# EQ Above 200-300hz



## jcmusic

I am curious as to why I have read many times over, you should not EQ any frequency's over the 200-300hz range??? Can someone explain???


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## Wayne A. Pflughaupt

If I could write a book on equalizing myths, this one would be in it. There only a couple of potential issues with equalizing above ~400 Hz or so. One I’ve seen is that you need to use matching filters for the left and right channels, otherwise it can do weird things to your imaging (of course, that would not apply to equalization for center or surround channels). However, when I made that determination I was using 1/3-octave equalizers, which are not nearly as precise as a parametric EQ. With a parametric it’s likely possible to push that threshold higher. Second, you shouldn’t try to equalize every little ripple in response, but only broader-bandwidth problems that would be audible, typically corrected with filters 1/3-octave or wider.

Regards, 
Wayne


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## blitzer

Sanity Check:

So the issue is that the wavelengths get so small that movements of an inch can make large differences in the notches you seen in the response graph. Nulls/nodes can happen from inch to inch instead of areas of the room with sub bass. So super tiny movements in microphone can give much different response graph notches the higher in frequency you go. Also, given that you are measuring with a single microphone and not 2 at the exact same spots as your ears, so right there is a problem. You are saying that at about 400Hz - this comes into play, or are there other considerations as well?


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## tonyvdb

Wayne is correct, adding any EQ adjustments above 400Hz that do not mach perfectly can mess up imaging. Even Adyssey is not great in that regard.


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## blitzer

It seems to me that Audyssey applies different curves to different speakers in my setup. Does Audyssey really do this knowing what can happen to imaging?


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## tonyvdb

blitzer said:


> Does Audyssey really do this knowing what can happen to imaging?


I dont think they care, (its not limited to Audyssey either all room correction formats do this). thats why its sometimes better to do all readings in a box about 2ft square around the main listening position rather than much farther apart.


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## blitzer

In thinking about it. Seems Audyssey's goals to to make movies sound better. Since the center channel centers the sound, you are right, why would they care? 

This seems like a gigantic hole in the value of these products for 2 channel listening, especially the ones that specialize on 2 channels. Seems that for 2 channel stereo, you should never check speakers individually....


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## jtalden

jcmusic said:


> I am curious as to why I have read many times over, you should not EQ any frequency's over the 200-300hz range??? Can someone explain???


It is important to differentiate between EQ to adjust the direct signal verses EQ to try to compensated for room effects.

I hope that even those that feel it is never good to EQ HF for room effects still would agree that there is nothing wrong in adjusting the direct signal response to fit a desired target - whether that is a flat SPL at 1m or some house curve (preference). After all, adjusting the direct signal response is exactly what is being done when the selection of drivers and the passive XO design is being done by a commercial speaker manufacturer.

Your posted SPL response indicates there is room for significant improvement by reducing the excess HF SPL in the direct signal. We just can tell you what the best target is as that is dependent on many factors.

Once we move past that into compensating for some room effects then the opinions get much more diverse:
> Never do it.
> You can do it with the following constraints...
> Best to convolve using the inverted IR taken at the LP so the resulting SPL and phase measures flat at the LP.


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## primetimeguy

blitzer said:


> In thinking about it. Seems Audyssey's goals to to make movies sound better. Since the center channel centers the sound, you are right, why would they care?
> 
> This seems like a gigantic hole in the value of these products for 2 channel listening, especially the ones that specialize on 2 channels. Seems that for 2 channel stereo, you should never check speakers individually....


I guess I don't see the harm just because of 2 channel listening. With no EQ are you getting the exact same frequency response from both Left and Right speakers? Doubtful because of room effects.


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## tonyvdb

primetimeguy said:


> With no EQ are you getting the exact same frequency response from both Left and Right speakers? Doubtful because of room effects.


At the speaker yes, thats why room treatment and speaker placement is so important. You dont know what your missing if you have not heard good imaging from speakers.


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## Wayne A. Pflughaupt

blitzer said:


> So the issue is that the wavelengths get so small that movements of an inch can make large differences in the notches you seen in the response graph. Nulls/nodes can happen from inch to inch instead of areas of the room with sub bass. So super tiny movements in microphone can give much different response graph notches the higher in frequency you go. Also, given that you are measuring with a single microphone and not 2 at the exact same spots as your ears, so right there is a problem. You are saying that at about 400Hz - this comes into play, or are there other considerations as well?


For the main channels, above or below 400 Hz, we aren’t trying to equalize all those little notches and ripples in response. We’re only interested in addressing significant trends in response that would sound better if improved. As perhaps an extreme “for instance,” the broad range between 1 – 10 kHz peaking at 4 kHz, in this graph of the OP’s response from another thread. A broad trend like that isn’t going to appreciably change if you move the mic a few inches one way or the other.












jtalden said:


> It is important to differentiate between EQ to adjust the direct signal verses EQ to try to compensated for room effects.


Exactly! Above ~400 Hz, it is the direct signal you’re dealing with via equalization. Below that point the room starts to greatly affect response, as we know. I intend to put this to the test someday, but I imagine that the transition frequency is higher in “live” rooms than it is in treated rooms, and for normal listening distances compared to near-field. Wouldn’t be hard to determine, all you need is something like 1/3-octave filtered pink noise test tones: Descending frequencies from say 2 kHz down, the point where you can no longer pinpoint the sound source is the transition point for your room. I have such a test disc; all I need is some time. 

It’s also above the transition frequency that we perceive stereo imaging, which is why you have to be careful applying separate, per-channel filtering above that point. Filters introduce unnatural phase changes at the adjusted frequencies (“unnatural” in that it’s phase that’s not introduced by normal room reflections), and that’s why independent filtering above 400 Hz or so can “whack” imaging. Thus above the transition frequency filters probably should be matching. Naturally, this only applies to the front stereo pair, not the center or surround channels, where imaging is not an issue.

Regards, 
Wayne


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## ajinfla

jcmusic said:


> I am curious as to why I have read many times over, you should not EQ any frequency's over the 200-300hz range??? Can someone explain???


Because a single mic that captures pressure at that single point, is quite different from your two spaced ears with a head in between.
..and it's not that you _should not_, but rather, use caution and understanding if you do.

cheers


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## ajinfla

jtalden said:


> I hope that even those that feel it is never good to EQ HF for room effects still would agree that there is nothing wrong in adjusting the direct signal response to fit a desired target - whether that is a flat SPL at 1m or some house curve (preference). After all, adjusting the direct signal response is exactly what is being done when the selection of drivers and the passive XO design is being done by a commercial speaker manufacturer.


True, all "EQ" should have taken place during design, though we can never escape the preference thing, be it designer, or user, "voicing".
When people speak of "EQ", I always think of sound power, which is a speaker/room issue.



jtalden said:


> Your posted SPL response...


In another thread??




jtalden said:


> Once we move past that into compensating for some room effects then the opinions get much more diverse:
> > Never do it.
> > You can do it with the following constraints...
> > Best to convolve using the inverted IR taken at the LP so the resulting SPL and phase measures flat at the LP.


Or, >Do as you prefer...

cheers


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## ajinfla

Wayne A. Pflughaupt said:


> I intend to put this to the test someday, but I imagine that the transition frequency is higher in “live” rooms than it is in treated rooms, and for normal listening distances compared to near-field. Wouldn’t be hard to determine, all you need is something like 1/3-octave filtered pink noise test tones: Descending frequencies from say 2 kHz down, the point where you can no longer pinpoint the sound source is the transition point for your room. I have such a test disc; all I need is some time.


Hi Wayne,
I would not use such a signal for such a test. Perhaps a low pass filtered click like Greisinger suggests here or other impulsive signal with a sharp onset.

cheers


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## jtalden

ajinfla said:


> True, all "EQ" should have taken place during design, though we can never escape the preference thing, be it designer, or user, "voicing".
> When people speak of "EQ", I always think of sound power, which is a speaker/room issue.


Yep, I agree. "EQ" is not a narrow concept. It covers too broad a range to be understood easily in context.

Just to expand:
The context is important - is this a commercial system? Is the skill of the manufacturer such that no EQ on the direct signal will improve the situation? Is it a DIY? My impression was that the OP's mains are DIY, or possibly a modified commercial design. Even if is not, is the original design optimized as an ideal system would be?

I did a DIY and could post my 1m "as found" response - it's pretty sad. :sad2: As the designer however it is my responsibility to correct it. Since it is an active system I can do that with XO, delays, levels and EQ. To EQ in this context is doing the "initial" design. 

I am not disagreeing with any of the above advice. I instead just wanting to point out that if it is a DIY or modified design then "EQ" has a different meaning. Many non-DIY users owning a modern idealize commercial design do not quickly consider this situation and thus comments like "no HF EQ should ever be done" shows up. These comments help to exacerbate the confusion over whether to EQ or not. DIY, older, or more modest systems can be improved even given the overall position is that "no HF EQ should be done". That position assumes that the proper EQ has already been done!

The OP"s SPL is now posted by Wayne above. It is pretty obvious that this is not the expected result of a good commercial design in a typical room. Although, it is very difficult to determine what is possible especially given no info on the actual situation.


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## Wayne A. Pflughaupt

Hey AJ, thanks for commenting!




ajinfla said:


> I would not use such a signal for such a test. Perhaps a low pass filtered click like Greisinger suggests here or other impulsive signal with a sharp onset.


Don’t make me read the whole thing!  On what page does he discuss how to determine what the transition frequency is in your room (that was the point in my quote that you highlighted, after all)? I’ll be surprised if that’s even discussed, since that isn’t exactly something that’s commonly discussed, recommended, or done.

