Professional Documents
Culture Documents
PART 1
This is the first section of an article published in the Vol. 3, Issue 6 of
Positive Feedback magazine. It is my *personal* view of the art of
designing high-end loudspeakers, and is not intended as any kind of "bible"
for folks new to high-end audio. Since I've gotten a lot of questions about
the kind of thing that appears in Positive Feedback, it's probably simplest
to demonstrate by reposting a complete article in this forum.
------------------snip-----------------
by Lynn Olson
* Introduction
These ancient questions of philosophy are the first questions you must ask
yourself if you are serious about designing audio equipment. These
questions repeat themselves in only slightly altered form:
* The Future
If you relax and make a mental journey to the far future, it is easy to
imagine the perfect loudspeaker. It would made of an immense number of tiny
point sources that would create a true acoustic wavefront (or soundfield).
Resonances due to massive drivers, cabinets, or frames would be a thing of
the distant past. A myriad of waveform distortions (harmonic,
intermodulation, crossmodulation, frequency, phase, and group delay) would
be utterly absent ... the sound would be literally as clear as air itself.
This perfect loudspeaker would be made of trillions of microscopic coherent
light and sound transducers, integrated with signal processing circuits all
operating in parallel. (This is similar to present-day military
phased-array radars, with tens of thousands of tiny antennas with
integrated electronics subsystems.) It would be constructed by a
combination of nanotechnology and genetic engineering and operate at the
molecular level, appearing simply as a very thin film when not in
operation.
Just for a moment, imagine a thin-film mirror for 3-dimensional sound and
images with a near-infinite random-access memory that is also connected to
all other mirrors; it would be "transparent" in a lot more ways than one.
* The Present
Whew! Now that we've glimpsed perfection it shows just how far we have to
go in 1994. Here's a partial list of the major problems we face now:
Even a simple central mono image has been shown to suffer from deep
comb-filter cancellation nulls between between 1kHz and 4 kHz, which is why
a solo vocalist sounds different coming from a single mono speaker and a
conventional stereo pair. It now appears that 2-channel stereo requires a
minimum of 3 speakers to faithfully represent the tonal quality of
centrally located sound sources, such as vocalists.
* The Past
Modern speakers are far better than the speakers of the Fifties (which I
still remember). Very few people had full-blown Altec "Voice of the
Theatre" A-5 systems, 3-way Bozak B-305's, or Klipschorns. Most "hi-fi
nuts" had to put up with University, Jensen, or Electro-Voice 12" coaxial
drivers in big plywood boxes with a single layer of fiberglass on the rear
wall. A large cutout served as the vent, resulting in boomy, resonant boxes
tuned much too high, with 6 to 12 dB peaks in the 80 to 150 Hz region.
(Have you ever heard a restored jukebox?)
The coax, or worse, triax drivers went into paper cone breakup at 200 Hz
and above, cavity resonances (due to the horn element mounted in the cone
driver) at 800 Hz and above, horn breakup throughout the working range of
the short horn, and phenolic diaphragm breakup at 8 kHz and above. A "good"
driver of this type usually had a plus/minus tolerance of 4 to 8 dB, and
it took a lot of judicious pen damping to get it to measure that well.
It does make you wonder about the resale value of the current crop of
multi-thousand dollar transistor amps and D/A converters - what will they
sound like on speakers ten years hence? By then we'll be listening to
speakers with evaporated diamond diaphragms and other materials with very
low distortion and stored energy, a long-overdue high-fidelity digital
storage medium with 24-bit depth and 96 kHz sampling, and the possibility
of replacing conventional 2-speaker stereo with something more natural. You
can be confident the sound will be far more transparent and realistic than
what we're accustomed to now.
What kind of sound do you like? People really do hear in quite different
ways, and different people assign importance to different qualities of
sound. Some audiophiles value timbral (or "tonal") qualities above all
else, treasuring the sound of their favorite instruments or voices; some
like a sense of immediacy, directness, and emotional impact; some like the
sensation of an immense 3D space; and others like a see-through
transparency, a palpable "you are there" quality.
It helps a lot if you know what's important to YOU, what you tend to
dismiss, and what you don't hear at all. We all tend to unconsciously
define the bounds of "reality" by our own personal perceptions and
thoughts, but this just isn't so. Reality is far, far larger than any
personal set of boundaries. That's why getting a second and third opinion
is so important.
Since all speakers have serious flaws in the absolute sense, it's up to you
to select the qualities that are most important, and most believable, to
you personally. "Perfect Sound Forever" is an arrogant marketing slogan,
not a realistic goal for an artist. For one thing, the materials to build
anything of the sort just don't exist. (Unless you've found a way to
generate a controllable room-temperature plasma. If you have, you'd better
talk to the Department of Energy first.)
In the section that follows, I'll describe the various paths that designers
must choose as they make their way to sonic perfection.
PART 2
This is a continuation of an article published in the Vol. 3, Issue 6 of
Positive Feedback magazine. It is my *personal* view of the art of
designing high-end loudspeakers, and is not intended as any kind of "bible"
for folks new to high-end audio. Since I've gotten a lot of questions about
the kind of thing that appears in Positive Feedback, it's probably simplest
to demonstrate by reposting a complete article in this forum.
------------------snip-----------------
Duntech, Thiele, Spica, and Vandersteen systems fall in this group. The
designer takes expensive steps to control diffraction, offset the drivers
for a coherent arrival pattern, and usually employs a first-order (6
dB/Oct) crossover. Some, such as Spica, may use 3rd (18 dB/Oct) or 4th
order (24 dB/Oct) Gaussian or Bessel crossovers.
Most British and Canadian speakers fall in this group. They have very flat
spectral responses, with the British paying more importance to the 1 or 2
meter on-axis response curve, and the NRC-influenced Canadians paying more
importance to the frequency response averaged over a forward-facing
hemisphere. These design priorities have been arrived at by BBC broadcast
professionals and NRC listening panels respectively, using statistically
verifiable double-blind listening tests.
To its great credit, the BBC is known as the first group to accurately
measure and identify sources of driver and cabinet resonances in the early
Sixties, and many British speakers still excel in this area. Since audible
resonances may be as far as 20 dB below a conventional sine-wave response
curve, the BBC was the first organization to identify and measure
colorations that were completely missed by the conventional sine-wave or
3rd-octive pink-noise measurements. These types of measurements are now
considered a standard part of FFT, TDS, or MLSSA systems.
