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What is your favorite house curve

OK. Next question for you folks. As far as I can tell there is no spinorama for the Focal Aria 926's, which happen to be my main speakers. However, Amir did do a spinorama for the Aria 906 standmount speakers. They use the same tweeter and mid driver as the 926, but lack the two woofers as they are not tower speakers. Can/should I use the spinorama data from the 906 to set higher frequency PEQ's for my 926's or is that a bad idea and until I get spinorama data would I be better off just going with Audyssey or even not EQ'ing them?
 
OK. Next question for you folks. As far as I can tell there is no spinorama for the Focal Aria 926's, which happen to be my main speakers. However, Amir did do a spinorama for the Aria 906 standmount speakers. They use the same tweeter and mid driver as the 926, but lack the two woofers as they are not tower speakers. Can/should I use the spinorama data from the 906 to set higher frequency PEQ's for my 926's or is that a bad idea and until I get spinorama data would I be better off just going with Audyssey or even not EQ'ing them?

You should use REW and see what's cooking in your room. I bet there are similarities between models, particularly in where there are directivity anomalies but high frequency EQ is a bit fussy. Not sure you can really fully compensate for directivity anomalies with EQ.
 
So, to summarize what I'm hearing/learning...and please correct me if I'm wrong, the general process would be:

1. Use spinorama to get the most accurate speakers you can afford (because EQ will not be able to fix everything, especially off-axis sound)
2. Position your speakers/subs optimally in the room
3. Use EQ (automated, manual, or both) below roughly 500Hz to flatten out the response curve

Below this step is where I'm getting confused

4. Above 500Hz should I be using EQ on the speakers based on what I measure myself in the room or what is visible in the anechoic spinorama data?


5. Increase/decrease bass and/or adjust tilt/tonality as desired

Thank you all for your sage advice.
(Why are you confused, I explained it to you already, also other people have explained it too)

1) Anechoic EQ on the entire speaker from 20Hz-20kHz.
2) Measure in-room response and then do roomEQ below 500Hz based on your measurements. (In my room I just do room EQ below 300Hz though, I prefer it that way.)

So you'd have a set of parametric filters from Step #1 above and another set of parametric filters from Step#2 above - you'd apply all of them. If you find you're running out of filters in whatever software or hardware you're using then you could think about saving some filters by not applying Anechoic EQ to the speakers below say 300Hz or 500Hz, but the higher you go up the frequency range then the more important Anechoic EQ will be as the room starts to have less influence, so try to keep your cut off point for not using Anechoic EQ as low as possible if you're running out of parametric filters.

All the other points you made are fine - in terms of speaker choice & positioning, subs, etc.
 
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So, to summarize what I'm hearing/learning...and please correct me if I'm wrong, the general process would be:

1. Use spinorama to get the most accurate speakers you can afford (because EQ will not be able to fix everything, especially off-axis sound)
2. Position your speakers/subs optimally in the room
3. Use EQ (automated, manual, or both) below roughly 500Hz to flatten out the response curve

Below this step is where I'm getting confused

4. Above 500Hz should I be using EQ on the speakers based on what I measure myself in the room or what is visible in the anechoic spinorama data?
Fix the on-axis frequency response as seen in the spinorama.
5. Increase/decrease bass and/or adjust tilt/tonality as desired
and this.
 
OK. Next question for you folks. As far as I can tell there is no spinorama for the Focal Aria 926's, which happen to be my main speakers. However, Amir did do a spinorama for the Aria 906 standmount speakers. They use the same tweeter and mid driver as the 926, but lack the two woofers as they are not tower speakers. Can/should I use the spinorama data from the 906 to set higher frequency PEQ's for my 926's or is that a bad idea and until I get spinorama data would I be better off just going with Audyssey or even not EQ'ing them?
Spinorama data is only valid for that model of speaker, so I wouldn't use the 906 spinorama data if you've got the 926. You could perhaps try to find a review site that has measured both speakers though, and then overlap the two measurements to see what the potential differences could be between the speakers in a bid to see if you could 906 spinorama data.....but thinking about it it's not valid to do that as you can't assume what the Anechoic On-Axis & Anechoic Listening Window would be from that info - yeah, don't do it.

