# Full range target curves



## Pio2001

Hi,
I have been experimenting with full range equalisation with REW for a few days, and I'd like to discuss the topic of Target curves for a stereo 2.0 system. Strangely enough, I wasn't satisfied at all with the proposed target curve in REW, and I ended with a somewhat different one.

I came to REW and parametric equalisation since I replaced my hifi speakers (Dynaudio Gemini, then Kef R-300) with Neumann KH-120 monitors. They are active monitors with a flat response until 50 Hz, which triggers two room modes at 55 and 70 Hz that are absolutely unbearable. 
The hifi speakers had an attenuated response at these frequencies, and I could play with the bass reflex opening, but with the Neumann, although the mids and highs sound much more natural, for low frequencies, equalisation couldn't be avoided.

The question of the target curve is especially interesting with the Neumann because they are supposed to sound completely flat in anechoic conditions, or that's what they claim. Here is an independent measurment of these speakers : http://kenrockwell.com/audio/neumann/kh-120-a.htm

Therefore what I measure with REW at the listening position should reflect the *room gain* more than the *speakers response*.

I use a Umik microphone in vertical position, with the standard calibration file (the Umiks are not calibrated one by one, instead, the file is supposed to fit a whole batch with the same serial number).

First, here are the measurements before equalisation. These are the average of 6 measurement for each speakers, separated by about 40 cm from each other, so as to match a wide range of possible listening positions.



















All of the next curves (except the 0.9 db slope ones) differ from the reality because of the activation of -1 dB treble correction directly on the speakers, that isn't taken into account in the predicted result. I set the treble at -1 according to my liking, as the user manual explains that the treble level must be set according to the kind of acoustic - bright or damped- the speakers are installed in.
Anyway, here is the measured effect of the switch. This is nothing to worry about. It might even be lower than the accuracy of the Umik (don't take the general look of the curve into account, the tone controls of the preamplifier were mistakenly turned on during both measurements).










I first targeted the curve proposed by REW, which lead to this : 



















On the graph, it looks ok. But in real life, there seem to be a good lack of low frequencies. The sound is thin... and at the same time, the bass, below 100 Hz are too loud.

Then I decided to try the following : since the Neumann are supposed to sound neutral, let them do, and just equalize the low frequencies flat.
I decided to ignore the high frequency part of the curve, and to align the target curve with the measured curve in the 1 - 2 kHz range, where they have exactly the same slope.

Following the measured curve towards the low frequencies, I tried this : a flat target curve with a knee at 400 Hz, then -1.8 dB / octave (the original slope proposed by REW). The target level for bass equalisation being given by matching the measured and target curves between 1 kHz and 2 kHz.



















And wow ! That sound much better. Although the bass are not perfect yet.

I then tried another idea : to completely follow the Neumann measured response in the mid-highs and continue with the same slope across the whole frequency range. That leads to a 0.9 dB / octave curve. Here is the predicted result :



















Aaaarg ! That sounds terrible ! Muffled.

I came back to the previous curve, with the knee at 400 Hz, and I tried to move the knee to 200 Hz (more bass), or 600 Hz (less bass). The result were audible inferior. The sound seems balanced when the knee is at 400 Hz.

This week, I went further and tried to fill the notches in the 100 - 200 Hz range. That lead to a great improvement. The bass were cleaner, and more natural.

Listening to my new equalisation today, I found that there was still a little annoyance in the low mids, especially in choral music, but also in metal. I again tried to move the knee of the target curve to the right, but that was not good, as the mids sound too loud on solo voices.
I wondered if the problems didn't come from the remaining little oscillations between 300 and 1 kHz. They might very well be high order room modes. The speakers are standing 80 cm away from the rear wall, which causes two dips, one just below 100 Hz and the other just below 300 Hz. Maybe there are other ones.

I then carefully equalized them, checking in the overlay view of REW the 6 separate measurements to ensure that the corrections I was doing were valid for a wide enough set of listening positions. 
The corrections are small, but I think that the problem in the low-mid is now reduced. The current result sound incredibly well balanced. Here are the predicted psychoacoustic curves :



















And the filters used :



















For the time being, the correction is done in Foobar2000's convolver, but I should soon get my MiniDSP and load the correction for all sources. I can't wait !


Now I'd like to hear about other people's experiments with the target curve. Do you actually feel the need to increase the bass level below 100 Hz ? Are you satisfied with the low-mids (200 - 800 Hz) ?


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## ajinfla

Pio2001 said:


> Therefore what I measure with REW at the listening position should reflect the *room gain* more than the *speakers response*.


Welcome Pio.
I understand what you are saying, but in reality, the room/speaker response is inseparable. The "speaker response" you refer to is the on axis onset taken at some distance. There are many more 3 dimensionally, the bass being pretty much omni at LF. If we compared it to a gradient loudspeaker with the exact same anechoic/free field response that you are referring to as the "speaker" response, the room response would be completely different. A subtle point often missed.



Pio2001 said:


> This week, I went further and tried to fill the notches in the 100 - 200 Hz range. That lead to a great improvement. The bass were cleaner, and more natural.


By now you should have read JJ's work.:smile: It is generally not advisable to fill notches, as these are points of energy storage in the room (where the mic is). Filling these "points" tends to create excess elsewhere (there goes your "flat" onset :smile....but if the EQ was used judiciously..and you are happier with the sound, ok. Just beware of headroom issues if you are applying several db of boost at those frequencies.



Pio2001 said:


> For the time being, the correction is done in Foobar2000's convolver, but I should soon get my MiniDSP and load the correction for all sources. I can't wait !