Unless one has a whole slew of clicks low-passed at different frequencies, you aren’t going to be able to figure out at what point your room transitions. Besides, filtered pink noise _is_ low-passed, and high passed as well, for a specific frequency. Maybe you’ve never heard of filtered pink noise – it’s not a common test signal – but it’s like frequency-specific sine waves, only pink noise. Pink noise would be a better test signal than sine waves for trying to figure out a room’s transition point, because it's a closer representation of program material. For instance, it’s difficult to localize a 1 kHz signal with a sine wave, but easy with 1 kHz pink noise. And I don’t see how an instantaneous impulse signal would be useful for this, as I imagine it will be harder – and take longer – for someone to localize the signal the closer to the transition frequency it is.

Regards, 
Wayne


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## ajinfla

Wayne A. Pflughaupt said:


> Hey AJ, thanks for commenting!


Probably a bad idea doing so from "work" .
I most likely misunderstood given your response. What is this "direct" signal you speak of? My confusion is most likely because the OP is following up on some previous thread. I now gather the measurement you show above is his. Is it nearfield (say 1m), or at the LP?
Maybe I should be asking him?

(Given what I see, I sure hope it's nearfield!)

cheers


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## jcmusic

I want to thank all for the replys and info, the measurement is from the LP. I thought if kinda strange to not EQ something like my curve is showing, how else would you be able to get a flat response?


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## tonyvdb

If you have a true digital EQ like the Yamaha YDP2006 you can apply equal filters to the same frequency on each channel and have no issues.


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## HifiZine

Hello there, I have an article in preparation that is somewhat around this issue and so I'd like to ask a question or two... (AJ is on my short list for reviewers but it was a while ago that I mentioned it and I didn't phrase it that way... in fact I didn't realize at the time...)

I'd like to ask AJ about the statement "all EQ should take place at design time". For one thing, we have the speakers that we have, and replacing all of them isn't always a very realistic position if we discovered some issue with them. But more to the point, we often read about a designer "voicing" a speaker, which is supposed to transcend the mere act of measuring them. Can we, as end-users, not have a hand in "voicing" our speakers for ourselves?


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## ajinfla

jcmusic said:


> I want to thank all for the replys and info, the measurement is from the LP. I thought if kinda strange to not EQ something like my curve is showing, how else would you be able to get a flat response?


It's not strange, because your curve is highly atypical. It looks like a system where the tweeter is waaay too hot. That isn't where the general recommendation not to eq > 300hz comes from. Caveats are required.
In your scenario, it's absolutely mandatory...or a proper crossover is needed.

cheers


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## ajinfla

HifiZine said:


> I'd like to ask AJ about the statement "all EQ should take place at design time".


Hi John,

It needs caveats and frame of reference (much like my "Treatments" position, etc.). I mean >500hz or so (similar to the thread topic)...with speakers having natively smooth on and off axis response (>500hz), in a "typical" room (no glass closet, iso-ward, etc.), with typical program material (not unbearable bright/dull, etc.), etc, etc.
So that...



HifiZine said:


> For one thing, we have the speakers that we have, and replacing all of them isn't always a very realistic position if we discovered some issue with them.


..and the recording we like, etc.
So yes, as I stated previously, that and plain 'ol "preference" should guide one as well. I speak of generally, rather than specifically.




HifiZine said:


> But more to the point, we often read about a designer "voicing" a speaker, which is supposed to transcend the mere act of measuring them. Can we, as end-users, not have a hand in "voicing" our speakers for ourselves?


Absolutely! The EQ police won't come take you away if you decide a speaker, speaker/room or recording is too bright/dull/forward, etc., last I checked.
As for *designer* "voicing", I'm very aware of it, though I choose not to myself. I prefer to let my own measured targets "voice" the loudspeaker. That is, as I get older and deafer, my designs will(should) sound same-ish. To each their own of course.

cheers


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## jcmusic

ajinfla said:


> It's not strange, because your curve is highly atypical. It looks like a system where the tweeter is waaay too hot. That isn't where the general recommendation not to eq > 300hz comes from. Caveats are required.
> In your scenario, it's absolutely mandatory...or a proper crossover is needed.
> 
> cheers


Ok thanks AJ I was just trying to find out why some people say not to Eq above a certain frequency!!!


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## ajinfla

jcmusic said:


> Ok thanks AJ I was just trying to find out why some people say not to Eq above a certain frequency!!!


No prob jc. I can't really speak for those people, but my advice is to first eq the native response of your system (perhaps this is what Wayne meant by "Direct" response?). Ideally anechoic (@ 2m), or outside, but if not, with a quasi-anechoic gated measurement, from at least 1m on the so called "design axis".
What we hear at the LP (where your measurement is from), is a combination of the native onset response (the first to physically reach your ear), followed by the delayed responses, the sum total being something like what you measured...except by two ears and a head connected to a brain, that can detect more than pressure at a single mic position.
Once you have "corrected" the native system response, listen, then, if necessary, apply the "LP" EQ response, to your preference.
My 2c.

cheers


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## jcmusic

Ok so what you are saying is i should take nearfield measurement of both speakers and eq them evenly if needed. Then eq from the LP correct?


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## Wayne A. Pflughaupt

That’s what I’m hearing. I’ve never tried that technique before but it sounds reasonable. The idea is that above the transition frequency – about 4-500 Hz – it’s the direct signal that dominates, so you’re equalizing the speakers’ signal with matching filters for both channels. Below that point the room starts to dominate, so you equalize that from the listening position. Down in this region separate (dedicated) filters can be used for each speaker.

Regards, 
Wayne


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## primetimeguy

Sounds interesting but I'm still struggling with why it makes sense, if it does at all. Would it not be any different than putting your ear next to the speaker and adjusting the treble until you like it and then just accepting how that sounds at the LP even if you don't like it?


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## Wayne A. Pflughaupt

Good point! That brings to mind something I’d forgotten, that high frequencies attenuate over distance. So it makes no sense to EQ based on a mic placed near field. That’s going to result in a measurement with exaggerated highs compared to what you’d be hearing at the listening position. If you equalize based on that, you’d attenuate the highs, which will result in things sounding really soft at the listening position. So I retract the statement I made in my previous post – I’m sticking with the “tried and true” method of measuring at and equalizing for the listening position. :T

Regards, 
Wayne


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## ajinfla

jcmusic said:


> Ok so what you are saying is i should take nearfield measurement of both speakers and eq them evenly if needed.


Yes. Decide what the design axis is (with your multi-driver speaker), then judiciously flatten the design axis response (be careful with sharp notches, don't try to fill them), while monitoring the off axis to see if it is staying relatively consistent. Try to make the interchannel (L-R) responses as close as possible. Listen.



jcmusic said:


> Then eq from the LP correct?


Mainly at LF (<500hz) and if necessary >500hz, after listening, yes. Then listen some more. Suit your preference.

cheers


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## jcmusic

Guys Iam willing to try a few things but, I don't want to waste a bunch of time if I can help it. There seems to be a bit of confusing advice here, can we get it sorted out before I go down the wrong road???


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## primetimeguy

In your case I think I would start with near field measurements and sort out what is wrong with your speaker or measurement system. I say that because a boost by that much at the high frequencies does not seem normal nor have I seen a plot like that before. So use near field measurements to try and sort that out first.


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## ajinfla

primetimeguy said:


> Sounds interesting but I'm still struggling with why it makes sense, if it does at all. Would it not be any different than putting your ear next to the speaker and adjusting the treble until you like it and then just accepting how that sounds at the LP even if you don't like it?


Not saying that you don't EQ at LP. I'm saying you fix the source first, thus fixing the onset response (or "direct" or whatever you want to call it). We are talking measurements here, so there is a "target" - "flat", or amplitude free of major deviations.
Once you move back in the room (to LP), the mic is now capturing both onset and room (the sum total pressure at that point), the sound power gives you the overall spectral balance...but it is not exactly what the ear "hears". So now what is the "target" response? If you EQ to "flat", the ear hears anything but! the sound should be unbearably bright. Propagation loss/room absorption is higher for HF. Measure at the source now...and the response is highly tilted up. Mic says "flat", ears say..NO! They hear that tilted up onset right through the "flat" pressure at the LP.
Fix the source...then tweak the LP. And then there is <500hz.

cheers


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## ajinfla

Wayne A. Pflughaupt said:


> Good point! That brings to mind something I’d forgotten, that high frequencies attenuate over distance. So it makes no sense to EQ based on a mic placed near field. That’s going to result in a measurement with exaggerated highs compared to what you’d be hearing at the listening position. If you equalize based on that, you’d attenuate the highs, which will result in things sounding really soft at the listening position. So I retract the statement I made in my previous post – I’m sticking with the “tried and true” method of measuring at and equalizing for the listening position. :T
> 
> Regards,
> Wayne


Hi Wayne,

See my post above, as I covered most of it. I am talking about EQing a _measured_ response target nearfield (again, depending on the 'design axis"). I think you may have it backwards above, unless I misread. Yes, there will be prop losses/absorption, so at the LP, if the "source" response is "flat" (both on and off axis [relative to on]), you should see a gradually tapered response at the LP.
Now you decide your own "house curve"...with your ears. Is it the B&K target curve (measured)? Or...?
That is where preference should come in.

cheers


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## ajinfla

jcmusic said:


> Guys Iam willing to try a few things but, I don't want to waste a bunch of time if I can help it. There seems to be a bit of confusing advice here, can we get it sorted out before I go down the wrong road???


