Amplitude Sensitivity of Human Hearing

gene

gene

Audioholics Master Chief
Administrator
<P style="MARGIN: 0in 0in 0pt"><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><A href="http://www.audioholics.com/techtips/roomacoustics/HumanHearingAmplitude.php"><IMG style="WIDTH: 125px; HEIGHT: 104px" alt=[chart] hspace=10 src="http://www.audioholics.com/news/thumbs/chart_th.gif" align=left border=0></A>The human ear has been held by armchair acousticians and physicists as The Ultimate Microphone Ever Created.&nbsp;Truth is there are today microphones that can, with ease, outperform the human ear. Where the ear may, however, lay claim to the ultimate mic award is when it’s considered in combination with the post-processor to which it is hard wired, namely, the human brain. Mark Sanfilipo explores our hearing capabilities in these series of articles on the Human Ear.<SPAN style="mso-spacerun: yes"></SPAN>First up is our sensitivity to amplitude variation.<SPAN style="mso-spacerun: yes"> </SPAN><SPAN style="mso-spacerun: yes"></SPAN><?xml:namespace prefix = o /></SPAN></P>
<P style="MARGIN: 0in 0in 0pt"><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></P>
<P style="MARGIN: 0in 0in 0pt"><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">[Amplitude Sensitivity Part 1]<SPAN style="mso-spacerun: yes"></SPAN>[ The Physics of Hearing]</SPAN></P>
 
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A

audist

Audiophyte
flat response curves!?

excellent article..

This makes me wonder, shouldnt the ideal speaker response curve be an inverse of the human sensitivity curve? If so why do we keep targetting flat response curves for speaker design?

:confused:
 

Dumar

Audioholic
Another good article. I enjoy reading information like this because it helps me understand what's going on at the most fundamental level. Plus I get to say I learned something new today. ;)

One thing I would like to say is it would be nice if a short bio was included with the article so we could read a little something about the author.

The same goes for the appended editorial note. I was going to ask who wrote it, but after reading it and coming across the then/than conundrum, figured out it was Gene. :p

Keep up the good work you guys. :)
 
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mtrycrafts

mtrycrafts

Seriously, I have no life.
audist said:
excellent article..

This makes me wonder, shouldnt the ideal speaker response curve be an inverse of the human sensitivity curve? If so why do we keep targetting flat response curves for speaker design?

:confused:

I would think that you want a flat curve as a standard. The music will be mixed and EQed at mixing to a subjective sound to the engineers and producers.
Which curve would you design the speaker to? The curv posted is an average of many people.
 
gene

gene

Audioholics Master Chief
Administrator
The same goes for the appended editorial note. I was going to ask who wrote it, but after reading it and coming across the "then/than" conundrum, figured out it was Gene.
I wish I could take the credit, but we hired on a new writer whose knowledge in these areas far exceed my own.

Please welcome Mark Sanfilipo to our staff!
 
mtrycrafts

mtrycrafts

Seriously, I have no life.
I would suggest two more refs for the author:

"Level Discrimination as a Function of Level for Tones from .25 to 16khz", Florentine, Mary, et al, Journal of Acoustic Society of America, 81(5) May 1987, pg 1528-1541.

"On the Relations of Intensity JND's to Loudness and Neural Noise", Zwislocki, J and Jordan H., Journal of Acoustics Society of America, 79(3), Mar 1986, pg 772-780.
 
gene

gene

Audioholics Master Chief
Administrator
shouldnt the ideal speaker response curve be an inverse of the human sensitivity curve?
I don't think that would be advisable since the room actually causes a bump in bass to begin with and a natural roll off in highs is actually more pleasing to the ear.
 
