Sorry, meant, "turn all amps except sub amp off". Changed my mind halfway through the entry. Still worth doing, and may help you get better quality sound, too.
Corner and Big Box data
Hi JAG, and all
I scanned through the other thread for information as to the triangular box, here are some links:
http://www.diyaudio.com/forums/subwoofers/272833-16-hz-church-organ.html
Post #565/584/585/598 basic forms
Post #650/666/738 for build notes/pictures
I did not have measured T/S parameters for the driver, and thus do not claim, that my approximate Hornresp simulation has much to do with reality. Especially the output loss below ~24Hz, and the bump @ ~32Hz that result from using the "Large Voice Coil" button are questionable, and can only be verified, or laid to rest with measurements. The attached text file is all I have been able to find as far as simulations for the SI 18" in the corner box are concerned. I'm also attaching a simulation for the Big Box w/ dual ST385-8s.
You really need decent measurements, and T/S data to simulate eq/filter settings. With dsp functionality it should be possible to get a flat response, and protect the driver below the passband.
I second the motion of measuring the subwoofer boxes: first each individually, and then both of them together, using REW, and having everything else turned off.
Regards,
Hi JAG, and all
I scanned through the other thread for information as to the triangular box, here are some links:
http://www.diyaudio.com/forums/subwoofers/272833-16-hz-church-organ.html
Post #565/584/585/598 basic forms
Post #650/666/738 for build notes/pictures
I did not have measured T/S parameters for the driver, and thus do not claim, that my approximate Hornresp simulation has much to do with reality. Especially the output loss below ~24Hz, and the bump @ ~32Hz that result from using the "Large Voice Coil" button are questionable, and can only be verified, or laid to rest with measurements. The attached text file is all I have been able to find as far as simulations for the SI 18" in the corner box are concerned. I'm also attaching a simulation for the Big Box w/ dual ST385-8s.
You really need decent measurements, and T/S data to simulate eq/filter settings. With dsp functionality it should be possible to get a flat response, and protect the driver below the passband.
I second the motion of measuring the subwoofer boxes: first each individually, and then both of them together, using REW, and having everything else turned off.
Regards,
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Thanks for all that research. If OP can verify the height and internal volume of the box (not including port) and verify the cross sectional area and length of the port I have all I need.
You might be skeptical about the "Large Voice Coil" results and I admit they may not be perfectly accurate but I will bet money that they are more accurate than not using that option. For this driver and drivers of this type, anyway. My research speaks for itself, I think. There's plenty of sims compared to real world measurements in my research paper.
I agree that having measured t/s and frequency response measurements (at a minimum) would help a lot to get more accurate data to derive decisions on, but I seriously doubt OP is going to measure t/s or cart the box outside and measure FR in any meaningful way that could help us. So we do what we can do and make do. If there's any doubt I can specify a conservative hpf setting, but unless OP can get a boatload more power to the amp it's not really going to matter as he'll never get anywhere near needing it.
Even if you don't trust my large coil adjustment it doesn't make a whole lot of difference.
I ran the sim through with inductance effects. Same sim but I changed Bl from 21.39 (I think) to 16.3, which is the curve fit I did in my research (a bit more accurate than the "Large Coil" Hornresp tool which uses a generic formula based on all tested drivers).
The first graph is the unfiltered response of the same sim as above but with the inductance effect included. The hump at ~30 hz isn't as pronounced as the "Large Coil" tool indicates, as curve fit shows a bit less Bl reduction than the tool calculates.
Second and third graph show frequency response and excursion with EXACTLY the same filters applied that were applied in the previous sim (the 20 hz 2nd order LR high pass, 21 hz low shelf and 20 hz parametric eq boost).
So you can see the exact same filter settings yield a reasonable frequency response (although 16 hz is 3 db down now) and still protect the driver from excursion below tuning. Simulated at same voltage so excursion is a bit higher, just need to turn the volume down a bit to stay within xmax. The point is that the excursion below tuning doesn't skyrocket above the excursion above tuning.
So either way, with or without inductance effects, for the purposes of protecting the driver it doesn't matter a whole lot either way.
The bigger question is which method to use (inductance effects included or not) and what target frequency response to shoot for, as we have no idea what the room is going to do with the frequency response.
For my recommendation I'm going to sim it both ways (I trust my inductance correction method but I understand others might not) and I'll design a hpf that protects BOTH sub boxes either way (whether you trust my inductance correction method or not).
I ran the sim through with inductance effects. Same sim but I changed Bl from 21.39 (I think) to 16.3, which is the curve fit I did in my research (a bit more accurate than the "Large Coil" Hornresp tool which uses a generic formula based on all tested drivers).
The first graph is the unfiltered response of the same sim as above but with the inductance effect included. The hump at ~30 hz isn't as pronounced as the "Large Coil" tool indicates, as curve fit shows a bit less Bl reduction than the tool calculates.
Second and third graph show frequency response and excursion with EXACTLY the same filters applied that were applied in the previous sim (the 20 hz 2nd order LR high pass, 21 hz low shelf and 20 hz parametric eq boost).
So you can see the exact same filter settings yield a reasonable frequency response (although 16 hz is 3 db down now) and still protect the driver from excursion below tuning. Simulated at same voltage so excursion is a bit higher, just need to turn the volume down a bit to stay within xmax. The point is that the excursion below tuning doesn't skyrocket above the excursion above tuning.
So either way, with or without inductance effects, for the purposes of protecting the driver it doesn't matter a whole lot either way.
The bigger question is which method to use (inductance effects included or not) and what target frequency response to shoot for, as we have no idea what the room is going to do with the frequency response.
For my recommendation I'm going to sim it both ways (I trust my inductance correction method but I understand others might not) and I'll design a hpf that protects BOTH sub boxes either way (whether you trust my inductance correction method or not).
An externally hosted image should be here but it was not working when we last tested it.