Horns typically have extremely low THD, IM, and FM distortion, reasonably
flat response, and sharp cutoff characteristics at both ends of the
frequency range, which may be fairly narrow. Historically, horns were
usually beset by intractable problems with impulse response, diffraction,
and smooth dispersion, which is why most high-end designers (in the West)
avoided horn systems, leaving them to the studio monitor and PA markets.
In the past decade, though, Dr. Bruce Edgar in the US, and others in Japan,
have made very significant improvements in horn design for high-end and
ultra-fi audio, which are just beginning to be recognized by magazines like
Sound Practices. According to the movers and shakers in the American
ultra-fi renaissance of the 90's(Joe Roberts, Herb Reichert, and Mike
LeFevre), these new horns, and the Edgarhorn in particular, are in a class
of their own, superior to ribbons, electrostats, planars, exotic dynamics,
etc.
My own opinion? Well, I feel that the trio I mentioned above have pretty
good taste, so it's entirely possible I'll also like the new horns. Right
now, I can't say anything until my own set of Edgar midrange horns arrive
and I design them into a full-range speaker system. Stay tuned.
* Electrostatic Planars
A few English, American, and Japanese companies make this class of speaker,
which I have to admit are old-time favorites of mine. A well-designed
electrostatic offers the most linear and completely uniform diaphragm
motion of any class of loudspeaker (and low IM distortion), as well as the
potential for the best pulse response. The first Quad electrostatic is the
most famous example of a speaker decades ahead of its time.
The old Quad pioneered the most widely used solution, a side-by-side 3-way
system, using progressively narrower panels for the higher frequencies. The
new Quad uses a complex phased array system which approximates a spherical
radiator. The Martin-Logan uses an unusual cylindrical panel relying on
curved damping pads stretched across the perforated high-voltage stators.
Problems still remain, though. All of the electrostats I have measured show
moderate resonances below 200 Hz (primary room-diaphragm resonance) and
multiple sharp resonances above 8 kHz (non-homogenous diaphragm motion
and
standing waves in the HV stators or metal grill-frame assembly).
Magnetic-planars use arrays of magnets on the back side of the film (not
good for distortion) or on both sides (operating in push-pull, but also
creating a cavity between front and rear magnet pairs). The arrays of
magnets provide a somewhat uneven drive field, so the uniformity of
diaphragm motion is not in the same class as an electrostat. Then again, HV
arcing is not problem, so the magnetic-planars can play a lot louder than
their electrostatic cousins.
The magnetic-planars have weak magnetic coupling, which is much lower than
a dynamic driver due to the large magnet spacing and shorter length of wire
in the gap. In a dynamic driver, a high "BL-product" means a strong
magnetic field in the gap (the "B") and a long helical voice coil immersed
in the field (the "L"). Dynamics with a high BL-product provide the
tightest amplifier-speaker coupling, which is why they are sensitive to
amplifier damping factor and wire resistance. By contrast, in a
planar-magnetic, damping is mostly provided by the stretched film and the
air load, and little by the amplifier. Both impedance and efficiency are
lower than dynamics, and attempts to raise both by increasing the amount of
aluminum wire on the film usually degrades the transient response.
The true ribbon is free of the stretched film resonances and obstructing
magnets of the planar-magnetic, so it offers outstanding pulse response,
uniform drive, and a good approximation of a line source, but the
efficiency and impedance are both phenomenally low and it is not usable as
a woofer due to the small area. Most practical ribbons either use a
stepdown transformer or ask the amplifier to drive a 1/2 ohm load (not a
joke, unfortunately).
These kinds of speakers aren't my cup of tea, but I know many people who
really enjoy the neutral, relaxed type of sound they can offer. In
addition, a true ribbon offers some of the best treble around, superior to
dynamics or electrostats, exceeded only by the "massless" exotics.
* Hybrids
Each type has a quite distinctive sound, and some, naturally, are hybrids.
This gets pretty tricky when the dispersion patterns are different, along
with different IM distortion spectra. It helps if the original designer is
the one making the decisions, because a consistent philosophy is then being
used for the entire system (we hope). Even so, if the designer is
unfamiliar with the strengths and weaknesses of each type, the result can
be the typical disjointed "hybrid sound", with the crossover region quite
obvious.
* "Massless" Exotics
One day, I'd like to design one of these myself. The "massless" speakers
fall into this category ... Ionovac, Magnat, and Plasmatronics (what a
name!) They DO sound exotic, and measure the same. No resonances at all,
and accurate pulse and frequency response up to 100kHz or more. Low
distortion too ... like an amplifier. Actually, the "diaphragms" do have
mass. But it's not much. It's the same as air, so the acoustic coupling is
1:1. Efficiency is a little difficult to state, though, since the output
tube plates of the power amplifier are providing a high voltage that
directly modulates a conductive gas.
I remember hearing the Plasmatronics at the 1979 Winter CES, and I must say
I've never heard a tweeter that even came close to that one. They darkened
the room, and you could see this weird purple glow through the grill cloth
that looked for all the world like a gassy triode ... but it was the
tweeter! It glowed and pulsed with the music!
The rest of the speaker, though, was a pretty mundane paper-cone setup in a
huge cabinet ... oh well. Even so, the Plasmatronics was a wild thing, a
glimpse of the future, a SR-71 Blackbird next to a bunch of Cessna 172's.
Not too surprisingly, the designer was a plasma physicist at Los Alamos
Labs. Talk about being ahead of your time! This was 12 years before anyone
was talking about turning swords to plowshares, or in this case, Dr.
Teller's atomic toy into the next breakthrough in audio!
The real-world problems? Well, if you ionize air (by using RF heating) you
strip apart oxygen-2 and get oxygen-3 (ozone gas). Not very healthy, and
illegal in the USA. The Plasmatronics ionized helium gas, which got around
that problem, but it required a fresh tank of helium every month (I'm
serious!). I remember seeing a full-size helium tank, gauges and all, in a
special compartment inside the subwoofer enclosure.
* Summary
In Part 2, I'll discuss the different qualities of dynamic drivers, the way
they affect the sound of the total loudspeaker, and the effect they have on
the sound quality of the listener's high-fidelity system.
PART 3
The second in a series called "The Soul of Sound" where I wear my professor
hat. These articles are my *personal* view of designing high-end speakers.
I expect this article to be thoroughly out of date in two years or less ...
so enjoy it while it's fresh!
-------------------snip-------------------------
* Drivers
We have a long road ahead of us ... but take heart! Major advances in
materials sciences are happening right now, with many improvements due to
occur in this decade. I confidently expect we'll be seeing breakthroughs
every 2 or 3 years at the present rate of progress.
