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From: Jeff Austin
OK, let's start with basics. What are the main differences in sound between
comparable speakers, one with a ceramic magnet, one with AlNiCo? Also, how
does the size of the voice coil affect the sound.
Jeff, good questions. The whole 'AlNiCo mojo' is about smooth compression at high average levels,
such as what you would have running the amp flat out. AlNiCo (Aluminum-Nickel-Cobalt) is an alloy magnet and all alloy
magnets are easier to demagnetize than comparable Ceramic (Strontium Ferrite) magnets. What this means
is that as the voice coil starts moving in response to the input signal, it generates a magnetic
field of its own that tries to demagnetize the magnet. As its effect lowers the available magnetic
field of the AlNiCo magnet, the speaker becomes less efficient, the voice coil moves less, etc.
The physics of it is that the small magnets near the surface of the magnet poles (called 'domains')
begin to change state, or flip directions. The result is smooth
compression, the same kind of operating curve compression that occurs in a tube amplifier.
The ceramic magnet, on the other hand, doesn't compress or demagnetize as easily, so the
voice coil moves to its mechanical limit and won't go any farther. This is why some players say ceramics
sound a little edgey at high average levels as opposed to AlNiCo. However, by properly designing
the entire magnetic circuit, Ceramics can be made to behave quite well for desireable guitar amp tone and dynamics.
You might compare the two magnetic circuits to solid state amps versus tube amps, where the
solid state amp gives it all its got then clips hard, while a tube amp compresses nice and
smooth. The extension of this idea, then, is that with the AlNiCo, like the tube amp, you can seem
to have a louder average volume since it gets compressed smoothly. By the
way, the compressing or demagnetization that occurs with the AlNiCo is not
permanent. It springs right back to its design operating point.
A voice coil is like an electric motor. The bigger the voice coil, the more wire used,
the more torque or pulling power you have to move the cone. With the proper
match of components, you can get more sensitivity, wider frequency response,
and more power handling ability.
From: Larry Barras
What's the difference in sound between paper and kapton/synthetic voice coil
formers? Is the choice of material substantial to the sonic character?
Larry, despite the fact that using a paper former is a great marketing tool for vintage
style speakers, the fact is it has no effect on the sound itself. It, like Kapton,
is a diamagnetic material, and it's effect on the magnetic field and the effect of
the magnet field on the former itself is negligible. The big difference is in the mass, or weight
of the formers. In the early 70's when the race was on to have the highest powered
solid state amp, speaker manufactures had to keep up by making their speakers take
alot of power. Therefore, Kapton was put into use. Because the Kapton is thicker,
heavier, etc. (a typical one looks like an orange plastic pill bottle), it takes
alot of power to get it moving. So, the movement was to heavier Kapton voice coils,
heavy, damped cones, and damped surrounds for a high powered, relatively low sensitivity
speaker. Now we're going the other way. Low powered amps, light weight speaker components,
and high sensitivity to make the lower powered amps sound huge. The one
possible exception to the coil former having an effect is the use of an
aluminum former. Some believe that since it is a closed loop of metal, there
are 'eddy' currents circulating in the former. These eddy currents could affect the
sound by causing a braking action to voice coil movement, thereby increasing the damping of the speaker.
From: Rich Santucci
Does having an aluminum voice coil cover affect frequency response?
I've always been told that it adds high end. Is this true?
Rich, let's look at the history of the voice coil cover, or 'dustcap'.
At first, its only purpose was to keep dust and debris out of the voice
coil gap. If you look at any speaker made in the early years, like a Jensen
P12R, it had a simple flat felt dustcap about the size of a quarter. Then,
as research on the perfect speaker proceeded, it was discovered that if you
use a domed dustcap made out of the same material as the cone, you could
change or smooth out some of the peaks and dips in the frequency response
of the speaker. Next came a combination marketing/engineering idea. A big
aluminum dustcap would certainly look cool, and since aluminum has high
thermal capacitance, it would suck heat out of the area around the voice
coil and radiate it into the air. It's a win-win-win. Cool look, adjust
the frequency response, and get heat out of the voice coil. So, to answer
your original question, yes, by sizing the dustcap appropriately, within
reason, you can affect the frequency response dramatically, including
extending the highs.
From: David Knechtges
I recently purchased a '62 Brown Princeton which has a noticeable speaker buzz.
I determined the problem is probably a warped voice coil or something in the gap,
because I can push in on the cone and hear a rubbing noise as the cone moves.
The speaker is an Oxford, dated '62. Should I buy a replacement speaker, have the
original reconed, or ?
David, congratulations on the purchase. Nice amp. The noise is definitely a rub,
either from, as you suggested, a warped (from overheating) voice coil, or flakes
of paper or other material stuck in the gap between the voice coil and pole
or front plate hole. There is a way to correct that if it isn't too severe.
I'll detail it here, then you can make the decision whether to try it or not.
The result is that you correct the problem without reconing the speaker, thus
preserving the value of the original speaker. First, since you will be performing
this operation without demagnetizing the magnet, make sure your work area is very
clean and you have plenty of light. Lay the speaker on its back with the cone facing
up and with a scalpel, carefully cut out the dustcap, leaving about 1/16" of dustcap
where it is glued to the cone. This is important because the voice coil wires
pass through this point and you want to make sure you don't cut them. Next,
use a vacuum cleaner or clean, dry pressurized air to suck or blow the dust
and other debris out of the gap. If you hold the speaker upside down with the cone
facing downward it will probably help getting the dust and debris out. Next,
take a 3x5 index card and cut it into a strip that is the correct length
so that you can form it into a circle and stick it down into the gap between the
inside of the voice coil and the outside of the pole. This will
help form the voice coil back into a circle. Next, lay the speaker back down
on its back. Take a Q-tip or small paint brush and dip it into a bottle of
acetone (finger nail polish remover). Spread a small amount of this acetone
on a couple of the rings of the spider, which is the brownish/yellow corrugated
disk attached to the backside of the cone at the base of the basket. Next,
place a jar lid or other disk on the cone where the dustcap was and let the
speaker set overnight. The lid or disk will prevent dust from getting into the gap
overnight, and the acetone causes the spider to relax and reposition slightly,
thus repositioning the voice coil. The next day, remove the lid and the index card
strip and see if you still have a rub. If you do, try the acetone again, same
procedure. If, after a couple of tries, it seems hopeless, then professional
reconing is the only resolve. I think it's worth trying though, to preserve
the value of the original speaker. If it works, contact me with the size of the
dustcap and I'll send you one to replace the one you cut out and instructions
on how to replace it. As far as using the speaker, if you plan to use it regularly,
at high volumes, I would suggest packing the original away and install a replacement
speaker. Many speakers would work well in that amp, such as a Mojo MP10R,
a Naylor 10, a Kendrick 10, or a WeberVST P10Q. If you want some british tone,
you might check out Celestion's new Silver series or WeberVST's Blue Pup and
Silver Ten.