Hi jc,

Of course you could just twist the knobs until you're happy! As I mentioned before, there is no EQ police that I know of.
I'm simply giving you my opinion, of how I would approach things, with a method in mind. YMMV.

cheers


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## jcmusic

Well thanks i am looking for a point to start at, the music already sounds good but i think it can sound better. Plus i want to know what the room is doing to the sound, so i think i am gonna try your method and see where it takes me.


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## skinnydoggy

I agree with this article... This may be of interest

http://www.soundandvision.com/content/rethinking-room-correction

I noticed when eqing higher frequencies, I hated the results. You can make beautiful neutral speakers sound fatiguing. What made the most difference to me was the bass...


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## Wayne A. Pflughaupt

Equalizing the main channels is tricky compared to the bass; it’s easy to make a mess of it if you don’t know how to do it properly. Plenty of people here have pulled it off and have been pleased with the results – here’s one case:

Spridle’s Experiment

Regards, 
Wayne


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## ajinfla

skinnydoggy said:


> I agree with this article... This may be of interest
> http://www.soundandvision.com/content/rethinking-room-correction


Nice. 
Laurie Fincham was very influential to my philosophy.

cheers


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## jcmusic

Ok guys I finally got off work before 9:00 pm so maybe these curves will help. I took these the same day as the one posted in my other thread, these are from the LP of each speaker and then both together flat no eq or subs. The red is left speaker, the green is right speaker, the purple is both together.


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## Wayne A. Pflughaupt

ajinfla said:


> Hi Wayne,
> 
> See my post above, as I covered most of it. I am talking about EQing a _measured_ response target nearfield (again, depending on the 'design axis"). I think you may have it backwards above, unless I misread.


I understand what you’re recommending - I just don’t see equalizing based on nearfield measurements as a “best practices” method. A speaker isn’t going to sound - or measure - the same at 3 ft. as it will at the typical 10-12+ ft. listening position – not even in an anechoic environment. 

Regards, 
Wayne


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## Wayne A. Pflughaupt

jcmusic said:


> Guys Iam willing to try a few things but, I don't want to waste a bunch of time if I can help it. There seems to be a bit of confusing advice here, can we get it sorted out before I go down the wrong road???


As AJ rightly indicated, there is no consensus on how to EQ, and indeed you might not be happy with the results you get following either of our suggestions! So, experiment. Ultimately, the results should net you an audible improvement in sound quality. If not, then try a different approach. There’s always a chance that you’ll think any equalization doesn’t sound as good as unequalized. 

Regards, 
Wayne


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## jcmusic

Hi Wayne,
Well all of this started because I was not happy with the boomy bass I had in my system. So i decided to get the gear to measure the room and acoustic treatment to go with it. All this was a learning expirence for me.
Once i got rid of the boomy bass I wanted to take it to the next level, now before i saw any of the measurements I loved the way the system sounded other than boomy bass. Now it more of a quest to get a good looking curve and have it sound god too!!! So that's where I am just trying get the ost out of the system...


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## Wayne A. Pflughaupt

jcmusic said:


> Ok guys I finally got off work before 9:00 pm so maybe these curves will help. I took these the same day as the one posted in my other thread, these are from the LP of each speaker and then both together flat no eq or subs. The red is left speaker, the green is right speaker, the purple is both together.


Regarding the red and green graphs, I think a filter at ~4000 Hz, with a bandwidth of ~1-octave or a bit less, and cut ~6 dB would make a significant improvement in SQ. 

As for as the purple graph, notice the overall loss of signal below 1000 Hz. That tells me your speakers are out of phase – check your speaker wiring to make sure the polarity isn’t reversed on one of them!

Regards, 
Wayne


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## jcmusic

Wayne A. Pflughaupt said:


> Regarding the red and green graphs, I think a filter at ~4000 Hz, with a bandwidth of ~1-octave or a bit less, and cut ~6 dB would make a significant improvement in SQ.
> 
> As for as the purple graph, notice the overall loss of signal below 1000 Hz. That tells me your speakers are out of phase – check your speaker wiring to make sure the polarity isn’t reversed on one of them!
> 
> Regards,
> Wayne


 OK Wayne,
I am confused by this, at 4000hz that looks to be a valley not a peak. Why would I want to cut that??? Also i check all the connections they are all correct!!!


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## HifiZine

Hi, you must be looking at the narrow notch, Wayne is referring to the overall broad peak ("bandwidth of ~1-octave or a bit less").


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## jtalden

jcmusic,
You may find it helpful to EQ to one of the "House Curves" that are commonly used. The B&K curve is often mentioned a good starting point for music listening. There are others that can be found as well. Your current curve is a very large departure from any of these common recommendations and, since you asked, we are all suggesting you try something more conventional to see if you like it. There is no guarantee that you will chose one of the common recommendations as your favorite as very situation is different. 

You can set a target house curve into REW and have REW automatically calculate the needed filters or you can manually adjust the filters within the REW's "EQ Filters" popup. REW will predict the new response overlaid with the target and the current response so that you can easily see the expected change. When the filter are implemented in your hardware the actual measurements will closely agree with the REW prediction. This makes any adjustments relatively quick and easy. The bigger problem will be deciding on which target you prefer. 

Good luck!


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## ajinfla

Wayne A. Pflughaupt said:


> I understand what you’re recommending - I just don’t see equalizing based on nearfield measurements as a “best practices” method.


How do you think DSP studio monitors end up with such smooth native response? What method do you think is used?



Wayne A. Pflughaupt said:


> A speaker isn’t going to sound - or measure - the same at 3 ft. as it will at the typical 10-12+ ft. listening position – not even in an anechoic environment.
> 
> Regards,
> Wayne


Unless it's a low diffraction point source in mono (anechoic). But yes, that's true 99% of the time...and in complete agreement with what I'm saying.
So what specific target (measured) curve does he EQ (measured) to at the LP Wayne? 
If it's just twist knobs to his ears (audible) "preference", exactly what do we need measurements for?

cheers


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## jcmusic

HifiZine said:


> Hi, you must be looking at the narrow notch, Wayne is referring to the overall broad peak ("bandwidth of ~1-octave or a bit less").


I am still not getting it!!!


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## primetimeguy

jcmusic said:


> I am still not getting it!!!


The range from 2.5-5khz is high in relation to everything else. So a broad drop in that range would probably be preferrable.

Do we know that your mic and measurement system are accurate? I would think that response curve would be very bright and almost painful to listen to. Voices would be very harsh.


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## jcmusic

Well the mic has a calibrated file and everything else seems to be correct. Now you say broad explain how this is to be done?


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## Wayne A. Pflughaupt

jcmusic said:


> Wayne A. Pflughaupt said:
> 
> 
> 
> As for as the purple graph, notice the overall loss of signal below 1000 Hz. That tells me your speakers are out of phase – check your speaker wiring to make sure the polarity isn’t reversed on one of them!
> 
> 
> 
> Also i check all the connections they are all correct!!!
Click to expand...

Well that’s curious. Try playing a song with some heavy bass, and disconnect one of the speakers. The bass level should drop. If it gets _louder_, then you definitely have something wired wrong.



jcmusic said:


> HifiZine said:
> 
> 
> 
> Hi, you must be looking at the narrow notch, Wayne is referring to the overall broad peak ("bandwidth of ~1-octave or a bit less").
> 
> 
> 
> I am still not getting it!!!
Click to expand...

As they say, a picture is worth a thousand words. As I think I mentioned somewhere in this thread (if I didn’t I should have), with the main channels we’re interested not with all the little ripples in response, but bigger issues that are significant enough to make a negative impact. You have a big, broad hump there in the 3-4 kHz range that could easily be tamed with a single broad filter. Get the idea? 







​

Regards, 
Wayne


----------



## jcmusic

Hey Wayne,
Thanks replying i will try playing a bass heavy song and see what happens. Also i understand about the broadness correction just not sure how to do it.


----------



## Wayne A. Pflughaupt

jcmusic said:


> Also i understand about the broadness correction just not sure how to do it.


 I gave the basic filter parameters in Post #43:




Wayne A. Pflughaupt said:


> Regarding the red and green graphs, I think a filter at ~*4000 Hz*, with a bandwidth of *~1-octave* or a bit less, and *cut ~6 dB* would make a significant improvement in SQ.


I actually based that filter recommendation on what I see in your combined (purple) graph because (as mentioned previously, I think) you want matching filters for _both_ channels up in that frequency range. Your Yamaha YDP2006 equalizer doesn’t give filter bandwidth in octaves, but Q; a setting of 1.4 Q would be the equivalent of a one-octave filter.

It’s really tough to do this with sweeps; I’d recommend using REW’s RTA feature. It’ll play a continuous pink noise signal while you equalize. Start with filter at 0 dB, and apply negative gain and you should see that hump flatten out. You might have to shift the filter frequency back and forth a bit, and/or the bandwidth as well, to accurately accomplish the objective. You want to stop cutting at the point where the hump is roughly in line with the rest of response. 

If you see the areas beyond the hump (i.e. in front of and behind it, such as at 600 Hz and 10 kHz) start dropping, then you've cut too far. As noted above, it probably won't take more than about a 6 dB cut, but that's just my best guess: If it takes more or less, no problem, that's perfectly fine. 

If the areas beyond the hump start dropping before you get the hump flattened, then the filter is too wide. 