So this means that you'd be just basing your EQ's from your own in-room measurements. In that case I'd correct any obvious large deviations above the transition frequency and listen to see if it's an improvement. You'd also be fine to apply broad Tone Control to the speaker based on your measurements (& listening) to bring it into line in terms of brightness or darkness.
 
The 906 HF measurements look nice enough that I don't think there's any need to EQ it other than broad tone controls -- so why even bother? I would assume the same holds true for its bigger brother.
 
To get an idea of the direct sound of your speakers you can do the following.

1. Make sure of the height of your speaker's acoustic center. Most likely at the height of the tweeter but can sometimes be at the height of the midrange driver, and sometimes between the tweeter and the midrange driver.

2. Place the speaker on something to elevate it so that the acoustic center is at an equal distance between the floor and the ceiling, and make sure you place the speaker at least at the same length or more from the walls or other reflective surfaces.

3. Make a regular sweep measurement of the speaker with REW at a distance of 2 meters and at the height of the acoustic center.

4. Open the "Impulse" window in REW, and also click on the "IR Window". Change the value for the "Left Window (ms)" to about 0.2 ms, and press enter. Now you can slide the small "L" above the window in the "Impulse" window left or right until it's just in front of the start of the response (including the small waves in front of the big peak).

5. Now change the value for the "Right Window (ms)" to something between 2 ms to 5 ms. You should clearly see the first reflection which you should leave out of the window by moving the small "R" from left to right, as you can see in the picture below.
When you move the "R" beyond the first reflection point you will see that the frequency response curve starts to change, back it up from that point until the response is stable.

1675452409910.png


6. When this is done, click on "Apply Windows" in the pop-up IR window.

7. Now you can open the "SPL & Phase" window and see what the gated response of your speaker looks like. It's not perfect and the resolution is very low when the window is just about 2 to 5 ms long, but you get an idea of the direct response of your speaker.

It's even better if you can do the measurements somewhere outside your house with the speaker placed high up from the ground, or if you have your own anechoic chamber. If not, try the above to get the short gated window. :)
 
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OK. Next question for you folks. As far as I can tell there is no spinorama for the Focal Aria 926's, which happen to be my main speakers. However, Amir did do a spinorama for the Aria 906 standmount speakers. They use the same tweeter and mid driver as the 926, but lack the two woofers as they are not tower speakers. Can/should I use the spinorama data from the 906 to set higher frequency PEQ's for my 926's or is that a bad idea and until I get spinorama data would I be better off just going with Audyssey or even not EQ'ing them?
If the spinorama exist at all, it is here with a good probability:


… and that’s why many of us have bought speakers with existing spins.
 
To get an idea of the direct sound of your speakers you can do the following.

1. Make sure of the height of your speaker's acoustic center. Most likely at the height of the tweeter but can sometimes be at the height of the midrange driver, and sometimes between the tweeter and the midrange driver.

2. Place the speaker on something to elevate it so that the acoustic center is at an equal distance between the floor and the ceiling, and make sure you place the speaker at least at the same length or more from the walls or other reflective surfaces.

3. Make a regular sweep measurement of the speaker with REW at a distance of 2 meters and at the height of the acoustic center.

4. Open the "Impulse" window in REW, and also click on the "IR Window". Change the value for the "Left Window (ms)" to about 0.2 ms, and press enter. Now you can slide the small "L" above the window in the "Impulse" window left or right until it's just in front of the start of the response (including the small waves in front of the big peak).

5. Now change the value for the "Right Window (ms)" to something between 2 ms to 5 ms. You should clearly see the first reflection which you should leave out of the window by moving the small "R" from left to right, as you can see in the picture below.
When you move the "R" beyond the first reflection point you will see that the frequency response curve starts to change, back it up from that point until the response is stable.

