# Cut or Boost? How Much



## mpompey (Jan 5, 2007)

I've been looking at some of the graphs folks have been posting up here. How do you get your sub responses so flat by cutting freqs alone. Do you ever give any a boost? If so, by how much. Is there a dB limit for boosting a freq range? I'm not talking about a room null. I understand that you can't electronically even that out with an eq alone.


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## brucek (Apr 11, 2006)

> Is there a dB limit for boosting a freq range?


Zero..........



> How do you get your sub responses so flat by cutting freqs alone.


Turn up the sub amp and cut....

There is really no upside to boosting in the BFD.... it decreases dynamic range, reduces headroom in the DSP and reduces signal to noise ratio. Any boost at any frequency before the crossover frequency requires reduction in the input level of the BFD to make room for the boost. 

brucek


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## mpompey (Jan 5, 2007)

brucek said:


> Zero..........
> 
> 
> Turn up the sub amp and cut....
> ...


Of course, I never thought to do that. I've left the sub amp at the same level I used prior to the BFD and REW. It never occurred to me to turn it up; thereby raising the response of the low points, and then cutting everything else. 

I feel so stupid!


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## cyberbri (Apr 27, 2006)

On the other hand, from personal experience, if you cut too much to get the majority of frequencies down to the level of a dip, then you have to turn the gain on the sub way up. When you turn the gain up, you are amplifying "noise" from the BFD to be louder. 

For example, in my case my HK receiver puts out a lot of voltage for the sub signal. So I had to greatly reduce the sub level in the receiver to make sure the signal doesn't clip going into the BFD. But that means I had to raise the gain on the sub a lot. With one pass at the BFD, I cut a ton of frequencies, and had to raise the gain on my sub past 1/2 way. Getting close to the sub I could hear noise/static coming through. Lowering the gain reduced this greatly, so I re-did my BFD filters with less cuts, and was able to get the same final level without the audibly loud noise coming through the sub.

Here's the thread where I talk about it:
http://www.hometheatershack.com/for...digital-meter-strange-results-high-freqs.html

Here's my results, current filters:
Freq BW Cut/Gain
74Hz 6 -13
45Hz 8 -7
66Hz 2 -4
54Hz 8 +4
37Hz 4 +4
53Hz 3 +2

Response w/out BFD, just bass traps in the room:









After above filters:











Before that, I was using these filters:
24Hz 8 -3
30 10 -4
48.5 3 -5
40 6 -7
44.5 6 -12
53.25 3 +6
72.35 10 -16
83 6 -5
51 2 +4
89 2 +6
37.88 3 +3
100 4 -7

And this was my response:








(Granted, my speakers are now in a different position from when this one was taken, reflected in the above graphs)


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## mpompey (Jan 5, 2007)

I think I got it. I raised the gain knob on my sub from a quarter turn to about a third and remeasured with REW. Here are the graphs of the initial (unfiltered) response and then the final one with the filter. (This is with a 5dB house curve from 30 to 100Hz) 

However, after I applied the filter and checked the level of the fronts versus the sub with Avia DVD, the sub was about 4 dB lower. So I raised the gain knob on the sub to just shy of halfway and the levels matched. Did I do that correctly? I didn't think I needed to touch the signal from the receiver, as to that would mean I would need to REW the whole thing over again.

Here are the graphs:


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## mpompey (Jan 5, 2007)

Thanks for the replies and the links to your personal history, but I noticed that you have a few boosts in your filter settings. I thought boosts were a No-no. How do you make sure you aren't taxing your sub/amp?


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## cyberbri (Apr 27, 2006)

mpompey said:


> I think I got it. I raised the gain knob on my sub from a quarter turn to about a third and remeasured with REW. Here are the graphs of the initial (unfiltered) response and then the final one with the filter. (This is with a 5dB house curve from 30 to 100Hz)
> 
> However, after I applied the filter and checked the level of the fronts versus the sub with Avia DVD, the sub was about 4 dB lower. So I raised the gain knob on the sub to just shy of halfway and the levels matched. Did I do that correctly? I didn't think I needed to touch the signal from the receiver, as to that would mean I would need to REW the whole thing over again.
> 
> Here are the graphs:



Yes, you did that correctly. After eqing the sub, you need to re-calibrate the sub's level.

But looking at your "after" graph, it looks like you could still use some more tweaking. Did you use REW's "find peaks" and the "suggest filters" and "optimize filters" options? Ie., some if your cuts are in the right spot, but could be widened (bigger bandwidth) for a larger bandwidth.

I think I've configured my BFD filters 6 times or so before I settled on my current filters. 

For example, it looks like you put a cut at 58Hz, but it's pretty narrow so it leaves some humps on either side. Same with the cut at 20hz, although you probably don't even need that. Depending on the final level/graph, you could widen the filter, raise the frequency it hits a little, not cut it quite so much, and smooth that out. You might want more of a house curve, so there's more boom down low compared to the 40~80hz range. 

Keep experimenting and you'll hit a combination of filters that really make it smooth.


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## cyberbri (Apr 27, 2006)

mpompey said:


> Thanks for the replies and the links to your personal history, but I noticed that you have a few boosts in your filter settings. I thought boosts were a No-no. How do you make sure you aren't taxing your sub/amp?



I cut some other frequencies quite a bit, so I'm not too worried about a few small boosts. I didn't want to go over 3~4dB boost, though.

And I checked with some very taxing scenes - The Haunting and WotW at -15 ~ -10, no chuffing, no audible distress. Those are about as hard-hitting as it will get, and it seemed okay.

