Tapped Horn Cabinet for 16 Hz. organ speaker

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If that measurement is accurate you have a lot of output below your enclosure tuning frequency. This could damage the drivers. Ideally you should be using a high pass filter or tuning the enclosures lower.

I'm wondering if I can do any adjustment to this using the DSP features. If not, it may require more extensive efforts. And that might entail getting those big boxes out of the chamber. The guys who helped put them up there would not be happy about that.

BO

P.S. I'm wondering if this "sample" might have been digitally enhanced somehow to increase the low frequency output? I wouldn't be shocked to discover that it was.
 
Simplest thing to set up is a high pass filter.

The high and low parts are bass ackwards but once you get used to it it makes sense.

It's a filter that will pass frequencies higher than it is set for. So to be dead nuts safe a 16 hertz filter would protect your drivers well.

Don't know the setup on the Amp, but could read the manual.

If there is anyone out there that is fluent with the Behringer Amp settings say something!
 
I'm wondering if I can do any adjustment to this using the DSP features. If not, it may require more extensive efforts. And that might entail getting those big boxes out of the chamber. The guys who helped put them up there would not be happy about that.

Yeah, hpf is dsp. And you could eq any problems too, that's also dsp. I don't think there's any need to move the boxes anywhere. You can measure the enclosure tuning and measure the signal (again, and at different levels maybe, to make sure it's accurate) without moving anything.
 
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Yep the fun and games of what actually happens versus what we thought might be happening.

If it has no obvious peak(s), it isn't a note. Looks like white noise to me - but with a well-filtered power supply.

It also looks like Bach On needs to take care to ensure the extra-low frequency noise doesn't proceed too far downstream (assuming it is coming out of the tone generator and not just measurement system noise).

B.
 
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Below 16Hz

Yes, anything that Large below 16Hz WILL cause BIG problems :worried:

@ Bach On

We can speculate forever, but that's not going to really help you. If you could get hold of a Hardware frequency generator & connect it to your laptop, then you could see Exactly what the frequency response etc is of the laptop.

Set one f @ a time from 10Hz to 20Hz, @ 1 Hz intervals. Whilst doing that have a AC multimeter connected in parallel to the FG output. Record each f, with for eg Audacity, & note down each f + voltage level from the FG output.

You only need to record say 5 seconds on each f. After all recordings have been done, compare the results in Audacity etc with the ones you took note from the FG's output.

This will leave no doubt as to what the fr etc of the laptop is.

Also i would ask Artisan about the below 16Hz stuff. It might not be them, but a laptop etc issue !
 
REW does all of that free and in a few seconds. Never found the pieces in my test system, even with stuff from the Salvation Army store, to be anything less than flat and clean as could be.

I have a closet full of the kind of hardware you are advocating. Under a layer of dust. Not to mention the hours of labor.

Bach On - it should be pretty self-evident when your RTA is working right. When you press the key, the RTA lights up like a Christmas tree (and when silence resumes, it relaxes down over 30 seconds, unless you freeze it by untriggering) or should I say, Easter egg. (In the settings, you need to make sure that Left is chosen, assuming you are connected to the Left cable.)

Ben
 
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Hardware

My using a hardware FG etc, it allows you to Actually see what's going in, by Totally eliminating the high pass rolloff of the laptops input capacitors.

I use hardware FG's & other equipment often, as well as software. If i didn't find them useful, i would only rely on software !
 
Lets look at the big picture.

Ron has finally done some testing.

We know that there is plenty of signal a way down low.

We know he wants more out in terms of Sound Pressure Level.

The only simple test I might ask for is an 8 foot c being played and a volt meter at the end of the connector he plugged into his computer from the Artisan.

That will give us an approximate 60 hertz clean tone and a voltage telling one and all if there is enough signal to push the amplifiers to their full output.
 
And since it takes a whole 30 seconds to set up REW, for Zero's interest, here's a loop in my 7 yr old Mac laptop with $20 Behringer DAC. 5-20kHz, no kidding. Close to ruler flat down to 20 Hz and droops a bit below.

The less straight line is phase. BTW, THD is down more than 40 dB.

I wonder if that meets Zero's standards or should I compare that to my substantially less accurate Heathkit hardware with matched resistors?

I'm puzzling over the meaning of "eliminating the high pass rolloff of the laptops input capacitors."

Ben
 

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Lets look at the big picture.

Yes, let's.