View attachment 262044

6. When this is done, click on "Apply Windows" in the pop-up IR window.

7. Now you can open the "SPL & Phase" window and see what the gated response of your speaker looks like. It's not perfect and the resolution is very low when the window is just about 2 to 5 ms long, but you get an idea of the direct response of your speaker.

It's even better if you can do the measurements somewhere outside your house with the speaker placed high up from the ground, or if you have your own anechoic chamber. If not, try the above to get the short gated window. :)
This is excellent advice, and if one wishes, it can be elaborated to estimate off-axis performance as well, simply by rotating the loudspeaker. In the extreme, it is possible to generate the data necessary to create a spinorama, but only for higher frequencies. Because of the time windowing the frequency resolution becomes so large as to be useless at lower frequencies. In the spinorama database there are examples of people who appear to have done this. One also sees examples of excessive smoothing - low resolution in the frequency domain - probably done to hide evidence of resonances that would be embarrassing. So, just seeing a spinorama does not mean that it is trustworthy evidence. People lie . . . surprise, surprise.
 
This is excellent advice, and if one wishes, it can be elaborated to estimate off-axis performance as well, simply by rotating the loudspeaker. In the extreme, it is possible to generate the data necessary to create a spinorama, but only for higher frequencies. Because of the time windowing the frequency resolution becomes so large as to be useless at lower frequencies. In the spinorama database there are examples of people who appear to have done this. One also sees examples of excessive smoothing - low resolution in the frequency domain - probably done to hide evidence of resonances that would be embarrassing. So, just seeing a spinorama does not mean that it is trustworthy evidence. People lie . . . surprise, surprise.

I have recently learned that procedure from a guy that calls himself I-or at the Swedish forum Faktiskt, so the credit for the advice should go to him instead. :)

But... I did this kind of measurement for my speakers and tried to equalize them flat based on the direct sound response, but unfortunately, I didn't like the sound of that and found it sounding way too bright in my setup and choice of speakers. Without any EQ adjustments, the direct sound of my speakers has a broad dip from about 2k all the way up to 10k as can be seen in the picture below, but I subjectively prefer this listening to music.

The reason for my preference may be that most music productions usually have some kind of lift in the upper mid/presence area, which is most likely to make some sound elements stand out/don't drown in a modern crowded mix. The other reason may be because of the somewhat uneven off-axis response of my speakers. I'm not sure why, but the overall balance simply doesn't sound right when I equalize it for a flat response. But the speakers do sound fantastic anyway, so I'm not that bothered. :)

1675535509913.png
 
I have recently learned that procedure from a guy that calls himself I-or at the Swedish forum Faktiskt, so the credit for the advice should go to him instead. :)

But... I did this kind of measurement for my speakers and tried to equalize them flat based on the direct sound response, but unfortunately, I didn't like the sound of that and found it sounding way too bright in my setup and choice of speakers. Without any EQ adjustments, the direct sound of my speakers has a broad dip from about 2k all the way up to 10k as can be seen in the picture below, but I subjectively prefer this listening to music.

The reason for my preference may be that most music productions usually have some kind of lift in the upper mid/presence area, which is most likely to make some sound elements stand out/don't drown in a modern crowded mix. The other reason may be because of the somewhat uneven off-axis response of my speakers. I'm not sure why, but the overall balance simply doesn't sound right when I equalize it for a flat response. But the speakers do sound fantastic anyway, so I'm not that bothered. :)

View attachment 262282
The reason to aim for a flat and smooth on-axis frequency response is that it is a good indicator that there are no audible resonances. Looking at off-axis responses can confirm that estimate. If the loudspeaker exhibits no audible resonances the next step is to examine the frequency dependence of directivity. That also should be smooth, and probably smoothly declining with frequency. Most rooms will need significant attention at bass frequencies - a separate story - and once that is cleaned up you are in good shape to get the spectral balance to your satisfaction. That will inevitably be affected by the program itself and your personal preferences - it should be a broadband "tone control" kind of adjustment, and it is unlikely that a single "calibration" will be satisfactory for all programs. Good luck.
 