You can cut and cut and cut until you get to the level of the lowest dips. But will the dips stay at the same level as the other frequencies? And since I had to reduce the level of the sub in the receiver so much (HKs put out a very strong LFE/sub signal) so I don't send a clipped signal to the BFD, I'm already pretty low. The more I cut, the higher I had to turn the gain on the sub itself, which caused audible noise. So I re-did it, used fewer filters, a big cut or two, a few small boosts, and I'm good to go.


BTW, I made a flickr account. 
Here's my room with speakers, sub, and treatments:
http://www.flickr.com/photos/[email protected]/sets/72157594507788330/

And here's my latest graphs:
http://www.flickr.com/photos/[email protected]/sets/72157594507798589/


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## lovingdvd (Jan 23, 2007)

brucek said:


> There is really no upside to boosting in the BFD.... it decreases dynamic range, reduces headroom in the DSP and reduces signal to noise ratio. Any boost at any frequency before the crossover frequency requires reduction in the input level of the BFD to make room for the boost.
> brucek


Can you please explain this in some more detail. In a couple of places I used a +2 or +3dB gain for a small bandwidth maybe 2-4 (IIRC on about 25hz and another about 60hz.

I'd like to better understand how this may be hurting things. If this one of those "in principle" things or am I definitely getting lesser dynamic range/sound from the sub as a result? Or is it something I'm really not going to notice sound or power-wise with small amounts?

In my case the sub has plenty of power. I have a nearly perfect house curve at +8dB 20 to +0dB at 80hz. The response follows the curve almost perfectly. The sub know is turned up only 1/4 of the way. When I listen to music or movies with heavy bass action scenes the sub sounds nice and strong with plenty seemingly left over if I wanted to go even louder (which I don't).

Is there a way to measure definitively whether any gains I've made no matter how small or large are hampering the subs ability to perform. Basically I turned the sub up considerably to begin with to get things at the right "starting point" and then made cuts throughout - but to balance it out I needed a couple gains here and there...

Thanks!


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## Ivaols (Dec 29, 2006)

brucek said:


> Zero..........
> 
> 
> Any boost at any frequency before the crossover frequency requires reduction in the input level of the BFD to make room for the boost.
> ...


I have read that boosting is not that bad, or "less worse", if you do it on a higher frequency. Is this true?


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## brucek (Apr 11, 2006)

> I have read that boosting is not that bad, or "less worse", if you do it on a higher frequency. Is this true?


It's fine in varying amounts on any frequency that will enter the BFD at a maximum voltage that is less than the BFD's maximum input.

This would be those frequencies around and above the crossover.

With an 80Hz crossover, as an example, a 120Hz signal will always enter the BFD ~24dB lower than a 60Hz signal. Problem is, equalization is less effective as you move along along the target curve. Any eq at 120Hz would be ineffective.

Think of using gain in the BFD this way. A 20Hz signal that enters the BFD at its maximum input level will drive the yellow LED on. That's fine. At this point, how much gain can you add to a 20Hz filter to clip that 20hz signal and turn on the red LED? Zero........ gonna have to turn that input level down. Now the softest 20Hz signal just went into the noise...... So, the DSP may have a little less to work with, the difference between the maximum and minimum signal that can pass through the BFD is less, so you have less dynamic range.......................... can you hear the difference? Maybe, maybe not.

brucek


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## mpompey (Jan 5, 2007)

Brucek, thanks for taking the time for that message. I think I'm beginning to understand. Do you have any links that might explain the whole dynamic headroom vs range thing a little more.

My next question, when setting the level between my subs and fronts, should I use the flat or the house curve filter?


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## lovingdvd (Jan 23, 2007)

mpompey said:


> Brucek, thanks for taking the time for that message. I think I'm beginning to understand. Do you have any links that might explain the whole dynamic headroom vs range thing a little more.
> 
> My next question, when setting the level between my subs and fronts, should I use the flat or the house curve filter?


House curve issues is addressed in the house curve sticky. The consensus seems to be that a flat response in the sub range is rather dull and I certainly found that to be the case. So at a minimum you'll want to experiment with a few various house curves and see how it sounds. Of course just for your own curiosity you may want to hear how things sound flat but be prepared to do this only as an experiment not as a "keeper" setting (most likely).


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## Wayne A. Pflughaupt (Apr 13, 2006)

brucek said:


> There is really no upside to boosting in the BFD.... it decreases dynamic range, reduces headroom in the DSP and reduces signal to noise ratio. Any boost at any frequency before the crossover frequency requires reduction in the input level of the BFD to make room for the boost.


Sorry, but this is functionally incorrect. You’re confusing the dynamic range of the DSP with the overall unit’s S/N, which includes other circuitry besides the DSP. 

The two are not the same. The BFD’s DSP has a theoretical dynamic range of over 140 dB. Reducing the input level 10 or 15 dB is not going to have an appreciable affect on the DSP’s dynamic range – not when the BFD’s rated S/N is a mere 94 dBu.




cyberbri said:


> On the other hand, from personal experience, if you cut too much to get the majority of frequencies down to the level of a dip, then you have to turn the gain on the sub way up. When you turn the gain up, you are amplifying "noise" from the BFD to be louder.


Exactly. You have to look at the whole picture; the BFD is only one component in the signal chain. Assuming you have your BFD’s input level set just below clipping, if your output level is –10 or -15 dB (or even less) after equalization, that’s 10 or 15 dB (or more) you reduce the sub’s S/N when you turn it up to compensate. Well, a high gain/low signal scenario is a know recipe for audible noise.

Does that matter much at the end of the day? As cyberbri and PeteD have discovered, sometimes it matters a lot.

With that approach it isn’t terribly relevant that you managed to maintain the BFD’s questionable >94 dB noise rating. Your overall S/N can never be higher than the weakest link in the signal chain anyway. (Not that noise is much of an issue with subwoofers anyway.)