* Uniform Motion
Rigidity means accelerations from the voice coil are accurately translated
into cone or dome acceleration over the entire driver surface; this
translates to ruler-flat frequency response, fast pulse risetime, low IM
distortion and a transparent, "see-through" quality to the sound.
As usual, both sides are right, and both sides are wrong. They're just
speaking about different things. What the audiophile is actually hearing is
uniform cone motion; this phenomenon can be measured by the absence of IM
distortion, a flat frequency response in the working range, and good pulse
response with a clean and quick decay signature.
Well, that's great, you might think, just make the cone, or dome, or
whatever as rigid as possible. How about a metal, like bronze, perhaps?
That's nice and strong, and it can be formed into nearly any shape.
You can see the direction this is taking. Bells are made out of bronze.
Another problem raises its head .... resonance! After all, why does a bell,
or any other rigid metal, ring so long, for many thousands of cycles?
The answer has two parts, one obvious, one not so obvious. First, the metal
is rigid, and formed in a shape that increases the rigidity even further.
Second, the only path for the bell to release mechanical energy is to the
air itself, which takes a long, long time, since the density of air and
bronze are quite different, resulting in very weak coupling, and very
little damping by the air load. This leads us to another desirable property
for the speaker driver, which is ...
* Self-Damping
We also want the voice coil to stop the cone or dome, not have the cone or
dome play a tune all by themselves. Unfortunately, the most rigid materials
(traditionally metals) have very little self-damping, resulting in
vibrations of very long duration (high Q). One way to control the problem
is to extend a heavy rubber surround partway down the cone, and pay a lot
of attention to the damping behaviour of the spider and surround materials.
At the present, though, even the best Kevlar, carbon-fiber, or aluminum
designs show at least one high-Q peak at the top of the working range,
requiring a sharp crossover, a notch filter, or preferably both to control
the peak. Unfortunately, this peak usually falls in a region between 3 and
5 kHz, right where the ear is most sensitive to resonant coloration.
This is a problem, by the way, that plagues all current 2-way Kevlar,
metal, or carbon-fiber loudspeakers ... at the current state of the art,
the 6.5" or 7" drivers are forced to operate right up to the edge of their
working ranges in order to meet the tweeter in a moderate-distortion
frequency range.
This presents the designer with a tough choice: rough sound in the entire
treble region, or the characteristic Kevlar forwardness, which can at times
actually give a snarly sound to the speaker system. At the present, the
best choice is a fourth-order (24dB/Oct.) crossover with a sharp notch
tuned to the Kevlar resonance.
I should add, by the way, that I like Kevlar and carbon-fiber drivers very
much ... but they are difficult drivers to work with, with strong resonant
signatures that must be controlled acoustically and electrically.
As mentioned above, rigid cones have advantages, but are difficult to damp
completely. A different approach is to use a cone material that is made
from a highly lossy material (traditionally, this was plastic-doped paper,
but this has been supplanted by polypropylene in most modern loudspeakers).
The cone then damps itself, progressively losing energy as the impulse from
the voice coil spreads outwards across the cone surface. The choice of
spider and surround are then much less critical.
This type of material typically measures quite flat and also allows a
simple 6dB/Octave crossover; personally, though, I don't care for the sound
of most polypropylene drivers, finding them rather vague and
blurry-sounding at low-to-medium listening levels. Without access to a B&K
swept IM distortion analyzer, I have to resort to guesswork, but I strongly
suspect that this type of cone has fairly high IM distortion since it is
quite soft. In addition, it is quite difficult to make a material that has
perfectly linear mechanical attenuation; in practice, distortion creeps in
when you actually want a progressive attenuation of energy over the surface
of the cone.
* Cavity Resonances
Even though the dust cap in a mid/woofer (or the dome in a tweeter) looks
pretty harmless, the space between dustcap and the polepiece of the magnet
creates a small resonant cavity. One example of this was the (in)famous KEF
B110 Bextrene midbass driver dating from the early Seventies (as used in
the BBC LS 3/5a).
Although this driver was probably the one of the first high-quality
midranges available, it also had a number of problems, such as low
efficiency, limited power-handling, a broad one-octave peak centered at 1.5
kHz (corrected by the crossover), and group of 3 very high-Q peaks centered
around 4.5 kHz (only slightly attentuated by the BBC third-order
crossover). These upper peaks, which reviewers mistakenly ascribed to the
tweeter, were also very directional, which is typical of dustcap
resonances.
The popular tweeters of the 1970's, including the Audax and Peerless 1"
soft-domes, also had similar resonances between 9 and 16 kHz, which were
partially damped by a felt pad nearly filling the space between the dome
and the polepiece. Since the soft-domes were much more lossy than the stiff
B110 dustcap, the resonances were much broader and only 1 to 3 dB in
magnitude ... but they were still there, and they were responsible for some
of the fatiguing quality noticed by attentive listeners.
Not surprisingly, the problems were much worse in the phenolic, fiberglass,
and hard paper domes used in the more mundane speakers of the day. (Ah yes
... who can remember such paragons of excellence as the BIC Venturis? The
Cerwin-Vegas? The Rectilinears? The JBL L100's? In a prior life, I actually
had to sell these awful things! "Wait'll you hear 'Dark Side of the Moon'
on these babies!")
Returning to the present, the best midbass and tweeter drivers now sidestep
this problem in two ways: a vented polepiece assembly, used by the
Scandinavian manufacturers Dynaudio, Scan-Speak, Vifa, and Seas; and a
bullet-like extension of the polepiece, which replaces the midbass dustcap
entirely, used by the French manufacturers Audax and Focal.
By contrast, the Focal T120 and T120K, which use a rigid fiberglass or
Kevlar inverted dome directly above an undamped polepiece cavity, show a
series of high-Q resonant peaks at the top of their operating range, which
are caused by the resonant cavity coupling to the first breakup region of
the rigid dome.
I must admit I was rather puzzled by the public acclaim for these drivers
when they first came out; I didn't like the way they sounded, and I wasn't
too impressed by their measurements.
>From all accounts, though, the new Focal titanium-dome T120Ti and
titanium-dioxide T122Ti-O2 are excellent, and I liked what I heard when I
auditioned a speaker that used the Focal T120Ti at a recent Triode Society
meeting.
Magnetic Non-linearities
Most audiophiles are aware that loudspeaker drivers are inductive; after
all, the voice coil is wound around a ferrous polepiece, and that's how you
make an iron-core inductor (or "choke"). Not as many audiophiles know about
the myriad of problems this creates.