The reason to aim for a flat and smooth on-axis frequency response is that it is a good indicator that there are no audible resonances. Looking at off-axis responses can confirm that estimate. If the loudspeaker exhibits no audible resonances the next step is to examine the frequency dependence of directivity. That also should be smooth, and probably smoothly declining with frequency. Most rooms will need significant attention at bass frequencies - a separate story - and once that is cleaned up you are in good shape to get the spectral balance to your satisfaction. That will inevitably be affected by the program itself and your personal preferences - it should be a broadband "tone control" kind of adjustment, and it is unlikely that a single "calibration" will be satisfactory for all programs. Good luck.
Appreciate you continued endorsement of tone controls that many preamplifiers and A/V processors have abandoned. I use the below quite often on both music and movies. Also handy for lower volume listening.


0D781BAD-9E6A-4D84-BC53-160EFB8AC77A.png
 
The reason to aim for a flat and smooth on-axis frequency response is that it is a good indicator that there are no audible resonances. Looking at off-axis responses can confirm that estimate. If the loudspeaker exhibits no audible resonances the next step is to examine the frequency dependence of directivity. That also should be smooth, and probably smoothly declining with frequency. Most rooms will need significant attention at bass frequencies - a separate story - and once that is cleaned up you are in good shape to get the spectral balance to your satisfaction. That will inevitably be affected by the program itself and your personal preferences - it should be a broadband "tone control" kind of adjustment, and it is unlikely that a single "calibration" will be satisfactory for all programs. Good luck.

Yes, the bass region is a separate beast to solve. I have managed to get that to my liking with bass traps, two active subwoofers, and EQ adjustments. It's not perfect but I'm satisfied with the result and accept that I probably can't make it much better without converting my living room into a studio. :)

Thank you for your response.
 
The "early reflected" curve in a spinorama is a good predictor of the shape of a steady-state room curve above about 500 Hz. Again, a room curve is NOT a target, it is a result, and the result of a well designed forward-firing loudspeaker is a relatively linear, slightly downward tilted steady-state room curve. If the curve you measure is not that shape one must suspect that the loudspeaker has some "issues". This is where only a set of anechoic measurements - like a spinorama - is needed to determine what might be wrong and to determine whether equalization is a possible remedy. Frequency dependent directivity variations are a common problem, and it is the off-axis sound that dominates the shape of steady-state room curves. A spinorama would have revealed the problem from the outset, and perhaps have modified a purchasing decision. Beyond that, one is free to employ broadband humps and tilts to compensate for program variations - a subjective judgement, not a "calibration".
Thanks for this. It makes sense that the room curve is a result and not a target, but I think part of the reason that it keeps coming up is that it's common for room EQ software to allow the user to specify an in-room target curve based on measurements in the room. So the natural question that comes up is, what target curve should I use, what is the "best" target curve?

I recently got some new speakers which have spins provided by the manufacturer, and the spins look good. I use Dirac Live which requires the user to specify a target curve, and I wanted to get these new speakers dialed in to sound their best, so I wanted to answer the question of what curve would be best for myself. I tried three different calibrations which used different target curves, and limited room correction of one of them up to 500 Hz and used full range room correction for the other two. Here is a measurement of each of them at the listening position, after level matching them to the best of my ability.
dirac-target-comparisons.png


The first thing I noticed was that all three of them actually sounded very good. Even with such drastic differences in bass and treble, differences between them were audible if you were looking for them, but they were a lot less noticeable than I was expecting based on these measurements. So I guess a well designed speaker sounds good even if you make big changes to the bass and treble.

After testing several songs, the Harman +8 curve (downloaded from https://mehlau.net/audio/dirac-live-2/) limited to 500 Hz sounded the best of these three targets, and produced the most realistic sounding vocals. The other curves sounded more constrained in the vocals, probably due to the treble rolloff as seen in the graphs, but I never found the uncorrected treble to be too harsh - it sounded just right. I found that I wanted just a little bit more bass boost and ended up using the Harman +10 curve downloaded from that same website, and that was just enough to get the best sound I've ever heard in my room, and probably any room.