A

audist

Audiophyte
mtrycrafts said:
I would think that you want a flat curve as a standard. The music will be mixed and EQed at mixing to a subjective sound to the engineers and producers.
Which curve would you design the speaker to? The curv posted is an average of many people.
ah..good point ! :)
 
M

MarkS

Audioholics Staff Writer
This makes me wonder, shouldnt the ideal speaker response curve be an inverse of the human sensitivity curve? If so why do we keep targetting flat response curves for speaker design?
For the sake of discussion, imagine you have a selection of music with a spectral balance that resembles the ANSI-A weighting curve (essentially the inverse of the 30-phon Fletcher-Munson equal-loudness contour).

Here in our example that means at 20 Hz the level will be ~ -50 dB when referenced to the level present at 1 kHz. At the other extreme, the level at 16 kHz will be ~ 7 dB down when referenced to 1 kHz.

Now further imagine you have at your disposal two separate loudspeaker systems: one sporting a flat response, the other sporting a response curve that resembles the ANSI-A weighting curve.

Playing your music sample back through both produces the following results:

Frequency (Hz) Music Sample (dB) Flat Resp. System (db) ANSI-A System (dB)
20 -50 -50 -100
100 -19 -19 -38
16k - 7 -7 -14
NB: dB levels referenced to level @ 1kHz

Played back through the system sporting the flat response, we find the levels at 20, 100 and 16 kHz are reproduced exactly as found in the original recording. Played back with the ANSI-A curve system and the results are very different: compare the original levels found in column 2 with those found in column 4.

Essentially, the flat response of a well designed loudspeaker is one indicator that it is doing its job of reproducing the spectral content of the original acoustical event without altering the instantaneous spectral balance in the process. The ANSI-A curved system did alter it - quite substantially. So, no, making a loudspeaker response curve the inverse of one of the equal loudness contours isn’t the direction you’d really want to go.
 

Dumar

Audioholic
gene said:
I wish I could take the credit, but we hired on a new writer whose knowledge in these areas far exceed my own.
Sorry about the mis-directed jab, Gene ... I stand corrected. :eek:

Thanks for posting Mark's bio, I expected it would be impressive.

We were recently involved in a discussion in Philosophers and Wisemen concerning our hearing ability and its inevitable degradation. After reading the article on the Physics of Hearing and noting that the threshold of pain is 120dB, I was wondering if there is a threshold of damage. :confused:
 
L

Leprkon

Audioholic General
Dumar said:
I was wondering if there is a threshold of damage. :confused:
damage is based on time and loudness. According to OSHA the maximum permissable limits without hearing damage are

|
Duration per day, hours | Sound level dBA slow response
______________________|_________________________________
|
8...........................| 90
6...........................| 92
4...........................| 95
3...........................| 97
2...........................| 100
1 1/2 .....................| 102
1...........................| 105
1/2 .......................| 110
1/4 or less.............| 115

Exposure to impulsive or impact
noise should not exceed 140 dB peak sound pressure level.
___________________________________________________________
Footnote(1) When the daily noise exposure is composed of two or
more periods of noise exposure of different levels, their combined
effect should be considered, rather than the individual effect of
each. If the sum of the following fractions: C(1)/T(1) + C(2)/T(2)
C(n)/T(n) exceeds unity, then, the mixed exposure should be
considered to exceed the limit value. Cn indicates the total time of
exposure at a specified noise level, and Tn indicates the total time
of exposure permitted at that level.
 

Dumar

Audioholic
Thanks

Thanks a lot for the info and links, you two.

My plan is to buy an SPL meter and monitor my listening levels to determine if I'm getting into the danger zone. I don't think I am, but want to make sure anyway ... and it would be nice to have the information as a reference baseline. I am aware that I need to keep the levels up to fully enjoy the sound, much louder than the wife would like. I tend to listen to my music when I'm alone so I don't feel like I'm driving her out of the house.

I believe music is a very personal indulgence anyway, don't you? :)
 
J

J Risch

Enthusiast
Regarding "Phase Distortion Audibility"

In your latest article Human Hearing, Part 2, Phase Distortion, at:
http://www.audioholics.com/techtips/roomacoustics/HumanHearingPhaseDistortio.php

there are some questionable presentations of technical information.