We have one measurement now. It doesn't look anything like what I was expecting. There's no huge peak at 16 hz and there don't appear to be any pipe harmonics at all.

We also don't know what the Artisan sample is SUPPOSED to look like. We have no idea what organ they recorded, what environment it was in, what the measurement conditions were or how they might have processed it.

For all these reasons I have qualified all my comments by saying "if the measurement is accurate". There's no way to know if it is or not, despite Ben's certainty.

It also doesn't look anything like white noise either, sorry Ben. Unless it was somehow bandwidth limited to something like 10 - 15 hz which is exceedingly unlikely. You would have to either know what you were doing or royally screw up to get a either the Artisan or the REW software to output a white noise signal to measure like that.

White noise - In signal processing, white noise is a random signal with a constant power spectral density.

2000px-White_noise.svg.png


It's impossible to even begin to guess how accurate OP's measurement may be.

And let's not forget this is probably the first time OP has ever measured anything, so let's temper our expectations and maybe we (and when we say we I mean Ben) should back off a bit. OP is well aware that some good accurate measurements could help a lot and he is trying. Maybe give him a bit of room to breathe between the constant harassments.

Looking at the big picture, OP should be able to get a very good result without ANY measurements. Ideal? Absolutely not, but still pretty good.

The first step as I see it is to get the Inuke in the chain. The Inuke should have it's own methods of letting you know if the input signal is strong enough. I don't own one and know very little about them but maybe it's got an input signal clip light. If you can't light that up you probably are underpowering the input.

Also the inuke instructions should specifically say what input signal strength it's looking for. Some pro amps are happy with a consumer electronics signal level, some want a fully powered (much stronger) balanced input.

If the inuke is not receiving a strong input signal the DI box must also be inserted into the chain to boost it.

That should solve ALL the input signal strength problems without ever measuring anything.

Next step, which should be done before the inuke volume is ever turned up, is to set the high pass filter. This should be set somewhere between 14 and 16 hz probably, depending on the actual box tuning. Info on expected box tuning should be included in tb46's design notes. Actual box tuning may be a bit different.

Actual box tuning can be found by listening to tones, one frequency at a time and visually looking for the frequency that causes the least cone motion.

So find expected (or ideally actual) box tuning and set the inuke high pass filter to the appropriate setting.

All of this so far can be done with no measurements of any kind.

Getting further in depth would require some type of measurements. Ideally the Artisan sample could be accurately measured beyond dispute, this is going to be tricky unless a pro (or at least someone with experience) does it, since we have NO IDEA what it's supposed to look like. At that point, we could consider retuning the box and resetting the high pass filter if there really is strong output below 16 hz. If it's there and it's meant to be there in the sample, ideally it should be included in the subwoofer output.

And going a step further, frequency response measurements of several different spots in the audience listening positions is the only accurate way to know what the listeners are actually hearing. And if they aren't hearing 16 hz at 120 db or so they aren't actually perceiving it at all unless they can feel the subwoofer physically vibrating the building.

There's levels to how far this can be taken. OP is not comfortable with measurements but he's trying, let's give him a break. There's a whole world of stuff that can be done with NO measurements, and beyond that a whole universe of stuff that can be done with a series of good accurate measurements.

One step at a time.
 
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Test

Who can guess which trace is the noise and which is the loud organ note (from recording of enormous pipe organ in St John the Devine, which claims to be world's largest stone gothic cathedral)?

OK, kidding, I admit this is not a very hard test. But obviously needed as a reality check.

12.7, 24, 37, 50, 100 Hz.... (the 12.7 Hz peak, if it is meaningful at all, is 22 dB below the 24 Hz peak and perceptually twice that down)

Anybody still doubt Bach On should eyeball the RTA for his tone generator? And points downstream (since it takes only a minute to set up)? And with a mic, as per the trace below? Ummm, guess he already has his hands full this holiday weekend.

Ben
 

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That measurement screenshot means ABSOLUTELY NOTHING unless you can answer a few basic questions.

What note(s) were being played?
Was the recording post processed?
If so, how?
What was the mic location?
Was the recording processed (filtered) at the time of recording?
What does a single 16 hz note played by that organ look like at any given mic position?
Why did you choose to show that particular clip to measure?
Are there other clips of the same recording or other organ recordings that show something a whole lot different?

I could go on, do you want me to? I could go on for quite awhile.

If we're just going to post pictures of organ measurements that we know little about, try these on for size, from this link.

