level of them. Everything flows, everything changes. Look at a snake in a cage if it does not move. You don't see it, I guarrantee, if it is still. But if it moves far from you, far from your direct sight, you will immediately turn your head, blood pressure will rise up a bit, adrenaline level will go up immediately. Similatry, switch on a light bulb in your bedroom in the middle of the night, to discover how bright the light is, despite it seems to be very dim during the day when the sun shines. Dynamics, this is what is more significant. Everything change, and we are trained to spot changes.

However, it is possible to measure sensitivity of human eyes to different light frequencies, also it is possible to measure gradations of light that seem to be different, depending on the background brightness. The same with sound... Human perceptions were trained a long time, generation to generation, no matter if to believe Darvin or Church clerics, anyway all our perceptions are best suited to live in the real world, where brightness, loudness, temperature, change in certain levels, and some levels are better recognizible, especially when they change.

Back in 30'th Fletcher and Munson experimented with loudness' perceptions and found that different objective measured levels of sound of different frequencies seems to be equal in loudness, and such loudness - frequency dependency curves depend on sound pressure, and such curves correllate well from listener to listener. I.e. subjective perception curves may be objectively measured!

You may experiment for yourself, and probably your perceptions are similar to perceptions of other people, no matter what kind of operational amplifiers do you prefer.

http://www.phys.unsw.edu.au/jw/hearing.html

 

[Guest]

[quote author="JohnRoberts"] For a really obscure reference the flanging on Heart's Barracuda track was accomplished with a $39 delay/flanger kit I sold in mid '70s.[/quote]

I heard the rumor, like one company advertized a premium to somebody who can fix a production line they built for the plant for some 3'rd world country. A man who did that used a hummer to knock once. Bookkeepers were greedy and have demanded a specification. The answer was, "$1.00 for knocking, $999,999.00 for the knowledge where to knock".

So, how loudly can we knock on different levels of sound before it will be audible?

 

[Kev]

but
that knock was a hummer knock
... it may not translate to a VW knock


distortions are usually measure with NOISE & Distortion measurement systems
AND
the measured distortions has NO regard for phase or group delay of the distortions ... if there are any ... and IF they matter at all.

That's what I have been on about
Have we identified the very broad subject of distortions in the audio band
and
which distortions effect the human collection and computer system

Fletcher and Munson was largley about levels and frequencies
:roll:
is there a Fletcher and Munson type of effect in the perception of the various distortions ?
OR
does it go deeper than levels and frequencies and which harmonic ? ... delay/phase


will we one day talk about the Wayne & Wavebourn distortion analysis
:wink:

 

[Guest]

[quote author="Kev"]

will we one day talk about the Wayne & Wavebourn distortion analysis
:wink:[/quote]

Until now, Wavebourn distortion analyzer was based on speculations, how do they corellate with natural audio phenomena in the real life. What I've found so far,

1) Natural distortions are asymmetrical. Only abused mechanical systems distort symmetrically.
2) The more sounds fade out, the less harmonics they have, and the lower in level and order they are. It work well both for sounding and reflecting mechanical systems.

3) Designed in 1,2 in mind sound tracks the more distort, the louder they sound subjectively, despite powers of signals are the same. I.e. they don't sound more distorting, they sound louder.

[quote author="Kev"]
Fletcher and Munson was largley about levels and frequencies
:roll:
[/quote]

Yes, but thanks to them noises are measured against levels of audibility, and nobody tries to achieve less level of noises on less audible frequencies increasing level of them in more audible band.

 

[Kev]

Natural distortions are asymmetrical.

already we have a problem in that we don't have a definition yet

Natural ?
symmetrical, asymmetrical ... in polarity and/or lobe structure ?

that's without looking at the other points

:roll:

sorry I can't let #2/ go
I assume you are on the Natural distortions thing
generally opamps and mosfet amps have GREATER distortion the lower the level
that is dis-counting the Just before and at clipping part of the graph

My issue is that, so far, everyone has brought a pre-conceived idea and/or agenda to the discussion
Analysys of distortion or any electronic circuit should work evenly and consistently for what you call natural OR abused mechanical
or any other type of distortion.

