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Genelec on audio science

NorthSky

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Blush! Yes, of course, ASR. I participate in both and my aging mind occasionally gets confused. Apologies! I appreciate a good discussion, wherever it originates. There is so much uninformed rubbish out there, it can be depressing.

Speaking of depression, a new scientific report has just been released:
https://www.bustle.com/p/depression...number-is-even-higher-for-millennials-9047775

Very recent news report, from yesterday in fact:
http://time.com/5271244/major-depression-diagnosis-spike/
 
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pirad

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I am wondering about this steady state...If auditory experience is mostly about transients how to measure and describe it best?
 

Soniclife

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I am wondering about this steady state...If auditory experience is mostly about transients how to measure and describe it best?
I was wondering the same, is this easy to do at home?
 

Cosmik

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I am wondering about this steady state...If auditory experience is mostly about transients how to measure and describe it best?
Steady state measurements make perfect sense when attempting to detect nonlinearities, or basic issues with amplifiers and DACs, etc. They don't map very well onto acoustics and hearing. This doesn't stop people from using them as the primary means of attempting to understand acoustics and hearing, though!
 

pirad

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I was wondering the same, is this easy to do at home?
It's easy to measure. Take REW free app and Minidsp USB mic, sweep with window 500ms.
But what does it tell us about real music with transients played on the system in this room?
 
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JohnPM

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It's easy to measure. Take REW free app and Minidsp USB mic, sweep with window 500ms.
But what does it tell us about real music with transients played on the system in this room?
There may be some differences in interpretation of 'steady state'. In this context it does not mean information about transient behaviour is omitted - the window you mention is being applied to the impulse response, which is a complete description of the behaviour of the system and by definition the response to a transient, hard to get more transient than an impulse :). Using a broad window includes the contributions of the environment in which the measurement was made, including the various reflections from surfaces. Since those have reduced HF content in most environments the overall response has a downward tilt. Using a progressively narrower window gradually removes those environmental contributions, eventually leaving only the direct response of the speaker once the window is narrow enough to exclude all reflections, at the expense of reducing frequency resolution. That does become less representative of perception of the sound at LF however, since the ear's integration time is longer there and the room contributions down there do not get filtered out by the brain's processing. You will find a frequency-dependent window option in REW which applies a window which becomes narrower as frequency increases, which comes a little closer to perception.
 
OP
svart-hvitt

svart-hvitt

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TOOLE WARMLY RECOMMENDED BY GENELEC'S MARTIKAINEN

When confronted with an uninformed reader on Genelec's forum pages in 2015, Ilpo Martikainen (1947-2017), Genelec's founder, recommended Toole's "Sound Reproduction: The Acoustics and Psychoacoustics of Loudspeakers and Rooms" to replace confusion with insight:

Quote:
"We are most interested in how the speaker performs, i.e. sounds. This includes lots of listening tests as well. However, there is lot of research evidence of what makes the speaker sound good and how this correlates with measured performance, i.e. specs. I warmly recommend Floyd Toole’s book “Sound Reproduction: The Acoustics and Psychoacoustics of Loudspeakers and Rooms”, it is an excellent representation of this subject".
Source: https://www.community.genelec.com/forum/-/message_boards/message/914379

I recommend reading all of Ilpo's response in the link above because it gives brief insights into the thinking at Genelec, how they combine science, listening tests and speaker production.

We have recently had discussions on the internet on who makes Genelec's drivers. According to internet rumors, all of Genelec's drivers are made in China. Martikainen cast light on this issue thus:

"The midrange driver is designed and manufactured in house since 1988, as there was simply not good enough midrange drivers available. Its performance and design processes were reported in AES preprint 2755, including sensitivity, distortion, power handling and compression with high levels. The claim that midrange could not handle much power is wrong. The driver was tested up to 1 kW power to make sure it is mechanically stable and reliable. The reliability track record of this driver is extremely good. It was first used in the largest monitor 1035A and since then it is used in all 3-way models down to 1037B".