Which MiniDSP, existing 2x4 or new 2x4HD?

cheers,


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## Pio2001

Thank you for the response, ajinfla



ajinfla said:


> By now you should have read JJ's work.


What is JJ's work ? I've read the threads by Wayne A. Pflughaupt about house curves, and also Floyd Toole's book about Loudspeakers and rooms.



ajinfla said:


> :smile: It is generally not advisable to fill notches, as these are points of energy storage in the room (where the mic is). Filling these "points" tends to create excess elsewhere (there goes your "flat" onset :smile...


I understand, although I wonder if this is also true for acoustic cancellations occurring at the speaker location (the 80 and 280 Hz dips caused by the front wall direct reflections). 

Anyway, this is where I used the "overlay" option in REW. First, in the averaged curve, I identify a dip that I would like to fill. Then I reset all smoothing and go to the overlay view. I ask for all the individual curves measured for the speaker (6 in my case).
If the dip is present in all 6 measurements -> I fill it (in consequence, the sound may become worse outside of the measurement zone).
If the dip results of the averaging of various behaviors -> I equalize according to the individual curve that has the least pronounced dip.
If one of the curves recorded at equal distance of the speakers (one of the possible exact sweet spots), or if two curves recorded sideways have no dip at all, I don't equalize this part.

Then, there is the second problem with dips: our audition is more sensitive to peaks than to dips. That's where the Psychoacoustic smoothing option is useful. It draws a curve that gives precedence to peaks over dips.
So, after having selected the dips that are eligible for equalization, and setup the right frequency and Q in the filters section, I fine-tune the amount of correction with the Psychoacoustic view so that half of the curve is below the target, and half is above.
Doing the same thing with any other kind of smoothing would emphasize the regions where the curve oscillates around a mean value over regions where it is smooth.



ajinfla said:


> Just beware of headroom issues if you are applying several db of boost at those frequencies.


Yes. I disabled the autolevel option in Foobar's convolver and set its gain to -8 db, as my highest boost is +8 dB.
I hope that there is an easy way to do this with the MiniDSP, as I don't see any master gain in the filters section of REW.



ajinfla said:


> Which MiniDSP, existing 2x4 or new 2x4HD?


I ordered the cheap 2x4 with the Advanced 2x4 software. I'll just have to reduce my equalization to 6 filters per channel, as it is the maximum that the MiniDSP can do, if I understand well.


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## jtalden

Pio2001 said:


> ...Now I'd like to hear about other people's experiments with the target curve. Do you actually feel the need to increase the bass level below 100 Hz ? Are you satisfied with the low-mids (200 - 800 Hz) ?


We all know there is only one correct house curve. :sarcastic: It's the one we each prefer with our particular setup, measuring practices and chosen set of program material. 
It's the one we choose is dependent on:
> Speakers used
> Room acoustics
> Chosen program material
> Smoothing type applied (psy, 1/6, var, etc)
> Mic orientation
> Averaging type (MMM, Multiple Sweeps, etc)
> EQ type and practices
> Our personal preferences

Now to answer your questions.
*'Other Experiments':*
Attached is versions 19-36 of the house curves I have tried. There are maybe 5 different measurement and averaging schemes that are represented there so there is no real direct comparison between them. I am currently using #31 as a reasonably good compromise for most material. It tends to be too sharp on some material for me, but is a good compromise.

*'Bass Level below 100 Hz':*
I leave 20-100 Hz target flat as shown. Note though that the relative level compared to the midrange varies depending on the House curve roll-off. More roll-off results in stronger bass. I just chose the bass range as the reference range for my trial curves. It made sense to me as it is a flat portion of the curves. Some of my earlier curves did not have this same standard and thus more confusing to display here.

*'200-800 Hz':*
No, not really. For me 100-600 is the range that is very difficult for me to get right. I have room influences that can only be solved with positional or room acoustic changes that I am not willing to make. At least in my case it seems the lower and higher freqs are relatively easy to control with EQ.


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## Pio2001

Thank you for sharing your data and experiences. I'm a bit relieved that I'm not the only one to choose a flat extension into the bass below 100 Hz.

I have pictured three curves together : your average one, the one used in REW, and what I ended with after my calibration (including the -1 switch on the speakers).










They seem quite different, but the vertical scale is wide. In fact, the difference between them is typically 1dB, which is small.


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## ajinfla

Pio2001 said:


> What is JJ's work ?


I thought you were the mod from HA. This JJ, this work, in particular: A Low-Complexity, Fast-acquiring Perceptually Tuned Room Correction Algorithm.
Yes Toole's work too, in particular: http://www.aes.org/tmpFiles/elib/20160707/17839.pdf



Pio2001 said:


> I understand, although I wonder if this is also true for acoustic cancellations occurring at the speaker location (the 80 and 280 Hz dips caused by the front wall direct reflections).


Not sure what you mean here. There should be no notches "at the speaker location", only away from speaker where the speaker/room response is measured, at least the pressure at that point.



Pio2001 said:


> If the dip results of the averaging of various behaviors -> I equalize according to the individual curve that has the least pronounced dip.
> If one of the curves recorded at equal distance of the speakers (one of the possible exact sweet spots), or if two curves recorded sideways have no dip at all, I don't equalize this part.
> Then, there is the second problem with dips: our audition is more sensitive to peaks than to dips. That's where the Psychoacoustic smoothing option is useful. It draws a curve that gives precedence to peaks over dips.
> So, after having selected the dips that are eligible for equalization, and setup the right frequency and Q in the filters section, I fine-tune the amount of correction with the Psychoacoustic view so that half of the curve is below the target, and half is above.
> Doing the same thing with any other kind of smoothing would emphasize the regions where the curve oscillates around a mean value over regions where it is smooth.