So, the best approach is to try to get the BFD’s output close to its input. Most response curves have both peaks and valleys, so the most efficient equalizing (i.e., using the fewest filters) means a combination of boost and cut filters applied to them. After EQ you’ll often end up with the BFD’s output close to its input, so you don’t burden your sub with a low-level signal.

Regards,
Wayne


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## brucek (Apr 11, 2006)

> So, the best approach IMO is to try to get the BFD’s output close to its input. Most response curves have both peaks and valleys, so the most efficient equalizing (i.e., using the fewest filters) means a combination of boost and cut filters applied to them. After EQ you’ll often end up with the BFD’s output close to its input, so you don’t burden your sub with a low-level signal.


Is "using the fewest filters" our ultimate goal?

Whether you apply gain filters to response dips (and turn the BFD's input level down to allow it), or you apply cut filters to areas around response dips (and setup the input level to maximum), it makes no difference to the output signal level of the BFD. The former example reduces dynamic range and degrades signal to noise ratio, the latter example does not. 

Which do you choose?

Setting a best input level maximizes the number of bits used in the BFD. The signal to noise ratio in a digital device is quite sensitive to input level. The BFD is a 24bit device. The ADC has a fixed theoretical noise floor of 147 dB that is a function of bit resolution (6.125x24=~147db). The spec for the BFD is ~94dB, so we know already that not only is the LSB hidden in the noise, but quite a few others. The spec of 94dB equates to between 15 and 16 bits of actual resolution. We have about 8 bits lost in the noise. 

As you lower the input signal to the BFD, the noise internally rises exponentially. We know this, since each digital bit we lose drops about 6dB from our noise figure. As we lower the input level, the softest signal we can resolve (and therefore more low order bits) are lost in the noise. The BFD system becomes a 15bit, 14bit, 13bit device. An exponential drop in dynamic range occurs.

Since we know that we can obtain the same output level whether we use cut or gain, why choose gain when it creates a problem as I've described.

brucek


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## Ayreonaut (Apr 26, 2006)

A bit of boost is not a problem if it is in combination with cuts so that the sum of all the filters results in no net gain.


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## cyberbri (Apr 27, 2006)

Ayreonaut said:


> A bit of boost is not a problem if it is in combination with cuts so that the sum of all the filters results in no net gain.



That's what I assumed and hoped was correct - that my large cuts and several small boosts would average out or cancel each other out.


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## Guest (Jan 31, 2007)

brucek said:


> It's fine in varying amounts on any frequency that will enter the BFD at a maximum voltage that is less than the BFD's maximum input.
> 
> This would be those frequencies around and above the crossover.brucek


brucek, perhaps you could share your thoughts on whether the following example (well below the x-over) would constitute a boost? 

For our example let’s say we have a peak at 35hz that’s 15db above the target line. When we cut it down to the target we now find the filter has created a new 6db dip at 28hz. 

So here lies the question: Is creating a filter at 28hz with a positive value (gain) to counter the effects of the neighboring cut filter considered a boost simply because the filter has been set with a gain? 

Obviously, if you add enough gain to push that filter above the “flat filter” target line that REW also displays, then I can see it clearly being a boost. But I’m not talking about that.

Rather, let’s say our 28hz filter has +5db gain, yet the net result still remains below the flat filter line (due to the effect of the neighboring cut filter), how would this compromise BFD headroom?

Thanks


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## brucek (Apr 11, 2006)

> that my large cuts and several small boosts would average out or cancel each other out.


Unfortunately, it doesn't work that way..............

brucek


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## cyberbri (Apr 27, 2006)

brucek said:


> Unfortunately, it doesn't work that way..............
> 
> brucek



Technically, no, because a sound effect or musical note could hit just the boosted frequency and not the cut frequency. So the benefit of the cut frequency is not there, with only the drawback of the boosted frequency. But overall, without pushing music to max volume, I would assume that the "torture test" scenes in movies would the WoTWs and Hauntings and LOTRS, where bass is hitting at many, many frequencies over the range.


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## Ayreonaut (Apr 26, 2006)

I should have added "at any frequency."

Keep an eye on the "filters" graph in REW and make sure that it doesn't go over the target level at any point.

You can use positive filters to help shape the EQ curve so that the wide cuts don't further attenuate your measured dips, but you can't use positive filters to boost the level at the dips.

For example here's one with two filters. Note that there is no net gain at 53.3 Hz despite the large boost, because the sum of the two filters is always less than (or equal to) zero. Thus, we do not further attenuate our original dip, but we don't boost out of it either.

41.5 Hz, -14 dB, 1/1 octave wide
53.3 Hz, +12 dB, 1/6 ocatave wide


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## brucek (Apr 11, 2006)

> I should have added "at any frequency."


Yeah, that's correct. 



> perhaps you could share your thoughts on whether the following example (well below the x-over) would constitute a boost?


There's a couple points to make. Indeed, as ayreonaut implies, if I have a +10dB filter at 50Hz and a -10dB filter at 50Hz, the resultant effect on the output signal is 0dB. In that regard, although I gave an extreme example, I can use gain to shape a negative filter without providing any gain to the input signal and as such, the output signal will not clip (requiring a reduction in the input level to compensate).

My point in my upthread posts is that if you provide any resultant gain in the input signal, you will have to reduce your input level to the BFD (with your processor subwoofer output trim). This lowered input level to the BFD causes negative effects as I discussed upthread. I am simply trying to get the maximum signal to the A/D converter. You can't do that if you apply gain because the output will clip.