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I've gone back and re-read some of the posts regarding the iNuke.
Many seem to feel the iNuke will need a lot of electrical power if it is run so it is approaching the clipping level. My early experience with it last night after adding the Samson bump box was that I could not turn the volume up without the pedal music line being too dominant for the other organ sounds.
JAG asked if I had turned UP the boost pot on the Samson. The answer is that I experimented with raising it, but then turned it back to the mid point setting. That's where it was when I removed it from the package.
I do have the option within the Artisan system of reducing the level of each individual sample. I can, for example, turn down the level of the 32 foot Contra Bourdon. That is the sample that is supposed to be able to produce a nominal 16 Hertz for the lowest note.
So my questions:
Is it better to run the iNuke so it is operating above the -24 dB point? I turned the gain up to the point where I was running at -12 dB for some material with only a few louder passages causing the -6 dB LED to briefly flicker. At no point did the material played cause the -24, -12 and -6 light to stay on constantly. And I never see the Clipping LED flash.
I am getting the impression that the amp will not provide enough power to get the speakers to produce 16 Hertz with authority unless I'm running the amp at the -6 dB level.
Yet when if I have the gain up that high, then the pedal is just too loud for the other organ sounds. What's the trade-off?
Sorry to be dense.
Bach On
Many seem to feel the iNuke will need a lot of electrical power if it is run so it is approaching the clipping level. My early experience with it last night after adding the Samson bump box was that I could not turn the volume up without the pedal music line being too dominant for the other organ sounds.
JAG asked if I had turned UP the boost pot on the Samson. The answer is that I experimented with raising it, but then turned it back to the mid point setting. That's where it was when I removed it from the package.
I do have the option within the Artisan system of reducing the level of each individual sample. I can, for example, turn down the level of the 32 foot Contra Bourdon. That is the sample that is supposed to be able to produce a nominal 16 Hertz for the lowest note.
So my questions:
Is it better to run the iNuke so it is operating above the -24 dB point? I turned the gain up to the point where I was running at -12 dB for some material with only a few louder passages causing the -6 dB LED to briefly flicker. At no point did the material played cause the -24, -12 and -6 light to stay on constantly. And I never see the Clipping LED flash.
I am getting the impression that the amp will not provide enough power to get the speakers to produce 16 Hertz with authority unless I'm running the amp at the -6 dB level.
Yet when if I have the gain up that high, then the pedal is just too loud for the other organ sounds. What's the trade-off?
Sorry to be dense.
Bach On
When we get to frequency response measurements (especially at different locations in the audience position) we'll see what's going on. I'm guessing you have plenty of spl in the 30 - 100 hz range from these subs but very little 16 hz. And you really won't know until you measure.
You might have to (probably will have to) do quite a bit of eq to get something resembling flat response in the audience. Level matching individual notes is not the same thing as getting a flat frequency response. These notes probably (should) have harmonics, and if so they may be as strong or stronger than the fundamental. So you may be hearing plenty of higher frequency harmonics and very little 16 hz.
All of this is speculation until you measure.
The correct level for the Inuke is whatever is required to get the spl you need to match the real pipes with flat frequency response in the audience location. At this point it's a complete unknown.
You might have to (probably will have to) do quite a bit of eq to get something resembling flat response in the audience. Level matching individual notes is not the same thing as getting a flat frequency response. These notes probably (should) have harmonics, and if so they may be as strong or stronger than the fundamental. So you may be hearing plenty of higher frequency harmonics and very little 16 hz.
All of this is speculation until you measure.
The correct level for the Inuke is whatever is required to get the spl you need to match the real pipes with flat frequency response in the audience location. At this point it's a complete unknown.
I think Art would be better qualified in this answer than myself.
But how I set up gain controls on amps to subwoofers is to set the gain just below clipping. And introduce a soft limiter to keep the output safe. This way all you volume adjustment is on the device sending the signal. You have almost all the power on tap if need be.
It works for the subwoofer applications I have done.
But as I said. I'm not the guy here with the greatest amount of live sound experience.
But how I set up gain controls on amps to subwoofers is to set the gain just below clipping. And introduce a soft limiter to keep the output safe. This way all you volume adjustment is on the device sending the signal. You have almost all the power on tap if need be.
It works for the subwoofer applications I have done.
But as I said. I'm not the guy here with the greatest amount of live sound experience.
Bach On,
1) There is nothing "better" about any indicator level, it just indicates output level. In automotive terms, think of the indicators like a tachometer, they just tell you how close to red-line your engine (amp) is. A level of -6 is like running an engine with a 5000 RPM red-line at 1250 RPM, the engine (amp) is loafing.
2) Since you can reach -6 dB, using the amplifier's EQ you should easily be able to hit "full tilt boogie" (clip/ limit, full power). The speakers can easily take all the peak power the amp can deliver above Fb (box tuning). The EQ settings JAG suggested earlier are a good start point.
Remember to set the amp's RMS limiters (with a long attack and release so they do not peak limit) to half the AES rating of your subs (this can best be done with a 60Hz sine wave, speakers disconnected while setting) and do not apply boost below Fb.
3) Using the PEQ you can selectively boost the low end of the spectrum without adding level in the upper, more sensitive range of the subs.
4)No trade off, just EQ for the balance of fundamental to harmonics you want.
The lower frequencies will not be reduced as much as upper when the "volume door"(forgot the proper organ term) is not wide open.
Art
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Art,
The Shades control the amount of sound that leaves the pipe/speaker chamber.
Thanks. I get the automotive analogy. Engines can provide more horsepower at higher RPMs. And for quicker acceleration and/or more speed - more HP is usually needed.
The reason I'm asking the question is that I'm trying to ascertain whether the Crown XLS1500 MIGHT be a better match for those homemade boxes. I may not need an amp with a much HIGHER amount of Horsepower if I cannot drive the speakers too loud for the other organ sounds. Using the Samson bump box may increase the output of the 1500 (as it did for the iNuke) so that Crown is a better match for providing the SPL I need - and can actually utilize.
