My sub sounds better after EQ'ing it using the BFD, but... (1 Viewer)

[DanSt]

Thanks for the info Wayne. Just for the heck of it I used another preset and recalibrated using the RS meter but not adding correction values (I read about this on the forum somewhere). Every freq was set to the same values. While it isn't a proper curve I think it sounds REALLY GOOD. I still want to add a few db here and there, but overall its good. The curve kind of looks like a combo of flat and house curve. Is there something wrong with that type of calibration?

 

[DanSt]

Thanks for all the info Mark. What an explanation. I still have 8 more presets on the BFD. I will try your idea on the next preset.

 

[brucek]

Good post Mark.

I have to agree, it's important that when people introduce a house curve (through an increasing cut above 30Hz) that the integration with the mains has to be rechecked.

By the time they've reached the crossover of the sub, the "lowered" level because of the new house curve may be many dB's below the mains and an overall wholesale increase in sub level is required...... It's also important to have the headroom to do so..

brucek

 

[jeff lam]

Boost ~30Hz a little more.

How loud were you listening at? You can't really expect chest pounding bass till above 100dB's.

Try turning it up but only if your sub can take the heat!

Maybe you should try calibrating the sub a little Hot and setting a new "False" reference.

 

[Martin Schmencke]

I think part of the "issue" comes from people going to concerts and feeling that "chest pounding bass" and then thinking "that's what great bass is". But that's not what great bass is. It's what bass that's more easily produced in a large venue and "thrown" a reasonable distace from the source. As previously stated, a lot of that chest pounding is coming from what is actually the upper range of HT sub frequencies. In mid to larger venues, you don't really get a lot of sound reproduction below 50Hz.
Although I will point out something that I don't hear that much about: a lot of HT systems don't do well around the crossover point between subs and mains, and have "holes". One symptom of this is the lack of "chest pounding", since that's exactly where the crossover point is in the standard HT system. Part of the reason for this is that a lot of the compact speakers used for "mains" really don't go dwon to the crossover point well, and that a lot of subs are designed to perform well on the lower end, without much thought to the entire spectrum which they must reproduce ( remembering that a crossover is not a brick wall, it's a slope ). I can tell you that almost to a one, the HT systems which I have heard which use more sophisicated subwoofer crossover points ( Meridian, etc. ) and have been set up with "full range" mains and set the crossover point lower ( say 50Hz or even 40Hz) have had more "present"/chest pounding mid-bass.

 

[Tom Vodhanel]

Dan,
what is the latest?
With Wayne and Mark here...I don't have much to do but *lurk*...its nice
TV

 

[DanSt]

Their advice has been AWESOME. They have helped me quite a bit and I thank them very much. Now all I need to do is order a CS-Plus or two

 

[DanSt]

Ok everyone. I redid my house curve again and here are my results. The pink line shows the curve with my mains ON and the dark blue is sub only. What does everyone think now?
PS-I have already used all 12 filters so anymore changes will be kinda difficult.
PSS-New graph on next page

 

[MarkFrab]

Out of filters? Nonsense. You could always loop the output from one channel into the other - and get another 12 filters!

Regards.

 

[DanSt]

True. However, I was under the impression that could cause the signal to possibly get worse since it is being routed through the DAC's again...or something like that.

 

[Mark Seaton]

Hey Dan,

Looks like you are making some real progress. My first question of course is how the changes corellated to what you were hearing?

The subwoofer only curve looks about as good as you can hope for in and of itself. What still has me curious is the pink curve with the main speakers. In particular the drop in output above the subwoofer's range. How were you measuring this? While hardly and uncommon problem in a real room, a deep depression in the 100-200Hz range can have a very significant effect on the sound. This is where main speaker placement is very valueable. Also, could you verify what the large/small and subwoofer settings are within your reciever, along with the crossover frequency? Interestingly, this sort of effort is where you will often get more benefit to the sound of your system than with most component swaps.