:sad:
now I can't let #3/ go
" ... the louder they sound subjectively ... "
TIM doesn't make things sound louder


and the last

but thanks to them noises are measured against levels of audibility, and nobody tries to achieve less level of noises on less audible frequencies increasing level of them in more audible band.

Less ... wasn't my point
it was more about your agenda or creating a pleasant sound
in that IF you wanted to add some cool distortion you may want to aim it in area that the human finds pleasurable

It would require a deep analysis of the human's perception of distortion vers frequency vers phase ... and perhaps even more parameters.

sorry no more net time

 

[Guest]

[quote author="Kev"]
Natural ?
[/quote]

Yes, I mean mechanical. Strictly speaking, spectrum of mechanical sounding and reflecting systems is the better word than "distortions".

symmetrical, asymmetrical ... in polarity and/or lobe structure ?

In the form of transfer function, however.

[quote author="Kev"]
it was more about your agenda or creating a pleasant sound
in that IF you wanted to add some cool distortion you may want to aim it in area that the human finds pleasurable
[/quote]

It was my agenda in 70'th when I designed musical synthesizers learning what kind of forms and spectrums how sounds. My current agenda is to design sound channels for live concerts and records of traditional musical instruments players and singers, so despite of electrical modifications they still sound like alive, it is the different story.

 

[JohnRoberts]

[quote author="Wavebourn"]

I heard the rumor, like one company advertized a premium to somebody who can fix a production line they built for the plant for some 3'rd world country. A man who did that used a hummer to knock once. Bookkeepers were greedy and have demanded a specification. The answer was, "$1.00 for knocking, $999,999.00 for the knowledge where to knock".

So, how loudly can we knock on different levels of sound before it will be audible? [/quote]

This must be one of them non-sequitors I've heard so much about. :?:

In the spirit of providing more references, maybe study the old Picasso joke this is based on.

"...an aging Picasso is sitting in a Paris cafe and a woman tourist begs him to draw something for her on a napkin. He complies and then asks her for a thousand francs... Why she exclaims, that only took you 30 seconds to draw? No, he replied it took me 65 years."

The Fonzi (hammer tap) variant usually involves a consultant of some sort, but it does reveal the nature of how little human's value experience and knowledge vs. visible toil.

JR

 

[Kev]

[quote author="Wavebourn"]... Strictly speaking, spectrum of mechanical sounding and reflecting systems is the better word than "distortions".[/quote]
:shock:
then why is the thread title
" Distortions' dynamics and human perceptions-what to measure? "
if we are to talk about these terms then they need to be defined
and
I don't think that distortion has been correctly defined
and
until it is defined ... how can you consistently test the human and their perceptions

... In the form of transfer function, however.

what ?
the distortion is a transfer function ... ?
what ??
I have no idea what you are trying to say

... My current agenda is to design sound channels for live concerts and records of traditional musical instruments players and singers, so despite of electrical modifications they still sound like alive, it is the different story.

:roll:
what ?
how does that relate to
" Distortions' dynamics and human perceptions-what to measure? "

once we have a measurement system and a repeatable analysis to the above, then you can perhaps you can find a way through
" ... despite of electrical modifications they still sound like alive "


AND George will disagree ...

http://www.3daudioinc.com/3db/showthread.php?t=6160

http://www.3daudioinc.com/040602_f2f_massenburg.mp3


even so I do see why people want to find and add colour ... so to speak

 

[mikep]

Distortion is any deviation from perfect reproduction. It can be time, amplitude, phase, waveshape (harmonics), and other things we haven't yet quantified. saying that distortion is a transfer function makes it sound like you think you figured out all the terms and are just searching for the coefficients. might lead to better understanding of the low-order effects, but if you are trying to judge the overall performance this approach is fundamentally flawed.