Ultimately if the bass sounds better to you with the EQ you have applied and there is no audible clipping of the signal (the speaker should handle those frequency boosts as they would not be excursion limited in those ranges, more amplifier power limited), then "correct" should not matter. Only your preference.



Pio2001 said:


> Yes. I disabled the autolevel option in Foobar's convolver and set its gain to -8 db, as my highest boost is +8 dB.
> I hope that there is an easy way to do this with the MiniDSP, as I don't see any master gain in the filters section of REW.


Good, I figured you might know how to use Foobar:smile:. I believe the Minidsp has limiter functions, but you could always split 8db into 4db gain and 4db cut across the main bandwidth.



Pio2001 said:


> I ordered the cheap 2x4 with the Advanced 2x4 software. I'll just have to reduce my equalization to 6 filters per channel, as it is the maximum that the MiniDSP can do, if I understand well.


Ok.

cheers


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## Pio2001

ajinfla said:


> I thought you were the mod from HA. This JJ, this work, in particular: A Low-Complexity, Fast-acquiring Perceptually Tuned Room Correction Algorithm.


Indeed I am. But I have been inactive for years in Hydrogenaudio. Thanks a lot for the link !



ajinfla said:


> Not sure what you mean here. There should be no notches "at the speaker location", only away from speaker where the speaker/room response is measured, at least the pressure at that point.


I was talking about the distance between the speaker and the front wall behind it. If it is 80 cm, a 100 Hz signal will bounce back on the wall and pass again through the speaker location, but with opposite polarity, as 160 cm is roughly half the wavelength of 100 Hz. In this case, shouldn't the sound level decrease in all the room, like when you put two speakers playing with opposite polarity front to front ?


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## ajinfla

Pio2001 said:


> Indeed I am. But I have been inactive for years in Hydrogenaudio. Thanks a lot for the link !


You coulda fooled me :smile:
https://hydrogenaud.io/index.php/topic,111267.msg916570.html#msg916570
Welcome on the links.



Pio2001 said:


> I was talking about the distance between the speaker and the front wall behind it. If it is 80 cm, a 100 Hz signal will bounce back on the wall and pass again through the speaker location, but with opposite polarity, as 160 cm is roughly half the wavelength of 100 Hz. In this case, shouldn't the sound level decrease in all the room, like when you put two speakers playing with opposite polarity front to front ?


It's a bit more complex than that as the speaker is omnidirectional in this range, so there will be a great many reflections and delays combining with the direct response to get that pressure minima you measure. All that should concern you is what is heard by your two ears...and what they prefer. A pressure mic should be used as a guideline, but as already noted, it may to be advisable not to try complete fill of those points of pressure minima/velocity maxima with more energy, as that more energy ends up elsewhere as well. Luckily those frequencies you are boosting should not overtax the driver. If the result to your ears is better bass...better yet.:smile:

cheers


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## Pio2001

By the way I've got a question. I read, and noticed, that it is better to equalize the lowest frequencies with both speakers active, because two independent correct equalizations don't sum up to give a correct one in stereo.

At what frequency should be the transition between common equalization (bass) and independent equalization (higher frequencies) ?


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## Pio2001

Thanks ajinfla,

About channel independence, I've run a small test : I turned my head to the left and the right while playing a sinewave with a software sine generator. It seems to me that the direction of the sound becomes perceptible between 70 and 100 Hz.


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## ajinfla

Pio2001 said:


> By the way I've got a question. I read, and noticed, that it is better to equalize the lowest frequencies with both speakers active, because two independent correct equalizations don't sum up to give a correct one in stereo.
> At what frequency should be the transition between common equalization (bass) and independent equalization (higher frequencies) ?


How far apart are the speakers? I would also use global EQ as you suggest, rather than independent CH.



Pio2001 said:


> Thanks ajinfla,
> About channel independence, I've run a small test : I turned my head to the left and the right while playing a sinewave with a software sine generator. It seems to me that the direction of the sound becomes perceptible between 70 and 100 Hz.


Try an impulsive signal instead, something with a sharp onset.

cheers,


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## Pio2001

ajinfla said:


> How far apart are the speakers? I would also use global EQ as you suggest, rather than independent CH.


The speakers are 130 cm apart from each other. 
I experimented with full stereo, partial stereo and full mono correction, but, except for the two peaks below 100 Hz, it seems that the result depends more on the way the corrections are made (they are manual above 100 hz) than on the stereo or mono choice.

Anyway there is not much difference in the left are right measurements above 100 Hz, as we can see looking at both curves, so it shouldn't matter.









I'm currently satisfied with a set of filters that are identical below 100 Hz and different above. I'm currently tuning the general amount of bass with an extra low shelf filter of -2 dB starting at 200 Hz.

There was too much bass, but anything I do at and above 400 Hz to modifiy the target curve decreases the quality of the medium frequencies. I prefer not correcting anything above 300 Hz and deal with the bass level alone, which gives me a strange shaped house curve, with a small peak at 400 Hz.


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## Wayne A. Pflughaupt

> By the way I've got a question. I read, and noticed, that it is better to equalize the lowest frequencies with both speakers active, because two independent correct equalizations don't sum up to give a correct one in stereo.