As for looping the outputs, I wouldn't think 2 conversions would be much more audible than one is now, especially for a subwoofer. 12 filters is already quite a bit though, so it is certainly debateable as to how much more you can do. Looking at the response, any more filters would probably be best used as wide band contours, not worrying about wiggles in the response.

 

[DanSt]

Hello Mark!

If I understand your question correctly you want to know if it sounds better then before? I think it sounds pretty good now. The bass seems to be just right.

My measurements were all taken the same way. I used sine wave tones off of the Stryke Bass Zone CD and measured it with my RS SPL meter. My current settings have all my speakers set to small with subwoofer to yes and my crossover set at 80Hz. I definitely see what you are talking about. Am I going to need to equalize my mains now?

PS-Do also need to do something about my peak at 80Hz when my mains are on (pink line)?

 

[Mark Seaton]

I would suggest you start looking at what is coming from the main speakers first. That said, from what you describe there may be another issue to consider. From the way you describe your test, it appears that you took the subwoofer measurement with the Stryke disc input through stereo inputs, and then just unplugged the mains. If this is the case, then your response should include the subwoofer crossover. I suspect you might have equalized out some of the bass management's intended rolloff for the subwoofer. With a classic THX 80Hz XO, you would want your subwoofer 6dB down at 80Hz, with the mains also being 6dB down at this point when set to small. This of course is in the ideal. Reality kicks in, and we find response problems, and non-ideal phase causing less than perfect summation. The end goal is to achieve smooth integration between the sub and the mains.

Basically the usual way to approach the process is by first looking at the sub and the mains separately relative to your target response, and then look at how they all integrate. As for the slight bump at 80Hz, yes, that is in a somewhat sensitive range, but first do some experimentation with the mains.

 

[DanSt]

That is exactly how I calibrated using the Stryke disc. Guess I need to do a little more callibration.
PS-When am I done calibrating? I never thought it would take me almost 2 weeks to calibrate.

 

[brucek]

Dan,

I was under the impression that could cause the signal to possibly get worse since it is being routed through the DAC's again...or something like that.

Not possibly, but absolutely, the signal would get worse if you fed channel two from channel one. I suspect Mark was kidding with you, since you've already used 12 filters, indicating that maybe you've gone overboard. Unless your frequency response is very bad, it's usually not necessary to use that many filters. If you need more than 12 filters, you probably should consider either moving your sub location or using some form of room treatment.
You see, if you fed the output of one channel into the input of the other channel, besides the fact that you would be subjecting your signal to another analog to digital and back to analog conversion, the input level optimization would be upset.
To take maximum advantage of the dynamic range provided by the analog to digital converter in the BFD or any ADC for that matter, you need to provide it with a maximum input signal level that takes advantage of the greatest number of bits to represent the resultant voltage samples.
If the range from your weakest signal to the strongest signal for example, was only half the required maximum input level, you would be robbing yourself of dynamic range and subsequently increasing your signal to noise level.
Any frequency that you had reduced the voltage gain by using a filter in the first channel, would enter the second channel at a much reduced level. Not really a good idea.
I wonder if you have taken a look at your phase curve created by your 12 filters. You can monitor this by entering your filter settings into the BFD design software.
You can download the design software from Behringer and enter your filters to see what your filter curve looks like. It's useful to understand the width and interaction between your filters, and also the resultant phase changes you're ending up with. Using it can sometimes show you that two filters are really fighting each other and you can just use one and accomplish the same result. Getting a feel for the phase changes you have introduced with your filters is also fairly interesting. It can cause some unwanted cancellations around the crossover overlap with your mains.
The web site is below:
http://www.behringer.com/05_support/...mid=4&lang=eng
Select FEEDBACK DESTROYER DSP1100P Design and download the zip file.
Or download the latest version that works with XP from here: Link Removed
The latest version has a smaller screen than the original, so I prefer the original.....
You have to play around and get used to the software, but it's useful once you get the hang of it just to see how filters work. When you first start this program it shows a MIDI port error. Click OK and the program will start. The error is because you haven't set the MIDI port parameters yet. You can set them once your in the program and it won't do that error the next time.
When I entered my filters into this software, the resultant graph shows a perfect mirror image of the frequency response of my sub without filters. Kind of expected, of course, but interesting to see.
Anyway, you may find once you play with this software that you can get the same equalization result with a lot less filters.
brucek