I guess the point I'm trying to make is that my experience is yet another data point that agrees with all of the research you and others at Harman did. The science was right, what a shocker. :) It really is as simple as buying speakers with good spins, getting the bass under control with treatments and/or room EQ, setting up bass boost to taste (for Dirac in particular those Harman curves are a good start), and letting the speaker do its thing in your room. Thanks for all of your contributions to the field.
 
Thanks for this. It makes sense that the room curve is a result and not a target, but I think part of the reason that it keeps coming up is that it's common for room EQ software to allow the user to specify an in-room target curve based on measurements in the room. So the natural question that comes up is, what target curve should I use, what is the "best" target curve?
When all you have is a hammer, everything looks like a nail. When all you can measure is a steady-state room curve (some algorithms can do more, but most do not) a "target" room curve has to be the recipe of the day. This is in spite of the extensive knowledge that it is not a reliable means of delivering neutral, uncolored, sound. Hence seemingly endless discussions in forums like this. Thanks for doing it the right way, and finding that there is more to it than somebody's aftermarket tweak that knows nothing about your loudspeakers, the devices mainly responsible for the sound you hear. Tone controls are still a requirement to compensate for occasional excesses and failings of recordings.
 
Could you point me in the direction of information on doing it the right way you referred to. I have only come across the many discussion on tutorials on target room curves. I have the same speakers as @dataman above. I have audyssey not Dirac but think I can get Dirac if needed.

Thanks!
 
Could you point me in the direction of information on doing it the right way you referred to. I have only come across the many discussion on tutorials on target room curves. I have the same speakers as @dataman above. I have audyssey not Dirac but think I can get Dirac if needed.

Thanks!
OK. I thought about opening a rat's nest by doing this, because it is a simplification of what sometimes has to be done. Some of the points have been made earlier in this forum thread and elsewhere, but it might be useful to bring the key factors in the process to one place. The marketing of room EQ algorithms often presents the impression that all combinations of loudspeakers and rooms can be "fixed", "calibrated" or the like, by means of measurements, math and equalization. In reality, much of the "math" does not include the exceptionally complex, non-linear and occasionally capricious psychoacoustics of human listeners. A critical missing element is that humans adapt to circumstances, bringing our perceptions into acceptable territory. Loudspeakers reproduce sounds. Musicians produce sounds. Both do it in rooms. We don't feel the need to "equalize" - even if we could - the instruments and voices of live music. Two ears and a brain separate the sources from the venue, and adapt to aspects of what the environment contributes to the overall performance. The venues vary, and some are even not ideal, but we manage to appreciate the excellence of fine instruments and voices in most of them.

The special problem with sound reproducing systems is that flaws get superimposed on everything that is played through them. These monotonous colorations can sometimes be beyond the ability of humans to adapt, and they need to be identified and attenuated.

Therefore, the "right way" begins with choosing well designed, timbrally neutral, loudspeakers. If the loudspeakers exhibit audible resonances and/or frequency-dependent directivity issues, it is not likely that measurements in a room will reveal such problems and that equalization is capable of compensating for them. It is often the case that the solution is better loudspeakers. Fortunately these can be identified with good reliability from competently made anechoic measurements presented in a "spinorama" format, following the industry standard. Amir, on this site, makes such measurements and others can be found at www.spinorama.org.

This done, set them up in your room and make a steady-state frequency response measurement at the prime listening position - the stereo seat. We will be paying close attention to the frequencies below about 400-500 Hz, where adjacent boundary effects and room resonances are active. Because much of the bass in recordings is mono (all of it in LPs) drive both loudspeakers simultaneously to evaluate what is happening at low frequencies. Measure them individually to find out what is happening at frequencies above about 400 Hz. If you are using bass management and one or more subwoofers the process is the same, and of course all subs should be running simultaneously. Why? Because multiple sound sources couple energy to room resonances differently when they operate in unison.