there is one uncommon measurement that takes *everything* into account, even the stuff we don't yet know we are looking for. The phase cancellation test. you take your output signal and mix it with an inverted copy of the input signal. the level must be set precisely (tune by hand) and the summing amp should be "free of distortion". I think Mr Hafler wrote about this technique for testing power amps. it works, you can listen to the residual or look at it on a scope. no notch flter is needed. Why is this test not used more? Why am I not using it? :roll:

I am guilty of using a traditional distortion analyzer. But I think there is alot more info in there than the meter/scope is telling you. just listen to the residual. you ear can do a pretty good job of ignoring the fundamental bleed-through, the analyzer's own distortiion AND noise. I can usualy hear some low level, higher order products that are down in the noise that don't show up on a scope. you listen to the looped-back residual for a few seconds and then switch to the input, even when the meter is at its measurement limits you can often hear a change. For what its worth, the frequencies I use are 400Hz and 200Hz. this way the harmonics are in a range where they are easy to pick out. I guess there is a bit of a learning curve to this method but for anyone with musical background or lots of critical listening experience should try it.

mike p

 

[Guest]

[quote author="Kev"]
AND George will disagree ...

http://www.3daudioinc.com/3db/showthread.php?t=6160

http://www.3daudioinc.com/040602_f2f_massenburg.mp3
[/quote]

Did you ask him? :grin:

even so I do see why people want to find and add colour ... so to speak

This topic is not about why other people want to find and add colour. It is about what to transfer without modifications. For example, once the whole party of assembled receivers come from an assembling line defective. Investigations reveiled that one senior lady who worked for years changed one capacitor. Why? Because new capacitors she got to install had the wrong colour: she used to install red capacitors for a long time, now she got gray capacitors. She decided that it was an error and corrected it. She took red capacitors from another bin, but they had wrong capacitance, despite they had "right" colour. She "transfered right colour, without modifications". And she believed she was right.

Now, about transfer functions: no matter how complex they are, they are still transfer functions that may be obtained by objective measurements. The question is, what deviations from straight line to ignore in order to assume that it is straight, because it can not be straight absolutely, any active elemnts are non linear, and any audio design is about finding right compromises. The question is, what may be compromised, and to what degree.

Now, speaking of human perceptions, suppose you are making video equipment trying to transfer all spector well, from radio waves to gamma-rays. Measurements are excellent, but the movie still looks bad. Why? Because trying to capture spector that is out of perception of eyes you loose focus on a visible part of the spector allowing more distortions inside of it's visible part thinking that it is not significant.

Very similar effect happens when people build preamps with the single button with THD of 0.02% or less on levels up to 10V, but when sounds fade out distortions rise up and their spector is wide, instead of concentrating on more significant parameters that are better perceived.

They sincerely believe that they build "straight amplifying wire" and convince others using good hypnotic voice. You know, all people who believe have a good hypnotic voice. When a seller in a high end audio boutique speaks of $20,000 speaker cables he uses such a voice so everyone around would believe that cables after certain number of hours of listening to proper music will magically break-in and start sounding better...

 

[Guest]

[quote author="mikep"]Distortion is any deviation from perfect reproduction.
[/quote]

Excellent. Copy will be always a copy. Reproduction is not the original. Birds can reproduce human speach, but do they transfer all meaning? Morse code don't sound even close to a human speach, but it reproduces well some valuable information.

For what its worth, the frequencies I use are 400Hz and 200Hz. this way the harmonics are in a range where they are easy to pick out.

Excellent. This reminds me a story about a man who've lost keys in a dark spot in the garden, but searched for them under the street light. Why? Because it was brighter there...

Ok, let's return back to the topic. Vocal, acoustic instruments, sound better with a single ended tube class A amp than with a modern AB class one designed like a typical operational amplifier. Why?
Contrary, modern music sounds better through a transistor AB class opamp, and horrible through a single ended tube class A amp.
Why?