On the issue of hifi speakers vs pro audio, and production principles in general, he wrote:

"I may not be aware what these rules may be, but the 1238A is designed as a professional monitoring speaker, where requirements are more stringent than in most hi-fi speakers. For example, monitoring speaker shall be extremely neutral, linear, reliable, serviceable, manufactured with very tight tolerances between units and batches over the years. Any speaker of same model may be paired with any other sample, even made years apart. As the “sound” of the recording is adjusted by listening it with the monitoring speaker, the monitoring speaker must be most revealing. Actually the monitoring speaker has to be more revealing than anything else in the reproduction chain after that. This ensures that end users will not find any surprises".

2017014963989.jpg
 
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pirad

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There may be some differences in interpretation of 'steady state'. In this context it does not mean information about transient behaviour is omitted - the window you mention is being applied to the impulse response, which is a complete description of the behaviour of the system and by definition the response to a transient, hard to get more transient than an impulse :). Using a broad window includes the contributions of the environment in which the measurement was made, including the various reflections from surfaces. Since those have reduced HF content in most environments the overall response has a downward tilt. Using a progressively narrower window gradually removes those environmental contributions, eventually leaving only the direct response of the speaker once the window is narrow enough to exclude all reflections, at the expense of reducing frequency resolution. That does become less representative of perception of the sound at LF however, since the ear's integration time is longer there and the room contributions down there do not get filtered out by the brain's processing. You will find a frequency-dependent window option in REW which applies a window which becomes narrower as frequency increases, which comes a little closer to perception.
I am not an engineer and my math education finished way before Fourier transforms, so I am seriously handicapped here. Please correct me if I go wrong somewhere. I understand steady-state as a kind of equilibrium reached by the accumulation of energy from direct sound and reflections. Toole (3rd edition 2016, p.383) recommends at least 500ms measurement window to allow this energy accumulation. The sweep is pitch based. Is the transient information somehow coded in this sweep or interplolated in analysis?
TIA for explanations.
 

JohnPM

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This may help to explain the underlying concepts, transfer function and impulse response are covered in it. Many types of signal can be used to determine the impulse response or transfer function, which describe the behaviour of the system in time domain and frequency domain respectively. To be most effective the signal should cover the full range of frequencies over which we wish to understand its behaviour. Measuring how the system changes each of those frequencies in amplitude and phase lets us determine how the system will respond to any signal.

To show the 'steady state' the behaviour of the system up to the point the impulse response has decayed into the noise floor is analysed - in a domestic room that takes around 300-500 ms or so, in a concert hall it can take many seconds. It means analysing a long enough portion of the impulse response that its contribution can no longer be distinguished from the noise in the measurement system. One could equally view that as waiting long enough that the transient effects of applying a signal have decayed into the noise. I think the terminology has a lot of scope to confuse, which this discussion seems to support :). An alternative might be to refer to the total contribution of speaker and room at the measurement position versus the speaker's contribution alone.
 

RayDunzl

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Is the transient information somehow coded in this sweep or interplolated in analysis?
TIA for explanations.

I can't explain it and didn't believe it.

So I performed an experiment.

In-room Sine sweep (nothing seems transient about it) generated and captured by REW with calculated impulse and step response in REW displayed, vs the in-room audio recording of an actual impulse and step.

To my lasting surprise and befuddlement, the two results are essentially identical.

https://www.audiosciencereview.com/forum/index.php?threads/impulse-response.1765/
 
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Frank Dernie

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I agree that ported designs are a compromise but all of the monitor designs we considered were ported -- even the JBL M2 is ported. Perhaps that is fashion rather than science at work.

I have both ported and sealed speakers for my home hi-fi. These particular sealed designs are passive and wonderfully coherent but restrained and anaemic. If I want an emotional charge, I will crank up the ported designs even if I know the bass is inaccurate. I have mixed music on both with satisfactory results, perhaps because I know their respective weaknesses.