If only one side or the other of the areas beyond the hump start dropping before you get the hump flattened, (like if it starts dropping at around 600 Hz, but not up at 10 kHz), then the center frequency is off and needs to be adjusted.

Keep in mind this will sound drastically different from what you’re used to hearing; it would be a good idea to live with it for a while to get acclimated to the sound. After several days, you might try switching that filter off, and see what it sounds like to you then. At that time you should be able to decide if you think the filter made an improvement in SQ or not.

Regards, 
Wayne


----------



## Wayne A. Pflughaupt

ajinfla said:


> How do you think DSP studio monitors end up with such smooth native response? What method do you think is used?
> 
> 
> Unless it's a low diffraction point source in mono (anechoic). But yes, that's true 99% of the time...and in complete agreement with what I'm saying.
> So what specific target (measured) curve does he EQ (measured) to at the LP Wayne?
> If it's just twist knobs to his ears (audible) "preference", exactly what do we need measurements for?


Wow, where to begin? Why don’t we just agree to disagree and part as friends. 

Regards, 
Wayne


----------



## HifiZine

Wayne, I didn't see you and AJ as being *that* much in disagreement. It's common practice to perform measurements for loudspeaker design at 1m and AJ would I have no doubt be the first to acknowledge that depending on the speaker design a greater distance may be appropriate in some cases. But there is the fact that anechoic and in-room measurements are substantially different... right?


----------



## jcmusic

I will be taking new measurements today in the hope I can make some progress!!!


----------



## ajinfla

HifiZine said:


> Wayne, I didn't see you and AJ as being *that* much in disagreement.


We aren't. We're both telling jc he needs some form of EQ.
It's the why and how....



HifiZine said:


> It's common practice to perform measurements for loudspeaker design at 1m and AJ would I have no doubt be the first to acknowledge that depending on the speaker design a greater distance may be appropriate in some cases.


Indeed, as I stated in post #24. Nearfield/gated/anechoic will show the issues being created at the source(s). LP is the whole mess. So now what is the target? A measured curve? Fiddle about until it sounds "good"?



HifiZine said:


> But there is the fact that anechoic and in-room measurements are substantially different... right?


Yes. I see potential problems beyond the 4k peaking. There also appears to be a depression centered around the 400-500hz region. Care to guess the Klipschorn XO frequencies?: http://www.klipsch.com/klipschorn-floorstanding-speaker/details
yep, 450hz and 4500k. I suspect the "modded" speaker might have some phase alignment issues...and that is not correctable via amplitude at the LP. Cutting the 4k range will help. But nearfield/gated/anechoic (the last rather doubtful) would tell far more of what's going on, than pressure at a single mic at the LP.
YMMV.

cheers


----------



## jcmusic

I am using the older type Xo and they are 400hz and 6000hz...


----------



## jcmusic

Ok nearfield measurements from today. Blue is the left channel


----------



## EarlK

Dr_Suess said:


> In reviewing many different graphs I have not yet seen what ideal graphs should look like. Can someone please post sample graphs for SPL, IR, and Waterfall.


* What do Ideal Graphs Look Like ? *












audiolin said:


> Hello All,
> 
> Getting used to looking at REW curves of my room ( SPL and Waterfall) and realizing that I need a point of reference. It would be really useful to have an example of something that would be considered a very good curve. Can this be found somewhere on the forum? If no example exists perhaps someone could post one up.
> 
> Thanks,
> gg


 *What does a good curve look like ?* 


















Read all posts from  *Mitchco* .

:sn:


----------



## ajinfla

jcmusic said:


> Ok nearfield measurements from today. Blue is the left channel


Hi jc, it appears there is phase misalignment between the mid/woofer. But I couldn't find much on original/unmodded Klipschorn measurements. I'm also puzzled as to why your HF taper is about the same NF as at the LP. It shouldn't be even if your mic exhibits some HF roll off.
At this point I would say check the phase on each driver on each speaker. Use a wide filter EQ like Wayne suggested to take down that 4k hump.

cheers


----------



## jcmusic

Hi AJ,
I check all the drivers yesterday for that and all were correct. Anyway here are the curves i ended with today it is sounding much better...


----------



## Wayne A. Pflughaupt

HifiZine said:


> Wayne, I didn't see you and AJ as being *that* much in disagreement. It's common practice to perform measurements for *loudspeaker design* at 1m and AJ would I have no doubt be the first to acknowledge that depending on the speaker design a greater distance may be appropriate in some cases. But there is the fact that *anechoic* and in-room measurements are substantially different... right?


But that’s just it: We’re not designing speakers here, or listening in anechoic rooms. To the extent that the room is influencing response at the listening position, we want equalization to compensate. 

The proof is in the pudding, as they say: You can see with jcmusic’s graphs that the nearfield graph has exaggerated highs, just as I predicted in Post # 28. Notice in the nearfield graph below that the upper-midrange peak is now roughly on the same plane as the bass frequencies, where in the LP graph that area is about 5 dB down compared to the bass. 

It should be blinding obvious: If you equalize based on a nearfield measurement it isn’t going to sound right at the listening position, and you’re going to end up re-equalizing. So why just start at the right place to begin with?










*Nearfield Measurement










Listening Position Measurement*​

Regards, 
Wayne


----------



## Kal Rubinson

Wayne A. Pflughaupt said:


> But that’s just it: We’re not designing speakers here, or listening in anechoic rooms. To the extent that the room is influencing response at the listening position, we want equalization to compensate.


This may be tangential to the discussion here but I want to add that all recordings are mastered in non-anechoic studios and, therefore, presume a similar acoustic contribution from the reproduction environment.


----------



## jcmusic

Wayne,
I think you said it first about being out of phase. Tell me what makes you and AJ think that I have a phase issue??? Can you point it out to me???


----------



## Wayne A. Pflughaupt

AJ’s observation on that was different from mine. Since he designs speakers he noticed that you had a big dip in the nearfield measurement precisely where you said the crossover frequency was. Since I’m not a speaker designer that one would have never clicked with me, but I’d certainly agree with AJ that it seems highly peculiar and just too much of a coincidence to ignore. I’ll leave it to him guide you on sorting that one out, since that’s his field of expertise and I know nothing of it.

As for what _I_ noticed, it was the graphs you showed us in Post #39. Specifically, that the L/R combined measurement showed less bass response than the two single-speaker graphs. Reduced bass response when the second speaker is added is a classic indication of a polarity issue. Typically it simply means one of the speaker wires is connected backwards (e.g. amplifier [+] to speaker [-]), but it could also be an internal problem, if the wiring is reversed on one of the woofers. There may be other causes, but again that would be AJ’s field of expertise. I can’t account for anything in the actual measurement process that would cause this.

Regards, 
Wayne


----------



## HifiZine

Wayne A. Pflughaupt said:


> It should be blinding obvious:




Oh.... not that obvious at all... look instead at 3k vs 600 Hz.


----------



## Wayne A. Pflughaupt

That’s a bit vague – can you be more specific?

Regards, 
Wayne


----------



## tonyvdb

Ya, looking at the graph I dont see anything that would indicate that something is out of phase. Even nearfield measurements unless done in a proper environment will be affected by the room.


----------



## jtalden

Regarding the phase alignment:
I took a look at the near field measurement of the Left Speaker. 

How we describe it depends on how we define "in-phase". If we are looking to have 2 drivers in-phase at the XO point such that the SPL does not have a significant dip in the XO range at the LP then both XOs appear to meet that criteria. 

[The near field measurement SPL dip at 400 looks suspicious, but may be due to floor bounce. The phase actually appears to be pretty well aligned right at 400.]


In this case however the phase between drivers is crossing at a very steep rate. This is true for both the 400 and the 6k XOs. The steep rate of phase crossing is indicative of a significant driver offset; either physical offset or delay added by a speaker management box, e.g., DCX2496, HTPC XO, or similar. With a KHorn the drivers are physically offset by a significant amount so it is not surprising that if passive XOs are being used this condition would be expected.

Even though the SPL is not expected to be impacted too much with a steep phase crossing like this, my experience is that the sound may noticeably different. The detail of the sound may be slightly reduced, the sound may seem more diffuse and may make the room and sound stage seem larger. Some listeners may prefer this type of sound. Of course if the excess delay gets too large the sound will deteriorate noticeably. 

If we define "in-phase" as close phase tracking throughout the XO range this setup would require significant delay adjustments to achieve that. It is very difficult to accurately tell how much delay would be needed without measuring the drivers independently, but the MR-TW appears to need about 0.890 ms more delay for the TW relative to the MR. The W-MR XO is even more difficult to read this way, but I would estimate that the MR would need an additional 5 ms or so. Again, accurate delays can only be determined with independent measurements of the drivers with loopback timing applied.

Below is the windowed response of the left speaker to focus on the upper XO at 6k. You can see large slope discrepancy between the MR and the TW phase and the resulting SPL dips in the XO area as the phase differential passes through the 180° out-of-phase points. These sharp near field (direct sound) dips will not be very apparent without the short window and with the mic at the LP. The room reflections will wash them out.


----------



## jcmusic

Hold everything new development!!! I may have found the problem i ned to measure to be sure but, I am 99% sure I have found what was causing all the problems!!!


----------



## EarlK

Hi John,

*re; Room Reflections ( & system wash-out ) :*

Here's an ETC ( IR envelope ) from one of Jay's measurements .

If this measurement is legitimate ( & considering that I'm no expert in parsing IRs ) I would respectfully suggest, that significant room treatments should be the priority here ( before any more talk of system EQ-ing or sub-synching can rationally proceed ) .

ie; I just don't see the point in EQ-ing reflections of this magnitude ( if real ) that are almost equal to the main impulse . This is best left to an acoustician to sort out .

