In consequence of my article about a "Time Correct Bass Reproducer" in ElektorLabs september/october 2019 the editors of Elektor asked me to write about the whole system. Well, here it is.
Please note that the information here is not intended to make an identical system yourself (you may do of coarse....)
It is more a bunch of ideas from which the reader may pick a few or many.
The theory behind the "Time correct Bass Reproducer" is not treated here, as it is in the Elektor Article under copyright.
I also took this opportunity to document the whole system in quite detail for my own purposes.
Most images on this site can be clicked for a larger version.
You may contact me by email: jan @ breem.nl, but please, first read the disclaimer.
You can download the BassCalc program as a .zip file with executables for Windows and Linux and the Source in FreePascal.
An unconventional loudspeaker system with as most important properties:
- 3-way, Bass, Mid, Tweeter. Each loudspeaker has its own power amplifier.
- Active filtering with compensation of loudspeaker / box properties.
- Reproduces frequencies well below 20 Hz.
- Concentric around a vertical axis, radiating over 360 degrees in the horizontal plane.
- Optimized time response.
- Subtractive crossover of second order for bass and tweeter, first order for mid range. Vector sum = 1.
- Bass and mid speakers are electro-dynamic, the tweeter is a monopole electrostatic unit.
- The relative volumes of the loudspeakers can be remotely adjusted by software.
- Class D power amplifiers located in a free space on the back of the bass unit. Distorsion spec <0.005% in range 1 .. 100 Watt
- The control amplifier from which the speakers are driven and the cabling support symmetrical transmission of the analogue audio signal, symmetric RS422 digital communication and a 5 Volt line to switch the power on / off.
An important source of inspiration was Hans van Maanen. Have a look at the "Diamond" system on his website: https://www.temporalcoherence.nl
The setup
All parts except the cone and the tweeter box have a pentagonal footprint.
The lower part has the bass speaker radiating upwards, above that the somewhat diamond shaped midrange compartment, also radiating upwards, and on top the tweeter radiating downwards. In between mid and tweeter sits a double cone approximately on ear-height when the listener is sitting.
The picture gives the current status, still in ground color.
Combined with video projection.
Of course the illumination on the loudspeakers is only for this photographing. Normally it is'nt there.
Important considerations:
Why 3-way?
Not any single loudspeaker is able to reproduce the full audio spectrum of over 3 decades with a good quality.
One could think of a full-range electrostatic speaker, ESL's are known for their low coloring in the mid range and their very transparent high. However they also have drawbacks. For the lower frequencies a large surface is required, and for real bass power a large excursion of the membrane. That is difficult. Additionally, they are almost exclusively constructed as dipoles, so reproducing very low frequencies is nearly impossible because of the shorting effect. For the highest frequencies the directivity is often a problem. In my concentric setup that is no problem at all.
With a two-way system the bass speaker must produce well into the mid range, and / or the tweeter must do well in the higher mid range. Such a combination is difficult, because the bass speaker must be quite large for enough power at extreme low frequencies, and for the tweeter I had already decided for an ESL because of the transparency in the higher frequencies.
So almost automatically I ended up with a 3-way system.
Why has each loudspeaker its own power amplifier?
This approach has so many advantages that it is astonishing that it is not done much more often, even in the very high priced segment where people easily pay thousands of euros/dollars/pounds for a speaker set or an amplifier.
- Every loudspeaker is damped well over the full frequency range by the low output impedance of the amplifier. In the conventional approach with passive filters the damping is impeded because the filter often represents a high-Q resonator and other reactive properties.
- With active filters one can realize much nicer filters with cheap components.
- Passive filters are often designed with the assumption that the loudspeaker impedance is nicely 8 Ohms or so. Far besides the truth; a 4 Ohm loudspeaker can easily reach 40 Ohm at resonance, and then your filter does something different.
- Equalizing the sensitivities of the loudspeakers can easily be done with a potmeter or so. In the passive situation one needs power resistors which have an effect on the filter characteristics, or certain combinations of loudspeakers do not fit because the sensitivities differ to much or the impedances do not match. 4-Ohm and 8-Ohm speakers often do not mix very well.
- The collection of power amplifiers is often placed nearby or inside the loudspeaker cabinet, resulting in very short cables.
- Distortion produced by one amp only appears in one speaker. E.g. high frequency distortion in the bass channel does not end up in the tweeter. Sure in the bass speaker, but that hopefully does not reproduce it very well.
- Per channel less power is required in comparison to a full-range amplifier.
- Experimenting with the filter properties is more attractive, because the components cost a few cents, while the passive components may cost tens of euros/dollars/pounds.
- In particular it is possible to make filters such that the summed output is perfectly flat in amplitude and phase response. Passive filters mostly produce phase errors resulting in inadequate addition of the sound outputs for certain frequencies, and degraded impulse response.
