#2 EXPANSION OF DYNAMIC RANGE
WHAT IS DYNAMIC RANGE?
Dynamic Range, DR, for a stereo, is the difference between the loudest and smallest sound that the
system can reproduce. For human hearing, DR is about 120dB, with OdB being the smallest audible
sound and 120dB being very loud. The scale is logarithmic, each 10dB = ten times, so 120dB =
100,000,000,000 times; which presents immense engineering challenges.
Early in High Fidelity, the Long Playing record achieved a DR of 75dB, on a good day. The signal
source was electromechanical, with stylus deflection being the analog of amplitude, which set the
upper limit. So a chain of components ending in a pair of speakers that could manage 75dB could
(possibly) reproduce the DR on the LP. As an industry, we did a good job at this.
When the CD came along, the upper limit for DR was bumped to 103dB, that limit being set by the
16 bit word standard. Wow! From 75 yo 103dB! That was huge! Nearly 30dB, so nearly 1000 times.
That upset the applecart. I was there, building speakers. It was a stretch. Some guys did OK, many
did not; their product was noticeably lifeless by comparison. The point then was: conventional tech
barely made the cut, with some effort.
Today, HD’s 24 bit enables DR that surpasses that of human hearing, meaning that it is finally
possible for recorded music to have ideal DR. The recording guys can capture it, but can we
playback guys transduce it? Can we possibly wring another 15dB out of conventional tech? Nope.
Ain’t gonna happen. Can’t.
The proof is out in the open. When you hear guys saying that they don’t hear much difference in the
dynamics between CD and HD, they’re right! But not for the reason they think. The bottleneck is not
on the recording side, it’s on the playback side. Their speakers are not dynamically capable AND
their passive filters are obliterating the improvement in resolution.
Let’s take a look inside DR reproduction in the speaker. A speaker’s job it is to transduce an
electronic time/voltage signal into time/sound pressure in your room. Perfect Dynamic Linearity
means that: for each doubling of the input, you’ll see a doubling of the output, across the entire
Here’s a 3D graph
Linearity in a
speaker would look
But what actually
than ideal. Instead
of a 1: 1
between input and
output, you get
typically not evenly
the audio band. In
most cases the
woofer gives up
first; it can’t keep
up with the rising
outputs of the
surprise, really, its
job is much harder.
This is the reason
why so many
when you crank
them: the tonal
balance is shifting
upwards. I’m sure
you’ve noticed that
presented to you in
the specs and
taken at modest
it’s very difficult to
measure, in room, at high pressures. But you should know what’s going on; you should not expect
that a “pretty” freq/amplitude curve will still be so at high amplitudes.
WHAT’S CAUSING IMPERFECT DR AT THE WOOFER?
#1 - The motor itself is nonlinear.
Motor force at the cone falls off on
both sides of center position. This
happens for two reasons: as the
voice coil moves fore and aft, more
and more of it moves outside the
magnetic gap, so the force
generated falls off. And a less-than-
perfectly elastic suspension
‘tightens’ as the moving system
excurds, mechanically limiting it.
This problem is even worse than it
looks because the two performance
functions that we want most are
governed by adverse square
functions. Bear with me, please.
First, because your perceptions are logarithmic, each time you want the system to sound twice as
loud, you actually need it to play four times as loud, so four times the excursion.
Plus, each time you ask for lower bass, let’s say you’d like to go from 40Hz to 20Hz, halving the
frequency, you’ll need four times the excursion.
So, as you ask your system to play some combination of bass heavy music and nice amplitude, you
are actually demanding huge amounts of excursion. That’s why your woofers bottom out. The
critical point, here, is that the greater the percentage of time the coil is in the declining force zones,
the less faithful it is to the signal’s DR across the entire range of the woofer. That’s why you hear
‘congestion’ on difficult music, which is not as evident on simpler tunes.
What you’d like to see, and happily
pay for, is motor behavior that looks
more like the second plot. A few of
today’s best makers of cutting edge
woofers are taking special care on
this aspect, knowing that customers,
like me, are discerning.
(Has anyone ever told you this
before? No? Because they don’t
know it, themselves.)
#2 - Thermal compression drags
down output, an effect that,
unfortunately, sums with
As you demand more from a woofer,
in addition to causing more excursion, you are also pushing more current thru the voice coil. So it
gets hot. Law of nature: heat generated is a square function of of current, so the effect come on
quickly. Problem? Yes. Resistance rises linearly with temperature. As the coil warms, the resistance
rises. Remember the woofer in series with the resistor (in section one)? Lower output, higher Q,
loss of amp damping. But it’s worse: like the coils in your toaster (which are supposed to get hot)
the rise time and cool-off times are slow. So when you punch a bunch of current thru your coil, the
hotter it gets, the more the output falls. It takes a few milliseconds to hurt you, then the effects
linger. As distortions go, this one is an real mess.
What can we do about it? Two things are obvious: keep the coils cool and to spread the heat out
among multiple woofers, thus multiple coils.
Examine these two Satori woofers, a 9.5” and a 7.5”. You can see the heroic means of keeping the
voice coil temperatures down. First, oversized coils spread the heat out and increase the radiating
area. All things being equal, a larger coil would also spread out the magnetic flux that motivates it.
So, by increasing the coil size, we do need to ramp up the magnetic circuit, and exponentially. So it
gets expensive. Then, generous venting thru the pole piece, thru the center of the coil with both
ends flared for smoother and quieter flow, a symmetrical spider that fully exposes the forward end
of the coil and a perforated coil former form a system that effectively pumps cool air across the
coil. Implicit in this thermal design is an extremely coercive neodymium boron drive system that
focuses an intense magnetic field across the voice coil, raising sensitivity, so requiring less current.
As you will see, one of the advantages that Next Gen tech brings is the ability to put more drivers
into less space. So we can go ahead and pack baffle with woofers, which both beneficial to the
system’s output/bandwidth envelope because they can do more work, but also to DR.
Completing the strategy for keeping both temps and excursion low is a DSP “trick” that results in
less current at the coil and less demand for excursion. Every bass system has a natural low
frequency roll-off; a region where it can make little useful output.
For all of these years, in conventional systems, we been sending full range signal into that region
(shown in red), both heating it and forcing useless excursion.
We can’t do this with passive filters, but the idea is simple, in the processor we filter out the
unwanted energy but just fitting a high pass filter to the acoustic output; the blue line shows the
filter. This approach cleans up the working part of the band and improves DR greatly because it
both helps keep the coil cool by reducing total current flow, plus eliminates unproductive excursion.
This is a rarity in engineering: all upside, no downside. A free lunch!
To recap: multiple advanced woofers designed for motor force linearity and coil cooling, plus a high
pass filter. It sounds simple, now that we understand it, no?