Acoustics of the PAB organ | Inspired Acoustics

An externally hosted image should be here but it was not working when we last tested it.


Magnitude of all stops and all notes on. Organ engine noise was removed from each single note prior combining them.

What does this mean? Not a whole lot if you don't know a lot of details.

But it APPEARS to show that this 16 hz low note organ peaks in spl in the mid frequencies below 1khz and it APPEARS that this organ is only down about 10 db down at 10 hz (reading the grey average line). And 10 hz is roughly the same level as 10 khz.

Small notes from the article -

... the loudest frequency component is not necessarily the fundamental frequency, a complex analysis algorithm was developed for the purpose of the present investigation. Additionally, some notes show slightly unstable fundamentals; ...

and

... the very approximate Schroeder frequency of the hall was found to be 15.5 Hz which corresponds to a B-1 note, one semitone below the lowest C key of the 32' foot’s fundamental frequency. This suggests that the lowest few notes, matching the experience of the organ builders, may have some audible spatial level dependency in the hall.

It would appear that these measurements were well done and took over a year to complete. But if you don't know the details, these pictures don't mean much. I barely skimmed the article, I'm not interested in researching this kind of thing. But clearly you aren't interested in knowing much about the things you are posting either. Anybody can post pictures of random stuff, you need to prove you know what the information you post really actually means.

The link to the article where the pictures and quotes were taken is above, research it if you like. Or don't, it doesn't matter. Just don't post random pictures of measurements unless you can provide a lot of details about what it actually means.
 
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Guys,

I did install the iNuke after our service last night.

The manual is mostly a "Getting Started" type of thing in six or seven languages.

I plugged my laptop with the Behringer software installed. The laptop connects to the amp via a USB cable. I was able to set the Highpass and Lowpass filters.

The bad news is that the lowest frequency for the software is 20 Hz., since that is the lowest listed frequency for the amp. There are several types of crossover parameters that can be set for EQ, such as Linkwitz, etc.

The software has a help system that may provide some assistance in addressing the lowest levels I'm trying to obtain. Too, Behringer has an online forum where questions about how to best accomplish details may be explained. I've not yet visited that site.

I plan to go back and tinker with it again today, and probably tomorrow.

The connections that I have now for the pedals are as follows:

There are two RCA outputs from the Artisan Sound Module for the pedals.

One of the Artisan outputs is connected directly to the iNuke Right input with an RCA to XLR patch cable. My understanding is that the XLR input is what is sometimes called a balanced input.

The other Artisan pedal output WAS connected to the Crown XLS1000 amps. That amp is set for highpass mode for signals higher than 96 Hz. That powers the Allen HR100 speaker. It has a frequency response from 40 to 20,000 Hz.

I kept this setup. BUT I added a splitter to the Artisan RCA jack. The splitter continues sennding the signal to the XLS1000 and HR100.

The other terminal from the splitter now plugs into the iNuke Left XLR input.

So the iNuke is now getting both pedal outputs from the Artisan through balanced inputs.

The Getting Started manual says the iNuke input can be adjusted. But it does not state how that is done. I'm guessing some research and experimentation will be required to learn how this is done.

The Samson Preamp I've ordered is a stereo unit. It will be placed between the Artisan pedal outputs and the iNuke XLR connectors. It will not be needed for the XLS1000 feeding the HR100 since the frequencies and required power are fine.

I'm HOPING that the preamp will increase the input voltage so the amp is getting what it designed for. That would mean the output would be more on the order of it's rated specs.

The other possible attribute of the Samson preamp is that it has a frequency response listed as extending down to 10 Hz. So, IF the Artisan is putting out frequencies down that low, they will not be filtered out due to the preamp's limited frequency range. (That was the problem with the Rolls preamp I tried to modify by changing out the capacitors. My efforts were NOT successful, though I haven't given up on that unit - though not for this purpose.)

My REW measurements did not focus on only one note. It probably should have. I'm going to attempt to measure one note only at a time. I'll repeat the process for the lowest four or five notes.

JAG has observed in the measurement what appears to be a spike or two below the 16 Hz. (approximate frequency) of the lowest note. This might be because my measurement was not accurate. Or, it could be something in the sample I was playing. This is a concern that will probably need research and attention lest our woofers become damaged.