The 1500 can provide 525 watts per channel at 4 ohms. The iNuke 3000dsp can provide 880 watts RMS per channel at 4 ohms. The iNuke is the clear winner in potential capacity. BUT - if I cannot turn the gain up without overpowering the rest of the organ mix, the iNuke is not actually going to have the opportunity of delivering anything close to it's published rating.
I was running the 1500 at maximum gain, but that was without the bump box increasing the input signal strength. If I run the 1500 with the bump box, it MIGHT have enough power for the SI HT18 and the box with the two Dayton Audio ST385-8 15" Series II Woofers. The Dayton woofers are rated for 300 watts RMS at 8 ohms. I'm running two of them connected for a 4 ohm load. I THINK this means that the Dayton's would be perfectly comfortable with 600 watts RMS. The HT18 is rated for 600 watts per voice coil. I bought the 2 ohm version. But the twin voice coils were connected so the circuit is a 4 ohm one. Doesn't that mean the HT18 connected as I've wired it is perfectly comfortable with 600 watts RMS at 4 ohms?
I have the general impression that the circuitry of more powerful amps generally "negotiates" the power with the speaker. And a speaker can accept more than it's RMS rating - at least for a finite period of time. The RMS number is just a suggested rating or general guideline.
Yes. I see the DSP as a definite benefit of the iNuke. The Crown XLS1500 obviously doesn't have that. But that aside, is the 1500 a better match in power than the iNuke3000? Do I need 300 more HP if the speed limit where I'll be driving never goes above 45 MPH?
Finally, I'm also between a "rock and a hard place" on the amount of electrical power (aka: watts) I can provide to feed all those amps. Do I have the luxury of having a more powerful amp when a lower powered amp would work and the amount of current is in short supply?
BO
The SI power rating is not a conservative one. If you actually feed it 600 watt rms sine wave the driver won't last long. It will start into heavy power compression very soon and after awhile it will fail thermally (voice coil will melt).
But if your signal is dynamic and the duty cycle is low it could take well over it's rated power handling.
If you don't know the crest factor and duty cycle of your signal you should probably follow weltersys' advice and limit conservatively.
The Inuke DSP is a clear and distinct definite advantage. You might not even realize how valuable it is until you measure your frequency response. And even if you didn't need any eq capability you still need a high pass filter which the crown probably doesn't have, so you would have to buy ANOTHER external unit for that. (Mini dsp is about $100.)
A quick search indicates that your Crown is smps power supply, so both the Crown and Inuke are very efficient, this is a good thing when your breakers are tripping. So there's no advantage (or at least not much) for either amp as far as efficiency goes.
So it really comes down to the dsp. I'd use the unit with built in dsp even if it's potential output exceeds what you may need. This isn't a bad thing, you don't need to "power match" the speaker and the amp. As long as you have enough power it doesn't matter if you have a bit extra power that isn't fully used. And you might find that once you measure you may need a lot of boost at 16 hz, so you may actually need quite a bit more power than you think. it's not like the Inuke was expensive or you need it for something else. You have it and it does the job you bought it for, might as well use it. What's the alternative? Stick it in a closet and not use it?
I'm just guessing, but I don't think you will have any room gain to speak of. And my sims indicate that 16 hz isn't very strong in your sub. And then add in the cumulative rolloff of all the electronics in the signal chain and you might need a hefty boost at 16 hz. That's my guess. And every 3 db boost requires double the power. This can add up really quick. If 16 hz measures several db below the higher frequencies you may need all the power the Inuke has and more just to get flat response.
I've said it about a dozen times now, I'll say it again. Don't make any decisions about changing stuff until you measure what you have now. Making decisions without having any idea what your frequency response in the audience is would be very short sighted.
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JAG,
Thanks for very lucid answers addressing my questions.
I think one issue that has been present within the recent threads and posts is the range of expectations about just how loud the 16 Hertz tones need to be for our organ. I have only played one pipe organ with real 32 foot pipes. Their sound is there. You can hear/feel the fundamental tone. But it doesn't have the SPL of, say, a 16 foot Diapason. I've played many electronic organs that had a 32 foot stop on the organ. But the speaker system often was inadequate to actually produce the fundamental of the lowest notes. Mainly what you got was some rumble and a lot of the first and second overtones until the notes moved up into a range where the speakers actually could produce the fundamental frequency of a note. I have never liked that kind of stop and I almost never used it.
That is part of the reason that I originally had not considered including 32 foot samples for our enhanced organ. But as I found out more from informed people - like you - I began to see that it MIGHT be possible to have an electronic system that actually could duplicate the sound of real 32 foot pipes. But like everyone who tries, I quickly discovered that it wouldn't be nearly as simple as building speakers that could play down to 32 hertz.
I'm not trying to have the lowest sounds reaching 100 or more decibels. I think it would be much too loud - especially in a Sanctuary the size of my church. And large pipe organs with 32 foot pipes don't have those lowest sounds at anything approaching 100 dB. I don't yet know just where the sweet spot is to balance those lowest sounds with the remainder of the organ notes. I suspect it may be closer to 40 or 50 decibels. But that is only a guess. I think I will know it when I hear it.
What I think I read in your post is that the DSP is worth the money. And the iNuke includes it "for free". I'd have to buy another unit if I wanted to have it to use with the XLS1500.
What I think I was sensing from some of the posts was that I needed to run the iNuke with an output in the -6dB range. And I was hearing that this was just going to be far too loud for the rest of the mix in our organ.
So, I'm guessing that our final pedal settings for a balanced mix will probably be that softer pedal sounds will be under -24dB. I little louder sounds may cause the -24dB LED to light. And with full organ sounds the -24 LED will be steadily lit and the -12 LED will flicker. I don't think our mix will be anywhere close to balanced if we have the amp's gain turned up to the point we are seeing the -6dB LED lit.