Are there solutions? Yes. The best drivers from Scan-Speak (SD System) and
Dynaudio (DTL-System) plate the polepiece with copper to short out eddy
currents induced within the magnet structure by the voice coil. The
specification that gives this away is the voice coil inductance.
The 8" Scan-Speak 21W/8554, probably one of the best 8" drivers in the
world, has an inductance of 0.1mH, which is far lower than the 8" Vifa
P21W0-20-08, which has in inductance of 0.9mH. Both are excellent drivers;
the Scan-Speak, though, is almost certainly going to have more transparent
sound when asked to carry bass and midrange at the same time.
The inductance figure also has a another hidden meaning; remember, the
upper rolloff frequency of the driver is the combined function of the
mechanical rolloff and self-inductance of the voice coil. If you calculate
the electrical rolloff frequency by using the VC inductance and the DC
resistance, a few drivers have an electrical rolloff well above the
measured acoustical rolloff. This is desirable; it means that the
interaction between the two rolloff mechanisms is going to be small.
Other drivers (and this is true of most drivers) are going to have an
electrical rolloff well below the measured acoustical rolloff. How is this
possible? The mechanical system actually has a broad peak which is masked
by the self-inductance of the voice coil. This is not good; any change in
either the mechanical system or the electrical system is going to strongly
modulate the frequency and transient response.
This, by the way, is the same kind of problem found in the old
moving-magnet phono cartridges. Most moving-magnets (typically Shure and
Stanton) were mechanically peaked, then rolled-off electrically by the
combination of cable capacitance and cartridge inductance. Not
surprisingly, this type of cartridge usually sounded much less transparent
than its high-end moving-coil brethren, which had less than one-tenth the
inductance and a much flatter, more accurate mechanical system.
In the section that follows, I'll show you how you can make your own
decisions about which drivers you like (and second-guess the manufacturer,
reviewer, and your audio friends).
* Selecting A Driver
I use a method that's so crude it might sound kind of dumb; I put the
driver on large, IEC-sized baffle (135cm by 85 cm) and listen to it. No
crossover, no enclosure, and if it's a tweeter, not very loud at all. I
listen to pink noise (to assess the severity of the peaks that may appear
in the sine-wave and FFT waterfall measurements) and music (to get a sense
of how much potential resolution the driver posseses).
This does take an educated ear, though, since you have to listen around the
peaks that the crossover might notch out, and not hold the restricted
bandwidth against it. However, this listening process tells you a lot about
how complex the crossover has to be, particularly if you remember that the
crossover can never totally remove a resonance ... it can just make it a
lot more tolerable.
2) The Group Delay vs. Frequency Response. (How ragged is the frequency
range above the first breakup? Can it be fixed in the crossover?)
Listening and measurements are equally essential. Both give only a partial
picture of the actual driver. Even the finest modern audiophile system will
have very serious sonic deficits 5 years from now; measurements provide a
reality-check on colorations that present-day equipment may not reveal. In
turn, the MLSSA system can point out troublesome colorations to listen for;
some are much more audible than others.
------------------snap-------------------
This is a revision of an article published in Vol. 4-2 of Positive Feedback
magazine, a publication of the nonprofit Oregon Triode Society located at:
4106 NE Glisan
Portland, OR, 97232
USA
PART 4
The second in a series called "The Soul of Sound" where I wear my professor
hat. These articles are my *personal* view of designing high-end speakers.
I expect this article to be thoroughly out of date in two years or less ...
so enjoy it while it's fresh!
-------------------snip-------------------------
* Types of Drivers
It helps when you start listening and comparing to have a good grasp on the
basic characteristics of the driver, so you can determine if it is a good
example of its type. By listening carefully and examining all of the
relevant specifications, you can find out just how well the driver
designers solved the problems of making a good driver.
This dates back to the original Rice & Kellogg patent in the late Twenties.
Paper ranges in quality from the worst TopTone clock/radio speakers to the
superb Scan-Speak 5" cone/dome midrange used in the Theil line of speakers
and the SEAS 6.5" midbass used in the Wilson Audio WATT. This oldest of
materials is actually a composite structure, and changes properties
significantly when it doped with an appropriate plastic (the choice of
doping is invariably a trade secret of the driver vendor). The doping is
quite important, since paper undergoes significant alterations with changes
in humidity and time if it is left undoped; the doping stabilizes the
material and typically improves the self-damping.
Weaknesses are: Not as rigid as the Kevlars, carbon fibers, and metals, so
it lacks the last measure of electrostatic-like inner detail. Doesn't go as
loud as the materials above, but the onset of breakup is much more gradual.
Paper-cone drivers usually require modest shelving equalization in the
crossover for best results.
Best Examples are: Scan-Speak 8640 5" cone/dome midrange, with linear
response out to 13kHz, very low distortion, excellent pulse response, and
excellent inner detail.
SEAS 6.5" midbass (as used in the Wilson Audio WATT, but possibly
modified).
This is an acetate plastic derived from wood pulp, not petrochemicals, and
is always damped by a layer of doping material to control the strong first
resonance it displays around 1.5 kHz. It was originally developed by the
BBC in 1967 to replace paper with a more consistent and predictable
material for monitoring purposes. It came into widespread use in the early
Seventies, with the typical audiophile speaker using a 8" KEF or Audax
Bextrene midbass driver with an Audax 1" soft-dome tweeter.
Best Examples are: None. Modern designers are not willing to tolerate the
complex notching and shelving equalization required to make these drivers
acceptable.
* Soft-Dome Tweeters
These came into common use in the early Seventies with the introduction of
the Peerless 1" soft-dome (remember the tweeters of the original Polk
speakers?), followed by the superior Audax 1" tweeter, which found its way
into many British and American designs during the Seventies and early
Eighties.
These designs fell into disfavor with the introduction of titanium and
aluminum domes and the Focal inverted-fiberglass domes in the mid-Eighties,
which swept the Audax-class soft-dome drivers off the audiophile market.
In the last two years, the soft-domes have made a surprising comeback with
the introduction of the Dynaudio Esotec D-260, Esotar T-330D, and
Scan-Speak D2905/9000 1" tweeters, which compete on even terms with any
metal-dome around. These new designs combine sophisticated
transmission-line back-loading with new dome profiles and new coating
materials. As a result, they have the sonic resolution and detail of the
best metal domes without the characteristic 22 to 27 kHz resonance.