Footnote 02-17-97 David followed the procedure outlined above and corrected
the rubbing problem, thus preserving the value of his vintage speaker.
From: Shawn Bolton
I think I understand your explanation of the differences between AlNiCo and
Ceramic magnets in speakers, but what do people mean when they say modern
AlNiCo is different than old AlNiCo or that it has a half-life?
Shawn, I hadn't heard about the belief that old and new AlNiCo are different,
however they are one and the same. For speaker applications, AlNiCo 5 is
the best choice in the AlNiCo family of alloy magnets. Its peak energy
product is just right for loudspeakers where we need to concentrate high
densities of magnetic flux in the gap around the voice coil. AlNiCo 5 is
an alloy made up of 8% Aluminum, 14% Nickel, 24% Cobalt, and 3% Copper.
The cobalt is what makes the AlNiCo expensive. Most of the worlds supply
comes from the African country of Zaire. Besides that country controlling
the market, cobalt is also a strategic metal used in missles and other
weapons systems. It currently sells for about US $32 a pound.
As far as it having a half-life, that's another new one on me. When a speaker
is put together, the magnet is initially uncharged, or unmagnetized. Then,
at the end of the assembly line, just before testing, the speaker is passed
under a huge electromagnet that zaps the magnet with about 10 to 20 times
the magnetism that would be required to saturate the magnet. After the
electromagnet is turned off, the speaker magnet immediately loses about
2% of its magnetism and then stabilizes. In the next year, it drops another
1%, and then is essentially stable for thousands of years. Unlike a flashlight
battery, the magnet is not being used up or depleted of its energy while in
use. All that has happened is that we have forced many of the tiny molecular
magnets called domains to realign themselves in one direction. Once 'flipped',
they reach an equilibrium and stay there. Besides intentional demagnetization
with a demagnetizer, there are three situations that can cause a magnet to become
partially or fully demagnetized, and this may be what the half-life comment is
all about. The first one is excessive heat. That's not a consideration for speakers
because the temperature to demagnetize an AlNiCo magnet (called the curie point)
is over 1000 degrees F. The second is from a large, changing magnetic force. This
could happen in a speaker. The typical case would be where a person blows out
the speaker by using excessive power. The high value of magnetism produced by
the voice coil could partially demagnetize the magnet. This is why you
should always ensure that anyone who is going to recone a speaker for you have
a magnetizer to renew the charge on the magnet, just in case it got partially
demagnetized. The third and final situation is physical shock. If you
dropped an AlNiCo magnet speaker and it happened to land on the corner of the
magnet frame, it could partially demagnetize the magnet.
From: Graham Kahnt
I need information on how to wire speakers to obtain 2, 4, 8, & 16 Ohm
loads.
A schematic for each configuration would be prefered. This has been a
puzzle to me for a long time.
Graham, let's look at the definition of impedance first, then go from there.
You'll often see a speaker or other device rated at a 'nominal' Z, or impedance.
The word 'nominal' comes from the Latin word 'Nomen' which means simply 'name'.
An example of where you may have heard this term used in another context is during
a space shuttle mission. During the initial ascent, you'll often hear the
astronauts say "all systems nominal", or "mission nominal". What they mean
is that everything is going as planned, as written, or as described. In the case
of a speaker, we are calling, or naming the device a certain impedance. The electrical
quantity of impedance is made up of resistance, which doesn't change
with frequency, and reactance which does change with frequency. So, impedance
is the combination of the two at a particular frequency. Remember in the movie
Wizard of Oz when the Scarecrow finally got a brain, he immediately started reciting
some bizarre formula about "The sum of the squares of the sides of a right triangle...."?
He was reciting the Pythagorean theorm for right triangles. We can use that formula
to calculate impedance. Think of a flagpole with the sun casting a shadow on it.
The height of the flag pole would represent the reactance, and a line between
the base of the flagpole and the point on the ground away from the flagpole
where the shadow stops would represent the resistance. If you connected a string
between the top of the flagpole and the point on the ground where the shadow stops,
the length of the string would be the impedance quantity and would be longer than
either of the other two. So, what am I getting at?
A speaker that is said to be 'nominally' 8 Ohms will have a resistance lower
than 8 Ohms. So, a rule of thumb is that if it measures lower than a commonly
used name, such as 8 Ohms, but not lower than the next lower commonly used name,
such as 4 Ohm, then you would call it the higher name, or 8 Ohm. Many 'nominal'
names have been used over the years, including 2 Ohm, 10 Ohm, and 15 Ohm. 4, 8,
and 16 seem to have been standardized though, for the past 30 years or so. The main reason
for the variance between several coils, all of which would be called 8 ohms, for instance,
would be because each might have a slightly different DC resistance due to the
length of the coil, size of the wire used, etc. In each case, though, the DC
resistance would be in the range of 5.5 to 6.5 DC Ohms, so it would be called
an 8 Ohm device. Another method for naming the impedance of a device is by
actually measuring the impedance, or AC resistance of the device with
special test equipment. Many have used 400hz as the test frequency, while
others have used 1,000hz as the test frequency. Some have derived the name
from an impedance plot such as the one shown in Figure 1 below. They
declare the nominal impedance to be the impedance at the 'first dip after
the first peak'. Notice the large peak at the speaker resonance around
100hz. Then, it drops dramatically, dips, then starts going back up.