The very first figure on page one (marked via hand notation "C"), the one showing the waveform distortion of what appears to be a square wave after it has been run through a 4th order Linkwitz-Riley crossover, has several misleading and omitted details.

First, it should clearly say that the bottom trace is the SUMMED output of the crossover outputs, that is, the summation of the high pass and the low pass outputs of the 4th order L-R crossover. Yes, the notation on the diagram indicates this to a professional, but not clearly for very many layman. It should be spelled out in the text.

Second, the waveform is NOT a portion of a square wave, even though many people would look at it and think that it was, it is actually two long duration "pulses", of unknown duration, separated by a similar amount of time as the pulse duration. The clue to this, is that the pulses are rising up from the baseline of the waveform, but never go below the baseline.
(The spectrum analysis also supports this, as the spectral pattern indicates raised pulses, rather than a portion of a square wave)

If the time scale is the same as for the 2nd figure, which is not an unreasonable assumption given the way they are presented, then it would appear that the duration of these pulses is 10 mS.

We also do not know what frequency the L-R crossover is set to, although it would appear to be some where higher in frequency than the apparent period of the waveform shown.

One could also look at these as if they were a sequence of 1 and 1/2 square wave cycles, that have been offset from the baseline of 0 volts by 1/2 the peak to peak value of the square wave.

Third, from looking at this kind of result, one could erroneously draw the conclusion that a 4th order L-R crossover has no difference between the steady state waveform distortion and the transient waveform distortion, while in fact, there IS a difference for these two conditions.

The bottom waveform shown in that first figure does not appear to be entirely correct, in that it does not show the transient distortion due to the start and stop of the pulse train (or off-set squarewave).

This transient distortion can be clearly seen at the very start or final discontinuation of a more or less steady square wave signal, as having a different appearance once the initial or final waveform transient has occured, and one is within the "steady-state" portion of the waveform signal, which would be all wave transitions within the confines of the waveform excluding those first and last transitions.

For an illustration of this, see:
"A Novel approach to Linear Phase Loudspeakers Using Passive Crossover Networks", Erik Baekgaard, JAES Vol 25, No 5 (1977, May), specifically Fig. 14.

Now it may be that the pulses will not show this to the fullest, and it would take a true square wave with the transistions going both above and below the 0 volt baseline in order to fully manifest, but the point is, that typical audio signals will exhibit this kind of transient behavior, rather than the totally asymmetrical pulse behavior shown.

Perhaps this is nit-picking, but I find it relevant in the way this data is presented, along with the various comments on the published articles, in that it appears to be presented from the point of view that phase distortion does not matter for typical musical material, rather than as an exploration of wheteher or not this is so. In other words, the editor and/or author had already made up his mind, and was presenting the data solely in a way to back up his POV.

Otherwise, why go to the trouble to use such an unusual waveform, rather than a more typical square wave?

The main reason that it seems to me that the article and data are presented with a twist to them, is the inclusion of selected comments by Dr. Floyd Toole, which I presume was the work of Gene as the editor. Dr. Toole is quoted in such a way as to minimize the importance of phase shift audibility:
Quote: "It turns out that, within very generous tolerances, humans are insensitive to phase shifts."

Now if one reads the original papers, and the conclusions of the original author(s), and even reads the writings of Mr. Sanfilipo, without these added comments by Toole, one could easily come to the conclusion that there were indeed instances where phase shift was audible, and that maintaining correct phase relationships seems to be a desirable thing to do, as long as doing so does not severly degrade other parameters.

But the added comments tend to muddy those waters more than they really should.

I feel that correcting and clarifiying the diagrams on the first page might help matters some, as well as including less added comments by Dr. Toole, without the counter-balance of another POV present, including those of the original authors of the referenced papers.
It should be noted for the record that virtually none of the speakers that Dr. Toole is associated with in the form of Harmon International, maintain correct phase relationships across the audio spectrum, inso much as it is possble to do so given the typical bandwidth limitations of loudspeaker systems. This might be a potential reason for him to minimize the importance of phase distortion.