A +10dB filter at 50Hz and a -10dB filter at 50Hz doesn't constitute gain in the output. I might point out though, excessive filter gain or cut value are better served divided into smaller bites. There will be less internal DSP distortion to two -5dB cuts at 20Hz than a single -10dB cut at 20Hz. Divide your big filters in half.

brucek


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## cyberbri (Apr 27, 2006)

Good advice about splitting large cuts into several filters. I may try that with my -13 cut at 74Hz. I'll also look into adding a bit of boost to any areas around the dip/null around 85~95Hz that were lowered because of the cuts and see what happens.

With regards to input level into the BFD, I used ch5 of WotW DTS to set the input level by checking for the peak. I think I turned the sub off, but had the volume at the max level I would probably play that scene at (-12 to -10 or so, IIRC) and watched the input level meter on the BFD.

I ended up raising the speaker levels and lowering the sub levels in the receiver to get proper calibration with a low enough sub output into the BFD. Before the sub was 3~4 notches below the level of the speakers in the receiver, with the sub's gain at about 1/2 way between 0 and the first tick on the HSU gain knob. But since my HK puts out a very strong sub signal, I raised the speaker levels to about +2~+3 all around, and have the sub level at -9 for DD/DTS. This is re-calibrating the receiver so that -5 on the dial is reference level (-10 = 80dB with Avia, so 5 under) rather than 0 equalling RL 0.


This is all great advice. I'll re-evaluate my settings, although I think my small boosts are fine as the sub hasn't shown any stress on even The Haunting DTS or WotW DTS at high volume (max volume I would watch at).


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## Guest (Jan 31, 2007)

Yes! It seems we have a complete answer to the original question that started this thread. 



mpompey said:


> I've been looking at some of the graphs folks have been posting up here. How do you get your sub responses so flat by cutting freqs alone. Do you ever give any a boost? If so, by how much. Is there a dB limit for boosting a freq range? I'm not talking about a room null. I understand that you can't electronically even that out with an eq alone.


And the following reply:



> Is there a dB limit for boosting a freq range?
> 
> Zero..........


and 



> How do you get your sub responses so flat by cutting freqs alone.
> 
> Turn up the sub amp and cut....
> There is really no upside to boosting in the BFD.... it decreases dynamic range, reduces headroom in the DSP and reduces signal to noise ratio. Any boost at any frequency before the crossover frequency requires reduction in the input level of the BFD to make room for the boost.


You can see how someone reading this thread from the top could get the wrong impression that any filter with “any amount” of gain is to be avoided…unless it were close to or above the x-over. 

That’s why it seemed further clarification of what defines “a boost” was in need. 
Many could assume that any filter with any amount of gain is a boost, when that is not necessarily the case.

Utilizing the filter graph in REW as a means to evaluate filters that employ a bit of gain (but no actual boost) to determine if you are within the safe zone is very useful technique in smoothing out your response after those larger cuts needed to tame peaks down to the target line have created some modest dips in the process.





brucek said:


> A +10dB filter at 50Hz and a -10dB filter at 50Hz doesn't constitute gain in the output. I might point out though, excessive filter gain or cut value are better served divided into smaller bites. There will be less internal DSP distortion to two -5dB cuts at 20Hz than a single -10dB cut at 20Hz. Divide your big filters in half.
> 
> brucek


brucek, In dealing with that last question you brought up the technique of splitting larger cuts into two. Do you have any basic guidelines as to when a cut should be split? 

I’ve found myself doing this of sorts by using two fairly close spaced cut filters when dealing with a large odd shaped peak. However, I never knew if that was good practice or not?


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## lovingdvd (Jan 23, 2007)

"A +10dB filter at 50Hz and a -10dB filter at 50Hz doesn't constitute gain in the output. I might point out though, excessive filter gain or cut value are better served divided into smaller bites. There will be less internal DSP distortion to two -5dB cuts at 20Hz than a single -10dB cut at 20Hz. Divide your big filters in half."

So in other words, if REW recommends say a -15dB cut at 25hz, use one filter for -7 and another for -8 at the same freq, correct?

Just how much difference does this make? I have lots of tweaks and as it is I'm always struggling to fit all the tweaks into the 12 filters. Speaking of which, is there a way to apply more than 12 filters to the same sub?


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## cyberbri (Apr 27, 2006)

For the second question, I believe it is possible to use both the left and right "engines" to have more filters.


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## lovingdvd (Jan 23, 2007)

cyberbri said:


> For the second question, I believe it is possible to use both the left and right "engines" to have more filters.


I was under the impression that the left engine only applied to Output 1 and the right only applied to Output 2. No?


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## Guest (Jan 31, 2007)

lovingdvd said:


> I was under the impression that the left engine only applied to Output 1 and the right only applied to Output 2. No?



That's why to use both you would simply connect the output of one to the input of the other with a short cable and run them in series. How much this would degrade the signal (and if it really matters for sub use) is something someone else might answer.


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## brucek (Apr 11, 2006)

> In dealing with that last question you brought up the technique of splitting larger cuts into two. Do you have any basic guidelines as to when a cut should be split?


Nope. But if I had filter slots available and I had a cut over 10dB, why not cut it in half.



> Speaking of which, is there a way to apply more than 12 filters to the same sub?


Yes, feed the output of one channel to the input of the other channel. Now you have 24 filters. You have also converted from analog to digital and back twice.... do you want to do that?

If you need more than 12 filters, you're doing something wrong........

brucek


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## lovingdvd (Jan 23, 2007)

brucek said:


> If you need more than 12 filters, you're doing something wrong........
> brucek


Part of what drives my filter needs up is that I lose the ability to Optimize my filters once they are in the BFD. This goes back to my whole measure vs. re-measure thread.

For instance I do a measure and then optimize it with filters according to REW. I copy all these filters into BFD. So far so good.

