M

mjharris41

Audiophyte
Hi all, I recentely purchased the Rives Audio test CD 2 to help me position a new subwoofer, and after running it I found I had some pretty horendous peaks and valleys throughout the rest of the system. With correcting those in mind, a couple of questions:

-I had run the auto setup on my Yamaha RX-V2600, which includes paramatric EQ; given that I'm seeing farily dramatic peaks after the fact, does it make sense for me to try and correct this manually?

-Which type of decoding should I use with for the test CD? I've tried "straight effect," "music" and standard (defaults to pro-logic II). The source is an audio CD with a series of test tones.

-Can anyone suggest a good getting started guide? Right now I'm isolating each speaker and graphing its reponse through the series of test tones (from 20 Hz to 20kHz).

Thanks sincerely for the help!
 
mtrycrafts

mtrycrafts

Seriously, I have no life.
mjharris41 said:
Hi all, I recentely purchased the Rives Audio test CD 2 to help me position a new subwoofer, and after running it I found I had some pretty horendous peaks and valleys throughout the rest of the system. With correcting those in mind, a couple of questions:

-I had run the auto setup on my Yamaha RX-V2600, which includes paramatric EQ; given that I'm seeing farily dramatic peaks after the fact, does it make sense for me to try and correct this manually?

-Which type of decoding should I use with for the test CD? I've tried "straight effect," "music" and standard (defaults to pro-logic II). The source is an audio CD with a series of test tones.

-Can anyone suggest a good getting started guide? Right now I'm isolating each speaker and graphing its reponse through the series of test tones (from 20 Hz to 20kHz).

Thanks sincerely for the help!
A few things to consider.
Sine waves over drives and energize the room modes and don't think you will get an accurate picture of your real room acoustic issues. EQ makers like Audio Control and others always talk of pink noise signals and warble tones in the bass region or pink noise for them too.

Second, where are your biggest peaks and dips? You may need more EQ than the built in can manage.

Pink noise signals needs an Real Time Analyzer as all the frequencies are played at once, 20Hz-20kHz.

http://www.rane.com/pdf/old/thxeq.pdf
 
B

bpape

Audioholic Chief
Exactly. Take some measurements and don't smooth them. Use a noise signal or a warble for the measurements.

Play with sub/speaker/seating locations first prior to any EQ. You'll get much more benefit from that overall. Select the smoothest curve that has the fewest deep nulls.

Next, treat the room to assist with smoothing things out even more. If you still need some help after that, you can use a minor amount of EQ on the sub to knock down the last few remaining peaks.

Bryan
 
WmAx

WmAx

Audioholic Samurai
Mtrycrafts, I don't understand why it would be preferred to use pink noise for low frequency response analysis, for example. Perhaps one of you can explain this to me. A swept sine wave or impulse response analysis will produce dead accurate measurements of the final response(assuming a flat response measurement microphone and/or calibration). I don't know if very accurate pink noise analysis system is available to equal that of swept sine/impulse[ and if it did, I imagine that it would have to average over a relatively long period of time because of the random spectral structure of pink noise at any given instance ], but what is the real point of using pink noise for low frequency analysis, except for coarse averaged measurements? Is the supposed advantage to use a seperate non-correlated stimulus signals in order to try to remove the left and right speaker cancellation from the measurements(I don't know if this is done, just wondering), and try to analyze only room placement/mode behaviour(s)? If so, one can analyze one channel at a time using sine wave/impluse response analysis. But then, this is not allowing for the actual cancellations that will apply in real use, since the left/right speaker cancellations are a part of real use. I am not familar with warble tones, as I have not had a specific use for them. But I usually see it stated that warble tone measurement is intended to remove/reduce room modal response behaviour from the measurement(s). Since I have not investigated this form of measurement, I am not sure if this is an accurate summation, but if so, what would be the point, except for basic level setting purposes?