Finally, I would like to point out that the final conclusion #3 at the end of the article is really only signficantly true for a room with no room acoustics treatment. In an acoustically well treated room, phase distortion, as well as envelope distortion, is more readily observed and noticed.

Jon Risch
 
M

MarkS

Audioholics Staff Writer
Human Hearing, Part II

Jon:

You’ve raised some very good points here, particularly where it comes to items such as notation.

One of the challenges faced when writing for an online forum that serves as the "town hall" for such a large and skills-wise, diverse, readership is making good choices where it comes to deciding to what degree the technical depth of an article should go.

Essentially, there exist two competing priorities: simplify too much the concepts being presented and you run the risk of trivializing or otherwise obscuring useful information. On the other hand, complicate it with too much technical minutiae and run the risk of confusing readers who don’t have a technical background.

Audioholics counts amongst the reader populace large numbers of professionals, as well as enthusiasts. Indeed, Audioholics enjoys a good deal of popularity with audio/video enthusiasts who arrive at the website with varying levels of technical skills and understanding. One of the many benefits enjoyed by the audio/video enthusiasts who do populate the forum is that they can directly query authors regarding the topics they have written about. As a staff writer I welcome their questions.

Having said all that, let’s look at the other points you’ve raised.

Good point about the “LR4 LP + HP.” I’m so used to the notation I hadn’t thought of elaborating. Though I do have to say, the Linkwitz-Riley 4th order(LR4) crossover is so well known today that I don’t think too too many people who have either a keen avocational or professional interest in audio wouldn’t have recognized “LR4”. At least, I’ve not yet gotten any questions from any reader asking for clarification!

“LP + HP, being addition means a sum and the “100Hz” indicates the crossover frequency. Once again, the notation is so familiar to me I hadn’t thought of elaborating. I’ve not yet gotten any questions about from any readers about it, either.

The waveform illustrated is indeed a square wave. The square wave source was an HP 33120A arbitrary waveform generator. The input and output signal measurements were taken with an HP 54616B 500 MHz digital scope and the graphics were taken from the 54616B. Pretty straightforward and almost trivial a task for the 33120A to create the 50 Hz square wave signal (given its 100 μHz to 15 MHz bandwidth) and even more trivial for the 54616B, given its DC to 500 MHz bandwidth. What marketing department wouldn't love to have loudspeakers with those specs!

The graphic presented in the lower portion of the 2nd graphic is indeed the FFT of the 50 Hz squarewave, passed through the LR filter. Nothing more or less.

You are correct about start/stop transients and how they would appear. However, the graphics showing here are but a sample, obviously taken from, as you call it, the “steady-state portion of the waveform signal.” I am mystified as to why you consider the waveform “does not appear to be entirely correct”. At no point in my article did I say this graphic illustrated a start/stop point for the entire signal.

Last, in this article, I’ve taken the position that phase distortion requires some very specific circumstances and conditions before it can become audible. As this is an open forum, this is but one of many points of view. I (and no doubt other readers) welcome other points of view, especially when presented with a foundation qualitatively substantiated by scientific research as well as opinion, when based on repeatable personal experiences. Its from the clamour of exchanging ideas that we all learn.

Mark
 
J

J Risch

Enthusiast
RE: Phase Distortion

MarkS said:
Jon:

You’ve raised some very good points here, particularly where it comes to items such as notation.

One of the challenges faced when writing for an online forum that serves as the "town hall" for such a large and skills-wise, diverse, readership is making good choices where it comes to deciding to what degree the technical depth of an article should go.

Essentially, there exist two competing priorities: simplify too much the concepts being presented and you run the risk of trivializing or otherwise obscuring useful information. On the other hand, complicate it with too much technical minutiae and run the risk of confusing readers who don’t have a technical background.