Weaknesses are: The older class of soft-domes had a dull sound with a
hard-to-pin-down fatiguing quality. Many had quite limited power-handling
and required a high-slope 18 dB/Octave crossover to minimize IM distortion.
The high-profile dome, required for rigidity, has more restricted HF
dispersion than the competing metal-domes, which have flatter profiles.
The new units mentioned above do not have these faults, however, with the
exception of dispersion.
Best Examples are: The Dynaudio Esotec D-260, Esotar T-330D, and Scan-Speak
D2905/9000 1" tweeters.
* Soft-Dome Midranges
These things are dogs! I've listened to the AR-3, AR LST, ADS systems,
Audax 2", and the Dynaudio D-52 soft dome midranges, and they barked, they
snarled, they chewed the rug, and made a mess on any loudspeaker they
approached! They measure flat, all right, but they sound opaque, fatiguing,
strongly colored, and 2-dimensional.
A third problem is that the doped silk dome is just, well, too soft for the
job it has to do in the power band of the midrange.
The newer class of cone-domes, such as the 5" Scan-Speak 13M/8636 and
13M/8640, and the 5" Dynaudio 15W-75, are another story. These 3 drivers
are actually constructed as high-precision cone drivers, not midrange
domes. The only thing they have in common with the soft domes is a large
dustcap, which does act as a dome at high frequencies.
This class of driver has much more excursion, much lower distortion, and
much wider frequency response than the older soft-dome midranges. The
cone-dome drivers are capable of realistic and transparent sound. They are
described in more detail in the other sections, since they use Kevlar,
paper, and polypropylene respectively.
Strengths are: None. Metal-dome midranges have some potential, but they
require sharp crossovers on both ends with an additional sharp notch filter
at high frequencies to remove the first (and worst) HF breakup mode. Note:
This does not apply to the ATC driver or the cone-domes.
Weaknesses are: High distortion, fatiguing sound, high crossover frequency,
limited bandwidth, limited power-handling, and misleading frequency
response measurements. It takes a detailed swept IM distortion measurement
and laser holography to get the goods on these drivers. Note: This does not
apply to the ATC driver or the cone-domes.
* Polypropylene Drivers
This material was developed by the BBC in 1976 (my dates may be off) as a
replacement for Bextrene. Since it is intrinsically highly self-damping, a
correctly designed polypropylene driver is capable of flat response over
its working range without equalization. In addition, it typically attains
efficiencies of 88 to 91 dB at 1 meter, which is a significant improvement
over Bextene.
Best Examples are: The Scan-Speak 18W/8543 7" midbass, as used in the ProAc
Response Threes, is probably the finest polypropylene driver in the world.
* Metal-Dome Tweeters
Advances in German metallurgy (at Elac and MB) resulted in thin profile
titanium and aluminum domes in the mid-Eighties, with drivers from several
vendors in Germany, Norway, and France now available. This type of driver
can offer very transparent sound, rivaling the best electrostatics if
correctly designed.
Best Examples are: Vifa D25AG-35-06 1" aluminum dome, which is even better
with the plastic phase disk removed. This dome has a vented pole piece, so
power handling is quite good, and the ultrasonic peak is only about 3 dB
even with the phase disk removed. Also, the brand- new Focal T122Ti-O2 is
reputed to be outstanding.
* Rigid Drivers
Aluminum
Expanded-Foam
The next generation were the expanded-foam bass units, with the KEF B139
being the most famous examples. This class of driver offered piston-band
operation through the midbass, but suffered from very low efficiency,
limited power-handling, and severe high-Q resonances in the midband. (It
was not generally known that the B139 had a 12dB peak at 1100Hz with a very
high Q. Many reviewers blamed the midrange for problems that were actually
caused by the B139 not having a notch filter in the crossover.)
Charlie, my boss: "Hey Lynn! You remember what Laurie Fincham was talking
about when we visited KEF last year? All that stuff about impedance
correction and frequency response target functions?"
Yours Truly: "Well, I didn't write it down, but I remember some of it."
Charlie: "Great! You can do the crossover for this!" ... pointing at a
massive six-foot-tall loudspeaker with the 4 aforementioned drivers. The
previous designer had left town without warning, leaving Audionics with a
monster transmission-line complete with drivers and no crossover.True
story, folks.
The next generation were the Japanese carbon-fiber units, which made their
first appearance in the pro studio monitor (prosound) 12" TAD units with
very high efficiencies and very high prices (around $300 each in 1980).
Carbon fiber prices have now dropped, and Vifa and Audax make good examples
of this type of driver. The Japanese make lots more of them, having
pioneered the technology, but they have been difficult to obtain if you are
a non-Japanese small-run specialist manufacturer.
These drivers have true piston action, outstanding bass and midbass
response (the best I have ever heard), but also have nasty, chaotic breakup
modes at the top of their range. Removing these breakup modes requires a
sharp slope and one or two very sharp notch filters (this type of driver
and filter is used in the top-of-the-line Linaeum speaker).
Kevlar drivers made their appearance in the mid-Eighties with the French
Focal and German Eton lines, with the Eton having superior damping due to
the higher-loss Nomex honeycomb structure separating the front and rear
Kevlar layers. The Eton and much newer Scan-Speak Kevlar drivers now share
the limelight as the worlds pre-eminent high-tech drivers.
Composite
The Future
Evaporated diamond coatings are now available at low cost with a recent
Russian breakthrough (cheap enough to coat floppy disks!), and I very much
hope that Scan-Speak and others pick up this technology quickly. As I said
earlier, things are changing fast.
Any loudspeaker that does not use a correctly designed notch filter with a
Kevlar or carbon-fiber driver can be considered faulty; since the HF peak
does not lend itself to correction with a conventional low-pass filter, and
will be quite obvious to any listener familiar with the sound of an
unequalized Kevlar or carbon-fiber driver. Unpeaked rigid drivers are not
currently available ... stay tuned, this will probably change in less than
a year.
Although these types play quite loudly, the onset of breakup can be quite
sudden and unpleasant, akin to clipping in a amplifier. Some Kevlar and
carbon-fiber drivers require an extremely long break-in period (>100 hours)
to soften the fibers in the cone and the spider; this is a fault, since it
indicates the materials may not be mechanically stable with extended use.
Best Examples are: The Scan-Speak 13M/8636 5" midrange, 18W/8544 7"
midbass, and 21W/8554 8" bass drivers. These are the only Kevlar drivers
that have reasonably well-behaved rolloff regions above the high-frequency
peak.