Now I remeasure and see I need a few tweaks. So then I make a few tweaks using new filters in the new Measurement window. REW doesn't know about the old filters so it can combine and optimize things based on the filters any more because the first set of filters is independent from the second set.

I guess the proper way to do this would be to remove all the filters in the BFD after each run, and go back to the original filters window and make the adjustments there, and then copy all x filters back into the BFD. A lot more labor intensive especially without the MIDI programming. But perhaps this would do the trick?


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## Otto (May 18, 2006)

I see what you're saying about the filters not following forward. Last night, I decided that I was going EQ my sub and mains at the same time. I basically just did a sweep to 200 Hz, set the crossover to 40Hz and started tweaking filters on my own. I did not make any attempt to have REW calculate the filters for me; basically I was just using it for a measurement and download tool. 

The filters not following forward _is_ a pain, IMHO. However, I just made two sets of filters that I saved to my PC -- one called "temp.req" and one called "CurrentBest.req". Everytime I did a new measurement (with the "temp.req" filters loaded to the BFD), I was able to load those same filters into that new measurement. Now, I have the filters and the measurement together. Yeah, the "corrected" plot in REW is now meaningless, but I was able to do most of it by hand. I just addressed the big peaks first, and tweaked little by little from there. When I decided that I liked the result, I saved the filters to "CurrentBest.req". Then I had a place to return to if I didn't like what happened to the filters I was playing with.

I screwed up one thing, so I'll be back doing it again. My sub level on my preamp was -10 dB for some reason, when it should be 0. Must have been watching something with lots of bass or late at night to have it turned down so far. It definitely took longer than doing it with REW doing all the calculations, but I'm going to continue moving forward with it when I have some more time.


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## brucek (Apr 11, 2006)

> Now I remeasure and see I need a few tweaks


You're making this too difficult.

Take your raw measurement and play with REW until your predicted corrected plot is what you want.

Enter the filters and remeasure. 

It needs a tweak? So tweak the filters directly in the BFD a bit. Surely you've played with this enough to have a feel for it now and don't need to start all over with REW filters. Why weren't they very, very close the first time?. Keep doing this until it's good. 

brucek


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## cyberbri (Apr 27, 2006)

Actually for me REW's predicted filter results weren't very close at all to what it came out as. I may have some graphs at home showing no-fliters, predicted, and final result of the filters. 

But when I run the sweep and do the filters, on subsequent sweeps I disable the filters in REW and see what new ones I can add to tweak the final output - or redo the filters all together. I got pretty good coming up with my own filters, even before using REW.


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## cyberbri (Apr 27, 2006)

lovingdvd said:


> Part of what drives my filter needs up is that I lose the ability to Optimize my filters once they are in the BFD. This goes back to my whole measure vs. re-measure thread.
> 
> For instance I do a measure and then optimize it with filters according to REW. I copy all these filters into BFD. So far so good.
> 
> ...



Keep an original no-filter sweep where you tweak your filters. When you run a new sweep to test the results of the filters, do it in a new window. Compare that to the original window with the filters. Tweak the filter settings in the original window to get closer to where you want to be in the measurement window. The results probably won't match up perfectly between the prediction and actual, but say you still need to cut 2dB at 48Hz. You just go to the first window with the no-filter sweep and the filter settings, cut 2dB at 48Hz, go to the measurement window and measure again.


For having multiple variations of filters, though, you can use the channels or whatever in the BFD (1-10?). Say you want one curve for music, and one for movies that's a bit heavier on the low end. Rotate the dial and switch between sets of filters. I used this just to store previous filter sets so I could compare and use the best one. I have the different filter settings saved in an Excel file.


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## Otto (May 18, 2006)

brucek said:


> You're making this too difficult.


I dunno. After iterating on it a number of times, I got pretty good at it.



> Take your raw measurement and play with REW until your predicted corrected plot is what you want.
> 
> Enter the filters and remeasure.


Yeah, I started there. Remember, I have my mains on now -- trying something different, ya know...



> It needs a tweak? So tweak the filters directly in the BFD a bit. Surely you've played with this enough...


Nah, I try to actually touch the BFD as little as possible. Also, although I don't get exact feedback, there is a lot of graphical information that's there that helps me realize what's going on.



> Why weren't they very, very close the first time?. Keep doing this until it's good.


They're kinda close, but I would say that I've not really had them ever be "very, very close" with my IB. The REW predictions were more precise pre-IB for me.


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## JohnM (Apr 11, 2006)

brucek said:


> I might point out though, excessive filter gain or cut value are better served divided into smaller bites. There will be less internal DSP distortion to two -5dB cuts at 20Hz than a single -10dB cut at 20Hz. Divide your big filters in half.


I wouldn't recommend that, the overall response shape of a pair of filters at the same centre frequency is not the same as a single filter of the same total cut, adjusting the bandwidths of the paired filters to get a similar shape to the single filter would not be as good a match to the modal resonance being countered.


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## brucek (Apr 11, 2006)

> I wouldn't recommend that


Makes sense John. I had always done that when I had a large filter following advice from Behringer. The 1100P manual discusses digital overflow and recommends this, so I had it in my head. It seemed to make sense to me, but I also see what you're saying and it does indeed prove out when I play with REW as shown in the picture below where I overlay a single -20dB cut filter into two -10dB cuts. there is indeed a small difference in the bandwidths.

Behringer writes:

_ When using the Feedback Destroyer as a parametric equalizer (Parametric EQ mode) and applying extreme attenuation values in the low end range of the frequency spectrum (below 50Hz), you should perhaps set several filters to process the same frequency. Depending on the amount of attenuation applied, the use of only one filter may lead to slight distortion, which produces interference in the signal path. This is a natural physical phenomenon which should be avoided. Use for example two filters adjusted to a signal attenuation of about -12dB (with the same frequency and bandwidth). This produces the same effect achieved with one filter set to -24dB_

I guess I've always felt better about smaller filters, but I guess it throws a wrench in the works.