It would be the impedance at the lowest point of the dip that would be the
declared 'nominal' impedance. It's an interesting exercise in using an impedance
bridge, but the old rule of thumb we discussed earlier works just fine for declaring
a nominal impedance.
From: Glenn Ewbank
Do the holes in the speaker frame affect the sound any?
Glenn, that's a good question. Technically, in an open backed or ported enclosure the size and shape of the cutouts would make a difference in the sound because there is actual movement of the air mass behind the speaker as the cone moves. The air movement consists of a pressure waveform that changes with the frequency and intensity or loudness of the sound from the speaker. In a sealed enclosure, such as a 2-12 arrangement in a piggyback amp, the cutouts don't affect the sound as much because the waveform reaches a constant pressure or equilibrium and remains fairly constant. Of course, even closed back, or sealed cabinets have to leak a little. If they were hermetically sealed, barometric pressure changes would bias the speaker cone in or out and disturb the operation of the speaker. Over the past 30 years or so, most speaker designers haven't given much thought to the issue of designing basket cutouts because the necessary metal stamping and tooling for speaker baskets is very expensive. Therefore, a few different types have been essentially standardized for the industry. You'll notice that most speakers made in America are either a 4 or 6 trapezoidal hole arrangement, and baskets from most manufacturers look about the same. British speakers tend to use the larger 4 hole trapezoidal arrangement.
From: Steve
I have a 20/50 watt THD 'plexi' amp which I run either 2-EL84s or 2-EL34s.
In order to get rich distortion and compression, it must be played loud. I would
like to get that sound at lower levels, so I intend to purchase a THD HotPlate
or similar attenuator. I understand that at lower volumes, the
sensitivity of the ear changes according to a Fletcher-Munson curve, so
I'm wondering what kind of speaker cabinet or system I should use to
compensate at the lower volumes?
Steve, that's true about the F-M curve and the correction needed at the lower
volumes. I would have to recommend though, before using a power attenuator that
you go through the calculations or have a professional amplifier technician
advise you on its use with regards to steady state power handling and thermal dissipation
capabilities of your output circuit. There's some 'sonic psychology' involved here, in
that as you are listening comfortably to the lower, attenuated level of acoustic power, the
output circuit is still giving it all it's got. So, since your ear hears the lower
level, it isn't quite so evident that the output circuit is working so hard.
The bottom line is that the output transformer is heating up, it might be in saturation
alot of the time, you lose some bass due to the saturation as well as increased resistance
of the windings due to heating, etc. As far as speakers go, depending on
how much attenuation you want to dial in, you'll probably want to get the most effecient
speakers you can find. The bottom line is that you are attenuating the big amp down to the power
level of a small amp, so you need a speaker that will make the small amp sound huge.
If you want to restore alot of the highs, you might go with a Kendrick 10, which is
very efficient at the high end. As far as AlNiCo vs Ceramic goes, since you aren't going
to really blast the speakers with a high average volume, that is to say alot of your
tone (distortion plus compression) will be coming from your output circuit, I think
Ceramics would work just fine, and save you some money.
Additional comments submitted by Joe Pampel:
Another option you might consider is the use of a closed back or sealed
cabinet. With an open back, you lose a little bass due to cancellation
between the rear and front. In a closed back, that problem diminishes,
however it takes more power to get the same loudness. Also, rather than using
high efficient speakers as suggested above, you might actually prefer lower
efficient speakers. Assuming you'll have plenty of power to put into either the
power attenuator or the speaker, you can put less in the attenuator and more
in the speaker for a given loudness. You might find that this arrangement will give
you more overall control of your tone at the loudness you desire.
From: Bill Abernathy
Tell me about speaker fatigue. I've heard that it has to do perhaps with
heat buildup in the voice coil after heavy use (i.e. after a 4 hour gig at volume).
Also, does an older speaker that's been used alot at volume necessarily see speaker fatigue
and need replacing?
Bill, speakers have always been quite resilient and forgiving considering
the beating they take in the way of cone flexing and vibrations. The weakest mechanical links in the chain are the corrugations in the surround
at the periphery of the cone. At this point, the cone is the thinnest, yet
takes alot of vibration. So, like bending a wire until it breaks, the corrugations
will often begin to tear after long term use. Since the cone is made from paper pulp,
other considerations such as temperature and humidity play a part in weakening the cone.
Many speakers are a little tight when new, and seem a little sterile. After
being driven at high levels for awhile, they change character slightly as the
surround corrugations begin to loosen up. The spider also loosens from the constant
flexing, so the net result is a little more low end and the cone breakup characteristics
change. Most people acknowledge that this change is for the better as far as enhancing
the desired tone for guitar work.
Heat buildup in the voice coil is definitely a problem, because it will eventually cause the
shellac (or other chemical used to hold the wires in place) to soften and allow
the voice coil former to deform. We can't afford to have the former deform much,
because the gap between the outside of the former and the plate hole (called 'ring')
is somewhere in the neighborhood of .010 to .012 inch. Many attempts have been made
over the years to get the heat out of the voice coils, including the use of
large aluminum dustcaps or regular dustcaps with vent holes. Several different
former materials have been used also, including aluminum and Kapton.
One of the big problems I've seen with vintage speakers is with the spider
shifting in position. This shifting will allow the voice coil to shift, causing a
voice coil rub. Many vintage baskets were painted, so the glue holding
the spider down wasn't applied directly to the metal. Over time and with alot of vibration,
the glue would actually pull up the paint.
Vintage speakers used various forms of glue, from animal based to other chemical
concoctions. Those glues were no match for the CA (super glue) type glues available today.
Today, we use a brown, rubbery glue we call 'elephant snot' to attach the
spider and the underside of the cone to the basket. For attaching the voice
coil to the cone neck, we use CA-77, a type of thick super glue. For the
top of the cone to the gasket, we use a white glue similar to Elmer's glue.