Definitely science. Ported speakers have more bass extension then a more rapid roll off than closed boxes of the same size.
The bass from my (ported) speakers is as good as any bass reproduction I have ever heard.
Maybe I need to get out more.
I have Tune Audio Anima (horns), Goldmund Epilog 1&2 (ported) and Devialet Phantoms (small, sealed with DSP correction) in this room and Yamaha NS1000M (sealed) in my bedroom.
All have superb bass.
I am storing active Harbeth Model 40s (ported) for an acquaintance. Their bass is execrable, boomy flabby and disconnected, can't bear listening to them on any music with bass.
 

Floyd Toole

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I am wondering about this steady state...If auditory experience is mostly about transients how to measure and describe it best?
Steady state in-room measurements are unreliable indicators of sound quality. They cannot be used as a criterion of performance. Well-designed loudspeakers yield similar-looking steady-state room curves in normally reflective rooms, but equalizing poorly designed loudspeakers to have the same steady-state in-room curve guarantees nothing.

This is a good question, and the answer is in two parts:
1. The loudspeaker is, or should be, fully characterized by anechoic measurements. These include the time domain (transients). If the recording is an accurate rendering of the music, good loudspeakers will reproduce the music with sustained tones and transients intact. Loudspeaker transducers are minimum-phase devices so a flat smooth anechoic frequency response indicates a flawless transient response (within the frequency range of each transducer). Directivity must be included in those measurements because loudspeakers radiate a 3D sound field. In rooms most of the sound arriving at the ears is reflected sound, which is totally dominant at low frequencies, and progressively reduced at higher frequencies because of the frequency-dependent directivity of the loudspeakers. The downward tilt of the steady-state in-room curves is really better interpreted as a bass rise, as the loudspeaker radiates more of its sound in all directions as frequency falls/wavelengths increase.
2. The human binaural hearing system - see the illustration in post no. 40 - is not at all the same as a measurement mic and analyzer. In everyday conversation and at live, unamplified, music events, voices and musical instrument sounds are easily separated from within the sounds contributed by the venue. We simply hear those familiar voices and instruments in different spaces. It is therefore not surprising that humans are able to recognize good and bad attributes of loudspeakers in different rooms. We have considerable ability to adapt to our listening spaces, perceptually streaming the essence of the music (the loudspeaker) from that of the room. As with any adaptation process, there are limits, meaning that it is important to know what we can and cannot compensate for. What we do absolutely know is that listening in a totally reflection free environment such as an anechoic chamber is not pleasant. Likewise, an empty, reverberant room is intolerable. Between those extremes is a significant range within which we are able to function without stress and derive substantial pleasure. It is interesting that when tested in an elaborate listening lab in Germany, listeners were displeased when the floor reflection was removed. It seems that humans have adapted to having a solid reflecting surface under us.

Measurements within a room cannot interrogate the performance of a loudspeaker in more than a crude sense. We must start with a good sound source, and the manufacturer should provide us with anechoic on and off-axis data to confirm that this is the case. If that is done, then in normally reflective rooms, the rest is made much simpler. We don't feel it necessary to equalize voices and instruments for different performance spaces. Why should it be necessary to do so for loudspeakers reproducing those voices and instruments? Acoustically poor rooms can exist in both live and reproduced performances, and in neither case is equalization a solution - that is for acoustical treatments. If one starts with flawed loudspeakers, which many do, unfortunately, equalization may or may not be able to help, but the best data upon which to base such equalization is anechoic data, and if one had that, surely the best thing to do is to avoid purchasing such loudspeakers.

As emphasized in the book, low frequency misbehavior in small rooms is a huge problem but there are solutions. As Olive found, bass accounts for about 30% of our overall factor weighting in evaluating sound quality. It needs to be attended to.
 