You can repeat this at different seats to see how much seat-to-seat variation there is - often quite a lot. Averaging several of these curves is a common practice, making the curves look much smoother, but hiding some awkward realities at low frequencies. Superimposing the curves on one graph is a more useful display of what is happening in your setup. You can then choose which humps/peaks to attenuate, depending on which seats are affected. Remember, at this stage we are looking only at bass frequencies. Narrow dips, however deep, should be ignored. Broad dips can be filled in, but keep the EQ boosts below about 6 dB. Aim for a smoothish curve that is tilted slightly upward at lower frequencies.

The benefits of this exercise will apply only to the seat or seats exhibiting similar shaped curves. That is why multiple-sub methods have been developed aimed at reducing seat-to-seat variations so that one equalization can deliver improved bass to several listeners. These are discussed in detail in Chapter 8 in the 3rd edition of my book.

Above about 400-500 Hz the "early reflections" curve in the spinorama should be similar to what you have measured. If you have well designed loudspeakers the room curve might have some smallish ripples caused by acoustical interference between and among the direct and reflected sounds - these are not problems to two ears and a brain and equalization is the wrong method of addressing them if they were - that is an acoustics issue. Spatial averaging over several microphone locations tends to smooth the room curve at middle and high frequencies, thereby reducing the likelihood that an auto-EQ algorithm (or a person) might try to "fix" something that can't be fixed, or that doesn't need to be fixed. Remember, any EQ applied to a room curve modifies the direct sound, and it the the direct sound that is a key factor in determining sound quality. If you began with loudspeakers designed to have the desirable smooth and flat on-axis/listening window response, they will be degraded.

Finally, pay attention to the overall shape of the room curve. Usually, at least for conventional forward-firing loudspeakers, the room curve will tilt gently downward. If the shape deviates substantially from the early-reflections spinorama curve then one can suspect something is amiss in the acoustical treatment of the room. If listening confirms a problem, then one is free to try modifying the shape of the spectrum with broadband, low-Q, tone-control kinds of equalization. When listening to recordings we get into the circle-of confusion dilemma, where it is difficult to know where the problem lies: the playback system or the recording.

Don't worry about little ripples. When I see exceptionally smooth high-resolution room curves I strongly suspect that something wrong has been done. The measurement microphone is no substitute for two ears and a human brain. Happy landings!
 
OK. I thought about opening a rat's nest by doing this, because it is a simplification of what sometimes has to be done. Some of the points have been made earlier in this forum thread and elsewhere, but it might be useful to bring the key factors in the process to one place. The marketing of room EQ algorithms often presents the impression that all combinations of loudspeakers and rooms can be "fixed", "calibrated" or the like, by means of measurements, math and equalization. In reality, much of the "math" does not include the exceptionally complex, non-linear and occasionally capricious psychoacoustics of human listeners. A critical missing element is that humans adapt to circumstances, bringing our perceptions into acceptable territory. Loudspeakers reproduce sounds. Musicians produce sounds. Both do it in rooms. We don't feel the need to "equalize" - even if we could - the instruments and voices of live music. Two ears and a brain separate the sources from the venue, and adapt to aspects of what the environment contributes to the overall performance. The venues vary, and some are even not ideal, but we manage to appreciate the excellence of fine instruments and voices in most of them.

The special problem with sound reproducing systems is that flaws get superimposed on everything that is played through them. These monotonous colorations can sometimes be beyond the ability of humans to adapt, and they need to be identified and attenuated.

Therefore, the "right way" begins with choosing well designed, timbrally neutral, loudspeakers. If the loudspeakers exhibit audible resonances and/or frequency-dependent directivity issues, it is not likely that measurements in a room will reveal such problems and that equalization is capable of compensating for them. It is often the case that the solution is better loudspeakers. Fortunately these can be identified with good reliability from competently made anechoic measurements presented in a "spinorama" format, following the industry standard. Amir, on this site, makes such measurements and others can be found at www.spinorama.org.

This done, set them up in your room and make a steady-state frequency response measurement at the prime listening position - the stereo seat. We will be paying close attention to the frequencies below about 400-500 Hz, where adjacent boundary effects and room resonances are active. Because much of the bass in recordings is mono (all of it in LPs) drive both loudspeakers simultaneously to evaluate what is happening at low frequencies. Measure them individually to find out what is happening at frequencies above about 400 Hz. If you are using bass management and one or more subwoofers the process is the same, and of course all subs should be running simultaneously. Why? Because multiple sound sources couple energy to room resonances differently when they operate in unison.