That brings me back to the electrical power issue. If I'm not trying to have the iNuke produce a -6dB output, then the number of electrical watts the amp needs will be lower than if it was trying to produce more SPL. I was seeing a nominal 140 watts last night with nearly all the pedal stops running. Thus, I just cannot imagine how this amp would ever need anything like 1,200 - 1,800 watts of electrical power to operate with the sounds it is going to get into our two speaker boxes.
I invite you or others to feel free to correct and educate me if you think I'm wrong.
Bach On
P.S. I do understand the concern that what people hear in various parts of the Sanctuary may be very different from what I hear sitting on the organ bench. And I will do REW measurements in various spots within the Sanctuary to compare the differences. I did these measurements before the boxes were put into the speaker chamber. I need to make new ones since the amount of sound that is actually making out of the speaker chamber is very different than having a mike just a foot in front of the speakers.
Bach On
Thanks for very lucid answers addressing my questions.
I think one issue that has been present within the recent threads and posts is the range of expectations about just how loud the 16 Hertz tones need to be for our organ. I have only played one pipe organ with real 32 foot pipes. Their sound is there. You can hear/feel the fundamental tone. But it doesn't have the SPL of, say, a 16 foot Diapason. I've played many electronic organs that had a 32 foot stop on the organ. But the speaker system often was inadequate to actually produce the fundamental of the lowest notes. Mainly what you got was some rumble and a lot of the first and second overtones until the notes moved up into a range where the speakers actually could produce the fundamental frequency of a note. I have never liked that kind of stop and I almost never used it.
That is part of the reason that I originally had not considered including 32 foot samples for our enhanced organ. But as I found out more from informed people - like you - I began to see that it MIGHT be possible to have an electronic system that actually could duplicate the sound of real 32 foot pipes. But like everyone who tries, I quickly discovered that it wouldn't be nearly as simple as building speakers that could play down to 32 hertz.
I'm not trying to have the lowest sounds reaching 100 or more decibels. I think it would be much too loud - especially in a Sanctuary the size of my church. And large pipe organs with 32 foot pipes don't have those lowest sounds at anything approaching 100 dB. I don't yet know just where the sweet spot is to balance those lowest sounds with the remainder of the organ notes. I suspect it may be closer to 40 or 50 decibels. But that is only a guess. I think I will know it when I hear it.
What I think I read in your post is that the DSP is worth the money. And the iNuke includes it "for free". I'd have to buy another unit if I wanted to have it to use with the XLS1500.
What I think I was sensing from some of the posts was that I needed to run the iNuke with an output in the -6dB range. And I was hearing that this was just going to be far too loud for the rest of the mix in our organ.
So, I'm guessing that our final pedal settings for a balanced mix will probably be that softer pedal sounds will be under -24dB. I little louder sounds may cause the -24dB LED to light. And with full organ sounds the -24 LED will be steadily lit and the -12 LED will flicker. I don't think our mix will be anywhere close to balanced if we have the amp's gain turned up to the point we are seeing the -6dB LED lit.
That brings me back to the electrical power issue. If I'm not trying to have the iNuke produce a -6dB output, then the number of electrical watts the amp needs will be lower than if it was trying to produce more SPL. I was seeing a nominal 140 watts last night with nearly all the pedal stops running. Thus, I just cannot imagine how this amp would ever need anything like 1,200 - 1,800 watts of electrical power to operate with the sounds it is going to get into our two speaker boxes.
I invite you or others to feel free to correct and educate me if you think I'm wrong.
Bach On
P.S. I do understand the concern that what people hear in various parts of the Sanctuary may be very different from what I hear sitting on the organ bench. And I will do REW measurements in various spots within the Sanctuary to compare the differences. I did these measurements before the boxes were put into the speaker chamber. I need to make new ones since the amount of sound that is actually making out of the speaker chamber is very different than having a mike just a foot in front of the speakers.
Bach On
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50 db is literally whisper quiet. And the only reason you can hear a person whisper at 50 db is because they vocalize in a frequency range that humans are sensitive to.
At 16 hz 50 db is inaudible. It's also too quiet to feel. So 16 hz at 50 db is literally worthless, it's not perceptible. If this is your goal you might at well not have 16 hz at all since no one would even notice if it was there or not.
Yes.
Just because you are playing a 16 hz note on your organ does NOT mean you are hearing or perceiving 16 hz. You may be hearing sounds that are much higher in frequency than 16 hz and be hearing very little actual 16 hz fundamental. We don't yet have a known good measurement of the spectral content of your 16 hz organ note sample.
Flat frequency response is exactly that. Flat. You can't measure or judge whether your system is flat by ear, especially not with organ notes. You need to do a frequency sweep and measure the results in the audience.
I'll try to explain this one more time with a picture.
That's a personal guess at what I think your SI 18 sub would measure like in the audience position. After you add inductance effects to the sim and add in a cumulative roll off of a few different electronic components in the signal chain I think this is somewhat close to how your system could be actually performing.
You see how 16 hz is about 9 db down from the higher frequencies?
Ok, so if it takes 500 watts to hit this level and you need to boost 16 hz 9 db to get flat (up to the level of the higher frequencies), that 9 db of boost means that you are actually using 4000 watts at 16 hz. (Double power for every 3 db of boost.)
So at 40 hz (which needs no boost) the sub might need 500 watts (or actually much much less as shown in the picture below). At 16 hz (which might need 9 db of boost to be flat) the sub might need 4000 watts.
This is further complicated by the fact that the sub will draw different amounts of power at different frequencies because the impedance is different at different frequencies.
That's how much the sub draws at different frequencies. It's shown without a hpf or any processing but you can see pretty clearly that if you compare it to the frequency response graph above, it's drawing very little power at say 40 hz where it's putting out the most output and it's drawing a HUGE amount of power at 16 hz where it's 9 db down.