Audioholics counts amongst the reader populace large numbers of professionals, as well as enthusiasts. Indeed, Audioholics enjoys a good deal of popularity with audio/video enthusiasts who arrive at the website with varying levels of technical skills and understanding. One of the many benefits enjoyed by the audio/video enthusiasts who do populate the forum is that they can directly query authors regarding the topics they have written about. As a staff writer I welcome their questions.

Having said all that, let’s look at the other points you’ve raised.

Good point about the “LR4 LP + HP.” I’m so used to the notation I hadn’t thought of elaborating. Though I do have to say, the Linkwitz-Riley 4th order(LR4) crossover is so well known today that I don’t think too too many people who have either a keen avocational or professional interest in audio wouldn’t have recognized “LR4”. At least, I’ve not yet gotten any questions from any reader asking for clarification!

“LP + HP, being addition means a sum and the “100Hz” indicates the crossover frequency. Once again, the notation is so familiar to me I hadn’t thought of elaborating. I’ve not yet gotten any questions about from any readers about it, either.
I think that my post was an adequate example of some points that needed clairification, if nothing else, for all concerned, for professionals, techies, and layman. For instance, is the waveform or the L-R filter at 100Hz? As I noted in my first post, I can tell the L-R filter is set above the period (frequency) of the waveform as implied, but only because I have seen these kinds of displays many times, I think that a great many folks would assume that both the waveform period and the filter center for the L-R filter would be at the same frequency.

MarkS said:
The waveform illustrated is indeed a square wave.
Neither the waveform as shown, or the spectrum analysis, show a square wave, they show a series of pulses (or the alternative way of looking at it that I stated in my first post). The fact that the waveform rises from the baseline, and never goes below it is the key to understanding this. A square wave would have an equal amount of waveform content above and below the baseline.

MarkS said:
The graphic presented in the lower portion of the 2nd graphic is indeed the FFT of the 50 Hz squarewave, passed through the LR filter. Nothing more or less.
As I noted, it is not entirely clear, and it should be clearly annoted or marked, or an accompanyng explanatory sentence or two along with it. How can it confuse to spell out the details? Those who sorta know what is going on will then have a better idea, and those already lost will not be any more lost.

MarkS said:
You are correct about start/stop transients and how they would appear. However, the graphics showing here are but a sample, obviously taken from, as you call it, the “steady-state portion of the waveform signal.” I am mystified as to why you consider the waveform “does not appear to be entirely correct”. At no point in my article did I say this graphic illustrated a start/stop point for the entire signal.
The waveform, as shown, shows a start, and a stop. In other words, it is not continuous. Yet the output of the summed L-R filter shows a waveform that belongs to a steady-state waveform. In my opinion, without the proper labeling, it is implied that the summed L-R output is the result of the waveform show as the input. I say the output waveform is not correct, because the correct summed output waveform is not what is shown relative to what is shown as the input.

MarkS said:
Last, in this article, I’ve taken the position that phase distortion requires some very specific circumstances and conditions before it can become audible. As this is an open forum, this is but one of many points of view. I (and no doubt other readers) welcome other points of view, especially when presented with a foundation qualitatively substantiated by scientific research as well as opinion, when based on repeatable personal experiences. Its from the clamour of exchanging ideas that we all learn.
I think that your stance is not unreasonable, given the articles cited, and the ensuing analysis. You did not overemphasize the "inaudibility" aspect very much, and presented enough of the original articles data for folks to draw their own conclusions if they wished. However, the comments from Dr. Toole inserted by Gene seem to be much more definite about phase distortion being 'inaudible", while not really presenting any further solid data, and without the comments from the original authors, or even the contents of THEIR original article conclusions, there is no counterbalance of differing opinion present.