This is for appearance only, since it dries clear. We still use the
flexible black rubbery glue to hold down the voice coil wires where they
pass through the cone and connect to the tinsel foil wires that run
to the terminals. The dustcap is held down with either the black glue or
the super glue, depending on dustcap style.
Your question about using vintage speakers and the resulting fatigue is
a tough one for me to answer. Because of the vintage glue and pulp cone ageing problems,
the chances of failure when used regularly at high average volumes is considerable.
If I had a vintage amp with original speakers, I'd be inclined to remove the
originals and, assuming they were in good shape, pack them away in safe storage.
Then I would install replacement speakers, such as Kendrick, Mojo, Naylor,
Celestion, or WeberVST. Because of the high intrinsic value of original speakers,
I'd be much more comfortable beating a replacement speaker with high volume.
From: James Miades
I recently purchased a '60s Ampeg SB-12 amp. As I understand it, the amp was
designed to be used for either guitar or bass. But, when I play bass through it
it sounds distorted. I had a tech check it out, replaced tubes, etc., but still
had the distortion. As it turns out, the 12" had a small tear at the outer
edge. I thought about reconing it to keep the amp original, but the tech said even
after reconing, it would still distort since it is not a 'bass' speaker.
Why would Ampeg design a bass amp with a speaker that was not designed to
handle the bass? Do I need to replace it with one specifically designed to
handle the bass frequencies? If so, what would be a suggested replacement?
My tech has a reconed EV-SRO for $150, but that sounds expensive to me.
Jim, while using a speaker with a low bass resonance and given the fact that
lower frequencies tend to require more power, it makes sense that you would get
more volume with that type of speaker, especially if you could get one with a high
SPL (more sensitivity). However, one of the most popular amps of all times --
the Fender '59 Bassman -- started out as a bass amp and then became popular for
guitar, and it has 4-10" low power speakers. If you were going to use the stock
speaker in the Ampeg for guitar only, I'd suggest trying to repair the small tear
with the popular concoction of tissue paper and Elmers or with silicon gel.
Try your best to close the gap in the tear before glueing in an effort to
maintain the same position and 'parallelagram' established by the cone, spider,
and voice coil. If you want to use the speaker for bass, though, I'd recommend
reconing it. The excursions (overall cone movement in and out) at bass
frequencies will probably strain the patch, causing the cone to shift
slightly at that point during peak excursions, and the result could be voice
coil rub during those peaks. This would put high frequency scratchy sounds
on your bass note and would be very annoying. If you do have it reconed,
you can ask the reconer to select a cone and spider that will yield a lower
resonance and better low end response at the expense of losing a little of
the highs.
From: David
When I was restoring an old phonogragh I noticed the magnets on
the speakers were those with the thin magnet inside a small metal box
with two sides missing. What are these speakers called? Are they worth
reconing?
David, I believe you are describing AlNiCo plug type magnet speakers. The
small metal box with two sides missing is called the field case. The sides
or legs of the field case direct the magnetic field from the top of that
round magnet around to the bottom or base plate. In the middle of the
base plate is a hole. Attached to the bottom of the magnet is a piece of
round steel, and that round piece fits inside the hole in the base plate.
There is a gap or space between that pole and the hole in the base plate.
The voice coil that is attached to the cone slides in and out of that
gap or space and fits around the round metal piece attached to bottom of
the magnet. The speaker is assembled very carefully so the voice coil
doesn't touch the pole or the sides of the hole.
All old AlNiCo speakers are worth reconing in my opinion, because they
are AlNiCo and have that vintage sound. The only drawback is the
expense of getting them reconed. Many record players of that era used
similar speakers, so you might check out your local thrift shop,
garage sales, etc. for a similar unit. You'll probably be able to buy
a complete unit with good speakers for less money than you'd spend
reconing the ones you have.
I noticed in your original note you said you are 12 years old.
Congratulations on your restoration project and your success with it.
You've chosen a great and rewarding hobby, in my opinion.
From: Gerald C. Lopez
I have two 8 Ohm speakers that I can wire either parallel or series to my
tube amp for an impedance of either 4 or 16 Ohm. My amp has both 4 and
16 Ohm output taps. Are there any sonic differences or benefits of series
over parallel wiring or vice versa?
Gerald, connecting two speakers in parallel is an old trick to smooth
out speaker response and enhance the damping of either speaker. HIFI designers
took it one step further by connecting two speakers of different sizes in parallel.
A speaker has a large impedance increase at its fundamental resonance, and
depending on the installation, this can cause the speaker to sound boomy or
out of control. By connecting two speakers in parallel, particularly two speakers
of different sizes with different resonant frequencies, each speaker will tend
to quench or dampen the boominess of the other. Since no two speakers are exactly
alike, even two of the same size, that damping will occur, however slight,
for any speakers connected in parallel. For speakers connected in series,
there appears to be less control, and more of what is called 'back EMF'
from the speakers fed back into the output circuit. While that seems rather
chaotic, many players prefer the series connection, as it gives them a
more textured tone, enhanced breakup, and overall a more desireable tone
for guitar work. It's totally subjective, of course, and many factors
affect the end result, such as voice coil size, gap energy, closed back/open
back, output circuit damping, etc. The best thing to do, in my opinion,
is try both arrangements since you have the luxury of impedance tap
selection, and go with the configuration you like the best.
From: David Abe
I am interested in AlNiCo magnets. There are many kinds of AlNiCo
magnets, my question is what kind are used in those old Altec, Western
Electric, JBL, etc. speakers?
David, an entire family of alloy magnets employing Aluminum, Nickel, and Cobalt were developed.