Below is an overlay of two filter sets. One with a single filter and another with two filters in half. You can see the small deviation...........










brucek


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## lovingdvd (Jan 23, 2007)

brucek said:


> It needs a tweak? So tweak the filters directly in the BFD a bit. Surely you've played with this enough to have a feel for it now and don't need to start all over with REW filters. Why weren't they very, very close the first time?. Keep doing this until it's good.
> 
> brucek


This is exactly what I do, and exactly why I wind up with 12 filters and wanting room for more! 

I can't quite put my finger on it, but there seems to be a major need for a Remeasure feature of some sort. Or at least such a filter would significantly reduce what's involved to keep tweaking back and forth. Others have chimed in over and over since I mentioned this saying they know what I'm saying, so perhaps someone can articulate the need to the developers better...


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## brucek (Apr 11, 2006)

> so perhaps someone can articulate the need to the developers better...


Everyone 'gets' what you're saying. The solution has already been proposed in the suggestion to output a sweep signal that is modified by the filters set in REW. This would allow continuous remeasures and tweaking until the plot is to your liking. Then you would only need transfer the filters to the BFD once, confident that they are correct........

brucek


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## terry j (Jul 31, 2006)

but wouldn't that then mean we wouldn't need a BFD?? all we would need to do is play our music thru the computer and REW to have everything to our liking. I mean, if REW had the ability to modify a signal going thru it it has essentially become an equalizer hasn't it??

I am probably missing something very simple here.

Just as an aside, I haven't mucked about with version 4 yet but I never experienced any sort of problems with the filters on the earlier version, I'll have to redo it and see if I can spot why I didn't experience those 'inconveniences' mentioned by others. I have always found the predicted response to be VERY accurate and so not a lot of tweaking on that front was ever required. 

My initial thoughts are that at times 'too much significance and complication' is being introduced into what is an essentially simple operation.


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## Wayne A. Pflughaupt (Apr 13, 2006)

brucek said:


> Is "using the fewest filters" our ultimate goal?


Getting the job done with the least amount of filters is standard equalizing protocol - common knowledge that can be found at any pro audio Forum.



> _Whether you apply gain filters to response dips (and turn the BFD's input level down to allow it), * or you apply cut filters to areas around response dips, it makes no difference to the output signal level of the BFD.* The former example reduces dynamic range and degrades signal to noise ratio, the latter example does not._


(Emphasis added.)
Applying only cutting filters does indeed lower the output signal level of _any_ equalizer, analog or digital. Anyone can verify this with their BFD. 

Lowering the equalizer’s output signal can only exact a noise penalty to the subwoofer when you (are forced to) increase its gain. The subwoofer is the weakest link in the signal chain anyway, WRT S/N, so it’s the component that can _least_ afford an increase in S/N. There is no point in rigidly maintaining the BFD's 94 dB S/N ratio if the best the average subwoofer has to offer is 90 dB. And trust me, it’s a _questionable_ 90 dB at that.

I suggest looking into the fundamentals of gain structure...

Regards,
Wayne


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## brucek (Apr 11, 2006)

> I suggest looking into the fundamentals of gain structure...


My limited knowledge of system gain structure keeps dragging me back to a basic tenet, in that when mixing digital and analog devices, that it's important to ensure the lowest of voltages in the digital device are greater than the noise floor and that the highest voltages are passed unclipped.

Again I defer to your better knowledge of this matter than my own. I find that after some thirty two years working in engineering that I would rather play with my grandson than study the fundamentals of gain structure. 

brucek


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## Peter De Smidt (Dec 22, 2006)

This is an interesting topic. Both side make important points, and I appreciate the discussion. Bruce certainly is correct in holding that the BFD will preform best with the maximum signal going in short of clipping, and others are just as correct in holding that increasing gain elsewhere in the system will increase noise. Clearly, what's optimum will be system dependent, and hence what might be "best" might differ.

What would be the best way to test this? I'm a newbie when it comes to using the BFD, but I'm assuming that one could make a cut only filter set and a boost and cut filter set and then evaluate them.


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## Wayne A. Pflughaupt (Apr 13, 2006)

brucek said:


> I find that after some thirty two years working in engineering that I would rather play with my grandson than study the fundamentals of gain structure.


Can’t say as I blame you there!  Sure wish mine lived closer... 




brucek said:


> Setting a best input level maximizes the number of bits used in the BFD. The signal to noise ratio in a digital device is quite sensitive to input level. The BFD is a 24bit device. The ADC has a fixed theoretical noise floor of 147 dB that is a function of bit resolution (6.125x24=~147db). The spec for the BFD is ~94dB, so we know already that not only is the LSB hidden in the noise, but quite a few others. The spec of 94dB equates to between 15 and 16 bits of actual resolution. We have about 8 bits lost in the noise.


Sorry, but that’s not the way things work with A/D converters. Reducing the strength of the input signal has no affect on the functionality of the converters. It does not reduce its bit rate or affect its ability to resolve low-level signals. There is more to the component’s ultimate S/N spec than just the converter’s bit rate. The rest of the circuitry in the component contributes as well – the power supply, balancing transformers/circuits, etc. The reason why no 24-bit digital processor is able to generate an actual 147 dB dynamic range is because the analog circuitry in the device can’t deliver that kind of performance. 




> As you lower the input signal to the BFD, the noise internally rises exponentially. We know this, since each digital bit we lose drops about 6dB from our noise figure. As we lower the input level, the softest signal we can resolve (and therefore more low order bits) are lost in the noise. The BFD system becomes a 15bit, 14bit, 13bit device. An exponential drop in dynamic range occurs.