 

[DanSt]

Alright I have reached the supreme point of frustration with this thing. It's time for me to ask for some help. I originally wanted to try this on my own, but I am tired of trying. While I have read this board and the BFD comprehensive guide (many times) I still can't get it quite right for bandwidth's and stuff. I am one of those learners that needs to be shown at least once how to do stuff like this and then I usually understand.
The last graph I had up was using all BW's of 5. That's why I had used all 12 filters.
Above is a graph of my sub response without mains (blue) and with mains on (pink). This is my response with NO EQ. Someone help me EQ this thing the right way. Pretty please with cherries on top

 

[MarkFrab]

Stab in the dark, I'd try 2 filters:

31.5Hz, 15/60 BW, +2dB

50Hz, 23/60 BW, -14dB

You might also set your xover at 90Hz, if it's an option - 100Hz if it's not. I'm assuming it's at 80Hz now?

If you don't want to mess with your xover, maybe an additional filter:

80Hz, 8/60 BW, +6dB

Just my thoughts.

 

[Chris PC]

Can the BFD be used as a crossover? No ?

 

[Mark Seaton]

Dan,
The fact you were using all the same BW filters is certainly why you needed all 12 filters. For most rooms, you won't need much more than 5-6 to tame the problems of interest, but there are certainly cases and reasons to use more. While I understand brucek's concern with looping the signal, as long as the EQing isn't super radical, and you maximize the input level without overdriving the input, I would expect the sonic effect from looping the signal would be quite hard to identify on just the subwoofer channel. Of course if one channel does the trick, no reason to loop them.
With your graph above, what you really want is one more curve up there. That would be the response of your main speakers with the subwoofer unplugged. Note that this response SHOULD have the 80Hz bass management crossover engaged. This will help us understand how your mains and sub are combining.
Quick observation of the curve above would suggest that you start with a filter centered at ~49-50Hz with around 15dB of cut. Then set the filter width to best match the curve between 50Hz and 40Hz. I would normally then spot check this frequency range with the filter enganged to confirm it is working as planned. Room modes can be very high Q in nature, and can be missed even with 1/6th octave measurement, so presume that the absolute peak is probably a bit higher than you have measured. Spot checking as you go will help you to understand what each filter is doing in the system. With this first filter, concern yourself only with flattening the area from 35-50Hz. Once you achieve this, then you can place the next filter which will probably need to be in the upper 60s to tame that section of the peak. You may require one more band of EQ in this range placed inbetween these first two to get the response you want. My next focus would be to slightly pull down the rise just above 20Hz with a wider band filter centered near 21Hz. I would aim to bring this peak to about 2-3dB above the minimum point you have indicated at 31.5Hz. Then take another plot and see how things are looking.
After you get the two general peaks mostly tamed, I would then use 2 more filters(because you can) to pull down the peaks above 80Hz in the sub's response. It looks like one around 105Hz could be helpful, and you could also notch out the peak in the 140s for peace of mind. You will have to also do some experimenting with flipping the phase on your sub to see how it affects the summation of the sub and mains.
As for the time involved in calibration, that's why guys like Keith Yates, Russ Herschellman, Toni Grimani, and others charge what they do to come in with a TEF 20 or 1/12th octave analyzer to calibrate and equalize a full surround system.
Regards,

 

[Wayne A. Pflughaupt]

Dan,

Mark is right, it would be rare to need more than 5-6 filters to smooth response. I’ve only seen a few instances where more were needed, where response was extremely ragged and uneven. Your unequalized response has only a few, broad problems, certainly not a situation where 12 filters would be needed – maybe not even six.

Perhaps a rudimentary explanation of bandwidth and filters would be helpful. Brucek recently offered an excellent technical elucidation about bandwidth and Q – perhaps he can provide a link to it for your edification. Being mathematically challenged and a musician to boot I find it easier to deal with bandwidth as octaves.