You mention experience, and scientific research. Having been a pro loudspeaker system designer for over 18 years, and a life-long audiophile/music lover, I have conducted many of my own experiments, as well as confirmed for myself many of the technical writings on the subject of loudspeaker system design. I have made the study of loudspeaker crossovers my special focus, and have determined through listening tests, as well as various means of technical analysis, that crossovers of an order greater than the 4th order L-R are audible on music and to the average person in most cases. I have also found that many music industry professionals can hear the presence of a 4th order L-R filter, compared to one with less "phase distortion". Does this mean that they are noticable by "dead people and industry reps"? Probably not, but given the current limitations of many pro sound reinforcement systems as a whole, and the limitations of our (still) early digital sound equipment, it is not inconceivable that minimizing "phase distortion" without incurring other parameter degradations, might yield a noticable and desirable improvement.

My own studies and experiments tend to support this, as when every effort is made to preserve overall waveform fidelity (not just amplitude accuracy), a great many things are more noticable than when the waveform is treated with a cavelier attitude, as if it were indestructible.

Note that home playback situations may actually be at a severe disadvantage compared to live sound situations, see:
http://www.audioasylum.com/audio/cables/messages/30013.html

I thank you for responding to my post, and hope to see the article tidied up a bit to further clarify the presentation. I doubt that Gene will remove or alter Toole's comments, which is a shame, as they do tend to slant the article much more than anything you wrote.

Jon Risch
(Senior Project Engineer,
Transducer Engineer Dept.,
Peavey Electronics Corp.)
 
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J Risch said:
I thank you for responding to my post, and hope to see the article tidied up a bit to further clarify the presentation. I doubt that Gene will remove or alter Toole's comments, which is a shame, as they do tend to slant the article much more than anything you wrote.
Presumptuous. Absolutely presumptuous. Of course we won't remove Dr. Toole's comments. Sometimes I just don't understand you, Jon. This is definitely a different crowd then you are apparently used to.

As for "tidying it up" - this is industry-standard notation and you should know that. The only person questioning this recognized notation so far is yourself. Nothing wrong with that, but I wanted to explain why the article isn't being "improved" per your requests.
 
J

J Risch

Enthusiast
Not presumption, experience

Clint DeBoer said:
Presumptuous. Absolutely presumptuous. Of course we won't remove Dr. Toole's comments. Sometimes I just don't understand you, Jon. This is definitely a different crowd then you are apparently used to.

As for "tidying it up" - this is industry-standard notation and you should know that. The only person questioning this recognized notation so far is yourself. Nothing wrong with that, but I wanted to explain why the article isn't being "improved" per your requests.
Hi Clint.
RE Dr. Toole's comments first, his comments are presented in addition to the article as authored by Mark Sanfilipo. Dr. Toole's comments state a conclusion that does not necessarily coincide with that of the author's of the original papers anaylyzed in Mark's summary and critique, and none of the original paper's conclusions are presented, one would have to actually go and obtain a copy (How many people actually DO that? At least I have read all of them in the first place, and have at my disposal copies to look up).

So Dr. Toole's conclusions are presented without any of his own corraborating evidence, and thus accorded the status of an incontestable truth?

Obviously, I dissagree with the finality of his conclusions, having read the original articles cited and analyzed in Mark's article, AND having conducted my own investigations into the subject.

It may help to recall, that I design and voice loudspeakers for a living, and am very, very good at it, while cables are only a hobby for me.

RE the notation used in Mark's article, I stated my problems with that succinctly and precisely, without rancor or excessively confrontational language.
The points that I raised are valid and relevant, especialy for folks who are not familair with such displays (most of your readers).

Despite what is there for those who know how to read such displays, there are still some questions, even for an experienced professional, due to the lack of any further explanation, notations, or labels.

For instance, the implied labeling of the waveforms in the top graphic is incorrect, for the previously stated reasons. The top waveform has a start and stop transient (because it is not continuous), and the waveform below that is supposed to show the summed output [stated only as "LR4 (LP+HP)"] does not reflect the transient start and stop of the waveform correctly. It shows the stop and start of the waveform as behaving the same as the "steady state" portion. Thus, the minimum amount of labeling is actually incorrect (or the waveform is incorrect).