As far as I know that doesn’t apply for main-channel speakers so much as people using multiple subwoofers in different locations. I’ve had good luck equalizing the lower frequencies of bookshelf-sized speakers.




> At what frequency should be the transition between common equalization (bass) and independent equalization (higher frequencies) ?


I would recommend matching EQ above ~400 Hz. Indepent EQ above that point does weird things to the imaging. If that’s what you’ve been doing, it could account for your not being happy with your results in that range. 

Regards, 
Wayne


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## Pio2001

Ah thanks,

Here are the filters I'm currently using. It should be ok as long as mono / stereo is concerned.

Filter #1 and #2 have been generated by the software from the measurements made with both speakers, then copied here.


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## jtalden

Pio,
I am curious to look at the phase relationship between L and R. Do you mind posting the .mdat?


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## Pio2001

Oops ! I completely forgot about phase ! 

Here is the file. I've removed all individual measurements to make it smaller to upload. 
The filters used are the ones that are active in the L and R measurements (Left and Right). The inactive filters are obsolete.
The St measurement (left + right speakers) is no more used.


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## jtalden

The attached file contains averages. They do not include the IR and thus no phase information. I was looking for 2 single sweep measurements at the LP; one each of the L and R mains. I just thought you may have that at hand.

From the discussion, I was just thinking of the common situation in the 200-800 Hz range where is difficult or inadvisable to EQ due to possibly creating more issues than resolving. I was just interested in seeing if you have as much trouble with strong reflections in this area as I do. I don't anticipate there is any helpful info to be found that would suggest a best EQ approach for in that range. I am on board with the general thoughts above regarding whether to / how to EQ in this area. It was just more for my continued learning.


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## Pio2001

jtalden said:


> The attached file contains averages. They do not include the IR and thus no phase information. I was looking for 2 single sweep measurements at the LP; one each of the L and R mains. I just thought you may have that at hand.


Sweep measurements or impulse measurements ? 
I didn't do impulse measurements. How do you do this, in two words ?



jtalden said:


> I was just interested in seeing if you have as much trouble with strong reflections in this area as I do.


I'm sure I do. Ceramic floor, concrete ceiling and a bare concrete wall 50 cm on the left of the left speaker ! These solid surfaces absorb no acoustic energy. The sound completely bounce back on them.


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## jtalden

I was just looking for 2 sweep measurements. Don't go to any trouble. It was just a thought.

To explain:
Given a normal sweep measurement, REW automatically calculates the IR and thus the phase trace is available. If several sweeps are taken in the listening area and the 'Average the Responses' button is selected then the IR/Phase data is lost in the average, i.e., the SPL alone is averaged without regard to the IR/phase of the original sweeps. 

It is possible to manually average several sweep measurements in REW using 'Trace Arithmetic'. That process will retain the IR/phase data in the resulting average, but that is quite a bit more complicated to do. It is also considered unnecessary for the purposes of EQ. It does have other uses however.


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## Pio2001

Aaah Ok.

Here are the individual measurement for the center front listening position (attached file at the bottom of the message). I don't know how to interpret the phase.



jtalden said:


> Don't go to any trouble. It was just a thought.


I'm interested in any info. I've just received my MiniDSP, so I could easily perform a measurement with the correction loaded. Here is the result :









A huge bump at 300 Hz. I then went into new trials to see if I could get something more balanced, but if I try to remove the bump, it doesn't sound right. I just rose the low shelf from -3 dB to -2 dB to get a bit more bass below 200 Hz.

My hypothesis, for the time being, is that the excess at 300 Hz comes from reflections in the room, and if I try to equalize them, the direct sound becomes unnatural. I've read that we can, rather unconciously that conciously, distinguish the direct sound from the reflections (through directional clues, and precedence effect / masking). The microphone doesn't distinguish between them. It records the sum of everything. 

It would mean that this bump would sort of "sounds right" as long as real audio sources would be expected to sound this way in my living room.

It could also mean that I did it all wrong from the beginning 0

Time will tell. I can't stand unbalanced sounds for a long time. If something is really wrong, I should get tired of it after some days.


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## Pio2001

PS : third hypoithesis: the bass sounds sooo bad in my living room (very long resonances) that the only thing that sounds right is to remove it completely. :dizzy:


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## Pio2001

Never mind. I was just using too narrow filters or unsuited low shelfs to adress the 400 Hz bump. With a parametric filter of q=0.77 it is good. 

Yes, I should stop spamming with every minor modification in my filters. I'm just in the part of the process where I'm drowning in a sea of filters :work:


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## jtalden

Thanks for the file. 

I didn't find anything particularly noteworthy relative to phase. Nothing beyond what is normal for a live room. 

I did look at to see what I would first try in regard to EQ. It appears that a common filter for both channels may work out pretty well. Most all the major discrepancy between channels is <100Hz where many have said that the bass is often mixed as mono anyway. I don't know how true this is, but that would help with the smoothness of the bass in this case. 

There may be better progress to be made with different speaker/LP positions in this room, or with some acoustic treatment. Setting those options that aside, you may want to try the attached filter set. HF filter (#5) just brings down highs a little to meet the house curve I prefer. If this is too much roll-off of the highs for your preference just reduce the level or eliminate this filter. Attached is the modified .mdat file. Note that, I summed the L, R and use that to create this filter set. Your L+R measurement was done with the mic a little off-center so the HF was artificially rolled off.









View attachment Pio2001-ja-1.mdat


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## Pio2001

jtalden said:


> Your L+R measurement was done with the mic a little off-center so the HF was artificially rolled off.