Basically, any doubling or halving of the frequency spectrum is an octave. Thus starting at a reference point of say, 100Hz, an octave down would be 50Hz and an octave up would be 200Hz. Likewise, from a reference point of 4kHz, one octave down would be 2kHz, one octave up would be 8kHz. And so on.

With that in mind, let’s take a look at your graph. Notice the end-to-end points of the 50Hz hump. You can see it starts at 31.5 Hz. Take a look one octave above that point, which is 63Hz, and you can see the hump has not fully “resolved” – it is still pretty high. Looking further “upstream” you can see the hump has pretty much “bottomed out” at about 80Hz. So to keep it simple let’s say our hump begins at 31.5Hz and ends at 80Hz. Thus:

From 31.5Hz to 63Hz = one octave.

From 63Hz to 80Hz = 1/3 octave.

Conclusion: your 50Hz response peak is 1-1/3-octaves wide.

So all you need to do is set a 1-1/3-octave filter for the BFD to eliminate this problem, right? Uh... no.

The tricky thing about bandwidth and filters is that a filter will affect its set (or so-named) bandwidth value by a like amount in both directions from the center frequency. So if you dial in a 1-1/3 octave filter centered at 50Hz, it will actually be moving frequencies from 25Hz (1-1/3 octave below 50Hz) to 125Hz (1-1/3 octave above 50Hz).

Of course, this is not what we want. We only want to effect changes for the 1-1/3-octave bandwidth between 31Hz and 80Hz. Thus to find the correct filter value you need to divide the total bandwidth by 2. So to address our 1-1/3-octave problem all we need is a 2/3-octave filter.

The next tricky thing about filters: The bandwidth they affect is not “set in stone,” but varies with amplitude (that is, the amount of boost or cut applied). For instance, let’s say you have a 1/2-octave filter that has been boosted 6dB, and let’s say it is affecting a true 1/2-octave above and below the center frequency. Let’s also say you have a spectrum analyzer so you can see what actually happens when you adjust the filter. If you raise the filter’s amplitude from +6dB to +12dB you will see on the analyzer that the outer fringes “drag up” wider and wider as boost increases. So at +12dB the total affected bandwidth is noticeably wider than it was at +6dB.

The inverse happens when you cut the filter from zero down to -6dB and lower.

Thus you have to keep in mind that after you analyze your response graph and determine how wide the area is that you want to equalize, that bandwidth is “theoretical” when it comes to actual action from the equalizer. You have to keep an eye on the beginning and ending points on your chart and see if they have changed after you use the filter.

For example, let’s go back to your 1-1/3-octave problem between 31.5Hz and 80Hz. If you use a 1/2-octave filter and cut 15dB as Mark suggested, you will probably see that 31.5 and 80Hz have also been “dragged down” considerably, perhaps as much as 4-5dB. This may or may not be a desirable effect, depending on one’s specific EQ situation; in your case it is not good. So to keep the values at 31.5Hz and 80Hz at the same pre-EQ levels, you would want to “tighten up” the filter’s bandwidth as much as needed to accomplish this goal. That’s the nice thing about parametric equalizers - the ability to dial in the exact bandwidth you need.

The next thing to keep in mind: Don’t be obsessed with perfect response. Deviations that are ±2-4dB are usually difficult to hear (depending on how broad they are). For instance, if you effect major improvements with only 3-4 filters, then “burn” another 4-5 filters chasing every last ripple in response, chances are if you switch between the 3-4 filters and the 8-9 filters you won’t hear any difference. It’s always best to use as few filters as possible to get the job done.

Regarding your specific situation, Dan, I think there are two ways you could approach it: You could smooth response while simultaneously dialing in a “guess” at a house curve (as you did with your first chart), or shoot for flat response (Mark’s recommendations should send you in that direction) and then use an overlay filter to dial in a precise house curve, as brucek did in the link I provided previously. Either way it should take only a few properly set-up filters.

Regards,

Wayne A. Pflughaupt