The most common ones are 2, 5, and 8. Each has its own unique characteristics
which generally has to do with how strong they are when magnetized (similar to
how many volts a battery has at its terminals), and how easily the magnet
can be demagnetized (how much current the battery can put out before its
voltage begins to sag). AlNiCo 5 was almost universally chosen for loudspeaker use
because it has a high flux (like high battery voltage), and under normal
circumstances of use, the speaker wasn't intended to be driven hard enough
to affect the magnetism of the magnet by the voice coil magnetism. Of course,
all guitar players discovered early on that driving a speaker hard into
this region of what is called 'recoil' yields a smooth compression of the
magnet field and as it turns out is very desireable for guitar tone. Taking
the magnet to the 'edge' of recoil is not a problem for AlNiCo 5, because it
springs back to its operating point as the signal to the voice coil is reduced.
However, excessive power to the voice coil, enough that would probably cause
damage to the coil in the form of excessive heat would most likely generate
enough magnetism to partially demagnetize the AlNiCo magnet. What's interesting
about magnets and magnetic circuits is that they have a load line just
like tubes and transistors, and these load lines are employed when designing
the loudspeaker, motor, or other magnetic device.
From: Alex Rush
I was wondering if you could tell me anything about magnetic sheilding
on speakers (i.e. center channel speakers) because I'm building my own
and I have no clue how to shield them.
Alex, the reason we have to shield the magnet on the speaker is because
not all of the magnetic flux is focused in the gap as useful energy for use
by the voice coil. The wasted, or unusable flux is called leakage flux.
Ceramic magnet speakers and modern speakers with large gaps tend to have
more leakage flux than vintage AlNiCo magnet speakers with tight gaps.
An example of a typical ceramic magnetic circuit and the location of the leakage flux is shown in
Fig. 1 below. Shielding the near environment of a speaker from the leakage
flux is often important, especially near a television since the picture
tube in a TV uses magnetism to direct the electron beam to draw the
picture on the screen. The constant leakage flux from the speaker would
affect the scanning beam and distort the picture as well as the colors, etc.
The best shielding is a canister or pot-shaped cover made of a high permeability
metal called mu metal. It is basically a short circuit to the leakage flux
so the flux can't get through the mu metal and radiate into the surrounding air.
What happens is that the leakage flux tries to magnetize the mu metal, the
mu metal generates a magnetic field of its own that resists the leakage flux.
All of the magnetism stays within the mu metal so it isn't radiated. The
foregoing was a simple explanation of how all magnetic shielding works.
With the popularity of surround sound, center channel, etc. for TV's today,
many speaker companies are starting to use 'pot' covers for their larger,
higher powered speakers that will be used in close proximity to a TV. If you
want to try to make your own, you can get a coffee can or other metallic canister
that will just slip over the magnet as diagrammed in Fig 1. Apply some
silicon gel to the perimeter of the magnet to hold the can in place. The
only precaution you need to take is that you want to make sure your canister
doesn't come in contact with both the back plate and the front plate that
sandwich the magnet, which would cause a shunting of the magnetic flux
away from the gap where it is needed. Another method of containing the leakage flux
that is gaining in popularity these days is the use of a bucking magnet.
The speaker manufacturer takes a magnet the same size of the magnet
used in the speaker, magnetizes it, and glues it to the back plate.
It looks strange as heck to see this ceramic ring magnet hanging off the
back of the magnet assembly, but it is quite effective in cancelling
the leakage flux from the main magnet and in some cases, can actually
add to the flux in the gap. Strangely enough, when you add up the
cost of a metallic pot cover of sufficient size, the cost to paint it, and
the cost of the glue or fastener required to install it,
in many cases the extra ceramic magnet used as a bucking magnet is
actually cheaper. TV's are not the only place mu shielding has been used.
I was recently told by an acquaintance that when he installed a new speaker
in his amp (which had a rather large magnet), his channel switching quit
working. Apparently, the speaker was located close enough to the channel
switching relay that the leakage flux affected its operation.

From: T. Naumann
I don't know alot about guitar speaker cabinet design. I take it that
open back cabinets take advantage of the drivers free air resonance, but
how do manufacturers come up with the dimensions of sealed enclosures?
It seems to me that the specs on the individual speakers don't matter
much to designers if you can just buy four new speakers and drop them into
your cabinet. Do they even try to figure that stuff out like HIFI loudspeaker
designers do? For instance, if I had a Marshall 4x10, how could I be sure
that four of your P10P's would sound right in it? Would it be worthwhile to
try to custom design and build a sealed cabinet based on the T-S parameters
of the P10P?
I sure wish I could state an answer as well as you posted the question. Unfortunately,
I have never met, talked to, nor read any articles written by the original
designers of speaker cabinets for guitar amps with regards to their philosophy
of design, etc. Like most, I do have my own observations and opinions on them, though.
No doubt, in a combo amp an open back enclosure is imperative to allow air
to flow in and out for cooling the tubes and other electronic components.
Having the back open also allows for a convenient storage space for the
power cord, etc. The amp would sound louder overall at the expense of losing
a little on the low end, the extra loudness being a plus for the marketing of the amp.
Most combos I've seen are designed with just enough room for the speaker to
be mounted with a symetrical border of baffle around it, while not interfering
with any of the chassis components. Having been involved in production engineering
as well as cost accounting, I can assure you there was also some thought
given to how much wood and covering would be used for a given size cabinet
using the standard sizes of material stock, while yielding the least amount of
scrap. Another issue would be total shipping weight, since that would have to
be factored in when arriving at a price for dealers. In a sealed enclosure,
it generally takes more electrical power to get the same acoustic power as
a similar sized open back. The sealed air in the cabinet behind the speaker
acts as a resistance to cone movement, the smaller
the enclosure, the more the resistance. Think of blowing up a small, tight balloon.
You have to blow like heck to get it started compared to a large loose balloon.
So, the bigger the box, the better, generally. Now think of the early '60s
'piggy back' amps. It wasn't long before players were standing them on end with the
amp head sitting on top. Was it for better sound projection or because it
looked cool? Of course, manufactures saw that, started making the 'stand-up'
models, and made the cabinets larger but still used the same size speakers.
Next is the question of designing the box to match the speakers technical specs.
I've often wondered that myself, and have never really arrived at a good answer.