The Scan-Speak drivers also have vented pole-pieces that are copper-coated,
reducing inductive types of IM distortion by tenfold or more. These drivers
are probably at the pinnacle of rigid-driver technology (in Spring of 1993
at least).
The Audax HM130Z0 5.25" midrange, HM170Z0 6.5" midbass, and HM210Z0
8" bass
drivers in the HD-A series also look very interesting.
The German Etons also bear close watching, since the manufacturer is
continuing to work on techniques to maintain the cone rigidity while
improving the self-damping characteristics.
The next two articles that follow aren't on the Net, unfortunately, since
they make extensive reference to construction drawings of the cabinet,
suggested placement in the room, and lots of MLSSA measurements. The two
articles (The Soul of Sound, Part III and IV) describe an efficient (92dB @
1 meter) 2-way transmission line suitable for use with triode amplifiers of
modest power (8 to 22 watts work just fine). Look for these articles in
Vol. 4-4 and 5-1 of Positive Feedback magazine - back issues are still
available, as far as I know.
If I can convert the Macintosh PICT artwork to something that will survive
the uuencoding process and figure out an appropriate FTP site, I may post
these in the future. In the meantime, if you good folks on the Net are
interested, I could post my review of the Reichert 300B and the Ongaku in
this forum. (The review articles are entitled "The Outer Limits" and are
*not* the Authorized Version from StereoPriest, Absolution Sound, or the
Audio Cleric.)
------------------snap-------------------
This is a revision of an article published in Vol. 4-2 of Positive Feedback
magazine, a publication of the nonprofit Oregon Triode Society located at:
4106 NE Glisan
Portland, OR, 97232
USA
Readers who are collecting the entire series might notice the absence of
Part III and Part IV ... well, those two parts are the construction
articles for the Ariel high-efficiency transmission-line loudspeaker, and
I'm still working on the politically correct way to transmit the rather
large graphics files over the Net (I may set up an FTP site here at
Teleport). If you just can't wait to find out about the Ariels, you're
probably better off subscribing to the magazine, which is chock-full of
tweaks, reviews, construction articles, and off-the-wall pieces of all
kinds.
-------------------------
The Soul of Sound, Part V
by Lynn Olson
lynno@teleport.com
For those who haven't followed these electronic discussions, here they are
in print. I also have a bonus feature at the end of the column, a
newly-arrived little brother for the Ariel.
I don't think "euphonic distortion" exists. Granted, some sonic flaws are
a lot more tolerable than others, but I've never found any class of
distortion or signal bending that makes the sound more transparent, more
real, and more lifelike.
People have been making gizmos like Aphex Aural Exciters, delay circuits,
and "T-function shaping" for decades now, and all they really do is make
music sound more like commercial FM radio. To my ears, at least, when I
remove a known problem (such as a resonant mode in a cabinet, a standing
wave on a driver cone, a problem in a power supply, or improve the
linearity of an amplifying element) it pretty much always sounds better
... more real, more truthful, more expressive, etc. In short, fixing
problems makes it sound truer to life.
So why do tubes typically, but not always, sound better than transistor
circuits, despite worse overall distortion? The answer doesn't lie in
circuit simplicity, since if that were true, first-generation transistor
units like the Dyna PAT-4 and the Marantz 7T preamps would be audio
classics. Well, they're not, and a quick listen on even a half-decent
system tells the tale.
I don't like measurements that only apply to the entire "black box", be it
a speaker, an amp, a CD player, or whatever. They tend to mislead both the
reviewer and the buyer. The measurements that tend to be the most
meaningful are ones you can make on elements of the black box ... an
active device, a single driver, the way the crossover behaves far
off-axis, a single reflection at a cabinet edge, the frequency vs. source
impedance of a power supply, etc.
Case in point: I've lived with the Audio Note Ongaku SE211, as well as the
Kassai PSE 300B and the Reichert SE 300B. I've also had access to my
trusty Audionics CC-2 (not a bad transistor amp), a modern multi-kbuck
Class A transistor unit, and a souped-up Dyna Stereo-70. They all sounded
different, particularly to non-audiophile friends.
The Ongaku, by far, had the worst THD and power measurements ... 22W at 3%
distortion. It also made the Ariel sound better than any electrostat I've
ever heard ... in fact, the best sound I'd heard in many years. It
certainly sounded better than anything I heard at the 1994 Winter CES. So
what's going on here? Maybe THD is simply measuring the wrong thing.
To test this, of course, you'd want to bias the device and measure the
base-collector capacitance with a high-quality bridge that measures DF and
DA. Even though the capacitance is only in the pF region, it is
Miller-amplified by the gain of the transistor. This means any capacitance
non-linearities are multiplied by the forward gain, which is also
non-linear. So we have two non-linearities multiplied by each other, and
worse, one is in the time domain, and constitutes a type of FM modulation
of the audio signal.
Now, this effect is probably very small, and hard to measure as well.
Nevertheless, Walt Jung successfully identified very small problems
related to the DF and DA of capacitors more than a decade ago, and it is
generally agreed now in the upper reaches of prosound and audiophile
engineering that the choice of dielectric is very important in speakers
and electronics. I'm just taking the same thing and applying to it to
active devices, and throwing in the implications of Miller-amplification
of DA and DF problems.
Note that tubes are very different in this respect. As capacitors go, the
dielectrics are basically a little mica, glass, and a high vacuum ...
pretty close to theoretically perfect. In addition, this capacitance is
unmodified by voltage, current, temperature, or even if the tube is stone
cold and completely inactive.
A: This isn't going to be the answer you want to hear. Using an unmeasured
"textbook" crossover is certain to produce poor results ... nowhere close
to the potential of the driver. Why? Drivers are nowhere close to a purely
resistive load and do not have a flat response ... both of which are
requirements for a "textbook" crossover. Using a stock crossover will
produce errors from 3 to 20dB, as well as wildly variable inter-driver
phase characteristic, resulting in nulls that sweep through the listening
position. Results? Grossly non-hifi sound. Don't do it.
The minimum system is probably the IMP FFT board published in Speaker
Builder magazine. The IMP and the Mitey Mike are about $400-500 from Old
Colony. Be aware that pure FFT systems have severe problems with
signal-to-noise ratio, requiring a quiet room and an amp that can throw
big pulses at the speaker. Old Colony Sound Lab is at: (603) 924-6371.