-Chris
 
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mtrycrafts

mtrycrafts

Seriously, I have no life.
WmAx said:
Mtrycrafts, I don't understand why it would be preferred to use pink noise for low frequency response analysis, for example. Perhaps one of you can explain this to me. A swept sine wave or impulse response analysis will produce dead accurate measurements of the final response(assuming a flat response measurement microphone and/or calibration). I don't know if very accurate pink noise analysis system is available to equal that of swept sine/impulse[ and if it did, I imagine that it would have to average over a relatively long period of time because of the random spectral structure of pink noise at any given instance ], but what is the real point of using pink noise for low frequency analysis, except for coarse averaged measurements? Is the supposed advantage to use a seperate non-correlated stimulus signals in order to try to remove the left and right speaker cancellation from the measurements(I don't know if this is done, just wondering), and try to analyze only room placement/mode behaviour(s)? If so, one can analyze one channel at a time using sine wave/impluse response analysis. But then, this is not allowing for the actual cancellations that will apply in real use, since the left/right speaker cancellations are a part of real use. I am not familar with warble tones, as I have not had a specific use for them. But I usually see it stated that warble tone measurement is intended to remove/reduce room modal response behaviour from the measurement(s). Since I have not investigated this form of measurement, I am not sure if this is an accurate summation, but if so, what would be the point, except for basic level setting purposes?

-Chris
Maybe I am misunderstanding the Rane paper on EQ then:

http://www.rane.com/pdf/old/thxeq.pdf

It talks about equal energy per octave closely reflecting our psychoacoustic expectation? Perhaps that is not the case at the low band?
How would single tone, long duration sine waves not cause false responses?
I compared an RTA from a 1/3 octave EQ job, flat, 20-20k, and using that 1/3 octave Rives continuous sine wave CD with no comparison. The CD was all over the place but flat. So which one is closer to reality then?
 
WmAx

WmAx

Audioholic Samurai
A sine wave sweep or FFT analysis of high resolution on an impulse response(or impulse derived from MLSSA signal) is absolutely accurate. BTW, an impulse response analysis is calculated from the time domain and frequency domain, and it will yeild the same frequency response measurement as a sine wave sweep. As for psycho-acoustic interpretations, as you know, we tend to most closely hear things in a 1/3 octave resolution equivalent. You can apply any smoothing algorithm you want to the sine wave/impulse result to achieve this averaging, if required. As for pink noise, it is great for level setting, and fine for low precision measurements. The paper you just referenced specifically referred to using pink noise for low precision 1/3 octave measurements and the paper stated that to get accurate measurements, you need to average measurements over a time period due to the random structure of the signal. I referred to these issues in my previous reply. Such low precision measurements are useless for analysis of a room in order to find and correct many low frequency problems. Now we circle back to hearing in 1/3 octave increments, roughly. So, the 1/3 octave analysis approximates how we would hear something in the frequency vs. amplitude domain, but you can not analyze and determine problems from such a low resolution analysis. This also does not address time domain problems, which are a separate entity. I have only questions, not answers, in regards to LF measurements vs. audibility. I am not aware of how humans relate to LF delayed resonances(which seem to be the case here more so than just frequency response concerns), or even non-delayed minimum phase LF resonances, as I am not familiar with credible perceptual research on this issue. Toole performed extensive research on resonance audibility in the late 80's, but the resonance research was not carried out at low frequencies, and in addition, his research dealt with minimum phase non-delayed resonances. At higher bass frequencies(>80 Hz), a normal size room may tend to support non-minimum phase delayed resonances, which only the frequency response would not be sufficient to analyze[ though it's only fair to point out that measurement/quantification of specific time domain/resonances in this band can not be correlated to audibility via perceptual research, since none exists, at least not of which I am aware; therefor it's usually just guesswork/speculation when relating said resonance measurements to audibility]. BTW, if you are aware of credible perceptual research dealing with LF resonances, I would very much like these references.