:sn:

PS : Here's a compilation mdat ( of Jay's measurements, starting last Sept ) . A quick perusal will show that the large reflection ( identified in the pic ) really came into it's own after Christmas .


----------



## jtalden

Earl,
That IR/ETC peak at about 1ms is the MR horn coming in late behind the TW. It appears so large because the TW level is reduced significantly compared to the TW level. The room is not causing it.


----------



## ajinfla

jcmusic said:


> Hold everything new development!!! I may have found the problem i ned to measure to be sure but, I am 99% sure I have found what was causing all the problems!!!


Ok, I'm curious to see.
Couldn't open your post #62 files, but it appears other have. Again, flying blind, because I have no idea what a Klipschorn is supposed to measure like. That 400hz on axis dip could well be part of the design, due to the horn loading prop delay of the woofer relative to the horn mid. Far field/in room would spatially average it out a bit as shown in your LP measurements, but I would go gentle EQing around it if you were tempted to fill the depression.

cheers


----------



## EarlK

jtalden said:


> Earl,
> That IR/ETC peak at about 1ms is the MR horn coming in late behind the TW. It appears so large because the TW level is reduced significantly compared to the TW level. The room is not causing it.


Thanks for that, your explanation makes sense ( once some TW & MR terms get clarified) .

Here's a look-see of "excess group delay" ( from a nearfield measurement ) that shows pretty clearly ( I believe ) that the speaker in question is an ( unaligned ) 3-way . 

The horn "lags" the tweeter by about 1 msec, while the woofer "lags" the tweeter by @ 6.0 msec .

:sn:


----------



## HifiZine

Wayne A. Pflughaupt said:


> That’s a bit vague – can you be more specific?
> 
> Regards,
> Wayne


Hi Wayne, squinting at it, it looks like 5-6dB difference in either case?

Regardless, a. the thread seems to have moved on and b. I think it would be interesting to explore some fundamentals... perhaps not here - I threatened AJ with an article to review and I'd certainly be very appreciative if you would too.. no obligation at all but it's an interesting topic...


----------



## jtalden

Earl,
Yes, that's a very good way to get ball park numbers! You can also get a similar hint in the step response (although the Excess IR is easier). I actually shifted the IR as needed to align the phase chart. It's more complicated, but more accurate. Correct delays can't be determined without measuring the drivers separately though. 

I guess the W, MR, TW abbreviations I use for the drivers in a 3-way are not universally used, but at least you figured it out.


----------



## Mitchco

jcmusic said:


> I am curious as to why I have read many times over, you should not EQ any frequency's over the 200-300hz range??? Can someone explain???


It’s a myth. Coming from a recording/mixing background, most of the studio/control rooms I have worked in, have a least one set of “calibrated” monitors.

Calibrated meaning a “house curve”. As many have pointed out before, there are several well-known house curves, B&K being one of them and there are others if you do the research.

However, most of the house curves are similar, usually a flat frequency response to some “hinge” frequency point (1 to 2 kHz) and then a roll-off to typically around -6dB at 20 kHz.

Depending on how dead or bright sounding the room is will determine how much roll-off towards the 20 kHz end point. Some like -3dB, some -6db, and some -10db at 20 kHz. The idea behind a house curve is to give the listener a “perceptually” flat or neutral spectral balance at the LP

Personally, I advocate using full range eq after acoustic room treatments that take care of early reflections, typically an ETC of -20dB or greater down from the initial response peak at the listening position over a 40 millisecond window is a good rule of thumb.

Also one wants to get the reverb/decay as even as possible over the entire range of frequencies. An oldie but goody is the EBU Tech 3276 attached. It also has a “house curve” for frequency response (again falling into the range mentioned above) and guidelines for RT60/decay time.

I have attached the stereo frequency response (unsmoothed) and RT60 for my 3-way system at the LP and it can be seen it falls within the 3276 guidelines.

So what is the point of all of this? When I was recording/mixing sound, I got to work in numerous professionally designed and built recording studios. Aside from the great care in NC ratings, http://www.engineeringtoolbox.com/nc-noise-criterion-d_725.html the control rooms were completely symmetrical. Meaning from the center line at the LP, the left side of the room was a mirror image of the right side. The purpose being that the frequency response and reflections of the left monitor was a mirror of the right monitor at the listening position. So if you listened to a “mono” signal it would appear dead center in the image. Ideally, dead center over the listenable frequency ranges.

The purpose of the “house curve” FR, ETC, RT60, specs was so that if I had to move from one studio to the next (which occurs a lot), then when playing back the mix, it sounded almost identical from a spectral balance perspective. There is nothing worse than spending hours and hours on a mix and then move to another facility and finding the tonal balance or imaging or whatever off, sometimes off by a lot. Then one has to guess which mix has the right tonal balance so that it “translates” correctly to other listening environments.

With respect to house curves, the reason for the gradual roll-off at the top end is directly related to psychoacoustics, which is a hard topic to cover. But if one wishes, JJ Johnstons papers at http://www.aes.org/sections/pnw/ppt.htm represent a good body of work.

Most people would agree that a flat frequency response out to 20 kHz is far too bright. Playing with the house curves as described above, one can tune ones room to provide a “perceptually” flat frequency response at the listening position, or to put it another way, as neutral as possible. So by listening to your brightest and dullest recordings in one’s music library, one can find a balance that will work for almost any recording without having to adjust anything. It is very likely that it will fall into the range of the prescribed house curves.

Given that most peoples listening environments are not professionally designed “symmetrical” rooms, then it is very likely that the FR of the left speaker is going to be different that the FR of the right at the LP. This “distorts” the imaging that is encoded on the source material.

So by eq’ing the left and right speaker so they are as close to identical as possible at the listening position will yield the best stereo image and one will start to really hear what’s on the recording. DSP software like Acourate have a measure for that called IACC.

Taking care of the room acoustics to establish a reflection free zone at the LP, along with enough absorption and diffusion for the rooms RT60 or decay, will get you in the ball park.

Then by using eq, one can fine tune the tonal (spectral) balance to achieve a perceptually flat frequency response at the LP and match the speakers so that a mono image is dead center across the full frequency range.

Of course, one can go whole hog like I did with my Klipsch Cornwall clones and create a 3 way digital XO which then allows one to time align the drivers, which when using horns like in my system, or the Khorn, where driver offsets are almost 12” apart makes a big deal both from a tonal and imaging perspective. Further, one can linearize each driver in the near field (like at 30” on axis to each driver), before applying an overall frequency response, and in the case of Acourate, excess phase correction at the lp. I wrote a long and boring article about it here: http://www.computeraudiophile.com/c...e-alignment-driver-linearization-walkthrough/

In the end, I am trying to replicate the spectral balance and stereo imaging of the recordings I mixed in a very expensive control room environment in my living room. Following the EBU Tech 3276 guidelines will go a long ways to getting you there. After room treatments, with proper stereo or 5.1 setup, and establishing a RFZ, one can use a variety of digital eq’s from simple parametric all the way to sophisticated digital audio toolboxes like Acourate to get you the rest of the way.

When I listen to my mixes I made some time ago, on my fine-tuned system, it sounds almost identical to the way I remember it. And when you have listened to a song several hundred times over 8 hours a day for several days, one does not forget what it sounds like 

Hope that helps the OP’s original question.

Best regards, Mitch


----------



## ajinfla

Hi Mitch, 

Thanks for your input. I figure while we wait on jc's discovery....
I agree with some of what you presented, but..
I am neither a "studiophile" , nor an "audiophile" (despite heavy immersion in both subcultures). I find too many similar fashion trends to be either. I simply like music, electro-acoustically being the 2nd most preferred method.
First, the thousands of recording that I own, are mine, for my enjoyment. I care not one bit what the artist/studio/producer "intended". I'm not good at channeling, nor was I there for any of the recordings creation (though I've been in plenty studios to see how, generally). I care only that_ I_ enjoy listening to them. I suspect I'm in the same boat as the vast majority of music listeners.
Floyd Toole states pretty much the same, that the needs of end user/home listener environments, are *not* the same as studios. Your quotes about "treatments" and "RFZ" etc are to me, refugees of studio fashions (which lead to concepts like LEDE) making their way into home environments. Those are simply _*preferences*_ ...and rather bizarre concepts for someone familiar with basic wave propagation.
Is this a RFZ?








If so, can you see the (reflective) elephant in the room? I can't see any sane person wanting a giant mixing console in the living room to recreate the "Reflection Free Zone" studio effect...but then again, I'm neither studiophile nor audiophile......
Of interest, I note that you room looks an awful lot like what I would prefer...zero "treatments"or "RFZ", normal furnishings...and controlled directivity speakers. 








Though I prefer a greater sense of spaciousness as provided by reflections, that enhance the "realism" of artificial stereo constructs, for me. My desire for reproduction is not what is constructed/heard in a studio, but presented in concert halls. 2ch fails miserably here, more so when robbed of reflections from various directions and limited to waves from 2 direct radiators. JJ does detail exactly why this fails...and what lead to 7 mic, 5ch PSR.
Going back to EQ, it's still going to come down to preferences.

cheers


----------



## Mitchco

Hi Aj, sure totally understand about preferences...

I once had a pair of Maggies 2.6R in a very live room setup so the back wave reflected off the side walls, bounced off the rear wall and came forward to the LP in kind of a diamond shape. Sounded huge, but not accurate.