The next step up is the LMS from LinearX, made right here in Portland USA
(I have no affiliation with LinearX ... I don't even get a discount). This
is a chirp-based system with excellent frequency resolution and noise
rejection, a very good integral mike, but no directly-measured phase or
group delay info.
I use MLSSA (so does Glenn Philips and plenty of big-name manufacturers).
The MLSSA has largely displaced the Crown TEF, since it delivers pretty
much the same performance at about 1/3 the price and doesn't force you to
buy a reworked Kaypro as a host computer. (I've heard the new TEF unit
uses a PC host and has a price comparable to MLSSA ... maybe even cheaper.
I haven't kept up on this.)
The MLSSA measures freq. response, phase, group delay, does waterfalls,
calculates polar response, sum-and-difference curves, washes the dishes,
rents the movies, etc. (Well, maybe the last two items are in the next
upgrade.)
Yes, there are instruments that cost more, with the Audio Precision coming
to mind. Since I remember an all-up price of >$12,000, I haven't checked
it out. I'm not sure if the AP offers true time-domain measurements, but
it does have a lot of slick automated test features that lend it to fast
production checking. The AP is just the thing for amps and CD's, though,
so if my little startup ever shows a profit, I'll buy one (after I pay
back my wife/sweetie/partner).
Now that I've thoroughly snowed many readers, I should stop and make a
point. Discussions of measurement systems quickly degenerate into
religious wars for many speaker designers, with distinct overtones of
Windoze/Macintosh, UFO/Phil Klass, Fundie/Materialist/NewAge wrangles.
Phooey on all of 'em. I use these machines to find out why the speaker
sounds strange, and I don't expect the instrument to think for me. There
are plenty of things that are plain as day and don't show up with any
measurement technique. Likewise, there are things that are just slightly
audible, yet look quite nasty on the screen. This is where experience
comes in.
Not good enough: General-purpose FFT systems, 1/3 octave analyzers, and
warble tones used with sound level meters. These systems do not have
adequate resolution to disclose narrowband resonances, are nowhere
accurate enough for crossover optimizing software, and do not provide
information on time-domain behavior (the FFT does, but it suffers from
serious problems with signal-to-noise ratio).
Well, how do you select these "known-good" components? There are two ways:
A) Pick out things you really, really like. (For example, I really, really
like the Audio Note Ongaku and the Reichert amps. You might like Krell or
Levinson.) Find out what's inside. Don't be a pest, just quietly track
down the capacitors, resistors, wire, etc. on your own.
B) Do your own sonic investigations. Go ahead, tweak away, and see which
parts you like! Just keep good notes, ask friends over and record what
they say, and tweak something that's already good to start with! Don't
design something new and start playing with parts! You'll just go around
in circles, since you'll be dealing with far too many variables all at
once.
Of course, once you start researching parts quality, you'll soon discover
that tweaking takes just much time as "real" designing!
That's why I ask around and find out what other designers are already
using, and start from there. Speaking for myself, I just find it easier to
start constructing with "known-good" parts that have worked well in other
designs that I respect.
Frankly, I have no idea why these archaic rectifiers with all their
problems sound better than solid-state rectifiers. I don't care. I want to
design a good tube amplifier, not become an expert on rectifiers. Now when
I'm happy with the overall sound of the amp, and not before, I might
explore other types of rectification ... and maybe not.
3) To audition the new device, you need a "standard of reference" for the
rest of your hi-fi system.
The real secret is to track down first-rate systems, hear for yourself,
and find out what others have discovered. Tune into pleasure ... the right
equipment will have the extraordinary ability to touch the emotional truth
of nearly any recording, regardless of "sound quality." Seek it out!
---------------------
--
Lynn T. Olson ........ lynno@teleport.com
Senior Editor, Positive Feedback magazine,
a publication of the nonprofit Oregon Triode Society.
For subscriptions, call (503) 234-4155 or fax (503) 254-3866.
-----------------------
Q: Why did you draw such big distinctions between schools of design in
Part 2 of Soul of Sound?
"In America, we assume that if it were to sound "just like live", all
of the emotional content of the performance would be conveyed
automatically.
As my patient readers have noticed, I digressed pretty far from the last
question. Well, it covered a lot of territory. It's hard to convey the
flavor of this industry until you spend a little time inside of it, which
is why I slide over into a kind of gonzo journalism when I describe the
social side. The dry, "this-is-what-I-saw-at-the-CES" stuff just doesn't
convey the zany aspect of the high-end business.
My heart is with the zany side ... I saw the Japanese eat Scott, Fisher,
and Marantz whole, watched as Gordon Holt, Bob Fulton and Audio Research
rose from the ashes of the American consumer electronics business to
create an independent high-end market, then saw the two "underground"
magazines mature and become the all-powerful gatekeepers for the high-end
industry, and now ... full circle again ... a new set of magazines and a
new philosophy is sweeping through!
Since the new arrivals are minimonitors, efficient, and 2-way, and are the
little brother of the Ariels, Karna suggested christening them the ME2.
Why not? So ME2 it is.
The ME2 is 18" high, 8" wide, and 8.5" deep, with an internal volume of
521 cubic inches, and a pair of 1" diameter vents that are 4.125" long. It
retains the large diameter radius (1.25" on top, bottom, and one side) of
the Ariels. Everything else is the same as the transmission-line Ariels
... same drivers, crossover, front panel layout, composite MDF/plywood
construction, etc.
Your initial reaction might be: 70Hz! That's not very much bass! Well,
actually, it's not too different than plenty of other minimonitors ... a
lot of them are actually spec'ed at the -6dB point, which gives a better
match to the perceived low-frequency limit.
Still, I agree, you're not going to shake the room with these speakers.
The ME2 aimed at three groups: folks who like the unadorned minimonitor
sound, multichannel systems, and/or systems that use one or two
biamplified subwoofers.
A Multichannel Digression
Part of the reason I designed the ME2 was listening to the Cogent Research
SPI, which is a modern sum-and-difference 4-channel decoder using
psychoacoustically optimum decoding coefficients (whew!) This unit offers
high-resolution "frontal quad" without the listening fatigue that most
people associate with surround and quadraphonic systems.
Long ago, I invented the Shadow Vector quadraphonic decoder for Audionics,
which became one of the first dynamic-matrix decoders ever built. (This
was far more advanced than the commercial CBS/Sony "Full-Logic" system,
which used gain-riding and a static matrix.) As a result of months of
fine-tuning the Shadow Vector, I became quite aware of the action of a
dynamic-matrix decoder on various types of recordings.