Oh, so this is the cause of all these variations that I get from a measurement to another ! I was beginning to wonder if my speakers had a different amount of treble everyday.

Thank you for your curve. It's interesting. 
I've tried your filter on my test tracks. First, I had to reduce the amount of bass for the two peaks at 55 and 70 Hz. I don't know what's going on here. I suspect that there is so much reverberation in the room at these frequencies that this part of the sound seems to play completely apart from the rest of the music. The new predicted curve is 









Now the result is interesting. Better on metal (Nightwish, Whithin Temptation). It doesn't have the cavernous effect of my equalization. But on the other hand, there is a harshness higher in the midrange that affects the voices.
With your curve, the music seem well-balanced, but with harsh vocals. With the one I currently use, the vocals seem better, but the overall balance is affected by an excess of low-mid frequencies.

For the time being, I think I prefer my curve, but I keep yours at hand just in case I get tired of listening to mine.


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## Pio2001

I'm so bad at evaluating frequecies ! :duh: 
The harshness was not in the mids, but at 8000 Hz. I removed the treble filter, as you proposed, and now the result is better. The vocals are good and the overall balance is good too !

Thank you very much. I'm going to load the new correction in the MiniDSP :smile:


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## jtalden

There are strong reflection nulls, both at 275 Hz and also near 400 Hz. These are possible contributors to the harshness. There is little PEQ can do about that. If the offensive frequencies can be identified then they can be suppressed a little in level via a house curve adjustment to reduce the emphasis on the problem. Here, I tried to softening the adjoining peaks with filters 3 and 4, but that didn't work out. It may well be better to just remove those filters, as is often advised. PEQ filters often tend to be counterproductive in this frequency range. Room placement and room acoustic changes are the far better approach in this range. Good Luck.


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## Pio2001

Yes, yes, it worked. Look at my second message. Your eq did the job just right.


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## jtalden

Oh, I missed that! Thanks for pointing it out. 
I had no real expectation that this would be that close your preference. It was just an alternate starting point in case it differed significantly from what you were already evaluating. I'm glad to hear you are getting close. :sn:


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## Pio2001

Hi,
My filters have not changed for two weeks. I think they are good now. I'd like to share an interesting finding with you.

Thanks to jtladen, I was able to have a good correction in the low-mid range with two filters at 327 and 580 Hz. Later, I found that a minor peak at 200 Hz was causing a major audible annoyance. So I added a third filter at that frequency. The three filters, isolated from the rest of the correction, look like this :









What's interesting is that they match the effect of the reflections of the acoustic waves on the wall behind the speakers :









Since the speakers are 80 cm away from the rear wall, there should be a cancellation at 100, 300, 500 etc, together with peaks at 200, 400, 600 etc.
However, this is true in the direction perpendicular to the wall. In reality, we must take into account the energy radiated around this direction :









Which means that the result should be dips just *below* 100 Hz, *below* 300 Hz, *below* 500 Hz... and peaks *below* 200 Hz, *below* 400 Hz etc.
...Which is exactly what can be seen on the measurement, except for the peak at 200 Hz, that is weak, and not really below 200. Maybe it is partially masked by some other reflections. 

Here are the 6 raw measurements (both speakers active) with 1/24 octave smoothing, which show exactly what's happening.









Below 100 Hz, the two main peaks are perfectly dealt with using the auto correction, provided that we know exactly what should be the target level. I used trial and error to find this, and, to get a better overall balance, I added two low-shelf filters. One at 200 Hz in order to restore the level around 100 Hz, and another one at 950 Hz to adjust the overall bass / treble balance.
Trying to apply a continuous slope rising from 1000 Hz to 20 Hz didn't do any good. The result is better with +1 dB from 950 Hz, then 2 more from 200 Hz.
Here are all the filters together. The interesting thing with the 200 Hz low shelf is that is allows to boost the 100 Hz zone without creating any annoying peak. It needs to be applied *before* adjusting the amplitude of the parametric filters, if we want to match their level with a given reference. 









And here is the final measured result, with the correction loaded in the miniDSP, in variable smoothing and psychoacoustic smoothing.

















I didn't manage to correct completely the dip at 100 Hz. Things sound worse if I rise the level more.

What's interesting is that dealing with much higher frequencies, until 580 Hz, worked well, while we could expect that it would be much more risky to change anything so high in frequency.

A possible explanation is that the reflections coming from the wall behind the speakers are affecting the sound *coming from the direction of the speakers*, which would mean that they can be safely corrected.
Another possible explanation is that it is always better to cancel peaks than to rise dips.

Overall, the final curve doesn't look much like the usual house curves seen here and there. I don't know what to think about it. But at the same time, I'm not sure that I can trust the Umik measurement microphone to this point. It can surely spot local peaks and dips, but is it really capable of reading a 2 dB imbalance between bass and treble ?


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## jtalden

Thanks for sharing. 
The common measuring mics are all pretty flat from maybe 40-4k Hz. I would also expect this to be the easiest range to calibrate also. So, if the MiniDSP cal data is fairly smooth and flat in this range then that would be normal and reassuring.
If there is a problem with the mic or the calibration then it is more likely to occur at the upper or lower frequencies.
I wouldn't be too concerned with the bass level you chose. While most prefer a stronger bass level there is large variability in preference there. I would think the room, program material and other factors would influence this so there is no right answer. The good news is you now have the facility adjust the house curve easily if you ever do decide to try something different.
Enjoy!


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## Tonto

As long as you are happy with the sound, that's all you can ask for. A couple of thoughts.