-Chris
 
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mtrycrafts

mtrycrafts

Seriously, I have no life.
WmAx said:
A sine wave sweep or FFT analysis of high resolution on an impulse response(or impulse derived from MLSSA signal) is absolutely accurate. BTW, an impulse response analysis is calculated from the time domain and frequency domain, and it will yeild the same frequency response measurement as a sine wave sweep. As for psycho-acoustic interpretations, as you know, we tend to most closely hear things in a 1/3 octave resolution equivalent. You can apply any smoothing algorithm you want to the sine wave/impulse result to achieve this averaging, if required. As for pink noise, it is great for level setting, and fine for low precision measurements. The paper you just referenced specifically referred to using pink noise for low precision 1/3 octave measurements and the paper stated that to get accurate measurements, you need to average measurements over a time period due to the random structure of the signal. I referred to these issues in my previous reply. Such low precision measurements are useless for analysis of a room in order to find and correct many low frequency problems. Now we circle back to hearing in 1/3 octave increments, roughly. So, the 1/3 octave analysis approximates how we would hear something in the frequency vs. amplitude domain, but you can not analyze and determine problems from such a low resolution analysis. This also does not address time domain problems, which are a separate entity. I have only questions, not answers, in regards to LF measurements vs. audibility. I am not aware of how humans relate to LF delayed resonances(which seem to be the case here more so than just frequency response concerns), or even non-delayed minimum phase LF resonances, as I am not familiar with credible perceptual research on this issue. Toole performed extensive research on resonance audibility in the late 80's, but the resonance research was not carried out at low frequencies, and in addition, his research dealt with minimum phase non-delayed resonances. At higher bass frequencies(>80 Hz), a normal size room may tend to support non-minimum phase delayed resonances, which only the frequency response would not be sufficient to analyze[ though it's only fair to point out that measurement/quantification of specific time domain/resonances in this band can not be correlated to audibility via perceptual research, since none exists, at least not of which I am aware; therefor it's usually just guesswork/speculation when relating said resonance measurements to audibility]. BTW, if you are aware of credible perceptual research dealing with LF resonances, I would very much like these references.

-Chris

OK. But, going back to that Rives audio CD with ordinary single tone, 1/3 octave sine waves and spl meter would not do the trick of getting the real room responses, or will it?

Impulse response is a very short duration signal of specific Hz, no? So would a sine wave sweep would be a short duration on each Hz, right? How is this differentiated from pink noise that randomly produces the different band signals Is the time duration that is different? Or, the power into each Hz over the entire band?

Maybe I just don't see how these different signals are produced, or perhaps the recording program is such that they measure for a very short time of a continuous signal sent to the speaker at whatever bandwidth the signal is?
I am here to expand on what I understand:D
 
WmAx

WmAx

Audioholic Samurai
mtrycrafts said:
OK. But, going back to that Rives audio CD with ordinary single tone, 1/3 octave sine waves and spl meter would not do the trick of getting the real room responses, or will it?
1/3 octave sine waves increments can not get a clear picture of room response. The increments are too coarse.
Impulse response is a very short duration signal of specific Hz, no?
Yes, it is a short duration instance signal. The key lies in the processing of this signal. The impulse is used as a stimulus, and it is recorded by a computer program. The program can then analyze this recording in order to calculate time and frequency domain behaviour.

So would a sine wave sweep would be a short duration on each Hz, right?
That depends entirely on the parameter settings used in the measurement software. I use a relatively long duration sweep in my measurements.

How is this differentiated from pink noise that randomly produces the different band signals Is the time duration that is different? Or, the power into each Hz over the entire band?
Pink noise varies the spectral distribution at any given instance. Eventually it repeats and covers all spectra over a long enough period of time. What is constant, is that it always produces a constant level of energy per octave. You have to analyze the signal for a long enough period to gather an accurate result. I don't know the specific time required for a specific level of accuracy so far as the methods used by the RTA hardware analyzers like you refer to using; after all, I asked you why pink noise would used instead of other forms or measurement for low frequency response.

Maybe I just don't see how these different signals are produced, or perhaps the recording program is such that they measure for a very short time of a continuous signal sent to the speaker at whatever bandwidth the signal is?
I am here to expand on what I understand:
I responded initially asking a question: Why is pink noise desired for low frequency system measurements? I still don't know the answer. I am certainly not an expert at acoustic measurements, and this one reason why I asked why it would be preferred. I have not noticed literature explaining why one would prefer to use this method for low frequency analysis over other possible methods. However, it is interesting to note that MLS is used to derive impulse response, and is a form of white noise(pink noise is simply filtered white noise) measurement, but I am not sure how this would apply to the situation being discussed at the moment, since this discussion seems to be about direct pink noise analysis as in what is used by the 1/3 octave analyzers and other coarse measurement systems, not a separate system/method based on the principle signal.