- Yes, one needs multiple output stages. That may look costly. But the actual component cost of an output stage is some 10's of euros/dollars/pounds.
Why to below 20 Hz?
I do have some recordings, CD as wel as vinyl containing such low frequencies. So I want to hear them. I also use this system combined with video projection, and quite some movies / TV programs have very low frequency audio content.
But more important, and this is valid for any system, if you want a very good impulse response you need a much larger frequency range than what is actually to be used. In the SACD system the frequency response goes far beyond the for humans audible frequencies, resulting in a better impulse response in the audible range. The same argument holds for the low frequency end of the spectrum.
Many loudspeakers make use of a property of our hearing system where, when only some harmonics of a basstone are presented, the bass tone itself is imagined in our brain. Hearing the real bass tones is a relief then.
Why radiating all around?
That is a matter of taste. The -not-so-strong- argument is that the same spectrum is radiated in all directions, as do many acoustical music instruments.
Another is the so called "sweet spot", the location where you have the best audio perception, is very large here. It does not matter much where you set yourself to listen, it's ok everywhere.
Sure there are people who prefer a very direct sound and accept a smaller sweet spot. They also could have profit from some of the reasonings here.
Why concentric?
Well, then radiating around works better. More important is that the times of travel from each loudspeaker to the listener can be made equal, especially for mid and tweeter, by adjusting the height of the cones. With conventional setups the distances are often different for different listening positions, especially in height.
Why optimizing the time response?
The experience of people who did this indicates that every improvement of the time response leads to better audible detail.
Why a crossover of 2nd order?
One would like to use higher order filters to prevent signals to reach speakers for which they are not meant.
However, with filters of order over 2 the time response cannot be made perfect anymore. (I have this thesis from a math expert but without prove)
Why Butterworth?
I think it gives a good compromise between on one side to much overshoot and ripple in the pass band like Chebychev and on the other side not such a smooth transition as with Bessel and Gauss.
Why has the mid channel a 1st order response?
That is a consequence of the subtractive filter. If you do Bass an Tweeter with second order filters and the mid range is derived by subtraction, you will end up with a 1st order characteristic. I do not know a proof of this but the simulations show it without doubt.
Why not for the mid range also an electrostatic speaker?
This has been considered. But because of the 1st order response the mid range speaker has to do quite some low frequencies, and that is not possible with an ESL of this size.
Why a mono pole ESL as tweeter?
I want it to only radiate downwards, and not upwards, that does not combine with the principle of radiating around with the double cone.
So it sits in a closed box with damping material. There is a risk that the membrane will resonate with strong basses and produce sparks. (as per January 2018 no problem found)
Secondly, ESL's are dust magnets and I don't want to worsen that by letting rubbish fall on them.
And remember, any loudspeaker (except maybe the plasma speaker) is basically a dipole. We put them in cabinets to make monopoles.
Why remotely adjusting the output levels of the loudspeaker by software?
I do not want to do that with potmeters which might be differently adjusted left/right. I found an outstanding chip for that in the PGA2311 from Texas Instruments and that requires a microprocessor for the control. Besides that a micro is also advisable for the class D power amplifiers.
Why class D power stages?
Well, the argument is not that strong, other than I'd like to try that too. From a viewpoint of audio quality there is no preference for class D or the usual class AB, both can result in an amplifier with little distortion and sufficient power.
THE big advantage of class D is it's very high efficiency so everything can be quite small, no large cooling bodies etc. But for hifi in the home this rarely a problem, no need for extremely small.
Jan 2018; The class D amplifiers are operational now. The listening experience indicates less distortion than before with the STK086 modules, especially heard with choral music.
Why symmetric transmission?
Symmetric transmission results in a much better suppression of disturbing signals like hum from mains power or ground loops.
Remind that the control amplifier and the loudspeaker-amplifiers are likely to be powered from different outlets and that there may be long cables in between.
Why remotely switching the power for the final amplifiers?
I do not want them to be continuously switched on and I also do not want to go there to switch them on every time, both is waste of energy.
Power switching based on a minimal audio signal is no option too, something must be continuously powered there, and I have bad experience with the sets of Philips Motional Feedback units I have in use, they switch off when you are playing a bit softly. (I bridged the relays in these units)
How do you reproduce frequencies well below 20 Hz?
That can only be done with a completely closed box. Any other housing would be much to large.
Partialy "open" cabinets like Bass Reflex or other "vented" housings fall with 24dB/octave below the resonance frequency, a 4th order high pass filter. That cannot be eliminated with a forward correction, all alone because of the phase shift.
The behavior of a closed box can be calculated quite easily, it has a second order slope which can be corrected with a forward filter.
Yes, there will be large membrane excursions, but you will not lose the pressure through the opening of a "vented" system.
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