Actual power requirements to hit a flat measured frequency response at the audience position is a pretty in depth topic that I'd like to avoid because it takes a long time to explain.
For now let's leave it at this. I don't think you have anything resembling flat frequency response at the audience position. I think you will need a lot of boost at 16 hz to get the response flat. The fact that you can play your 16 hz organ note and it seems subjectively loud means nothing (yet), it could be higher frequencies in the 16 hz sample that are loud, not the 16 hz fundamental.
This is all speculation until you actually do some frequency response measurements at the audience position, but I think this is how it's going to play out.
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16Hz
@ just a guy
Hi, that's why i uploaded a clean ONLY 16Hz .WAV with NO harmonics file for him to play & Loop in Audacity etc. This is to test the Whole audio chain, from the input to the Samson etc unit, to the Amp & then Speaker. Not only that, but to test his computer for frequency response @ Low frequencies.
Also someone kindly gave him links to 2 software Frequency Generators so he can test different f's too.
From what he said earlier about the LF's being too loud when the amp/speaker wasn't been driven much, i "think" if he sets the levels in the Samson right with the Amp vol fully up, combined with a correct HPF & EQ in with the Amps DSP, he "should" be ok !
@ just a guy
Hi, that's why i uploaded a clean ONLY 16Hz .WAV with NO harmonics file for him to play & Loop in Audacity etc. This is to test the Whole audio chain, from the input to the Samson etc unit, to the Amp & then Speaker. Not only that, but to test his computer for frequency response @ Low frequencies.
Also someone kindly gave him links to 2 software Frequency Generators so he can test different f's too.
From what he said earlier about the LF's being too loud when the amp/speaker wasn't been driven much, i "think" if he sets the levels in the Samson right with the Amp vol fully up, combined with a correct HPF & EQ in with the Amps DSP, he "should" be ok !
Thanks again JAG,
One of the charts that attempts to compare sound pressure levels looks like this one:
Sound sources (noise)
Examples with distance
Sound pressure
Level Lp dB SPL
Jet aircraft, 50 m away 140 Threshold of pain 130 Threshold of discomfort 120 Chainsaw, 1 m distance 110 Disco, 1 m from speaker 100 Diesel truck, 10 m away 90 Kerbside of busy road, 5 m 80 Vacuum cleaner, distance 1 m 70 Conversational speech, 1 m 60 Average home 50 Quiet library 40 Quiet bedroom at night 30 Background in TV studio 20 Rustling leaves in the distance 10 Hearing threshold 0
My understanding is that such charts are relatively useless since the distance for many of these items is not provided.
Here's a chart that provides a bit more information:
190 dBA Heavy weapons, 10 m behind the weapon (greatest level) 180 dBA Toy pistol fired close to ear (greatest level) 170 dBA Slap on the ear, fire cracker explodes on shoulder, small arms
at a distance of 50 cm (greatest level)
160 dBA Hammer stroke on brass tubing or steel plate at 1 m distance,
airbag deployment very close at a distance of 30 cm (greatest level) 150 dBA Hammer stroke in a smithy at 5 m distance (greatest level) 130 dBA Loud hand clapping at 1 m distance (greatest level) 120 dBA Whistle at 1 m distance, test run of a jet at 15 m distance Threshold of pain, above this fast-acting hearing damage in short action is possible 115 dBA Take-off sound of planes at 10 m distance 110 dBA Siren *) at 10 m distance, frequent sound level in discotheques and close
to loudspeakers at rock concerts, violin close to the ear of an orchestra
musicians (greatest level) 105 dBA Chain saw at 1 m distance, banging car door at 1 m distance (greatest level),
racing car at 40 m distance, possible level with music head phones 100 dBA Frequent level with music via head phones, jack hammer at 10 m distance 95 dBA Loud crying, hand circular saw at 1 m distance 90 dBA Angle grinder outside at 1 m distance Over a duration of 40 hours a week hearing damage is possible 85 dBA 2-stroke chain-saw at 10 m distance, loud WC flush at 1 m distance 80 dBA Very loud traffic noise of passing lorries at 7.5 m distance,
high traffic on an expressway at 25 m distance 75 dBA Passing car at 7.5 m distance, un-silenced wood shredder at 10 m distance 70 dBA Level close to a main road by day, quiet hair dryer at 1 m distance to ear 65 dBA Bad risk of heart circulation disease at constant impact is possible 60 dBA Noisy lawn mower at 10 m distance 55 dBA Low volume of radio or TV at 1 m distance, noisy vacuum cleaner at
10 m distance 50 dBA Refrigerator at 1 m distance, bird twitter outside at 15 m distance 45 dBA Noise of normal living; talking, or radio in the background 40 dBA Distraction when learning or concentration is possible 35 dBA Very quiet room fan at low speed at 1 m distance 25 dBA Sound of breathing at 1 m distance 0 dB Auditory threshold
What I THINK I understand is that the distance from the ear is a significant factor in determining the SPL as perceived by the ear (or a microphone).
So a 100 dB pipe sound one foot from the speaker will normally be perceived as being much softer 50 feet away unless room gain impacts in some way to mitigate this. There are complex formulas that can show these relative values. AND the wave pattern and length can also greatly impact how loud the lowest sounds are perceived or measured.
I do get that REW measurements are a better tool to use than just trying to determine how loud it sounds to my ears. And I will make those measurements.
But voicing an organ (balancing all the various sounds it produces so they blend well together) is largely dependent of the taste of individuals. They guy who likes loud bass in his car audio system just seems to like his music better with more bass. Others find this balance far less desirable. All of this CAN BE highly subjective and is often just a matter of one's personal opinion.
The organist, Virgil Fox, might have a different concept of balance than the organist, E. Power Biggs did. And Cameron Carpenter would have yet a different concept. This isn't scientific as much as a matter of personal preference.