My preference is to try and reproduce as accurately as possible what is stored on digital media to my ears at the LP. My preference is to listen to what the artists, producers, and engineers intended as some have spent a great deal of time creating a specific audio illusion. I like listening to and getting lost in that illusion and not have the speakers or room color that illusion too much.

With respect to my room, I have added room treatments at the first reflection points, some bass trapping behind my speakers, absorption in front of the speakers to reduce floor bounce, celling absorbers and on the back wall. I could add some theater curtains to cover the front windows which is likely my next step.



















I wrote an article on the process and results at: http://www.computeraudiophile.com/blogs/mitchco/importance-timbre-sound-reproduction-systems-222/
I am happy with the sonic improvements that are both measureable by REW and audible to my ears. 

The point of what I wrote is to say that if one’s preference is for accurate or transparent or neutral music reproduction, then there is a great body of work and prescriptive guidance (one e.g. is EBU 3276 guidelines I linked to earlier) that will help in achieving that goal. 

Cheers.


----------



## ajinfla

Mitchco said:


> Hi Aj, sure totally understand about preferences...
> I once had a pair of Maggies 2.6R in a very live room setup so the back wave reflected off the side walls, bounced off the rear wall and came forward to the LP in kind of a diamond shape. Sounded huge, but not accurate.


Yep, it's all preferences really. Speakers, including large planar panels like Maggies, have complex 3 dimensional radiation. A "diamond shape" is more a 2D description of what is actually happening, our ears hear it all. Yes, added reflections will increase the sense of spaciousness, then it comes down to what one prefers. There is no "accurate", except in the mind of studiophiles and audiophiles, as I will explain shortly.
I highly doubt all those Maggie owners consider them less "accurate" than your Klipsch's. Quite the opposite actually. Even know of a studio guy who uses Maggies for mastering!



Mitchco said:


> My preference is to try and reproduce as accurately as possible what is stored on digital media to my ears at the LP. My preference is to listen to what the artists, producers, and engineers intended as some have spent a great deal of time creating a specific audio illusion. I like listening to and getting lost in that illusion and not have the speakers or room color that illusion too much.


What is stored on your media, is, in extremely rare cases, an acoustic event, like a live concert/recital recording. _Far_ more likely, it is a whole series of acoustic events. From a vocal booth, kick drum mic, etc, etc., that has been [stereo] "constructed" into an electronic event. The only way for your ears/brain to "hear" that electronic transcription, is to play it over an electro-acoustic system. So now the music "event" is an electro-acoustic reproduction....of which you have zero idea what would be an "accurate" reproduction. Do your floor residing Klipsch's sound like a pair of tissue covered NS10s on a console at your sofa? Or a pair of soffit mounted Urei's in a "LEDE" room? *NO*. They don't (thank goodness!). It is absolute futility to believe that you could possible recreate an "accurate" reproduction of some sonic construction that took place somewhere, for all your recordings.
There is absolutely no way to A/B compare anything. Not measurements. Not perception.
So now, what is this "accurate" reproduction studio/audio philes speak of?




Mitchco said:


> With respect to my room, I have added room treatments at the first reflection points, some bass trapping behind my speakers, absorption in front of the speakers to reduce floor bounce, celling absorbers and on the back wall. I could add some theater curtains to cover the front windows which is likely my next step.
> I wrote an article on the process and results at: http://www.computeraudiophile.com/blogs/mitchco/importance-timbre-sound-reproduction-systems-222/
> I am happy with the sonic improvements that are both measureable by REW and audible to my ears.
> 
> The point of what I wrote is to say that if one’s preference is for accurate or transparent or neutral music reproduction, then there is a great body of work and prescriptive guidance (one e.g. is EBU 3276 guidelines I linked to earlier) that will help in achieving that goal.
> 
> Cheers.


Well, seem like you liked the sound prior to all the "treatments". But maybe it didn't sound enough like a studio. To each their own preference. Nothing about it is more "accurate" or "neutral", unless we have some perceptual/measured A/B ability. We don't. We just have memory based preferences.
Btw, regarding reflections, Wayne posted in interesting article by JJ's old boss. Take a look if you have a chance: http://www.madronadigital.com/Library/RoomReflections.html
We can quibble about the 1/3rd vs 1/6th oct resolution of the ear after. The basics remain.
Regarding EBU 3276, what percentage of your recordings adhere to it? What about stuff recorded prior to 1998?
I've got tons of those.
See why I'm neither studiophile nor audiophile now?

cheers


----------



## jcmusic

Ok guys I am sitting here listening to the new settings i have as of today, things are sounding much better and the curves looks alot better as well. I will ask the experts to comment on the curve i am just happy with the sound right now. Please have a look and tell me what you see or think...


----------



## primetimeguy

jcmusic said:


> Ok guys I am sitting here listening to the new settings i have as of today, things are sounding much better and the curves looks alot better as well. I will ask the experts to comment on the curve i am just happy with the sound right now. Please have a look and tell me what you see or think...


Where are the curves?


----------



## jcmusic

primetimeguy said:


> Where are the curves?


You have to click on the link that says attached files....


----------



## primetimeguy

Using my phone and don't have that option.


----------



## primetimeguy

Looking pretty good now. Some EQ in the 40-80hz range would help your bass.


----------



## ajinfla

Definitely better. What did you do? How does it sound?

cheers


----------



## jcmusic

Hi AJ,
It sounds really good now much better than before, I had to move the LP around a bit until I found the flatest response I could. Then I started with Eqing. I am still not convinced that it can't sound better but, for now i amleaving it as is. AJ if you would hve a lok at th impulse and other parameters of this curve and tell me what you think.


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## Wayne A. Pflughaupt

Still think a filter or two to tame that persistent hump above 2 kHz would result in an audible improvement...

Regards, 
Wayne


----------



## ajinfla

Agree with this ^


----------



## jcmusic

Wayne A. Pflughaupt said:


> Still think a filter or two to tame that persistent hump above 2 kHz would result in an audible improvement...
> 
> Regards,
> Wayne
> 
> Deleted


----------



## Mitchco

ajinfla said:


> There is no "accurate", except in the mind of studiophiles and audiophiles, as I will explain shortly.
> I highly doubt all those Maggie owners consider them less "accurate" than your Klipsch's. Quite the opposite actually. Even know of a studio guy who uses Maggies for mastering!
> 
> What is stored on your media, is, in extremely rare cases, an acoustic event, like a live concert/recital recording. _Far_ more likely, it is a whole series of acoustic events. From a vocal booth, kick drum mic, etc, etc., that has been [stereo] "constructed" into an electronic event. The only way for your ears/brain to "hear" that electronic transcription, is to play it over an electro-acoustic system. So now the music "event" is an electro-acoustic reproduction....of which you have zero idea what would be an "accurate" reproduction. Do your floor residing Klipsch's sound like a pair of tissue covered NS10s on a console at your sofa? Or a pair of soffit mounted Urei's in a "LEDE" room? *NO*. They don't (thank goodness!). It is absolute futility to believe that you could possible recreate an "accurate" reproduction of some sonic construction that took place somewhere, for all your recordings.
> There is absolutely no way to A/B compare anything. Not measurements. Not perception.
> So now, what is this "accurate" reproduction studio/audio philes speak of?
> 
> I've got tons of those.
> See why I'm neither studiophile nor audiophile now?
> 
> cheers


With respect to accurate and transparent reproduction, Bob Katz said the same thing I am saying: http://www.hometheatershack.com/for...acourate-acourate-convolver-2.html#post686904 

My impulse and frequency responses are almost the same as Bob's and my IACC is 87% I already linked to the article that has the measurements and listening impressions. My speakers are custom designed using a similar Klipsch Cornwall cab, but with a Bob Crites woofer, and different waveguides/compression.

@jcmusic - looks good. As others have mentioned, bringing down the hump at 2 to 3 kHz will help.


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## ajinfla

Mitchco said:


> With respect to accurate and transparent reproduction, Bob Katz said the same thing I am saying: http://www.hometheatershack.com/for...acourate-acourate-convolver-2.html#post686904
> My impulse and frequency responses are almost the same as Bob's and my IACC is 87% I already linked to the article that has the measurements and listening impressions.


Hi Mitch, with all due respect to both of you, none of that addresses what I stated above.
Do you have said in situ measurements of playback for each (constructed) recorded sonic event in your collection, to verify the "accuracy" of your reproduced sonic event? Do you have perceptual data from the artists/producers of the original vs your reproduction, to confirm their "intent"?
Does either of your systems sound like tissue covered console mounted NS10s, or soffit mounted Tannoys, or Uries...?
If not, how is it "accurate and transparent reproduction"?



Mitchco said:


> My speakers are custom designed using a similar Klipsch Cornwall cab, but with a Bob Crites woofer, and different waveguides/compression.


Still a controlled directivity monopole box. And still doubt any Maggie owner would trade _their_ "accurate and transparent reproduction", for those.