As I noted in a previous post, the transient start and stop of the waveform will evince a slightly different summed output waveform, and thus, the reality is that a 4th order Linkwitz-Riley crossover will respond with a summed output that is DIFFERENT for transients than for a steady state waveform. This can certainly be an important aspect of the subject, and was only partially dealt with in the articles that were analyzed by Mark S.

There is a simple "100 Hz" notation, but we do not know if that is the equivalent of the frequency of the waveform period, or the center frequency of the LR crossover. It is NOT clearly stated. As I noted in an earlier post, I could tell by examination that the frequency of the waveform is lower than the center frequency of the LR crossover, and Mark has stated that the waveform frequency is indeed 50Hz, and the crossover center is 100 Hz. But nowhere in the article is that spelled out.

Yes, this is a different crowd than I am used to, I typically work in a much more precise environment, with other transducer engineers, where precision and accuracy in technical matters is expected at all times. While much has been made of my stance on cables, and thus my credentials by certain isolated individuals, I am a professional in the field of audio, and was merely pointing out a few things that I thought could use some further clarification.

I felt that Mark took my posted comments well, and in the right frame of mind, and I respect him for that. If you wish to dismiss my comments as merely presumptuous, then fine, but I have a far greater background in this subject matter than even a great many of my peers in the industry, and was taking Mark's article, and it's presentation quite seriously, and posting in the same serious manner.

I still feel that the addition of Dr. Toole's comments adds an uneccessary slant to Mark's article, as well as the original papers he discusses, and thus are not called for. Perhaps if Dr. Toole's comments were just another posted comment, like everyone else has to do, then there would be a more even overall presentation. I realize that is not likely to occur, but I felt that it was worth mentioning for those who do go past just reading the article, and delve into the forum comments.

Jon Risch
 
mtrycrafts

mtrycrafts

Seriously, I have no life.
Since this thread is ongoing, I re-read the original articles and the Daisuke Koya article discussed. This citation has been discussed elsewhere on the net in the past.
You may want to ask Mark S, the author to check the statistics used in the Koya paper, specifically the author's acceptance of only a 69% confidence level, unheard of in science. The minimum is 95% confidence level.
I would highly doubt his audibility conclusions based on such a piss poor confidence level.
 
M

MarkS

Audioholics Staff Writer
Article Graphics

In order to dispel any doubt that might arise when considering the graphics presented in the article - and the square wave data encapsulated within these - I thought it best to let the graphics (drawn from the website cited in the resource section of the article) original creator describe them.

Excerpted from private correspondence, the "SL" respondent is Siegfried Linkwitz, probably best known to Audioholics as the Linkwitz of "Linkwitz-Riley" crossover fame. Here, Siegfried offers some succinct responses as to the nature of the data presented. Hopefully this will clear up any doubts anyone may have regarding the graphics.

From Siegfried Linkwitz:

"Hi Mark,

Please see below my response to your questions.

Have fun,

-Siegfried Linkwitz

My questions center on the 3rd and 4th from the top graphics, where you illustrate the allpass 4th Order LR network and its effect on an input signal (Found in: 'http://www.linkwitzlab.com/phs-dist.htm')

MarkS: Is the input signal used an actual square wave or some sort of pulse? What is the frequency?

SL: It is a portion of a 50 Hz square wave as you can also see from the FFT of that signal in the bottom graph.

MarkS: Is the crossover frequency of the filter network considered by both graphs 100 Hz?

SL: Yes. The circuit which generated the outputs to the square wave is shown at the bottom. [Referring to a page in his website - Mark]

MarkS: What did you use to generate the data? Perhaps an HP (or Agilent) instrument?

SL: The square wave source was an HP 33120A arbitrary waveform generator. The input and output signal measurements were taken with an HP 54616B 500 MHz digital scope.

MarkS: Was the signal showing in each graphic the actual measured combined output of a woofer & midrange?

SL: It was the output from the circuit that is shown below. It simulates the allpass behavior of a B1, -B3, -LR2 and +LR4- crossover at 100 Hz.
[Referring to a page in his website - Mark]

MarkS
 

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