Some recent amp manufactures, such as Dr. Z are using ported cabinets and
are getting very high marks from reviewers. Nevertheless, these designers,
like all guitar amp designers, are designing and building for a particular
sound and performance as a guitar amp, not a HIFI speaker enclosure. In other words,
the issue here is music 'sourcing', not music 'reproduction'. Your last
question is one I don't think I'll ever be able to answer, even as a speaker
manufacturer. I've had people call me and tell me "I gotta tell you, this is
the worst speaker I've ever heard in this amp, way too dark". Yet, I've had
others tell me the same speaker is the best they've ever had, and they've been
trying for a long time to find the right speaker for that particular amp.
To design a cabinet that matches the speakers technical specs will certainly
smooth out its performance, make it a better mathematical model, so to speak,
but on the other hand, much of the desireable guitar tone is in the comb filtering,
cone breakup, and other distortions that occur when a speaker is taken to the edge
of its design operating parameters. It's apparent this is why serious players buy,
sell, and trade speakers, like amps, until they get the tone they are after.
Additional comments submitted by Joe Breher: Jim Marshall, the
man who gave us the world's most emulated sealed box guitar amp
speaker enclosure, was recently interviewed by a popular guitar
magazine. In the interview, he stated that the dimensions for his
cabinet were dictated by the smallest practical cabinet that would
fit 4-12" speakers. Evidently, no thought whatsoever was given to the
acoustical properties of the cabinet.
From: Bob Thomason
I'm building a 1x12 guitar cab and have made the baffle from 3/4"
plywood. I have not yet installed it and just read online about a guy using
3/8" ply for his baffle. I have a 70s Fender Princeton and it's baffle is
3/4" that's where I got the figure from. I have never built a cab so what
difference is this going to make in the tone? I plan to enclose the back.
The head is 50w all tube. Any help appreciated.
Bob, let's look at the reasons for the different woods, construction techniques,
and how they affect tone. In a HIFI enclosure, we want the cabinet to be
sonically dead, so some cabinets are felt lined, or other forms of deadening
material are used. In a guitar amp, to a certain degree, we want the opposite
characteristics. Because, as we discussed in the previous note, we're sourcing
music rather than reproducting it, a live cabinet adds to the texture of the guitar
tone by generating standing waves and minor resonances that produce a comb filtering
effect, hence the added texture of the overall tone. That's why there has
always been an ongoing debate on the use of plywood versus pine board,
finger joints versus lap or butt joints, etc. Of course, we don't want the cabinet
to be so lively that it buzzes on big bass notes, etc. I wouldn't recommend
the use of a 3/8" board for the baffle, though, especially with vintage speakers,
because of the tight gaps and lightweight speaker components. Any flexing in the
baffle could twist or strain the basket and cause a voice coil rub.
Another consideration when using a thicker baffle board is that since
the speaker is located farther back from the front surface of the
baffle board, it tends to decouple the speaker from air mass
resonances that occur at the opening. The 3/8" difference between a
3/8" and a 3/4" board may not seem like much, but it can and does make a difference.
Those who remember the tone ring used in some 60's amps are aware
of this effect. Vibrations on the surface of the baffle board can
intensify the effect of the air mass resonances. The thicker baffle board
increases the mechanical resistance to those vibrations, thereby
reducing their intensity.
I've built solid pine, particle board and plywood cabinets and have been happy with the results,
however, I have always used a 3/4" baffle board. Perhaps some other readers can
give us their opinions on this, which I would be happy to post here.
From: Mike
I would like to get into reconing speakers and was wondering if you
could tell me how to get started. Where to get parts, can you order
parts from the speaker companies,etc. If you knew the answers to these
questions, it would be great!
Mike, there is a company in Ft. Wayne, Indiana called WVS which sells an
instructional video tape on reconing. They can also provide parts and customer
support. Their web page is www.reconer.com
You can also contact Image/Waldom in Chicago to see about purchasing speaker
reconing products from them. They require a membership and have protected
territories for existing members, so your success in being able to buy
from them will depend on how many of their members are already established
in your area. Technically, speaker reconing is very easy, and with practice,
you can recone a speaker in about 15 to 20 minutes. It seems, though, with
any business, there is always a 'gotcha!' In the case of speaker reconing,
it is imperative (in my opinion) that you have a demagnetizer/magnetizer.
It's extremely important to get any debris out of the voice coil gap prior
to rebuilding the speaker. If any of the debris happens to be ferrous,
(material that is attracted to a magnet) the only way you will be able to
remove it is with the magnet demagnetized. After the rebuild, you recharge,
or remagnetize the magnet to it's full state of magnetism (saturation). This
is also very important, especially in the case of an AlNiCo magnet. If the
speaker was blown by the use of excessive power, the magnet could have been
partially demagnetized. So, you need to ensure the magnet is 'renewed' in
any case. At the very least, your customer deserves to have his or her rebuilt
speaker as clean as when new, and a full charge on the magnet.
The 'gotcha!' I referred to is the cost of the magger/demagger. A commercial unit
to do all of the more popular speakers today costs about $14,000. You can buy them
used or as surplus for about half that, and can get some even cheaper that need
repaired. Talk about lethal energy, though, WOW! Let me give you an idea of
how much energy is in a magger/demagger. If you're familiar with a typical
Fender 40 Watt amp's power supply, you might see a high voltage circuit
of around 350-450V with 40 to 80uF (microfarads) of filtering.
Our big magger for production goes up to 800 Volts and has over 80,000uF
of filtering (storage). Needless to say, that's alot of juice, and it all
gets dumped into a coil around the speaker magnet in less than 1/50 of a
second. Rare earth magnets require even more energy than that. The coil and
wires shake violently, and we have to hold down the speaker with quite a bit
of pressure so it won't get tossed around. The whole thing sounds like a
muffled explosion when the coil gets blasted with that much energy.
Best of luck to you with your reconing business. If we can be of any help,
please don't hesitate to ask. I don't think there will ever be a shortage
of speakers that need to be reconed.
From: Richard Canton
One of the two speakers in my amp has apparently developed a buzz or rattle.
I don't think it is a voice coil rub because I can't hear the noise at low
volume and it only does it when I hit certain notes. Any idea what the problem
might be?
