Beyond the Ariel

I totally agree with Lynn, that 15 Watt is high power region ;)

A good loudspeaker only needs a few hundred mW (MilliWatt) for satisfactory listening levels.

For the average (RMS) level, maybe.
But each +10dB dynamic swing requires 10x the power, and good recordings have 20+ dB of crest factor.

So even starting with "a few hundred mW" = say, 0.5W, you quickly end up with a peak demand for 50+ Watts. Then, you either have those clean Watts, or you run into amplifier clipping = heavy, nasty compression. Even though the "soft" clipping of a DHT amp may not sound particularly objectionable, it still is significant distortion of the original signal, i.e., not "Hi-Fi" anymore.

M.
 
Even though the "soft" clipping of a DHT amp may not sound particularly objectionable, it still is significant distortion of the original signal, i.e., not "Hi-Fi" anymore.

M.

In elaborate tests done decades ago I showed that "soft clipping" was barely audible while hard clipping was seriously offensive. Both sources had the exact same max-voltage (rails). Everyone thought that the soft clipped amp sounded louder and better. But, of course, if the signal never went to the rails then both amps sounded the same.
 
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I use Matlab, but the same thing as excel other than that it is doing it real time and making decisions to continue or not based on the measured performance. The ability to script your own tests is very nice. For this I just measure response at a low level, measure response at a higher level, calculate the difference between the curves, and subtract out the amount I increased the output by between tests. If there is anything left (non zero), that is compression or expansion. I set up that test to increase by 3dB each time.

Some years back I tried a similar test but using short term pulses of ever-increasing magnitude. I was unable to detect any changes. Hence, the question becomes how much time does the signal have to spend at the higher level before this compression takes hold. That, to me, becomes the key factor as music is seldom a constant high level.
 
During the built-up of my Decorator LS kit I decided after the very first measurements to completely redesign my own cross-over using the lower woofer to yield a 2.5 or more or a 2+ way loudspeaker.

The response of the lower woofer is designed to fill up what the upper woofer cannot deliver in the LF region due to baffle step and slightly rising response to higher frequencies. The lower woofer is running with considerable less field coil current than the original suggestion and the field coil currents are part of the design parameters.

It is very easy with a field coil loudspeaker to assess what lack of substance you have if you shut-down the field coil supply of the lower woofers ... so no more doubts on my side.

At the beginning of my thoughts on letting the lower woofer run, of course properly attenuated, to much higher frequencies than the original, I was in doubt if it could disturb the sound or stereo image. But to my surprise, it does not.

Field coils have an interesting effect. if you shut down the current there is an rest magnetization and you still have some sound, maybe 20 or more dB lower than before. You can then shut-on the current for lower woofer and shut-off the current for the middle woofer and the compression driver. In this way everything still sounds right with respect to stereo image or the sound stage. A really interesting effect.

Respect! An all-field coil loudspeaker system!!! That goes a lot further than anything I've been discussing.

Field coils have the interesting ability to change BL product on the fly ... of course, that changes not just efficiency but Qt as well. The mechanical system (Qms, HF resonances, etc.) stays exactly the same, but the electrical system is profoundly altered.

So if I'm reading your letter correctly, the residual overlap between the lower and upper woofer had no harmful consequences, and you've been using the lower woofer to shape the overall system response to correct for BSC and assorted other LF errors that are inherent to a single-point woofer in a cabinet. It does confirm my intuition that the response and level adjustments for the lower woofer need to be rather fine-grained, so the overall impression at the listener's position is subjectively flat (with most recordings).

P.S. I am puzzled why field coil loudspeakers are priced as high as they are. They should be much cheaper to make than Alnico or Neodymium magnets (which are brittle and difficult to machine), and not much more than molded ceramic magnets. True, they are aimed at a very narrow specialist market these days, but all loudspeakers used field coils back in the Twenties and Thirties, and they weren't anything special back then. (Yes, that includes the famed Lansing Iconic, which set the pattern for all 15" paper-cone woofer + MF/HF horn + 700~800 Hz crossover studio-monitor systems that followed.)
 
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... as music is seldom a constant high level.
At least the music I listen to isn't. :)

But that got me thinking; "How often does music hit the peaks?" I looked a a few dynamic recordings, 1812 Overture, Jupiter from the Planets and some others.
Answer: It depends - a lot. It depends on what is defined as a peak and it depends on the music and recording. What I found was that during loud, dynamic passages the signal is peaking 8 to 20 times per second. Those peaks are brief, between 4 and 12 samples each.

It really depends on where you set the threshold for the peak.
 