Btw, just checked out your blogs, found this http://www.computeraudiophile.com/blogs/mitchco/criteria-eval-sq-high-res-masters-181/. Which one contains the "accurate and transparent" capture and "intent"? 
Also, "I believe every piece of audio equipment has its own sonic signature" is verifiably false, via controlled (blind) testing. Something terribly lacking amongst both studiophiles and audiophiles. Certainly there are audible differences amongst some, but not all. You might be surprised if you did blind test of you favorite Class A amp...
Btw, how much time do you studio guys spend _outside_ listening to/recording folks like the Northern Pikes in studios, 









and instead, inside concert halls, where there is neither a class A amp nor klipsch type loudspeakers "reproduction"?









cheers


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## jcmusic

Mitchco said:


> With respect to accurate and transparent reproduction, Bob Katz said the same thing I am saying: http://www.hometheatershack.com/for...acourate-acourate-convolver-2.html#post686904
> 
> My impulse and frequency responses are almost the same as Bob's and my IACC is 87% I already linked to the article that has the measurements and listening impressions. My speakers are custom designed using a similar Klipsch Cornwall cab, but with a Bob Crites woofer, and different waveguides/compression.
> 
> @jcmusic - looks good. As others have mentioned, bringing down the hump at 2 to 3 kHz will help.


Thanks Mitchco I plan on fixing that hump next time out, any other advice??? Cn you take a look at the mdat and see if all else is ok? I am trying to learn how to read the inpulse curve and anything else that will help me to understand...


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## Mitchco

ajinfla said:


> Hi Mitch, with all due respect to both of you, none of that addresses what I stated above.
> Do you have said in situ measurements of playback for each (constructed) recorded sonic event in your collection, to verify the "accuracy" of your reproduced sonic event? Do you have perceptual data from the artists/producers of the original vs your reproduction, to confirm their "intent"?
> Does either of your systems sound like tissue covered console mounted NS10s, or soffit mounted Tannoys, or Uries...?
> If not, how is it "accurate and transparent reproduction"?
> 
> 
> Still a controlled directivity monopole box. And still doubt any Maggie owner would trade _their_ "accurate and transparent reproduction", for those.
> 
> Btw, just checked out your blogs, found this http://www.computeraudiophile.com/blogs/mitchco/criteria-eval-sq-high-res-masters-181/. Which one contains the "accurate and transparent" capture and "intent"?
> Also, "I believe every piece of audio equipment has its own sonic signature" is verifiably false, via controlled (blind) testing. Something terribly lacking amongst both studiophiles and audiophiles. Certainly there are audible differences amongst some, but not all. You might be surprised if you did blind test of you favorite Class A amp...
> Btw, how much time do you studio guys spend _outside_ listening to/recording folks like the Northern Pikes in studios,
> 
> 
> 
> 
> 
> 
> 
> 
> 
> and instead, inside concert halls, where there is neither a class A amp nor klipsch type loudspeakers "reproduction"?
> 
> 
> 
> 
> 
> 
> 
> 
> 
> cheers


Hi Aj, I have no idea what you are going on about. It would be helpful if you did not label people you don’t know and provide objective proof while making unfounded claims.

You said you found my blog. I also write articles for Computer Audiophile. If you spend any time reading some of the objective measurement articles, you would see that I provide a repeatable procedure for each of the measures. For each of the measures, I provide all tests files. Finally, I ABX every one of the tests and provide the ABX results, that anyone can download and listen for themselves. Here is a typical example: Fun With Digital Audio – Bit Perfect Audibility Testing.

With respect to accuracy and transparency, again, if you spend time reading about sonic signatures, you will see I am discussing transfer functions. When I speak of accuracy and transparency, I am speaking in the context of a transfer function.

Using REW to measure my AD DA converter using analog loopback, I can see that the converter has flat frequency/phase response/GD and virtually unmeasurable distortion. So, the converter for all practical measurement purposes is transparent and the output accurately tracks the input. Here is the step response of the converter using a 48 kHz sample rate. Textbook response.










Now add amplifiers, speakers, mic preamp, and measurement mic into the measurement loop. Here is the step response of Magneplaner MG 3.6R from Stereophile. Btw, with respect to Class A amps having different sonic signatures is “verifiably false”, feel free to take that up with Nelson Pass.










That’s far away from an accurate step response. The Maggies are not time aligned and the midrange diaphragm is negative polarity relative to the bass diaphragm and ribbon tweeter. So the output is not accurately tracking the input.

Now have a look at my right speaker's step response at the LP some 10 ft away:










That is much closer to the step response of the electrical step response of the converter. So, my speakers are measurably more “accurate” than the Maggies.

Some might say, well that’s fine at one measurement position. To bust another myth, here is an overlay of 6 different REW measures around a 6ft x2ft listening area that covers the couch area. The step response of the right speaker virtually remains the same across the couch area.










Same goes for the 6 frequency response overlay measures of the right speaker across a 6ft x 2ft matrix around the couch area.










The room starts having its way with the one measurement that is from the back left of the couch (furthest away from the speaker).

With respect to recording techniques and recording events, which I believe is what you’re on about, I have recorded many live classical events in a number of concert halls. But so what. My point, again, is that many artists, producers, and engineers have spent a great deal of time painting the sonic event for you and stored on digital media. What I want is to accurately reproduce the waveforms stored on the digital media without the speakers/room altering the waveforms or as little as possible. I believe I just showed that with objective measures.

If you wish to refute any of these measurement claims, feel free to post your own objective measurements of your own gear/speakers (step response at the LP would be a good start


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## ajinfla

Mitchco said:


> Hi Aj, I have no idea what you are going on about.


Apologies for not being clear to you Mitch. Your claim:



Mitchco said:


> My preference is to try and reproduce *as accurately as possible what is stored on digital media to my ears at the LP. My preference is to listen to what the artists, producers, and engineers intended as some have spent a great deal of time creating a specific audio illusion*.


I have stated several times that you (nor any end user/consumer) have no idea "what the artists, producers, and engineers intended" on any recording (with the possible exception of your own!), of the millions of recordings out there. You have provided nothing to support that claim.
The step response of a Lynx Hilo does not support that claim. Nor does the acoustic step response of your system, as captured by a single microphone, have anything to do with that claim.
If you have some testimonials from a whole slew of artists, producers, and engineers, based on their very human memory, upon listening to your system, that might count, slightly.
How many artists, producers, and engineers have viewed your step response and declared, "that's it!!" "just the step response we intended!!"?
I won't ask you for any psycho*acoustic* perceptual papers related to step response, as I think we both know the answer there.



Mitchco said:


> You said you found my blog.


Yes, via the link you provided in post 80. Thanks.
Which is where I found this verifiably false belief:


> I believe *every piece of audio equipment has its own sonic signature*. E.g. CD transports, preamps, amps, cross overs, interconnects, basically every component, part, and wire in (and around, e.g. power supplies) the audio signal path will have its own *sonic signature*, whether designed or not.


The fact is, some pieces could very well have a "sonic" signature. But that absolutely does not mean *all* do. Quite the opposite. Again, those are audiophile/studiophile beliefs. The ABX null scrap heap contains a mountain of them.
A "Bit perfect" ABX is a complete red herring regarding that claim above. Any ABX of: CD transports, preamps, amps, cross overs, interconnects, basically every component, part, and wire in (and around, e.g. power supplies) to share? No need to appeal to Nelson Pass' authority, a simple positive ABX of your class A amp vs any non-pathological class AB would suffice. By anyone. TIA.



Mitchco said:


> When I speak of accuracy and transparency, I am speaking in the context of a transfer function.


*Audible*, *sonic* TF, or....???




Mitchco said:


> Here is the step response of Magneplaner MG 3.6R
> That’s far away from an accurate step response.


I thought you had a 2.6? And what would that have to do with "accuracy and transparency" to a recorded sonic event, or more likely, series of events, that you avoid addressing? 



Mitchco said:


> The Maggies are not time aligned and the midrange diaphragm is negative polarity relative to the bass diaphragm and ribbon tweeter. So the output is not accurately tracking the input.


Right...and neither is your system "time aligned" to anything but a single spatially positioned mic...which is not your two ears with a head between.



Mitchco said:


> That is much closer to the step response of the electrical step response of the converter. So, my speakers are measurably more “accurate” than the Maggies.


One is *electrical*, the other a single *acoustic* spatial pressure position in a 3D soundfield. Mitch, is really behooves you to read and understand the entirety of those jj links you yourself provided.
Start with "Soundfields vs. Human Hearing": A (good) playback system would have both direct *and indirect radiation capability* in each loudspeaker.
How is your Klipsch more "accurate" than the Maggie here?
Also check out "A Low-Complexity, Fast-acquiring Perceptually Tuned Room Correction Algorithm."
Compare to your methods.



Mitchco said:


> If you wish to refute any of these measurement claims, feel free to post your own objective measurements of your own gear/speakers (step response at the LP would be a good start


Unfortunately, red herring is not on tonights dinner menu.

We need only examine your claims of "accuracy", for accuracy.
I of course, make no such claim, as I'm neither audiophile nor studiophile. I'll simply say, _I prefer_ some "indirect" radiation capability, though not Maggies. It sound more "realistic" to me. When compared to my memory of concert halls/jazz clubs, etc. Not to some wholly artificial stereo construct, made somewhere, by someone, who I have no clue, or care, of their "intent".
Sure don't want it to sound like the console NS10s they may have been listening to!

cheers


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## jcmusic

Ok guys did a little work on the curve and this is what I have now...


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## Wayne A. Pflughaupt

Looking pretty good! Here are the graphs with 1/12- and 1/3-octave smoothing. IMO 1/3-octave for full range graphs gives a better representation of what you’re actually hearing. Some additional work on the bass between 30-80 Hz would also be beneficial. :T















​

Regards, 
Wayne


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## ajinfla

Looks ok, will post later...unless you learn how to use the capture function between.
Ok, how does it sound? Is the bass clear and defined? For example, when you play well recorded stand up bass, is their definition and good pitch? Full bodied? Can you hear the string plucks separately?
Do drum tracks have a nice "thwack", like your memory of real (non-amplified) drums?
At some point, you have to let your ears do more work than eyes.

cheers

Edit: Wayne to the rescue as I'm viewing data/typing


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## ajinfla

Hi Mitch,

Some further thoughts, now that I have a bit more time :



Mitchco said:


> Here is the step response of Magneplaner MG 3.6R from Stereophile.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> That’s far away from an accurate step response.