-Chris
 
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mtrycrafts

mtrycrafts

Seriously, I have no life.
WmAx said:
1/3 octave sine waves increments can not get a clear picture of room response. The increments are too coarse.
WmAx said:
This I know but it is easier to find if you don't have the right setup for a full frequency stimulus/


Yes, it is a short duration instance signal. The key lies in the processing of this signal. The impulse is used as a stimulus, and it is recorded by a computer program. The program can then analyze this recording in order to calculate time and frequency domain behaviour.

I thought it might be. Need a computer setup to use this whole process with speaker analysis.



That depends entirely on the parameter settings used in the measurement software. I use a relatively long duration sweep in my measurements.

Would it over energize the room modes, or is that what is needed for an accurate analysis? Or, just long enough without over pressurizing it?


I asked you why pink noise would used instead of other forms or measurement for low frequency response.

Ease of setting it up with minimal equipment for many here who would not invest in the software program for the proper MLSSA or impulse even though one might probably find it on line.


I responded initially asking a question: Why is pink noise desired for low frequency system measurements? I still don't know the answer.

It wouldn't be if one has the proper test software but that Rives CD is not the right disc as it has continuous sine waves test tones. I suppose the pink noise is better in that case?
 
T

tbewick

Senior Audioholic
I'm sorry to butt in on this discussion, but a book I'm looking at at the moment has some stuff on this. It's easier and quicker for me if I just type down the relevant quotes and not bother quoting from the thread:

"High resolution measurements in rooms include details of the numerous acoustical interferences caused by reflections within the boundaries. The resulting visible 'grass' may disguise underlying trends so it is common practice to reduce the resolution in the frequency domain by using, for example, 1/3-octave analysis or smoothing. Unfortunately, removing the visible clutter also removes some of the useful information becuse of the spectral averaging.

It is possible to suppress the effects of acoustical interference, while retaining high resolution in the frequency domain, by using spatial averaging: the energy average of data taken at several microphone locations within a designated listening area. With computer-based measurement systems, such measurements are straightforward, and several of these systems allow for the energy-averaging of multiple measurements. It should be noted that a simultaneous combination of the outputs of several microphones does not yield an equivalent result unless the outputs are individually rectified and filtered prior to the summation".

- Loudspeaker and Headphone Handbook, 2nd edn, edited by John Borwick, director of Gramophone magazine. Butterworth-Heinemann Ltd 1994. Chapter 11, 'Subjective evaluation', written by Floyd E. Toole (Harman International Industries Inc.). 11.2.5 'Acoustical measurements in rooms', p474.

This chapter goes on to say that "bass response should be evaluated with both woofers operating". (p476)

"For measurement of the reverberation time of listening rooms of modest dimensions, the resident loudspeaker systems can often be used to provide the sound source. As a rule, the loudspeaker system is driven with a pink noise signal which is filtered by an octave or third-octave wide filter centred on the frequency range of interest. Alternatively, a sine wave signal may be used (if one particular frequency is of interest) or the sine wave frequency can be modulated (or wobbled) to cover a certain range. Whichever signal is chosen, it must first be applied until the sound level in the room reaches a steady value before switching the signal off to observe the decay".

- As above, Chapter 7 'The room environment', written by Glyn Adams, revised by Colin Bean (B&W Loudspeakers). 7.4.1 'Measurement of reverberation time', p320.

"The measurement of 'frequency response' implies measurement of the variations of both amplitude and phase... but in room measurements it is normal to measure the amplitude/frequency response only.

In electrical engineering it is common practice to use a sine wave signal whose frequency is slowly varied to measure the frequency response; this method may also be applied here. However this method is not convenient for measuring the frequency response in rooms because the relatively long delay time of the room resonances requires an unusually slow sweep.

Because of this random nature of the music signal, the room modes, which take a finite time to build up and decay, are not excited as strongly as they are under steady-state conditions. To simulate this effect, the sine wave signal can be modulated or wobbled rapidly about the measurement frequency so that the room modes never become completely established. Alternatively, a random noise signal can be used to excite the sound source, and the microphone signal analysed with some form of frequency-response analyser.

The frequency analysis is most commonly achieved by filtering the microphone signal through a band-pass filter of octave or third-octave width with an adjustable centre frequency. Some instruments employ a multiplicity of such filters, with fixed centre frequencies to cover the entire audio range and so provide a 'real-time' display of the frequency response. These instruments are particularly valuable when investigating the variation of frequency response as a function of microphone position within the room. A pink noise excitation signal is normally employed in this type of measurement because its octave or third-octave frequency spectrum is flat".