Speedskater

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It seems to me that Dr. Toole's two books:

1] "Sound Reproduction" Loudspeakers and Rooms 2008

2] "Sound Reproduction" Third Edition
The Acoustics and Psychoacoustics of Loudspeakers and Rooms 2017

Are two different books with almost the same title.
 

Floyd Toole

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It seems to me that Dr. Toole's two books:

1] "Sound Reproduction" Loudspeakers and Rooms 2008

2] "Sound Reproduction" Third Edition
The Acoustics and Psychoacoustics of Loudspeakers and Rooms 2017

Are two different books with almost the same title.

The original 2008 book was released as a second edition when Focal Press was purchased by Tayor and Francis - it was their mistake. The new Third Edition is completely rewritten. It contains a lot of content from the earlier book, but necessarily some has been omitted (mostly esoteric psychoacoustic content) to make room for new material. There is also an open-access companion website with additional material (www.routledge.com/cw/toole).
 

oivavoi

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Steady state in-room measurements are unreliable indicators of sound quality. They cannot be used as a criterion of performance. Well-designed loudspeakers yield similar-looking steady-state room curves in normally reflective rooms, but equalizing poorly designed loudspeakers to have the same steady-state in-room curve guarantees nothing.

This is a good question, and the answer is in two parts:
1. The loudspeaker is, or should be, fully characterized by anechoic measurements. These include the time domain (transients). If the recording is an accurate rendering of the music, good loudspeakers will reproduce the music with sustained tones and transients intact. Loudspeaker transducers are minimum-phase devices so a flat smooth anechoic frequency response indicates a flawless transient response (within the frequency range of each transducer). Directivity must be included in those measurements because loudspeakers radiate a 3D sound field. In rooms most of the sound arriving at the ears is reflected sound, which is totally dominant at low frequencies, and progressively reduced at higher frequencies because of the frequency-dependent directivity of the loudspeakers. The downward tilt of the steady-state in-room curves is really better interpreted as a bass rise, as the loudspeaker radiates more of its sound in all directions as frequency falls/wavelengths increase.
2. The human binaural hearing system - see the illustration in post no. 40 - is not at all the same as a measurement mic and analyzer. In everyday conversation and at live, unamplified, music events, voices and musical instrument sounds are easily separated from within the sounds contributed by the venue. We simply hear those familiar voices and instruments in different spaces. It is therefore not surprising that humans are able to recognize good and bad attributes of loudspeakers in different rooms. We have considerable ability to adapt to our listening spaces, perceptually streaming the essence of the music (the loudspeaker) from that of the room. As with any adaptation process, there are limits, meaning that it is important to know what we can and cannot compensate for. What we do absolutely know is that listening in a totally reflection free environment such as an anechoic chamber is not pleasant. Likewise, an empty, reverberant room is intolerable. Between those extremes is a significant range within which we are able to function without stress and derive substantial pleasure. It is interesting that when tested in an elaborate listening lab in Germany, listeners were displeased when the floor reflection was removed. It seems that humans have adapted to having a solid reflecting surface under us.

Measurements within a room cannot interrogate the performance of a loudspeaker in more than a crude sense. We must start with a good sound source, and the manufacturer should provide us with anechoic on and off-axis data to confirm that this is the case. If that is done, then in normally reflective rooms, the rest is made much simpler. We don't feel it necessary to equalize voices and instruments for different performance spaces. Why should it be necessary to do so for loudspeakers reproducing those voices and instruments? Acoustically poor rooms can exist in both live and reproduced performances, and in neither case is equalization a solution - that is for acoustical treatments. If one starts with flawed loudspeakers, which many do, unfortunately, equalization may or may not be able to help, but the best data upon which to base such equalization is anechoic data, and if one had that, surely the best thing to do is to avoid purchasing such loudspeakers.