The lack of equalization below 20 Hz. in the iNuke DSP software is a definite negative. I had read in one of the reviews that this could be modified. I don't know if that is true or not. The amp may prove to be the wrong tool for my purposes. I can repack it and send it back. But I'm hoping that the cutoff at 20 Hz. is not like a brick wall. Perhaps experimentation with some of the various crossover parameters can partially overcome this negative. Too, perhaps I can use the notelevel adjustment in the Artisan software will allow me to offset this negative for the pitches below the 20 Hz. frequency.

Did I mention that I was going to be experimenting?

Thanks for the input. And I'm open to any and all ideas.

Bach On
 
Have you guys ever considered the following possibility:

All organs run on blowers. Large motors turning a large fan. The are actually a type of compressor the fan is a rotary vane type. Not the normal push air around type fan.

Anyone that has been around a pipe organ knows the feeling of the blower running and not running.

That blower noise is almost certainly the ultra low frequency content in the measurement.

All the comparison in the world will not get to the nit of the issue regarding what has been measured.

We have a measurement and we have the measurement conditions stated.

From here we have to think. Not dissect to make our own points.

Take as much information as you can from the knowns and make sense of it.

Pertaining to the iNuke. I think there is a way to do what is required.

It is a little trick in the parametric EQ settings if it is similar to the DSP that I have worked with.

I'll have to look.

Out for a while.

But I'll be back.

Arnold Black Person
 
Mark could be right about those spikes being the blower noise. Some blowers are loud. Some are not so loud. Too, you can hear the sound of the air in the pipe chamber. I dunno.

The iNuke software can be downloaded for free from the Behringer site. If anyone wants to examine it out of curiosity. I'm not sure if it is Windows only, but it seems like I read somewhere that it was. Don't know if it will run in any virtual setups on a Mac.

BO

Arnold Black Person???
 
The bad news is that the lowest frequency for the software is 20 Hz., since that is the lowest listed frequency for the amp.

Bach On

And this is why I've been saying research first, purchase later. Same thing happened with the DI box.

But there are ways around this, entire discussions. As I said before, avs is the place for all things Inuke.

But here's a quick quote, it's about the DCX but I'm pretty sure it also applies to the Inuke.

FYI, you can use the DCX to high pass lower than 20Hz. There are two means to get an effective lower frequency high pass filter. If you look at both 12 & 18dB/octave Bessel filters, you will see the sharper roll off is in fact a good bit lower than the nominal Fc. Addition of 1-2 PEQ's makes a lower effective XO pretty easy. The other, more straight forward option is to use a Butterworth or LR with a simple 12dB/Oct Low Shelving filter (would have to check the "Q" of the shelf to see if or LR matches best). In simple terms, you are basically applying a Linkwitz Transform to the high pass filter. Works just fine within reason. There is probably some point where the cut and boost exceed the working range of the DSP math, but I've done this in practice a few times with good results.

WmAX--Well, that's actually quite easy. The DCX does not have ability to set filters below 20Hz. While this may seem to high, it's not really. You can easily extend the actual filter by for example setting high pass filter to 20Hz, 2nd order, Butterworth. Now add a single parameteric band at 20Hz in the EQ section on the unit, set to Q of 1.5 and gain of +4 dB. This will counteract and extend the roll off, giving you an effective roll off on the 13w7 of -3dB around 17Hz and -6db around 14hz, with rapid roll off after this; exactly what you want with this 10 cubic foot cabinet tuned to about 16Hz to prevent excessive excursion under this tuning frequency.

Essentially what they are doing is using the 20 hz hpf but using various dsp methods to boost and cut and push the ACTUAL high pass filter slope further down in frequency.

This is a major issue and so it's been discussed quite a bit.

DCX 2496 - High Pass Filter Setup - AVS Forum | Home Theater Discussions And Reviews

If you do a search you will find newer threads that deal specifically with the Inuke, not the DCX, but the process is the same.

There's also a petition to change the 20 hz hpf limit. Petition to Behringer - Please Remove 20Hz High pass - AVS Forum | Home Theater Discussions And Reviews

None of this is news the 20 hz highpass issue has been around for years with the Behringer products, this should have been researched before buying a Behringer product (or at least before expecting the Behringer dsp to do something it's not supposed to be able to do).

Luckily for you there is a workaround, you just have to search for the answers or ask for the answers in the appropriate forum.

You could even ask on this forum, there are probably people here that know how to overcome this issue but are not reading this thread. Personally I'm not prepared to find the answers to manipulate a product to do something it's not supposed to do when I don't even own the product and can't test the info I'm giving. So it's up to you.
 
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