Respect! An all-field coil loudspeaker system!!! That goes a lot further than anything I've been discussing.

So if I'm reading your letter correctly, ...

Lynn, as you showed deeper interest here are some more infos...

The woofers were already entirely tested and documented in german Hobby HiFi 4/2015. I will show the parameters only for two field coil currents here and additionally my measuremnts for frequency response at these (1.5A and 3.0A):


[A] --Qts -Rms --dB ---BL
1,5 0,499 1,59 83,5 13,68
3,0 0,164 1,48 88,5 26,22


Here are my own measurements in the open baffle (job_for_lower_woofer.gif).

The red line is the resonse for 3.0A and the black line at 1.5A up to 700Hz. Normally, one would say, hey the woofer has a more flat response for 1.5A and has a slightly better extension to LF. But in this way you give away 4-5 dB effiency in the upper region. Ok, let's go to 3.0A with slightly rising response from about 80 Hz on. I hope you clearly see that the baffle step (baffle is about 1m width) coming from 20 Hz with about 6bB per octave is turning around 70-80 Hz in a slightly rising response. And that's where the blue curve is coming into the game. It is the cross-over function for the lower woofer. A big sereis coil is doing the job to bring the woofer down but a parallel arrangement of an resitor in series with a much smaller coil resembles the curve to fill up to a nearly flat response. After that all, a parallel capacitor is doing it's job to resemble the roll-off of the middle woofer. I thought that is is a good idea that the the lower woofer shows the same acoustic roll-off where the compression driver comes into the game as the middle woofer but at much lower in level. In this way it works for me. Maybe because both woofers have comparable properties even with different field coil current. And the best thing is that you can lift up or down the woofers using the field coil supply to flat response.

Just to mention that it was a hard job to measure the responses as comb filter effects with the floor are horrible. I used all damping material on the floor that could be found. But comb filter is a good point. At the listening position every speaker shows these effects most often with the floor. The lower woofer is at another height with respect to this and should make these comb filter effect smaller. Where the middle woofer shows a comb filter to the listeners ears the lower does not and vice versa.
 

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Got you Pano. Whether that was an improvement over the original multi-cell horn is the question or just different. That horn is just to small to cover the top end of a 15" cone and have a chance.

I was thinking of the standalone big Manta's that is what I was around when I was working on horns, Manta's, Bi-radials, EV's CD horns, Community, EAW, saw it all.
 
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At least the music I listen to isn't. :)

But that got me thinking; "How often does music hit the peaks?" I looked a a few dynamic recordings, 1812 Overture, Jupiter from the Planets and some others.
Answer: It depends - a lot. It depends on what is defined as a peak and it depends on the music and recording. What I found was that during loud, dynamic passages the signal is peaking 8 to 20 times per second. Those peaks are brief, between 4 and 12 samples each.

It really depends on where you set the threshold for the peak.

Just curious, is that 1812 overture from Telarc?
LP or CD?
There is a big difference.
 
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For this I just measure response at a low level, measure response at a higher level, calculate the difference between the curves, and subtract out the amount I increased the output by between tests. If there is anything left (non zero), that is compression or expansion.
Thanks John, that makes sense and is what I would expect. I was just hoping that maybe it was a feature I had missed in some off the shelf software.

I liked your measurements enough to take a serious look at them, and try to get an idea of what they might tell us. Earl Geddes is correct, music isn't loud all the time, so pink noise may not be the best way to test this, it being a rather continuous level. Noise like that also is not a good match for the spectral content of music.

Pink and Brownian noise contain much more energy below 100Hz than most music does. I've run spectral analysis on many recording and genres before, and did again. A good approximation of big classical recording would be Brownian noise high passed filtered at 52 Hz, 2nd order. Some like Pink Floyd also fit that shape, with other rock and Pop - ACDC, Metallica, Black Eyed Peas, Green Day, the Eagles and others - rolling off faster under ~52Hz, perhaps 4th order. Testing with the heavy bass content of pink or brown noise may be showing more compression in your 15" sub than you are actually getting.

Looking at your graphs shows that with pink noise your sub starts compressing 10dB before your 15" mid bass. That made me wonder how much your subs would be into compression for various types of music, if your mid-bass is just at the onset of compression. Looking at various recordings, I'd say that the sub is into your compression zone about 2% of the time when the mid-bass is just touching that region. But on Pink noise? The sub would be into the compression zone 25% of the time. A major difference.

I made some graphics to show the compression differences of the sub in music vs pink noise. I can post them if anyone is interested.
 
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