From your own link:


> _As I have written before in these pages, measuring physically large speakers with in-room quasi-anechoic techniques is in some ways a fruitless task. The usual assumption, that the measuring microphone is very much farther away than the largest dimension of the speaker being measured, is clearly wrong. Yet without access to a large anechoic chamber costing many hundreds of thousands of dollars, in-room measurement techniques are all we have to rely on - JA_


There are differences between a (more) point source design like your Klipsch vs a large line source like the Maggie, especially how they can be measured. The measurement by Stereophile is at 50" and non-anechoic. Do you have any for your 2.6's to compare to the Klipsch under same conditions (your room/setup)?



Mitchco said:


> Same goes for the 6 frequency response overlay measures of the right speaker across a 6ft x 2ft matrix around the couch area.


...and the 2.6 measurements at couch? It would be interesting to do more of an apples to apples comparison...if you did so.

cheers


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## Wayne A. Pflughaupt

Zooming in on the lows, bass response looks unusually ragged. Not used to seeing so many cancellation nulls. Was the measurement taken with both speakers operating, or just one of them? I don’t recall, is there a subwoofer in use?







​

Regards, 
Wayne


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## jcmusic

Yes Wayne two subs and both mains a HPF @ 80hz and the subs are crossed @80hz.


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## jcmusic

Ok new measurements from today, changed the xover setting to move the mid range area from 400hz
to 6000hz up about 4db so now there is no boosting in that area only cuts.


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## Wayne A. Pflughaupt

That looks like it should sound pretty good. Now you need to work on the subs! 







​

Regards, 
Wayne


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## jcmusic

Hi Wayne,
Yes that is the plan I need to time align the subs with the mains also, I just don't know how to do yet...


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## Wayne A. Pflughaupt

You could easily do it with separate YDP2006s for the mains and subs.

Regards,
Wayne


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## jcmusic

Wayne,
I don't know how to do it so some advice would help.


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## ajinfla

jcmusic said:


> Ok new measurements from today, changed the xover setting to move the mid range area from 400hz
> to 6000hz up about 4db so now there is no boosting in that area only cuts.


You've been using a DSP XO the whole time? The Klipsch XO isn't passive?


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## jcmusic

AJ I am not sure what you mean, I don't have a DSP XO. The Klipsch XO is passive but, it has the ability to attenuate the mid driver in 1 db increments...


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## ajinfla

jcmusic said:


> AJ I am not sure what you mean, I don't have a DSP XO. The Klipsch XO is passive but, it has the ability to attenuate the mid driver in 1 db increments...


Ah, gotcha. There's an L-Pad that allows passive level adjustment. How did you get rid of the original 400hz-ish notch? Or did it fill in at the LP?
Definitely looking better overall.

cheers


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## jcmusic

AJ when I moved the LP everything got better. Just had to make some filters. Now can you tell me about time aligning my sub's with the mains.


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## ajinfla

jcmusic said:


> AJ when I moved the LP everything got better.


Ok.



jcmusic said:


> Just had to make some filters.


Unclear what you mean here. EQ adjustment?



jcmusic said:


> Now can you tell me about time aligning my sub's with the mains.


Worry only about frequency domain for now. Fixing that will fix your "time" issues (and vice versa), if indeed the dip around 60hz is a phase misalignment issue. Are you using the sub(s?) internal low pass XO (set at what frequency?) and running the Klipsch's full range, or...?

cheers


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## jcmusic

AJ yes Eq adjustments. Yes both the sub's x at 80hz the mains a hpf at 80hz


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## Wayne A. Pflughaupt

jcmusic said:


> jcmusic said:
> 
> 
> 
> Hi Wayne,
> Yes that is the plan I need to time align the subs with the mains also, I just don't know how to do yet...
> 
> 
> 
> 
> 
> 
> Wayne A. Pflughaupt said:
> 
> 
> 
> You could easily do it with separate YDP2006s for the mains and subs.
> 
> Click to expand...
> 
> Wayne,
> I don't know how to do it so some advice would help.
Click to expand...

Assuming your pre amp doesn’t have two sets of outputs, you could use “y” splitters to feed the inputs of the two equalizers. The output of EQ #1 would go to the main speakers (or rather, the amp for the main speakers), and the output of EQ #2 would go to the sub. The YDP2006 has delay settings that can be accessed by pushing the ”PEQ” button twice (you can set the display to read in feet in the Utilities menu). So you’d dial in the appropriate delay for whichever is closer (sub or mains) to the seating position.

BTW, in case you didn’t know the YDP also has high and low pass filters as well, so you can (for instance) cut the lows from the Klipsch speakers and the highs from the subs at whatever frequency you deem appropriate.

On a related note, did you see the review for the YDP2006 that I posted recently in the Equalizers forum?

Regards, 
Wayne


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## jcmusic

Hey Wayne,
My pre does have two sets of outs. I am using one set for both Eq's left and right channels. The other set the sub's are using, yes I know about the hpf and lpf's. I am using a hpf at 80hz on the mains and the sub's are xo at 80hz also.


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## Wayne A. Pflughaupt

jcmusic said:


> Hey Wayne,
> I am using one set for both Eq's left and right channels.


So you’re using one equalizer for the right channel and one for the left? And none for the subwoofer?

Regards, 
Wayne


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## jcmusic

That is correct Wayne.


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## Wayne A. Pflughaupt

The only good reason to do that would be of you needed more than six filters for the main speakers. That would be pretty unusual and indeed would most likely be an indicator of over-equalizing. I’ve seen nothing in your graphs that would require more than a few filters at best.

I’d suggest re-configuring at least one of your YDP2006’s for stereo six-filter mode, and using that one for the mains. You could use the other in mono 12-channel mode if your subs are mono; otherwise re-configure it for stereo mode as well.

Regards, 
Wayne


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## jcmusic

Wayne the reason I have them that way is because someone here told me that was the better way.


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## Wayne A. Pflughaupt

I can’t really see how that arrangement is serving your needs. It gets you 3-4 times more filters on the main channels than you’ll ever need, and nothing for the subs.

Regards,
Wayne


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## jcmusic

Wayne if I reconfigure the setup to the way you said how would I run the sub's? Turn off the xo and run them full and adjust with the Eq?


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## Wayne A. Pflughaupt

Not sure what “turn off the do” means...

Regards, 
Wayne


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## jcmusic

Sorry it was suppose to be xo.


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## Wayne A. Pflughaupt

And “turn off the so” means... :scratch:

Regards, 
Wayne


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## jcmusic

Wayne,
I am sorry man I was texting on my phone from work I am home now on the PC. I meant to say turn off the xo on the subs and run them full range???


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## Wayne A. Pflughaupt

Yes, that’s what I’d do, and set the YDP2006 for the sub to low pass at the same frequency you have the mains high passed for. The frequency setting on the Yamaha will be more accurate than the sub’s anyway (assuming it has the typical knob adjustment), so your HP and LP will be exactly the same. Just another benefit you’ll get with this configuration, in addition to the time alignment and EQ options.

Regards,
Wayne


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## jcmusic

OK sounds like I need to make the change. Now do I run the sub's full range by turning off the xo?


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## jcmusic

Deleted.


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## Wayne A. Pflughaupt

Yes, run the subs full range and let the Yamaha do the crossover duties. :T

Regards,
Wayne


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## jcmusic

OK Wayne will do thanks.


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## jcmusic

Ok Wayne I changed my setup to what we talked about and here is the curve from today. it sounds good a little bassy in the room but, at the LP it sounds better the right amount of low. If see room for improvements let me know.


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## Wayne A. Pflughaupt

Is the sub response with EQ? I don’t recall, do you have more than one sub and if so, are they all playing in this measurement? You have some horrific cancellation going on in the low end...







​

Regards, 
Wayne


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## jcmusic

Is that the measurement from today? If so yes 2 identical sub's playing and eq'ed with one of the Yamaha's.


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## Nyal Mellor

Hi Jay, thanks for your PM. I thought I should answer here as well.. please take a look at my Acoustic Measurement Standards whitepaper for comparison of your measurements to targets.


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## jcmusic

Wayne A. Pflughaupt said:


> Is the sub response with EQ? I don’t recall, do you have more than one sub and if so, are they all playing in this measurement? You have some horrific cancellation going on in the low end...
> 
> 
> View attachment 48762​
> 
> Regards,
> Wayne


Gonna try adjusting the phase on the subs to see if I can get improvement...


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## jcmusic

This is what I have today sounds good no more boom...


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## Wayne A. Pflughaupt

All those little nulls are still there (as seen at 49, 73, 98, etc. Hz in the graph in my previous post). 

Regards, 
Wayne


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## jcmusic

Wayne when adjusting for these should I be using smoothing or not?


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## Wayne A. Pflughaupt

No, not for bass-response graphs. Smoothing will merely remove the nulls from the graph, but they’ll still be there.

Regards, 
Wayne


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## jcmusic

Plan on doing a little more work on it this week, it is really sounding good right now almost don't want to change anything.


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## jcmusic

Ok this is where I am now and it sounds awesome, don't know if it can sound better...


----------