> I'm interested in any info. I've just received my MiniDSP, so I could easily perform a measurement with the correction loaded. Here is the result :
> 
> Full range target curves-19-final-ls-3.png
> 
> A huge bump at 300 Hz. I then went into new trials to see if I could get something more balanced, but if I try to remove the bump, it doesn't sound right. I just rose the low shelf from -3 dB to -2 dB to get a bit more bass below 200 Hz.


1) It was my understanding that phase needed to be checked without any correction applied.



> Name: Acoustic1.png Views: 24 Size: 9.3 KB


2) I don't see any mention of speaker "toe-in." The diagram has them pointing straight ahead. I suggest determining the best angle first, as the starting point. It will affect SS&I + room response. Should be optimized before attempting filters.



> Quote:
> 
> jtalden wrote: View Post
> 
> Your L+R measurement was done with the mic a little off-center so the HF was artificially rolled off.
> Oh, so this is the cause of all these variations that I get from a measurement to another ! I was beginning to wonder if my speakers had a different amount of treble everyday.


3) The mic should never be moved during the entire set of measurements. If anything, you could play around with small changes in mic position with each set of measurements to get a feel for needed filtering.

4) Use this information to apply absorption/diffusion to get the best room response (speaker postion is first). Then start applying filters.

I am by no means an expert on this subject, but I think these are basic principles. Please correct me if I'm wrong! :smile: 

Nice thread!


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## jtalden

Just my thoughts on a couple of points:



Tonto said:


> 1) It was my understanding that phase needed to be checked without any correction applied.


My experience is that the phase tracking in the XO range can be done prior to EQ. The impact of judicious EQ is not a big impact to the phase of a driver. It does have some effect however. After EQ is applied and the SPL response is closer to target it is easier to get a cleaner look at the final phase response of the 2 drivers. At that point we may find that a further adjustment to the timing will optimize the phase tracking. I would expect that small differences in timing in an SW XO range will not have a significant impact on the SPL or sound quality, so this is more just to fine tune the setup for the sake of it. 

In many (most?) cases of SW XO there are room modes in the XO range of at least one of the drivers and the modes often fall at different frequencies for the different main channels. This complicates the situation and often the timing selection is thus a compromise to achieve the best overall SPL response. I would think the SPL response is the main focus here and addressing phase is more to provide confidence that the best solution has been found.



> 3) The mic should never be moved during the entire set of measurements. If anything, you could play around with small changes in mic position with each set of measurements to get a feel for needed filtering.


Averaging in the LP area is a good way to helps smooth the midrange and HF SPL response in a given listening area and thus help with choosing an effective EQ solution. Whether a single LP measurement or an average measurement is used, measurements should be done one channel at a time. An SPL average of the 2 channels can be calculated in REW to address common SPL issues with a common EQ solution for both channels if that is the plan. The reason we don't measure with L+R active using sweep measurements is that the HF response is strongly impacted if the mic is even slightly off center due to phase differences. If a single measurement is taken on each channel and summed in REW it will closely agree with a measurement of L+R if the mic is exactly equal distance from the 2 channels. If the mic is off center the HF will be suppressed artificially.

If the RTA function is used for measuring with L+R operating it is likely the same or very similar to averaging the 2 channels independently and then summing them. The RTA only captures the SPL and thus the phase differences are not an issue. The RTA MMM method can be used it achieve quick and efficient averaging in an area and it provides very high repeatability.


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## Tonto

jtalden wrote:



> Averaging in the LP area is a good way to helps smooth the midrange and HF SPL response in a given listening area and thus help with choosing an effective EQ solution. Whether a single LP measurement or an average measurement is used, measurements should be done one channel at a time. An SPL average of the 2 channels can be calculated in REW to address common SPL issues with a common EQ solution for both channels if that is the plan.


Agreed, and all these things should be done first before EQ. I get the feeling our thread starter is missing these kind of basic set up priorities & jumping right into EQ. We really need to maximize his in room response 1st. What kind of treatments is he willing to get? I can't tell if he wants the room to be an average of several LP's or just one. Need to know that in order to proceed since they both have their own & different process.

Maximizing his room will save power for headroom when applying filters later.


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## AudiocRaver

Fun thread! Lots to think about.

My 2-cents worth, most of this has already been covered:

Never be afraid to try EQ whatever the state of your room and treatments.-Room treatments will almost always improve the sound, often dramatically, do as much as you can.
Do not over-do filtering. A little goes a long ways. I have been very happy with 3 to 6 bands total per side with some setups.
Do not sweat the sharp dips. You will want to push them up, until you hear how awful the result can sound. Leave them alone. I am not saying that you cannot hear them, only that you cannot hear them MUCH, and they sound less objectionably than the alternative.
All target curves are wrong. To be more accurate, no target curve is "right." Some have reasons, some depend on someone's research, all are based on assumptions which do not apply to you or your situation. Here is my own favorite target curve, which will also not apply, but here it is anyway. Try a number of "favorite target curves," then take what you think is best in them and create your own.
Speaker position, including toe-in, can make a big difference in soundstage & imaging (SS&I). Set speakers for best SS&I, which will probably cause some HF loss, then correct the high end with EQ.
Here is mine:
Hz Level
20 0
24 +3
60 +2
90 +1
120 0
1000 0
2000 +1
3000 +1
5000 0
12000 -2

Reasons:

I am not a monster bass fan. I like the sound pretty flat. But the Equal Loudness Contours certainly justify SOME bass boost, although it is compensated for already in the mix in many/most cases. A little lift below 120 Hz adds a pleasant amount of solidness to the low end without being distracting.
Any boost above 120 Hz contributes to a "boxy" sound very quickly, so the bass boost of the curve is over and down to zero by that frequency.
Having auditioned many models, most speakers have a little emphasis in the region around 1.5 to 3 kHz. The few models I remember hearing that were completely flat through that range sounded drab and lifeless. I little lift, only a dB or so, really livens the sound, yet is small enough to still consider the sound very accurate. Note that this much HF energy will sound harsh with tweeters that are not super clean, super fast, and super smooth in their response. I believe it is tweeter designs rather than acoustics that have listeners shying from extended high frequencies most of the time. If you do not have great tweeters, ditch your speakers and get some that do. You will be glad for many years to come.
The 2 dB drop from 5 kHz to 12 kHz takes the edge off of the highs, reducing listener fatigue, yet still sounds very flat.
The result still falls within a +/- 3 dB tolerance, so can still be considered flat by all but mastering lab standards.


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## Pio2001

Sorry, I didn't see this page of the discussion !



Tonto said:


> 2) I don't see any mention of speaker "toe-in." The diagram has them pointing straight ahead. I suggest determining the best angle first, as the starting point. It will affect SS&I + room response. Should be optimized before attempting filters.


The diagram is wrong. I setup the toe-in quite early in the process so as to get a large stereo image, although the speakers are closer to each other than they should be if the rule of the equilateral triangle would have been followed.
All trials were done after the toe-in was fixed. I can't remember if the whole set of measurements was done after, or before. All that I know is that if the measurements were done with a different toe-in, then I have checked the effect with additional measurements.
I did some additional measurements, at least to test the effect of the shallow table standing before the sofa (mostly nothing, just a little peak in the mids, among dozens of other ones), of a different speaker position, of a different setup of the acoustic controls on the monitor, and to compare left+right vs left alone and right alone, but i don't remember if I tested the toe-in.



Tonto said:


> 3) The mic should never be moved during the entire set of measurements. If anything, you could play around with small changes in mic position with each set of measurements to get a feel for needed filtering.





Tonto said:


> We really need to maximize his in room response 1st. What kind of treatments is he willing to get? I can't tell if he wants the room to be an average of several LP's or just one. Need to know that in order to proceed since they both have their own & different process.


Following the advice of a french professional, I made 6 sets of 3 measurements (left speaker, right speaker, both speakers), from 6 different locations : at the listening position, laid back (above the back of the sofa), at the listening position, bent forward (above the front side of the sofa), 40 cm on the left (back and front), 40cm on the right (back and front). Then I made two additional control measurements, one 15 dB SPL below, to check the effect of background noise on the measurement (none, the result was nearly the same), and another one identical to the first one, to check the stability of the measurement chain over time (none, the last measurement was mostly identical to the first one).
From these 20 measurements, I made an average of the 6 ones from the left speaker alone, the 6 ones from the right speaker alone, and the 6 ones with both speakers.
I made two separate sets of filters, a stereo one from the left/right averages, and a mono one, from the average of both speakers. And listened to the result. The difference was small compared to the adjustments that could be made to the target curve.
I decided to go for a mono correction because, listening to sweep tones, the stereo filters introduced localized imbalances in the sound at some given frequencies, but before all because I've read that because of phase issues, in low frequencies, the total SPL of both speakers is by no means equal to the sum of the SPL of each speaker playing alone. Since low frequencies are more often mixed in mono than in stereo, starting from the left+right measurements should lead to a better result in low frequencies.



jtalden said:


> My experience is that the phase tracking in the XO range can be done prior to EQ. The impact of judicious EQ is not a big impact to the phase of a driver. It does have some effect however. After EQ is applied and the SPL response is closer to target it is easier to get a cleaner look at the final phase response of the 2 drivers.


Wow, I didn't think about that. Fortunately, I did no correction around 2 kHz, where my crossover is working.



jtalden said:


> In many (most?) cases of SW XO there are room modes in the XO range of at least one of the drivers and the modes often fall at different frequencies for the different main channels. This complicates the situation and often the timing selection is thus a compromise to achieve the best overall SPL response.


Since I have no sub, I can avoid this problem too 



jtalden said:


> The reason we don't measure with L+R active using sweep measurements is that the HF response is strongly impacted if the mic is even slightly off center due to phase differences. If a single measurement is taken on each channel and summed in REW it will closely agree with a measurement of L+R if the mic is exactly equal distance from the 2 channels. If the mic is off center the HF will be suppressed artificially.


Yes, I experienced this, and didn't understand what was happening. It was you who gave the answer, above in this discussion. 

Although the measured response in high frequencies is not very smooth, I decided not to correct it. The speaker, measured by two independent reviewers, are supposed to be flat from 50 Hz to 20 kHz, and I rather trust these measurements, done with calibrated mics, than mine, done with the Umik. 

But here is an interesting point : if the room itself causes accidents in the mids or highs (measured by myself at the listening location, but not by the reviewers), should they be corrected ?
It all depends on the way we define "high fidelity". If the goal is to have the feeling that the musician have come into the living room, then the acoustics of the room should not be corrected. Only the native speaker response (and its development in the 3d space all around) should be, except in the lows, where even the musicians would protest about the sound quality of the room before starting to play.
If on the other hand the goal is to have the feeling that we left the living room to be transported to the concert hall, then the room acoustics should be corrected together with the speakers response.



AudiocRaver said:


> [*]Do not sweat the sharp dips. You will want to push them up, until you hear how awful the result can sound. Leave them alone. I am not saying that you cannot hear them, only that you cannot hear them MUCH, and they sound less objectionably than the alternative.