So, I'll do the REW measurements. But I'm going to ultimately set all the voicing to match my personal tastes. The next organist at my church may want to change the balance, and so on.
I get that the low frequencies like 16 Hz. require that a speaker get enough power to produce the sounds. And providing that power is the duty of the amp.
But then personal tastes enter the picture.
I do not think 16 Hz frequencies at even 80 or 90 dB at 12 feet is going to be a match the rest of the organ. And I may be wrong. I again remind everyone that I cannot raise or lower the SPL of the real pipes. THEY provide and establish the baseline SPL for the organ. Then the digital sounds must be produced at a SPL that is at that baseline.
I believe I did follow what you wrote. But I repeat that most organ pipes I've heard that produce 16 Hz. (those big 32 foot long pipes) do not have a SPL that is as loud as a 16 foot Diapason producing 32 Hz. I have not made REW measurements to prove this. But that is what my ears have been telling me for many years.
BO
One of the charts that attempts to compare sound pressure levels looks like this one:
Sound sources (noise)
Examples with distance
Sound pressure
Level Lp dB SPL
Jet aircraft, 50 m away 140 Threshold of pain 130 Threshold of discomfort 120 Chainsaw, 1 m distance 110 Disco, 1 m from speaker 100 Diesel truck, 10 m away 90 Kerbside of busy road, 5 m 80 Vacuum cleaner, distance 1 m 70 Conversational speech, 1 m 60 Average home 50 Quiet library 40 Quiet bedroom at night 30 Background in TV studio 20 Rustling leaves in the distance 10 Hearing threshold 0
My understanding is that such charts are relatively useless since the distance for many of these items is not provided.
Here's a chart that provides a bit more information:
190 dBA Heavy weapons, 10 m behind the weapon (greatest level) 180 dBA Toy pistol fired close to ear (greatest level) 170 dBA Slap on the ear, fire cracker explodes on shoulder, small arms
at a distance of 50 cm (greatest level)
160 dBA Hammer stroke on brass tubing or steel plate at 1 m distance,
airbag deployment very close at a distance of 30 cm (greatest level) 150 dBA Hammer stroke in a smithy at 5 m distance (greatest level) 130 dBA Loud hand clapping at 1 m distance (greatest level) 120 dBA Whistle at 1 m distance, test run of a jet at 15 m distance Threshold of pain, above this fast-acting hearing damage in short action is possible 115 dBA Take-off sound of planes at 10 m distance 110 dBA Siren *) at 10 m distance, frequent sound level in discotheques and close
to loudspeakers at rock concerts, violin close to the ear of an orchestra
musicians (greatest level) 105 dBA Chain saw at 1 m distance, banging car door at 1 m distance (greatest level),
racing car at 40 m distance, possible level with music head phones 100 dBA Frequent level with music via head phones, jack hammer at 10 m distance 95 dBA Loud crying, hand circular saw at 1 m distance 90 dBA Angle grinder outside at 1 m distance Over a duration of 40 hours a week hearing damage is possible 85 dBA 2-stroke chain-saw at 10 m distance, loud WC flush at 1 m distance 80 dBA Very loud traffic noise of passing lorries at 7.5 m distance,
high traffic on an expressway at 25 m distance 75 dBA Passing car at 7.5 m distance, un-silenced wood shredder at 10 m distance 70 dBA Level close to a main road by day, quiet hair dryer at 1 m distance to ear 65 dBA Bad risk of heart circulation disease at constant impact is possible 60 dBA Noisy lawn mower at 10 m distance 55 dBA Low volume of radio or TV at 1 m distance, noisy vacuum cleaner at
10 m distance 50 dBA Refrigerator at 1 m distance, bird twitter outside at 15 m distance 45 dBA Noise of normal living; talking, or radio in the background 40 dBA Distraction when learning or concentration is possible 35 dBA Very quiet room fan at low speed at 1 m distance 25 dBA Sound of breathing at 1 m distance 0 dB Auditory threshold
What I THINK I understand is that the distance from the ear is a significant factor in determining the SPL as perceived by the ear (or a microphone).
So a 100 dB pipe sound one foot from the speaker will normally be perceived as being much softer 50 feet away unless room gain impacts in some way to mitigate this. There are complex formulas that can show these relative values. AND the wave pattern and length can also greatly impact how loud the lowest sounds are perceived or measured.
I do get that REW measurements are a better tool to use than just trying to determine how loud it sounds to my ears. And I will make those measurements.
But voicing an organ (balancing all the various sounds it produces so they blend well together) is largely dependent of the taste of individuals. They guy who likes loud bass in his car audio system just seems to like his music better with more bass. Others find this balance far less desirable. All of this CAN BE highly subjective and is often just a matter of one's personal opinion.
The organist, Virgil Fox, might have a different concept of balance than the organist, E. Power Biggs did. And Cameron Carpenter would have yet a different concept. This isn't scientific as much as a matter of personal preference.
So, I'll do the REW measurements. But I'm going to ultimately set all the voicing to match my personal tastes. The next organist at my church may want to change the balance, and so on.
I get that the low frequencies like 16 Hz. require that a speaker get enough power to produce the sounds. And providing that power is the duty of the amp.
But then personal tastes enter the picture.
I do not think 16 Hz frequencies at even 80 or 90 dB at 12 feet is going to be a match the rest of the organ. And I may be wrong. I again remind everyone that I cannot raise or lower the SPL of the real pipes. THEY provide and establish the baseline SPL for the organ. Then the digital sounds must be produced at a SPL that is at that baseline.
I believe I did follow what you wrote. But I repeat that most organ pipes I've heard that produce 16 Hz. (those big 32 foot long pipes) do not have a SPL that is as loud as a 16 foot Diapason producing 32 Hz. I have not made REW measurements to prove this. But that is what my ears have been telling me for many years.
BO
Bach On, you are completely missing the point. Flat response at the listener position is not subjective or subject to personal taste.