The bit rate is static; it has no bearing on the signal strength. I.e., a 24-bit converter is not “downgraded” to a lower bit rate with a reduction on signal strength. If this were true, the BFD would be virtually screaming with noise when idling (i.e. no signal present).

Regards,
Wayne


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## cyberbri (Apr 27, 2006)

cyberbri said:


> This is all great advice. I'll re-evaluate my settings, although I think my small boosts are fine as the sub hasn't shown any stress on even The Haunting DTS or WotW DTS at high volume (max volume I would watch at).




I re-did my settings, yet again. I switched my cables around, because I realized before I was only working with the right channel/speaker and the sub. This time I input into both left/right channels of the receiver and ran those and the sub.


Smoothing is 1/12 octave for all of these. 

First, here's speakers only, with the receiver crossover on/off, and then speakers and sub combined, before eq. I have 2x GIK Acoustics 244 panels in each front corner.












Here's the sub + speakers response, filters, and predicted results. 
Note that the filter line makes it look like I'm boosting over the original output. In the end I am just a bit, but the actual response didn't match up exactly with the predicted results, so it's off a bit there. 










And finally, here's before/after for the eq results, sub plus mains. 











Filters used are as follows - most of the +s are bringing back narrow areas dropped by surround cuts, as can be seen in above graph - so net gain is 0 for most of the boosts:
Hz BW Gain
22 9 -2
30 9 -3
36.6 4 +6
40.5 3 -3 
45 11 -12
52 4 +8
66.4 4 -5
72.3 5 -10
75.75 4 -3
91 3 +9
87 3 +4
106 20 -4


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## brucek (Apr 11, 2006)

> 91 3 +9


What's the purpose of this?

brucek


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## Wayne A. Pflughaupt (Apr 13, 2006)

That’s an awful lot of filters for a graph that’s not all that bad to begin with. Your two big peaks are fairly symmetrical in shape and should respond well to a single filter set for the right bandwidth, and maybe one more at each of the three low points for “clean up.”

I suggest switching to 1/6-octave smoothing. Otherwise all the little ripples are “magnified,” and you’re tempted to chase every one of them. That’s not really productive and it doesn’t make any audible improvements.

Like brucek said, the 91 Hz filter is a head-scratcher. Also, maybe the 106 filter. I did a cut to my subs up in that range, only to find when I added the mains that it re-appeared. Actually, it was worse with the mains, which meant they were contributing more than the sub was. So I eliminated the filter, as it wasn’t doing anything, functionally-speaking.

Maybe you could e-mail me your response file and I could play with it.

Regards,
Wayne


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## cyberbri (Apr 27, 2006)

brucek said:


> What's the purpose of this?
> 
> brucek


Cutting the 74Hz area really drops the dip at 90~Hz, so it's to bring it back up to where it should be.


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## cyberbri (Apr 27, 2006)

Wayne A. Pflughaupt said:


> That’s an awful lot of filters for a graph that’s not all that bad to begin with. Your two big peaks are fairly symmetrical in shape and should respond well to a single filter set for the right bandwidth, and maybe one more at each of the three low points for “clean up.”
> 
> I suggest switching to 1/6-octave smoothing. Otherwise all the little ripples are “magnified,” and you’re tempted to chase every one of them. That’s not really productive and it doesn’t make any audible improvements.
> 
> ...



Like my last post, the 91Hz filter is to bring that range back up because the cuts below it drop it by 10~dB. It's somewhat predicted by REW, but the actual depth of it is deeper when actually measured.

The 106Hz filter is to reduce the 100~110 range of the sub+mains (graphs are all sub + mains). I listen to more 2.1 music than anything else, probably about 50% music, 35% movies/TV, 15% games. 

The peaks are all big and I used a number of filters to reduce them, and then some filters to bring up spots that ended up being dropped below the target level as a result. 

For example, this was to get rid of the peak centered at 45hz and then bring back up the edges of it lowered by the cut filters:
36.6 4 +6
40.5 3 -3
45 11 -12
52 4 +8

And this cuts the big peak (a 40Hz range, 50Hz~90Hz) but brings up the high end of it that drops 10dB or so, which was visible in graphs from previous filter applications:
66.4 4 -5
72.3 5 -10
75.75 4 -3
91 3 +9
87 3 +4

That's 9 filters there, plus two to tame the high 20s ~ 30s range:
22 9 -2
30 9 -3


You can see the net result of the filters in the green line:









And how the final results aren't really boosted over the original output except a tiny bit in a few spots:










Actually I didn't save the files, so I don't have those to play with. Redoing the sweeps doesn't take too long, though...


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## Wayne A. Pflughaupt (Apr 13, 2006)

> For example, this was to get rid of the peak centered at 45hz and then bring back up the edges of it lowered by the cut filters:
> 36.6 4 +6
> 40.5 3 -3
> 45 11 -12
> 52 4 +8


Your 45 Hz filter is really wide, which is why you needed such big “re-adjusts” at the fringes (36.6 and 54 Hz) to “re-equalize.” Also, you shouldn’t need a second filter so close to the center (that 40 Hz one). Next time you try it, try adjusting the bandwidth of the 45 Hz filter as you cut, so that those outer frequencies don’t get sucked down so much.



> 66.4 4 -5
> 72.3 5 -10
> 75.75 4 -3
> 91 3 +9
> 87 3 +4


As above - you shouldn’t have needed two filters so close to the peak. As with the other one, pick the center frequency, and play with the bandwidth as you cut. Also, tinker with moving the center frequency back and forth, to get the maximum effectiveness to the whole area without sucking out the fringe frequencies. Then maybe another filter in the 80-100 range to “clean up.”