Good advice. For shallow dips in low frequencies, I found a trick : I set a low shelf filter that boosts all the lows, and adjust the nearby peaks equalization in accordance. The result is that all the frequency zone between the peaks is raised in a way that can't be done with a parametric filter.



AudiocRaver said:


> [*]All target curves are wrong. To be more accurate, no target curve is "right." Some have reasons, some depend on someone's research, all are based on assumptions which do not apply to you or your situation. Here is my own favorite target curve, which will also not apply, but here it is anyway. Try a number of "favorite target curves," then take what you think is best in them and create your own.


Thanks. Eventually, I lost the idea of a "target curve". All I did was 
1-Adjust the treble setting on the speakers themselves (affects the frequencies above 8 kHz, the manufacturer says that this control is useful if the acoustics of the rooms are overly damped, or bright).
2-Cancel the two first room modes at 55 and 70 Hz (according to the SPL measured level at 1 kHz for the time being).
3-Equalize carefully the acoustic resonance with the wall behind the speakers : reduce the level at 200, 330 and 580 Hz, boost the level at 100 Hz.
4-Adjust the subjective bass / treble balance by ear, using a low shelf at 1000 Hz (my set of filters act over the whole 50-1000 Hz range).
5-Adjust the 55 and 70 Hz filters again in the continuity of the rest.

In the end, the "target curve" was decided during step 3, when I began to try the two filters proposed by jtLaden, and a third one, and fixed their values listening to a playlist of various recordings with voices.









Filters 1, 2 : room modes
Filters 3, 4, 5, 6, 7, 8 : acoustic interferences between speakers and wall
Filter 9 : personal house curve


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## Tonto

Sorry, I kinda lost track of this conversation. Are there any kind of treatments in the room (don't remember & haven't gone back to verify). It would be appropriate to treat the room before any eq (absorption & diffusion). Play with this until you get it the best you can & then eq off the final measurement. Sounds like you are getting a handle on it. And I think we all feel that treating/eq'ing the room such that it minimizes the affect it has on the speakers frequency response is the goal. Nice thread.


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## Pio2001

Sorry. No there are no treatments in the room.

The speaker placement was decided before setting up the furniture, long before I started equalizing, when I moved in this flat (I had different speakers at that time, a pair of modified Dynaudio Gemini). 
There were only the empty shelves, tables, chairs, a single armchair, and a lot of unpacked boxes all around. The first thing I did was setting up the CD player, amplifier, and speakers on their stands, and try all the dispositions in all the rooms.

The best sound was in the living room, by far. But in this room, anything, speaker or listener, placed in a given half, produced terrible resonances in the low frequencies, so both speakers and listener positions were chosen to be in the other half of the room. The direction was decided so that people sitting in front of the speakers could see the window and the trees outside. The good thing is that this way, there is no wall behind the listeners.

The first position for the speakers was 25 cm ahead of the current one, farther away (about 1 meter) from the wall... that is not a wall, but a large window, in fact. The 55 and 70 Hz resonances were weaker.
But when I got the Neumann speakers, the manual stated that they should not be positioned more than 80 cm away from the wall, to avoid acoustic interferences. With a native response of 0 dB at 50 Hz, and this close to the wall, the 55 / 70 Hz peaks were too strong, so I decided to equalize. 
I managed to manually cancel them without a measurement microphone, using a sine generator to find the right frequencies to correct, and creating a convolution file with Rephase to load in Foobar2000. But the dip at 100 Hz didn't sound right, and other audio sources (Youtube, Blu-rays, CD Player) could not be equalized. That's where I discovered the world of digital equalization, and went for the Umik and the MiniDSP.

Now that I can equalize, I could move the speakers ahead again, but all the filters from 3 to 8 would have to be redone, and that's too much work. Maybe the peaks number 7 and 8 would just have to be translated to the left, but the peak 5 is more complicated. It is an average of two different peaks at two different frequencies on the left and right channels, because the left speaker is near a lateral solid wall, and not the right one, and the 100-200 Hz zone is a complete mess anyway.

Setting up the 55 / 70 Hz cancellation is a piece of cake. REW does it automatically and it does it right too. But finding a proper, naturally balanced correction above 100 Hz is an awful lot of work ! It took me 40 days, and around 100 hours of measurements, trials and critical listening to get to this point, and I won't get through the whole process again this year !

About acoustic treatments on the walls, as long as the speakers don't move, there should be no need of resetting completely the low-mid filters. But I'm a tenant, and the physical modifications I can make to the walls are very limited. 
Besides, I'm just allergic to manual work :mooooh:


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## AudiocRaver

Midrange response problems are often the result of reflections. EQ will not help, only reflection control.


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## ajinfla

Pio2001 said:


> But here is an interesting point : if the room itself causes accidents in the mids or highs (measured by myself at the listening location, but not by the reviewers), should they be corrected ?


Only if you trust single mic pressure readings by your eyes over your 2 ears/brain.
Science has found that the latter is not the same as the former. There is a great deal of "listen through" and subconscious adaptation involved that doesn't show up in pressure readings.
It has also found that if your onset (free field response) is smooth over a wide arc, the "room" diminishes as a factor. So just like the sound of a relative or a Steinway sounds like a Steinway in your living room, a small lounge or a concert hall, a good loudspeaker remains "good" in varying environments (outside reductio absurdum). At least to ears. YMMV.



Pio2001 said:


> It all depends on the way we define "high fidelity".


Or "preference".

cheers


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