- 7.4.2 'Measurement of frequency response', p322.
 
WmAx

WmAx

Audioholic Samurai
tbewick said:
" 1/3-octave analysis or smoothing. Unfortunately, removing the visible clutter also removes some of the useful information because of the spectral averaging.
Absolutely correct, as per my previous posts.


"For measurement of the reverberation time of listening rooms of modest dimensions, the resident loudspeaker systems can often be used to provide the sound source. As a rule, the loudspeaker system is driven with a pink noise signal which is filtered by an octave or third-octave wide filter centred on the frequency range of interest. Alternatively, a sine wave signal may be used (if one particular frequency is of interest) or the sine wave frequency can be modulated (or wobbled) to cover a certain range. Whichever signal is chosen, it must first be applied until the sound level in the room reaches a steady value before switching the signal off to observe the decay".
This article may be dated, because the preferable way to do LF analysis today is with long sequence MLS signals, calculated into impulse response, and then analyzed. Everything is present in this analysis.

In electrical engineering it is common practice to use a sine wave signal whose frequency is slowly varied to measure the frequency response; this method may also be applied here. However this method is not convenient for measuring the frequency response in rooms because the relatively long delay time of the room resonances requires an unusually slow sweep.
I don't know what they mean by relatively long delay. One can, for example, sweep 20-100Hz in a 15'x14'x7.5' average room, in the center, and a 0.5 second sweep time will not yield much less information than a 5 second swee[in fact, nearly identical, except at the lowest frequencies, around 20Hz and below, where some differences can be noted]. You can reduce the sweep to something absurd, that is not even enough time to get proper samples, like 0.05 seconds, and then you can observe the effects of too short of sweep(room modes not being picked up accurately in the response and not sufficient time for analysis of lower frequencies based on their time length, for example).

The frequency analysis is most commonly achieved by filtering the microphone signal through a band-pass filter of octave or third-octave width with an adjustable centre frequency. Some instruments employ a multiplicity of such filters, with fixed centre frequencies to cover the entire audio range and so provide a 'real-time' display of the frequency response. These instruments are particularly valuable when investigating the variation of frequency response as a function of microphone position within the room. A pink noise excitation signal is normally employed in this type of measurement because its octave or third-octave frequency spectrum is flat".
Right, and the resulting measurements are only useful for general tonal balance, not proper room analysis in order to accurately evaluate and correct modal problems.
-Chris
 
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WmAx

WmAx

Audioholic Samurai
mtrycrafts said:
Would it over energize the room modes, or is that what is needed for an accurate analysis? Or, just long enough without over pressurizing it?
Depends on what you are comparing the signal with. If you want to compare to music, which music? One can analyze the spectral and time distribution in music samples to draw a proper comparison of one to the other. But for the most part, much of the bass stimulus is pretty long in terms of excitation in music as compared to test signals; otherwise it would actually be pretty hard to hear it. I can, for example, look at high resolution FFT analysis of the CD (Valery Gergiev) - Dmtri Shostakovich Symphony No. 4 in C minor, op.43 - Dmtri Shostakovich Symphony No. 4 in C minor, op.43 II Moderato con moto, track 3(I know you have this CD, so it will be good for you to refer to for reference), and the bass signals last for several tenths of a second. Plenty of time to excite the room modes. Remember, 0.001 second is approximately equal to 1 foot of distance [roughly] travelled in air, for sound waves, on average. A 30 Hz wave is about 0.033 seconds in duration to complete one cycle. In several tenths of a second, it has had plenty of time to reach the boundaries, and then cycle through many times.

It wouldn't be if one has the proper test software but that Rives CD is not the right disc as it has continuous sine waves test tones. I suppose the pink noise is better in that case?
The only problem with that Rives disc is that I believe it has coarse increments. If it in fact has fine increments, then there is no problem with it, assuming one wants to take the time to plot LF response in 1 Hz increments.