As emphasized in the book, low frequency misbehavior in small rooms is a huge problem but there are solutions. As Olive found, bass accounts for about 30% of our overall factor weighting in evaluating sound quality. It needs to be attended to.

Thank you, dr Toole! These are very interesting and enlightening points. I just want to comment on this one point: "We don't feel it necessary to equalize voices and instruments for different performance spaces". From the perspective of an active (amateur) musician, my experience is a bit different - if I understand you correctly, that is.

I have for some years been singing with a semi-professional choir, and we have performed in different kinds of acoustic spaces. I have also had my fair share of acoustic jazz jams (on piano). When singing with the choir, our conductor does in fact make sure that we adapt to the acoustics of the venue. I guess you could call it "equalization", in some sense. He is adamant that different kinds of dynamics and overall tonal balance is appropriate for different venues.

Acoustic instruments have also become tuned more brightly, because concert halls have become larger, and more treble energy gets lost on the way to listeners seated far away. Closed-mic'ed acoustic recordings therefore have a tendency to sound too bright to my ears.

My experience from jamming with other musicians is also that we more or less automatically adapt our playing to the acoustics of the room. Very bright rooms leads to another kind of playing than darker rooms, etc.

Does this have any bearing on the room eq discussion? Perhaps, perhaps not. My intuitive sense is that broadband adjustments of bass, treble and midrange may be useful if a room leans very much to one side acoustically (bright vs darker). In addition to equalization and other remedies in the bass, which you cover in an excellent way in your book!
 
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pirad

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Steady state in-room measurements are unreliable indicators of sound quality. They cannot be used as a criterion of performance. Well-designed loudspeakers yield similar-looking steady-state room curves in normally reflective rooms, but equalizing poorly designed loudspeakers to have the same steady-state in-room curve guarantees nothing.

This is a good question, and the answer is in two parts:
1. The loudspeaker is, or should be, fully characterized by anechoic measurements. These include the time domain (transients). If the recording is an accurate rendering of the music, good loudspeakers will reproduce the music with sustained tones and transients intact. Loudspeaker transducers are minimum-phase devices so a flat smooth anechoic frequency response indicates a flawless transient response (within the frequency range of each transducer). Directivity must be included in those measurements because loudspeakers radiate a 3D sound field. In rooms most of the sound arriving at the ears is reflected sound, which is totally dominant at low frequencies, and progressively reduced at higher frequencies because of the frequency-dependent directivity of the loudspeakers. The downward tilt of the steady-state in-room curves is really better interpreted as a bass rise, as the loudspeaker radiates more of its sound in all directions as frequency falls/wavelengths increase.
2. The human binaural hearing system - see the illustration in post no. 40 - is not at all the same as a measurement mic and analyzer. In everyday conversation and at live, unamplified, music events, voices and musical instrument sounds are easily separated from within the sounds contributed by the venue. We simply hear those familiar voices and instruments in different spaces. It is therefore not surprising that humans are able to recognize good and bad attributes of loudspeakers in different rooms. We have considerable ability to adapt to our listening spaces, perceptually streaming the essence of the music (the loudspeaker) from that of the room. As with any adaptation process, there are limits, meaning that it is important to know what we can and cannot compensate for. What we do absolutely know is that listening in a totally reflection free environment such as an anechoic chamber is not pleasant. Likewise, an empty, reverberant room is intolerable. Between those extremes is a significant range within which we are able to function without stress and derive substantial pleasure. It is interesting that when tested in an elaborate listening lab in Germany, listeners were displeased when the floor reflection was removed. It seems that humans have adapted to having a solid reflecting surface under us.