You can repeat this at different seats to see how much seat-to-seat variation there is - often quite a lot. Averaging several of these curves is a common practice, making the curves look much smoother, but hiding some awkward realities at low frequencies. Superimposing the curves on one graph is a more useful display of what is happening in your setup. You can then choose which humps/peaks to attenuate, depending on which seats are affected. Remember, at this stage we are looking only at bass frequencies. Narrow dips, however deep, should be ignored. Broad dips can be filled in, but keep the EQ boosts below about 6 dB. Aim for a smoothish curve that is tilted slightly upward at lower frequencies.

The benefits of this exercise will apply only to the seat or seats exhibiting similar shaped curves. That is why multiple-sub methods have been developed aimed at reducing seat-to-seat variations so that one equalization can deliver improved bass to several listeners. These are discussed in detail in Chapter 8 in the 3rd edition of my book.

Above about 400-500 Hz the "early reflections" curve in the spinorama should be similar to what you have measured. If you have well designed loudspeakers the room curve might have some smallish ripples caused by acoustical interference between and among the direct and reflected sounds - these are not problems to two ears and a brain and equalization is the wrong method of addressing them if they were - that is an acoustics issue. Spatial averaging over several microphone locations tends to smooth the room curve at middle and high frequencies, thereby reducing the likelihood that an auto-EQ algorithm (or a person) might try to "fix" something that can't be fixed, or that doesn't need to be fixed. Remember, any EQ applied to a room curve modifies the direct sound, and it the the direct sound that is a key factor in determining sound quality. If you began with loudspeakers designed to have the desirable smooth and flat on-axis/listening window response, they will be degraded.

Finally, pay attention to the overall shape of the room curve. Usually, at least for conventional forward-firing loudspeakers, the room curve will tilt gently downward. If the shape deviates substantially from the early-reflections spinorama curve then one can suspect something is amiss in the acoustical treatment of the room. If listening confirms a problem, then one is free to try modifying the shape of the spectrum with broadband, low-Q, tone-control kinds of equalization. When listening to recordings we get into the circle-of confusion dilemma, where it is difficult to know where the problem lies: the playback system or the recording.

Don't worry about little ripples. When I see exceptionally smooth high-resolution room curves I strongly suspect that something wrong has been done. The measurement microphone is no substitute for two ears and a human brain. Happy landings!
Wow. Thank you so much for the extensive write up and summary.
 
OK. I thought about opening a rat's nest by doing this, because it is a simplification of what sometimes has to be done. Some of the points have been made earlier in this forum thread and elsewhere, but it might be useful to bring the key factors in the process to one place. The marketing of room EQ algorithms often presents the impression that all combinations of loudspeakers and rooms can be "fixed", "calibrated" or the like, by means of measurements, math and equalization. In reality, much of the "math" does not include the exceptionally complex, non-linear and occasionally capricious psychoacoustics of human listeners. A critical missing element is that humans adapt to circumstances, bringing our perceptions into acceptable territory. Loudspeakers reproduce sounds. Musicians produce sounds. Both do it in rooms. We don't feel the need to "equalize" - even if we could - the instruments and voices of live music. Two ears and a brain separate the sources from the venue, and adapt to aspects of what the environment contributes to the overall performance. The venues vary, and some are even not ideal, but we manage to appreciate the excellence of fine instruments and voices in most of them.

The special problem with sound reproducing systems is that flaws get superimposed on everything that is played through them. These monotonous colorations can sometimes be beyond the ability of humans to adapt, and they need to be identified and attenuated.

Therefore, the "right way" begins with choosing well designed, timbrally neutral, loudspeakers. If the loudspeakers exhibit audible resonances and/or frequency-dependent directivity issues, it is not likely that measurements in a room will reveal such problems and that equalization is capable of compensating for them. It is often the case that the solution is better loudspeakers. Fortunately these can be identified with good reliability from competently made anechoic measurements presented in a "spinorama" format, following the industry standard. Amir, on this site, makes such measurements and others can be found at www.spinorama.org.