AFTER you achieve flat response at the audience position, THEN you can adjust things to your subjective personal taste.
UNTIL you achieve flat response at the audience position you have no idea what you are actually doing. You can easily get something that sounds pleasant but it might (probably will) be full of peaks and dips in frequency response and it probably will sound better if you start with a flat frequency response in the first place before you start adjusting things.
Your chart is very clear. 50 db is the spl of a fridge at 1 meter or a quiet library. Or like I said, a whisper. Barely audible. And ONLY audible because it's in the frequency range that humans are very sensitive to. 50 db at 16 hz is useless, it's not audible, it's not something you can feel, it's basically not there for all intents and purposes. If you are subjected to 16 hz at 50 db for long periods of time (like hours or weeks) it can have an effect on some people but it's too low in level to do much of anything in the short term. Besides, if you don't measure you won't know if you even have 50 db at 16 hz.
What you are trying to do right now with your note level adjustments is get a flat frequency response by ear in a remote location (not near the audience position) using sampled pipe organ notes as a signal. IT ISN'T GOING TO HAPPEN. Again, you might get something that sounds pleasant but it's not going to be flat response and it could be improved greatly if you just do the measurements and start with flat response and adjust from there.
Even at high spl 16 hz isn't very audible, it's more of a sensation. There's no way you can level match a sensation of 16 hz to the audible tones of higher frequencies by ear. It's just not possible. Especially when the signal is a sample of unknown spectral content and you are making adjustments in a whole other location than the audience position.
The only way you are going to get anywhere near flat response at the audience position is measuring from the audience position. Measuring discrete tones or frequency sweeps.
What you are trying to do right now is very similar to trying to drive to a very specific destination like an appointment with a blindfold on. Sure, you know how to drive, you know where you want to go, you know the route you want to take. But if you can't actually see where you are going you are not ever going to reach your destination. You might get somewhere very pleasant but it won't be the specific destination you were targeting and you might waste a lot of resources in the process while completely missing the original target goal.
AFTER you achieve flat response at the audience position, THEN you can adjust things to your subjective personal taste.
UNTIL you achieve flat response at the audience position you have no idea what you are actually doing. You can easily get something that sounds pleasant but it might (probably will) be full of peaks and dips in frequency response and it probably will sound better if you start with a flat frequency response in the first place before you start adjusting things.
Your chart is very clear. 50 db is the spl of a fridge at 1 meter or a quiet library. Or like I said, a whisper. Barely audible. And ONLY audible because it's in the frequency range that humans are very sensitive to. 50 db at 16 hz is useless, it's not audible, it's not something you can feel, it's basically not there for all intents and purposes. If you are subjected to 16 hz at 50 db for long periods of time (like hours or weeks) it can have an effect on some people but it's too low in level to do much of anything in the short term. Besides, if you don't measure you won't know if you even have 50 db at 16 hz.
What you are trying to do right now with your note level adjustments is get a flat frequency response by ear in a remote location (not near the audience position) using sampled pipe organ notes as a signal. IT ISN'T GOING TO HAPPEN. Again, you might get something that sounds pleasant but it's not going to be flat response and it could be improved greatly if you just do the measurements and start with flat response and adjust from there.
Even at high spl 16 hz isn't very audible, it's more of a sensation. There's no way you can level match a sensation of 16 hz to the audible tones of higher frequencies by ear. It's just not possible. Especially when the signal is a sample of unknown spectral content and you are making adjustments in a whole other location than the audience position.
The only way you are going to get anywhere near flat response at the audience position is measuring from the audience position. Measuring discrete tones or frequency sweeps.
What you are trying to do right now is very similar to trying to drive to a very specific destination like an appointment with a blindfold on. Sure, you know how to drive, you know where you want to go, you know the route you want to take. But if you can't actually see where you are going you are not ever going to reach your destination. You might get somewhere very pleasant but it won't be the specific destination you were targeting and you might waste a lot of resources in the process while completely missing the original target goal.
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This will let OP know what a clean 16 hz tone sounds and feels like. But what else can you do with it?
Even if he uses other tones he can't level match and get flat response by ear.
Measurements at the listening position have to be done or the response will never be anywhere near flat.
The Inuke has plenty of dsp power to flatten things out but if you don't know what you are working with it's just shooting in the dark.
Anthony.
Don't miss the forest for the trees sitting right in front of you.
THis is a reproduction of a musical instrument.
It is voiced, or the relative qualities of individual pipe sounds and combinations of sounds are adjusted so that they work well in a combined setting.
It is not in any way shape or form a flat frequency response.
You yourself posted a hearing threshold curve.
Now use your powers of reason to grasp this little point.
The low end will be what sounds best.
Not flat as is proven by the hearing threshold.
I agree that it would be nice in a perfect world to see some frequency response graphs.
But ultimately what Ron is looking for he is going to get.
It is Ron's project. We are just aiding and abetting.
Don't miss the forest for the trees sitting right in front of you.
THis is a reproduction of a musical instrument.
It is voiced, or the relative qualities of individual pipe sounds and combinations of sounds are adjusted so that they work well in a combined setting.
It is not in any way shape or form a flat frequency response.
You yourself posted a hearing threshold curve.
Now use your powers of reason to grasp this little point.
The low end will be what sounds best.
Not flat as is proven by the hearing threshold.
I agree that it would be nice in a perfect world to see some frequency response graphs.
But ultimately what Ron is looking for he is going to get.
It is Ron's project. We are just aiding and abetting.
1. The Real Time information we got from BO isn't very useful, he says he played one note after another, and gave us the graph including all of them. This includes fundamental tones from all of the pipes, and their overtones.
2. I'm going to suggest this again. Turn all the amps except the sub amp off. Play some pedal, and see what you get from the sub without all the harmonics. That will enable you to get a feel for what the subs are really putting out, and then adjust the level based on that, plus decide what kind of eq you might want.