Richard, it sounds like the magnet assembly has worked loose and is resonating
against the basket plateau. Let's look at how the magnet assembly is attached, then
we'll try to find a resolve for the problem. For the past thirty years or so,
most speaker manufacturers have used a staking process to attach the magnet assembly
to the speaker basket plateau. When the front plate of the magnet assembly is
fabricated, a punch press is used to punch out the round front plate. Then,
a second operation punches the hole in the middle where the voice coil fits.
A third operation punches either four or six holes in a three inch pattern
around the voice coil hole. In this operation, the four or six punches don't
go all the way through the plate, only about 3/4 the way.
This results in four or six plugs or stakes to appear on the other side
of the plate. At the speaker assembly shop, the basket is placed down over these
stakes and a press flattens or mushrooms them out to fasten the plate to the
basket. Some speaker manufacturers use a gasket between the plate and the
basket, while others use a bead of glue. These are used in an effort to
stop any rattle that may occur. The problem is that due to production tolerances,
punch wear, etc., some of the stakes may not get pressed or mushroomed all the
way out. After awhile, with vibration, these can work loose enough to cause
the buzz or rattle.
Traditionally, speaker manufacturers tested each speaker by sweeping a frequency
from low to high and then checked the response with a standard as a pass/fail
indication. This was time consuming, and with the advent of fast computers capable
of doing fast mathematical analysis, most speaker manufacturers today use
a composite test tone package, display the result on a computer screen, and compare
the plot to an upper and lower plot also plotted on the screen. The composite
test only takes about 1/2 second and resembles the sound that would occur
if you could push all the numbers on your touch tone telephone at the same time.
While this method saves alot of time, it doesn't always catch the problems
such as the one you are probably experiencing with the magnet/basket rattle.
So, what do we do about it? If you have an audio oscillator, connect it to
one of the amp channel inputs. Next, turn up the amp to a comfortable level and
slowly sweep the audio oscillator from a low to a high frequency. If the buzz
doesn't occur, try turning up the volume and do the sweep again. Speaker
manufactures tend to get carried away with their power ratings, so you don't want
to leave a steady tone at high power level driving the speaker for very long,
just long enough to detect the buzz or rattle. OK, let's assume you found
the frequency at which there is a pronounced buzz. Place your fingers on the front
side of the magnet and your thumb against the basket plateau. Slowly move
your hand around the perimeter of the magnet while maintaining a constant
pressure with your thumb against the basket plateau. When you find a spot
where the rattle of buzz stops, mark it or remember it, and turn off the audio
oscillator and amp. Next, get a safety razor blade, and with two pair of
pliers, break the razor blade in half lengthwise. With one pair of pliers,
or needlenose pliers, force the sharp side of the razor blade into the gap
between the front plate and the basket plateau. If it's too loose,
you may have to double up on the blades, or use a thicker piece of metal.
I like to use a razor blade since the edge is sharp and will work
into the gap much more easily than a blunt edge. Do the sweep test
again and see if this cured the problem. If it did, use some silicon
gel or other thick glue to hold the razor blade in place.
If, by the
way, you don't have access to an audio oscillator, you can use a
microphone and whistle a tone from low to high to perform the sweep test.
A few manufacturers still bolt their magnets to the baskets. The reissue
Bulldog, for instance, uses brass bolts to attach the plates
and large ring magnet to the basket. WeberVST uses bolts and glue on all
their products out of necessity since their plates and poles are
machined rather than stamped.
From: Chuck Minahan
How do I determine the best match, power wise, between amp and
speakers? For example, I have a 100 watt SF Twin and want to replace
the two 8 Ohm speakers. How do I calculate the wattage ratings needed
for each speaker to best match the amp? Also, how do I determine the
specifications of the two speakers I have in there now? They are
different in appearance and have different model numbers. I'd also like
to know if it is OK to mix different types of speakers in the same amp.
Chuck, normally you would use two speakers of the same power rating.
For instance, in your situation, you would use speakers which would be
rated at 50 watts minimum. Together, whether they are wired series or
parallel, they would handle the 100 watts since the power would be divided
evenly between them. Speakers with higher power ratings generally
have larger magnets since the wire in the voice coil is a larger diameter,
is heavier, and requires a larger magnet to get the magnetic energy required
to get the heavier voice coil moving to achieve high frequency response.
Low frequency response is also improved.
As far as determining the specifications of an unknown speaker, about
the only way to do that is to take it to a professional reconer who can
perform sweep and T-S parameter tests on it.
Mixing different types of speakers in the same installation will work
as long as you know the power ratings of each speaker and don't exceed
those ratings in either speaker. Many players I have talked to
achieve the tone they are looking for by mixing speakers with different
characteristics. For instance, one might have clean, tight low end,
with smooth top, while the second one would be very efficient in the upper
mids and high end. I know of one player who has a 2x12 amp with a
P12B Blue Dog on one side and a P12N on the other. According to him,
it gives him just the right blend and texture for his particular
playing style.
From: Albert Tejera
I have seen special devices that mimic the frequency response of vintage
Celestion speakers, etc. for the primary purpose of running a guitar
preamp directly to a mixing board for recording or P.A.
It seems that you could accomplish the same goal by either using a parametric
EQ or just a multiband EQ if you knew where to start. I know that a
Celestion starts increasing in output around 100hz or close to its resonant
frequency, and starts dropping off about 5Khz. Is there a concise source of
frequency response specs for all the speaker voices we have learned to love?
Can you produce such a source? I have searched for hours on the web and no
luck. I am mostly interested in the graph of vintage Celestion and Jensen
guitar speakers.