Regards,
Wayne


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## cyberbri (Apr 27, 2006)

Wayne A. Pflughaupt said:


> Your 45 Hz filter is really wide, which is why you needed such big “re-adjusts” at the fringes (36.6 and 54 Hz) to “re-equalize.” Also, you shouldn’t need a second filter so close to the center (that 40 Hz one). Next time you try it, try adjusting the bandwidth of the 45 Hz filter as you cut, so that those outer frequencies don’t get sucked down so much.
> 
> 
> As above - you shouldn’t have needed two filters so close to the peak. As with the other one, pick the center frequency, and play with the bandwidth as you cut. Also, tinker with moving the center frequency back and forth, to get the maximum effectiveness to the whole area without sucking out the fringe frequencies. Then maybe another filter in the 80-100 range to “clean up.”
> ...



Thanks for the advice. I'll definitely try that. But I have been anyway, trying different center frequencies and bandwidths, at least with the "predicted results. Must have spent an hour last night measuring and working with REW and tweaking the results. 


But at least with the predicted results, if I narrow the bandwidth on the large cuts, it turns into a bowl shape, with the edges (like 40-41 and 49-50Hz, just as an example) still high. Widening the bandwidth to bring those edges of the bowl down causes the frequencies just past that to go down. So I can narrow the bandwidth, but that still requires narrow cuts to bring the sides down as well. 

It just takes a lot of experimenting... :crying: :surrender:


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## Wayne A. Pflughaupt (Apr 13, 2006)

> But at least with the predicted results, if I narrow the bandwidth on the large cuts, it turns into a bowl shape, with the edges (like 40-41 and 49-50Hz, just as an example) still high.


Yup. It’s really nice that we can tweak the filters in “real time” with REW. Back on the old days, you’d have to try a filter, take new 1/6-octave readings in that area, re-adjust the filter, more readings, re-adjust, and on and on. You don’t know how good you have it! 

Personally I don’t even fool with the predicted results. I just identify the center frequency where I think it needs equalizing, set a filter for it, and start playing with the filter cut (or boost) value, and bandwidth. Sometimes nudging the frequency center up or down gets the filter looking better.



> Widening the bandwidth to bring those edges of the bowl down causes the frequencies just past that to go down. So I can narrow the bandwidth, but that still requires narrow cuts to bring the sides down as well.


Don’t forget to set your smoothing for 1/6-octave – that’s the one I like to use (if others here prefer a different setting, they can chime in with their reasons). Don’t worry if you end up with little bumps at each end that are a couple of dB – not really worth worrying about.

Regards,
Wayne


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## cyberbri (Apr 27, 2006)

Flickr is currently "having a massage" so I can't save new graphs to my account just yet.

I re-worked the filters, and was able to get very similar results, plus be able to turn down the sub's gain a bit, with these filters:

Freq BW Gain
29.88 8 -2
44.5 11 -9
72.35 9 -11
37 3 +4 
52 4 +6
57 4 +3
85  5 +4
91 3 +6


These are the best results I could get.
I started with the first three, the cuts, getting the peaks down (and smoothing the up-ramp in the lows at 30hz). See *graph 1*, raw + 3 cuts prediction, and *graph 2*, the actual reading of the results of the three cuts.

Then I added the next 3 boosts to bring back up areas dropped by the first three cuts, which unavoidable to get the peaks down. See *graph 3 *- actual reading of results of first 6 filters.

Then I experimented with center frequency, bandwidth, and boost level to bring up the dip around 90Hz caused by the 75~Hz cut. That resulted in the final two filters. See *graph 4*, raw measurement + 8 filter predictions.

*Graph 5* is the final measurement. Again, this is the sub and the mains. I recalibrated the sub after this, lowering the gain on the sub about 2~3dB, which started at just under 1/2 between 50% notch and next notch, dropping under 50% notch to about 1/3 of the way to the next lowest notch.

I will post my re-calibrated measurements in the next post.


BTW, all graphs are 1/6 octave smoothed.


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## cyberbri (Apr 27, 2006)

After I got the filters set, I re-calibrated. Here's how it now looks - *graph 1* is no smoothing, and *graph 2* is 1/6 octave smoothing. Again, this is the response of the sub and both mains together. 


And for kicks, before I started doing all this tonight, I took 2 readings and compared them. I took my bass traps (2x 244 GIK Acoustics panels per front corner) out of the room and measured, then brought them back in. They are staggered, at different angles in the corner, so there is open air between each panel. The results are *graph 3*. Red is with no bass traps, blue is with. I'm not sure how to do all the decay, waterfall, impulse response, etc. to get data on the effects besides what they do to the frequency response. I'd be happy to do that as well, once I figure out how to do all that in REW...


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## brucek (Apr 11, 2006)

> Don’t forget to set your smoothing for 1/6-octave – that’s the one I like to use (if others here prefer a different setting, they can chime in with their reasons). Don’t worry if you end up with little bumps at each end that are a couple of dB – not really worth worrying about.


I know you're aware of this Wayne, but for others who may not be. Be sure to only use smoothing after you've found your filters and have your final filtered response graph. The smoothing is only a final straw to get a look at what your ears may perceive. 

John's been fairly specific about this point. I quote - _"Don't use smoothing on low frequency measurements though, filters optimised against a smoothed response will have settings that don't accurately match the room's modes."_

brucek


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## Wayne A. Pflughaupt (Apr 13, 2006)

Hmm, interesting. I wasn't aware of that - thanks. 

Am I the only one tempted to "overequalize" the rougher-looking graph? :scratch: 

Regards,
Wayne


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