2. I'm going to suggest this again. Turn all the amps except the sub amp off. Play some pedal, and see what you get from the sub without all the harmonics. That will enable you to get a feel for what the subs are really putting out, and then adjust the level based on that, plus decide what kind of eq you might want.
Just another stab at it.
Hi Bach On,
Let me take a wack at the amplifier, etc. thingy too:
Post #789: "...I have the general impression that the circuitry of more powerful amps generally "negotiates" the power with the speaker..."
A power amplifier is a device with voltage oriented gain, e.g.: if the gain is set to x10, then for 1V input - a 10V output. Within the limits of the amplifers current output capability it does not care what the load impedance is, it's still 1V in and 10V out.
A vented (bass reflex) speaker is in general terms a difficult load as it has a strongly varying impedance particularly around its tuning (resonance) point, and as if that is not bad enough the phase of the acoustic output shifts considerably even if just looking at the driver without any eq or filters added.
So, in simple terms, the ouput power the amplifier tries to deliver changes with frequency (as the impedance changes with frequency). Some amplifiers are better at that than others.
The general rule of thumb for amplifer selection used to be 2x the speaker rated Prms (of Paes), so for your Dayton ST385-8 box, with 2ea. 8_Ohm/300W rated speakers connected in parallel for a 4_Ohm/600W load you should look for an amplifier with at least 1200W (1.2kW); for your SI 18Ht the data sheet just stated 600Wrms, so about the same power handling.
DSP functionality should enable you to arrive at a flat subwoofer system response above the the low cut corner frequency (or so). You want a flat response, as the signal that you will ultimately send to the speakers already has had all the processing it needs (or more); it has been recorded (sampled) in the real world, and then tuned to provide a suitable signal that after reproduction from you speakers should sound "correct" provided that the whole signal chain (including amplifers/speakers/chamber) produces a remotely flat response.
Post #794: "... I'll do the REW measurements. But I'm going to ultimately set all the voicing to match my personal tastes..."
I second that, get a good measured starting point, and then season to taste. 🙂
P.S.: The screen shots are from the Hornresp simulation of the corner box that I posted for JAG in Post #782. Bye the way, I do strongly believe, that JAG's "Large Voice Coil" method works, I just don't believe in the T/S parameters for the SI 18HT. 🙂
Regards,
Hi Bach On,
Let me take a wack at the amplifier, etc. thingy too:
Post #789: "...I have the general impression that the circuitry of more powerful amps generally "negotiates" the power with the speaker..."
A power amplifier is a device with voltage oriented gain, e.g.: if the gain is set to x10, then for 1V input - a 10V output. Within the limits of the amplifers current output capability it does not care what the load impedance is, it's still 1V in and 10V out.
A vented (bass reflex) speaker is in general terms a difficult load as it has a strongly varying impedance particularly around its tuning (resonance) point, and as if that is not bad enough the phase of the acoustic output shifts considerably even if just looking at the driver without any eq or filters added.
So, in simple terms, the ouput power the amplifier tries to deliver changes with frequency (as the impedance changes with frequency). Some amplifiers are better at that than others.
The general rule of thumb for amplifer selection used to be 2x the speaker rated Prms (of Paes), so for your Dayton ST385-8 box, with 2ea. 8_Ohm/300W rated speakers connected in parallel for a 4_Ohm/600W load you should look for an amplifier with at least 1200W (1.2kW); for your SI 18Ht the data sheet just stated 600Wrms, so about the same power handling.
DSP functionality should enable you to arrive at a flat subwoofer system response above the the low cut corner frequency (or so). You want a flat response, as the signal that you will ultimately send to the speakers already has had all the processing it needs (or more); it has been recorded (sampled) in the real world, and then tuned to provide a suitable signal that after reproduction from you speakers should sound "correct" provided that the whole signal chain (including amplifers/speakers/chamber) produces a remotely flat response.
Post #794: "... I'll do the REW measurements. But I'm going to ultimately set all the voicing to match my personal tastes..."
I second that, get a good measured starting point, and then season to taste. 🙂
P.S.: The screen shots are from the Hornresp simulation of the corner box that I posted for JAG in Post #782. Bye the way, I do strongly believe, that JAG's "Large Voice Coil" method works, I just don't believe in the T/S parameters for the SI 18HT. 🙂
Regards,
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I think you are missing more than the forest, you are missing whole point.
ANY sound reproduction system should have flat response at the listening position. This IS a reproduction system, it is reproducing samples. THIS SYSTEM SHOULD HAVE FLAT FREQUENCY RESPONSE AT THE LISTENING POSITION.
If we can't agree on that we shouldn't even be discussing this, as this point is the absolute basics, beginner level stuff.
The fletcher munson curves and the hearing threshold stuff was shown by me to indicate what is audible and what is not. If it's not at least 80 db at 16 hz then there's no point in even bothering to reproduce 16 hz at all. You could get by with a system tuned to 32 hz that's much smaller, much more sensitive, uses much less power, and would essentially give the same presentation.
The fletcher munson curves are NOT used to eq the system so that all notes sound equally loud, that would be so far from flat response that it would be unlistenable.
Maybe try using your own powers of reasoning to grasp this little point. This system should have a flat frequency response as it is an audio reproduction system. OP is trying to get a flat response by using Artisan note level adjustments as eq and pipe organ note samples as the signal, and doing it in a location nowhere near where the audience sits.
He has about as much chance of making an accurate music reproduction system with flat response in this manner as he has of winning the powerball.
Make no mistake, this is pro audio, this is science, not hippy dippy audiophile tweak until it sounds ok garbage. You can do it either way but using the science and measurements will give you a vastly superior result in very little time.
Read tb46's post. Notice how many times he says "flat response". He knows what he is talking about. This is a reproduction system, it needs to have flat response as a starting point before tweaking for personal preference. Otherwise it could end up (and probably will end up) so far off the mark that OP never should have bothered trying to produce 16 hz in the first place.
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