Many years later, after quad disappered from the map of hi-fi, the Dolby
Pro-Logic Surround decoders became popular for theatres and home video
systems. The modern Pro-Logic family of decoders all use a dynamic-matrix
technique closely related to the Shadow Vector, the CBS Paramatrix, and
the Sansui Variomatrix of the quad era. Since they use the same basic
operating principles as the previous devices, movie soundtracks are mixed
down in the presence of an engineer from Dolby Labs, to ensure that the
mix doesn't baffle the logic circuits of the decoder used in the theatre.
(Yes, it is quite possible to encode a mix that can knock the logic
circuits for a loop. Happens all the time in multimiked classical
recordings.)
Also, as with stereo, all of the speakers and amps must be *precisely*
matched. Unlike stereo, though, quality mismatching doesn't result in a
left/right asymmetry, but instead, a very rapid increase in "phasiness"
and fatigue. This is why so many people who use mismatched low-cost rear
speakers eventually give up on quad. Using a "logic" system to force the
localization into the mismatched speaker simply results in more listening
fatigue, not less, as frequency response, phase, and polar patterns go
through extremely rapid changes as the sound source is steered around the
room.
2) Any system that requires the user to flip through mode switches for
each type of program (mono sources, movie surround, classical, jazz, rock)
is fundamentally flawed. You don't need to do this in stereo, after all. A
correctly designed multichannel system should provide stable, accurate,
and fatigue-free localization of all possible encoding possibilities ...
mono, panpot stereo, multimike stereo, Blumlein stereo, Dynaquad, EV-4,
Ambisonic, Dolby Surround, etc.
My prelimary listen to the Cogent showed a lot of promise ... very wide
and spacious soundstage, stable imaging, with reasonable but not great
sound. BUT we were listening to a quad mid-fi Rotel amp and four
Definitive Audio speakers at $300 each. Comparing it to the two Ariels and
the Kassai 300B was ludicrously unfair ... the Kassai is a superb $40,000
amplifier, and the Ariels would have to sell for $4000/pair in a retail
environment.
Which brings us to the ME2 speaker. It's more compact, less expensive to
build, and offers the same sound quality as the Ariels, with 25 Hz less
bass. It's a prime candidate for an audiophile-quality multichannel system
using the Cogent decoder, particularly with a high-quality pair of stereo
subwoofers, which brings us to ...
The ME2 and Powered Subwoofers
Stereo subwoofers are a good idea, but not for the reasons you might
think. It is indeed true that localization is very difficult to perceive
when the wavelengths are longer than 10 feet. However, there is an effect
known as "Spatial Impression", which conveys the impression of size, or
space, and it is quite important at very low frequencies. A mono
subwoofer, in other words, won't distort the image, but it won't sound as
*spacious* as stereo subwoofers! In addition, it has been shown that a
2-speaker stereo image grows narrower as the frequency is lowered ...
which means that it is desirable to place the stereo subwoofers much
further apart than the normal 60 degree stereo-pair. Good results have
been reported with 120 or even 180 degree spacing of the stereo
subwoofers.
Level-setting shouldn't be a big chore ... select some piano music that
uses the bass register (I use the full-keyboard scale on the ProSonus test
CD) and adjust the subwoofer level and phasing for the most natural and
"fast" sound. Using piano music as a reference should curb the natural
tendency to let the subwoofers creep up in level.
The crossover is the same as the Ariel, since the factors that affect the
mid and upper regions are unchanged (cabinet width, driver layout, edge
radius, etc.) I would stick with North Creek Audio as a supplier,
particularly for the resistors ... do not use the typical sand-cast power
resistors found in most crossovers.
Cabinet Construction
The drawings pretty much tell the tale. Use Baltic Fir or Apple-Ply
plywood for the internal members, and premium-grade MDF for the outer
shell. The 1.5" thick front panel plays a significant role in quieting
down the cabinet and is worth the hassle of bonding two 0.75" panels
together. Be sure to make the speaker in mirror-imaged pairs and try to
achieve the 1.25" radius shown on the drawings ... I found out that there
is a pretty big sonic difference between 0.75" and 1.25" radius when I
made the Version 1 and Version 2 Ariels.
Lightly fill the V-shaped rear half of the cabinet with long-fiber wool
(best for mids) or crimped Fortrel (like Acousta-Stuf), but don't cover
the inlets of the vent tubes. If you're using the ME2 with subwoofers,
though, feel free to stuff the vents themselves (this slightly improves
the midrange at the expense of bass response).
You'll notice that there really isn't any room inside the cabinet for a
crossover ... that's intentional, since vibration of crossover components
can degrade the clarity of sound. A small external box for the crossover
is the best solution ... but keep it at least 18" away from steel stands
and transformers.
Setup
Next, adjust the height of the stand so the centerline of the tweeter is
between 38" and 40" high. If the tweeter is much below ear level, the
stability and overall size of the stereo image will be degraded, so adjust
the height so the tweeter is at or above ear level.
When the ME2's are correctly set up and used with the right electronics,
they will have a very big, spacious sound, a stable stereo panorama well
off-axis, and will sound quite natural and lifelike even from another
room. They are "tuned" for the best possible voice reproduction, so use
recordings of male and female singers to do the final in-room tuning with
crossover levels, cables, damping material, location from the back wall,
etc.
When you are all through with the tune-up process, the ME2's will have all
of the sonic quality of the Ariels you've read about in previous issues of
PF ... they're just as well braced, use the same drivers and crossover,
and just as efficient at 92 dB/metre. So turn on your favorite 2A3 or 300B
triode amp, get out your favorite sides, and have some fun!
Suppliers:
Cogent Research
25145 Manzanita Drive
Dana Point, CA, 92629
(714) 493-0721
Many, many drivers, all with spec sheets. Also, all of the little
bits-n-pieces needed for a complete loudspeaker.
A&S Speakers
3170 23rd Street
San Francisco, CA, 94110
(415) 641-4573
Even more drivers, but no spec sheets unless you ask. Good info on which
vendor is using what drivers in their commercial speakers. VMPS kits as
well as their own.
ORCA
1531 Lookout Drive
Agoura, CA, 91301
(818) 707-1629
They sell the very effective "Black Hole Five" damping material as well as
the interesting Axon speaker cables. All kinds of interesting odds-n-ends
from these folks.
--
Lynn T. Olson ........ lynno@teleport.com
Senior Editor, Positive Feedback magazine,
a publication of the nonprofit Oregon Triode Society.
For subscriptions, call (503) 234-4155 or fax (503) 254-3866.