Measurements within a room cannot interrogate the performance of a loudspeaker in more than a crude sense. We must start with a good sound source, and the manufacturer should provide us with anechoic on and off-axis data to confirm that this is the case. If that is done, then in normally reflective rooms, the rest is made much simpler. We don't feel it necessary to equalize voices and instruments for different performance spaces. Why should it be necessary to do so for loudspeakers reproducing those voices and instruments? Acoustically poor rooms can exist in both live and reproduced performances, and in neither case is equalization a solution - that is for acoustical treatments. If one starts with flawed loudspeakers, which many do, unfortunately, equalization may or may not be able to help, but the best data upon which to base such equalization is anechoic data, and if one had that, surely the best thing to do is to avoid purchasing such loudspeakers.

As emphasized in the book, low frequency misbehavior in small rooms is a huge problem but there are solutions. As Olive found, bass accounts for about 30% of our overall factor weighting in evaluating sound quality. It needs to be attended to.
Thank you dr Toole.
I tried to understand quantum mechanics for many years and finally got it: it's about the cat.
I am afraid that with my poor mathematical apparatus an attempt to understand Fourier/transform functions might bring similar results.
That's why I am so happy to get the clear explanations in your post about what I need to know.
I take in-room measurements and always wonder: how this little application knows about all those sound waves coming to the microphone while knowing nothing about their convoluted way to get here? Isn't it a bit like the travelling salesman problem but with gazillions of salesmen and cities? I also make quasi-anechoic measurements outdoors (pic). Here the curves look very different.
Would you be so kind to comment on the usability of power response curves obtained in the Spinorama manner but outdoors? Www.thxstandard.com started including power response curve from 70 points in their loudspeaker measurements. The number 70 cannot be accidental, can it? Is Spinorama doable manually outdoors and does it make sense? What kind of useful info can come from it? Thank you for you time.
DipoleA1.jpeg
DipoleA1.jpeg
 

Floyd Toole

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Thank you, dr Toole! These are very interesting and enlightening points. I just want to comment on this one point: "We don't feel it necessary to equalize voices and instruments for different performance spaces". From the perspective of an active (amateur) musician, my experience is a bit different - if I understand you correctly, that is.

I have for some years been singing with a semi-professional choir, and we have performed in different kinds of acoustic spaces. I have also had my fair share of acoustic jazz jams (on piano). When singing with the choir, our conductor does in fact make sure that we adapt to the acoustics of the venue. I guess you could call it "equalization", in some sense. He is adamant that different kinds of dynamics and overall tonal balance is appropriate for different venues.

Acoustic instruments have also become tuned more brightly, because concert halls have become larger, and more treble energy gets lost on the way to listeners seated far away. Closed-mic'ed acoustic recordings therefore have a tendency to sound too bright to my ears.

My experience from jamming with other musicians is also that we more or less automatically adapt our playing to the acoustics of the room. Very bright rooms leads to another kind of playing than darker rooms, etc.

Does this have any bearing on the room eq discussion? Perhaps, perhaps not. My intuitive sense is that broadband adjustments of bass, treble and midrange may be useful if a room leans very much to one side acoustically (bright vs darker). In addition to equalization and other remedies in the bass, which you cover in an excellent way in your book!

All good points. As discussed early in my book, performance venues are inseparable parts of live performances. Capturing those combined elements is the task of the recording engineers. For the performers/conductors, the dominant factor is reverberation time, which requires some accommodation in performance by way, mainly of temporal pacing and/or instrumentation that can render the music more intelligible to audiences. In concert halls with too much bass absorption the bass viol section may be augmented, a kind of equalization by orchestral composition. In large venues, directional instruments like horns will penetrate to more distant seats and these may be augmented and/or aided by their locations in the orchestra. All that said, the timbral identities - the sound qualities - of the voices and instruments are fundamentally unchanged.

The loudspeaker parallels we have been discussing apply to small, relatively well damped acoustical spaces - reverberation times typically being 0.3-0.4s for domestic rooms and home theaters, but often much less for recording control rooms where 0.1-0.2 s are not unheard of (the "truckload of fiberglass" approach to acoustical design :). Performance venues range from 1 to almost 3 s. These are huge differences.