-Chris
 
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mtrycrafts

mtrycrafts

Seriously, I have no life.
WmAx said:
Absolutely correct, as per my previous posts.
This article may be dated, because the preferable way to do LF analysis today is with long sequence MLS signals, calculated into impulse response, and then analyzed. Everything is present in this analysis.

I don't know what they mean by relatively long delay. One can, for example, sweep 20-100Hz in a 15'x14'x7.5' average room, in the center, and a 0.5 second sweep time will not yield much less information than a 5 second swee[in fact, nearly identical, except at the lowest frequencies, around 20Hz and below, where some differences can be noted]. You can reduce the sweep to something absurd, that is not even enough time to get proper samples, like 0.05 seconds, and then you can observe the effects of too short of sweep(room modes not being picked up accurately in the response and not sufficient time for analysis of lower frequencies based on their time length, for example).
Right, and the resulting measurements are only useful for general tonal balance, not proper room analysis in order to accurately evaluate and correct modal problems.
-Chris
You can see how confusing this can be with different info presented by the well known folks:D

From above post:
To simulate this effect, the sine wave signal can be modulated or wobbled rapidly about the measurement frequency so that the room modes never become completely established. Alternatively, a random noise signal can be used to excite the sound source, and the microphone signal analysed with some form of frequency-response analyser.



This would imply to me that using a continuous sine wave, single tone would perhaps over energize the modes, if that is possible? And, that is not a desired way to measure.
But, pulsed tones, those are short enough and long enough to do the right job then.
No wonder I am scratching head :confused:

Thanks for all the info.
 
T

tbewick

Senior Audioholic
"The paper you just referenced specifically referred to using pink noise for low precision 1/3 octave measurements and the paper stated that to get accurate measurements, you need to average measurements over a time period due to the random structure of the signal. I referred to these issues in my previous reply. Such low precision measurements are useless for analysis of a room in order to find and correct many low frequency problems." - WmAx

I think the problem was that the THX/Rane manual is aiming for:

"Remember that we are trying to achieve a response in the LCR channels of ±1 to 2 dB from 100 Hz to 1 kHZ without drastic EQ shifts."

and -

"When equalizing the Subwoofer Channel, you should concentrate on reducing the serious peaks. You may find that because of the depth of the room modes a ruler flat response is not within the range of the equalizer. This not a major concern since a response within ± 3 or 4 dB is very acceptable."

Perhaps then the use of pink noise is done with the knowledge that accurate low frequency equalisation is not required. Considering the accuracy and performance of most domestic subwoofers, this is probably a perfectly fair thing to do.

The book I referred to earlier does make the point that a flat response is not necessarily desirable in a normal domestic, or even professional, set up. This is because rooms with low reverberation times in the mid-range and treble aren't very pleasant to work in. Therefore the bass increase which is normally present in normal living rooms is in some ways a good thing, as it helps to hide treble/mid-range reverberation. In a real musical performance however, at a good venue, the sound field would be diffuse at all frequencies. It does go on to say that many professional studios do take measures to make reduce large peaks in low frequency reverberation.

As for the psychoacoustics of pink noise, mentioned earlier:

"Research done a decade earlier by C.P. and C.R. Boner defined a need for a "house curve". They based this on the fact that a flat electro-acoustic frequency response in a large room sounds too bright on well-balanced program material. Or in simpler phrase, we perceive sound in a large room to have more treble."

- http://www.hometheaterhifi.com/volume_9_2/feature-article-curves-6-2002.html

This is with regard to the 'x-curve' used in cinemas.

"The measurement of EQ is done using wideband pink noise excitation through the speaker. The larger the room, the longer and greater the reverberation buildup over time, resulting from this steady-state signal. If we measure EQ on a fixed bandwidth basis, like full octaves or third octaves, the reverb which tends to be stronger at low frequencies will tend to make the SPL read louder there than at the higher frequencies where it is better absorbed. By knowing how much the meter reading is influenced by this effect, a curve can be used to compensate."

Maybe this is why the THX/Rane manual says -

"Because of this perception of flat tonal balance, pink noise is a very useful tool when using a spectrum analyzer with 1/3 octave or octave measurement intervals, and when comparing loudspeakers for spectral similarity by ear".

As Chris said earlier, the key thing here with perceived tonal balance is only useful for loudspeaker comparision by ear. It's all a bit confusing really and not very important.
 
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