This done, set them up in your room and make a steady-state frequency response measurement at the prime listening position - the stereo seat. We will be paying close attention to the frequencies below about 400-500 Hz, where adjacent boundary effects and room resonances are active. Because much of the bass in recordings is mono (all of it in LPs) drive both loudspeakers simultaneously to evaluate what is happening at low frequencies. Measure them individually to find out what is happening at frequencies above about 400 Hz. If you are using bass management and one or more subwoofers the process is the same, and of course all subs should be running simultaneously. Why? Because multiple sound sources couple energy to room resonances differently when they operate in unison.

You can repeat this at different seats to see how much seat-to-seat variation there is - often quite a lot. Averaging several of these curves is a common practice, making the curves look much smoother, but hiding some awkward realities at low frequencies. Superimposing the curves on one graph is a more useful display of what is happening in your setup. You can then choose which humps/peaks to attenuate, depending on which seats are affected. Remember, at this stage we are looking only at bass frequencies. Narrow dips, however deep, should be ignored. Broad dips can be filled in, but keep the EQ boosts below about 6 dB. Aim for a smoothish curve that is tilted slightly upward at lower frequencies.

The benefits of this exercise will apply only to the seat or seats exhibiting similar shaped curves. That is why multiple-sub methods have been developed aimed at reducing seat-to-seat variations so that one equalization can deliver improved bass to several listeners. These are discussed in detail in Chapter 8 in the 3rd edition of my book.

Above about 400-500 Hz the "early reflections" curve in the spinorama should be similar to what you have measured. If you have well designed loudspeakers the room curve might have some smallish ripples caused by acoustical interference between and among the direct and reflected sounds - these are not problems to two ears and a brain and equalization is the wrong method of addressing them if they were - that is an acoustics issue. Spatial averaging over several microphone locations tends to smooth the room curve at middle and high frequencies, thereby reducing the likelihood that an auto-EQ algorithm (or a person) might try to "fix" something that can't be fixed, or that doesn't need to be fixed. Remember, any EQ applied to a room curve modifies the direct sound, and it the the direct sound that is a key factor in determining sound quality. If you began with loudspeakers designed to have the desirable smooth and flat on-axis/listening window response, they will be degraded.

Finally, pay attention to the overall shape of the room curve. Usually, at least for conventional forward-firing loudspeakers, the room curve will tilt gently downward. If the shape deviates substantially from the early-reflections spinorama curve then one can suspect something is amiss in the acoustical treatment of the room. If listening confirms a problem, then one is free to try modifying the shape of the spectrum with broadband, low-Q, tone-control kinds of equalization. When listening to recordings we get into the circle-of confusion dilemma, where it is difficult to know where the problem lies: the playback system or the recording.

Don't worry about little ripples. When I see exceptionally smooth high-resolution room curves I strongly suspect that something wrong has been done. The measurement microphone is no substitute for two ears and a human brain. Happy landings!
This is amazing. THANK YOU!!!!
 
Thank you Dr. Toole, as always, for your wonderful educational message!

When I see exceptionally smooth high-resolution room curves I strongly suspect that something wrong has been done.

Yes, I always feel the same.

In this perspective, I usually do not like the "psychoacoustic smoothing" after the very short-time sine wave scanning, as I wrote here.

At least for me, my "cumulative white noise averaging method" (please refer to here and here) would much better fit with my actual listening sensations, and the so-given "house curves" are highly reproducible, less statistical fluctuations/deviations all over 20 Hz to 25 kHz. Furthermore, I can choose proper FFT size (as smoothing intensity) depending on the Fq zones of interest. Of course, the quality (QC assurance) of the "flat white noise" should be critical; I use the very strictly well prepared and QC-ed "flat white noise track" of "Sony Super Audio Check CD".

Let me share an example of such "house curve" measured by "cumulative white noise averaging method" in my next post bellow.
 
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