An Unconventional Loudspeaker system Latest change 2019-07-06
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
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
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
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 /
- 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
All parts except the cone and the tweeter box have a
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.
picture gives the current status, still in ground
Combined with video projection.
Of course the illumination on the loudspeakers is only for
this photographing. Normally it is'nt there.
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
- 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
- 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
- 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
- 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
- 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
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.
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
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)
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
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
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
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
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
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
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
Why remotely switching the power for the final
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
Yes, there will be large membrane excursions, but you will not
lose the pressure through the opening of a "vented" system.
Divine Laws is punished only if you are a believer.
Trespassing Government Laws is punished only if they catch
Trespassing Laws of Physics is punished automatically and