In sound reproduction the final performance occurs in a recording control room or mastering room, both of which are dimensionally similar to domestic listening spaces. Control rooms, by tradition, can be quite dead, but mastering rooms should be closer to typical playback environments, otherwise the "circle of confusion" becomes a factor - and that assumes that all parties employ timbrally neutral loudspeakers, which is not at all certain. Equalization is part of that process. I agree that close mic'ed recordings can be problems, and if they are, it is an indication of differing tastes (or hearing performance) between the recording staff and the listener, or, equally likely, monitoring through spectrally colored loudspeakers - compensating errors. Excessive brightness may also be the result of non-optimum mic placement. This happens in concert performances with elevated mics picking up more high frequencies from violins than is heard in the audience. There is a loudspeaker that is preferred for "classical" recordings by some experienced recording engineers - it has a sagging frequency response in the upper midrange/lower treble and makes the strings sound more natural. This is a case of the monitor loudspeaker being deliberately incorporated into the recording - dumb. Listeners will not hear what they heard unless they have the same idiosyncratic loudspeakers.

A lot of high end audiophile demonstrations employ close mic'ed recordings of simple voice and instrument combos because it is believed that they reveal "resolution" more clearly. Chacun a son gout.

In your examples of live performance, some amount of broad spectral adjustments are sometimes employed - mostly not in my observation, and not possible with all instruments and not with individual voices. Something similar also applies in sound reproduction. As I state several times in my book, tone controls are still useful to compensate for variations in recordings and personal preferences.

Starting with neutral, resonance-free, loudspeakers is a good beginning. Fix the bass, and enjoy.
 

Floyd Toole

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The number 70 cannot be accidental, can it? Is Spinorama doable manually outdoors and does it make sense? What kind of useful info can come from it?
Nice back yard!

THX uses a Klippel near-field scanning measurement system and the spinorama data format is one of the output options. I wish they would just show the spinoramas and spare us their unexplained interpretive/scoring scheme. The wide range of prices and sizes is another problem when there is only one list of ratings. But, business is business. As shown in Chapter 5 in my book, the 70 measurement curves result from measurements at 10 deg intervals in horizontal and vertical orbits. Yes you could do it in your outdoor setup - it is no more than some loudspeaker manufacturers have. The instructions are in a standard: ANSI/CTA 2034A. Sound power, per se, is not really very useful, as I explain in my book. Early reflections are much more closely related to what is measured and heard in small rooms.
 

pirad

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Nice back yard!

THX uses a Klippel near-field scanning measurement system and the spinorama data format is one of the output options. I wish they would just show the spinoramas and spare us their unexplained interpretive/scoring scheme. The wide range of prices and sizes is another problem when there is only one list of ratings. But, business is business. As shown in Chapter 5 in my book, the 70 measurement curves result from measurements at 10 deg intervals in horizontal and vertical orbits. Yes you could do it in your outdoor setup - it is no more than some loudspeaker manufacturers have. The instructions are in a standard: ANSI/CTA 2034A. Sound power, per se, is not really very useful, as I explain in my book. Early reflections are much more closely related to what is measured and heard in small rooms.
Thanks again. I should have mentioned that the speakers I build are dipoles with a mirror back tweeter and I have some problems interpreting their anechoic on axis measurements. I was hoping that power response would tell me some more about their predicted behavior in rooms.
 

Floyd Toole

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I was hoping that power response would tell me some more about their predicted behavior in rooms.

Power response describes sound radiated in all directions, without any discrimination. That means that such data are most useful at very low bass frequencies where loudspeakers are omnidirectional, or dipoles have figure 8 directivity. For dipoles the situation is complicated by the fact that the front and rear radiated sounds are of opposite polarity, which affects how they couple to room modes - i.e. very differently than monopoles. I really don't see how this can help you be predictive, other than noting - assuming that you have created a true dipole - that there will be little sound radiated at 90 deg.
 
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