Albert, the speaker plots we are all familiar with and often see
published are single tone sweep plots, usually performed with 1 watt
of power applied to the speaker. This has become a standard in the industry
and allows us to compare one speaker to another on the same terms. Other
tests performed include 2nd and 3rd harmonic distortion plots, and single
tone sweep plots at higher levels, e.g. 25 and 50 watts. While these tests
allow us to compare apples to apples, so to speak, they don't really address
the issue of how the speaker reacts to a composite signal made up of
many frequencies. When you apply a composite signal to a speaker, the
low frequencies may move the cone a certain amount, while the higher
frequencies riding on the low frequencies move the cone in small steps
back and forth. The cone system itself, because it has mass (or weight),
will continue to move by itself and actually works against the signal
commanding it to move by generating a voltage in the voice coil that
counteracts the signal from the amp telling the voice coil to move.
This is called motional impedance. As you can imagine, this is a very
complex effect and is not very friendly to mathematical analysis. This is
why the selection of speakers, by necessity, is very subjective. What
appeals to one player may not appeal to another. This motional impedance
also plays a big part in how a speaker responds to a particular
amplifier output circuit or enclosure. Many speaker emulators on the market
today use inductors, capacitors, and resistors in an effort to model a
speakers voice coil, but they stop short of emulating the motional
impedance. Others use opamp filters as well as other shaping networks
in an effort to put peaks or dips in the response curve. The only way
you could truly emulate a speaker would be to have some kind of
electromechanical device in the signal path that would generate the
'back EMF' and in effect, the unpredictable motional impedance. Several
methods could be used to do this. You could build a dummy load/voice coil
emulator and use a small portion of the signal to drive two
4 or 6" speakers mounted face to face. The low powered signal
would drive one of the speakers, while the other one acts as a microphone
to pick up the acoustic signal of the driven speaker. This 'microphone'
signal would then be amplified and mixed in with signals from other parts of
the circuit, and you'd then have a fairly complex signal at the
output. Another method would be to remove about half of the
laminations from an iron core choke and use that in the design of the
dummy load/voice coil emulator circuit. The laminations would
vibrate and emulate motional impedance. Of course, the laminations
would need to be fairly small and lightweight so they would vibrate
sufficiently to affect the inductance of the choke.
The foregoing description of emulating motional impedance assumes you
have a power amp to drive the emulator circuit. Since your question
was about using a preamp output for DI, the only choice you really
have is to use filters or EQ to emulate the single tone sweep
response, short of using a complex algorithm employing a Digital
Signal Processor IC. As far as plots on the Internet, I have seen
plots for HIFI drivers, but as you have discovered, I haven't found
any for musical instrument speakers. I characterized and studied
every vintage loudspeaker I could get my hands on while doing the
research for our speakers, and plan to add a page to our website to
display those plots and their T-S Parameter info. I will also post as
much information as I can about guitar amp speakers available today.
We also plan to have a page that will provide information for
building guitar amp speaker enclosures using the T-S parameter info
for these speakers. We will have info for sealed (closed back) as
well as ported cabinets.
From: Ron
I have a ceramic Jensen gold and brown label speaker (220523) that is
stamped a "C12PS". I have seen 12P's, N's, Q's, etc., but what does the
"PS" mean?
Ron, according to the old Jensen data that I have collected, the AlNiCo 'P'
model (P12P) is similar to the 'N' model in that it uses a 1-1/2" voice coil
and has essentially the same cone dimensions. In your case, it is a later
model employing a ceramic magnet. For this reason the energy in the gap of
the 'P' model is about 2/3 that of the 'N' model. It also has a power rating
a couple watts less than the 'N' model. The 'S' suffix is something I'd have
to guess on, though. For special or custom design applications, many
manufacturers used one of their base models and made changes in the
magnet, spider, etc. to get the frequency response or other
characteristic the customer desired. In this case, the customer
probably specified a looser spider in an effort to get the same
sensitivity with a smaller (less expensive) magnet. That could be
where the 'S' came from. It could also designate a particular
customer, or simply the next letter in a sequence for modifications
of a particular base model. If I run across some difinitive
information on this particular model suffix, I'll post it here.
Incidentally, Jensen made a model C12N in later years, but if I'm not
mistaken, it is quite a bit different than the P12N in most
respects.
From: Chris Eades
Why is it that some speakers sound great in a closed back or sealed
cabinet, but sound terrible in an open back cabinet, and why is the
opposite true with other speakers?
Chris, excellent question. When a person designs a speaker, one of
the considerations is how much the cone will move and the subsequent loudness
for a given amount of power driving it. If the designer uses a loose
spider and fairly flexible surround on the cone, the voice coil will
have an easier time moving the cone and the speaker will be loud
with a small amount of power. The problem with a loose spider and
surround is that as the power is increased, it takes the cone to its
mechanical limit of movement and it gets harsh. This is an
underdamped system and can also sound ringy. If you put this speaker
in a sealed or closed back cabinet, the air that is trapped in the cabinet
acts as a spring or big balloon that the back of the cone pushes against.
This air mass helps control the movement of the cone and also affects
the damping of the system, making the speaker sound more in control.
A designer who uses a big magnet, tight surround and tight spider is designing
for a low system Q, good damping, and good transient response.
In other words, when you hit it with a sharp attack note, it responds
and then stops very quickly rather than ringing on and on. This
speaker would work well in an open back. However, in a sealed or
closed back, the air mass spring we discussed earlier would add to the
damping, potentially causing the speaker to be overdamped, compressed,
and lifeless. It's a difficult design task to design a speaker that
is a good compromise for all shapes, sizes and types of cabinets.
From: Bill Whicker
I saw a ceramic magnet speaker the other day that had two ceramic magnets
stacked together rather than the usual single magnet. Does that mean
the magnet is twice as strong or is there some other reason for that?
Thanks.
Bill, that sure looks impressive, doesn't it? In order to get more bass
you either have to go to a larger cone or you have to move the cone farther
in and out. This is because the radiation impedance of air
is very low at low frequencies. If you stick your hand in a tub of
water and move it forward and backward very slowly, you'll notice the
water doesn't resist your hand movement, nor do you create much of a wave.
So, you either need a bigger hand, or you need to move your hand farther
with each stroke. The same thing happens in a speaker. With a small
speaker such as a 6" or 8" you typically see connected to long tubes
in car audio for subwoofer applications, the cone has to move alot to
reproduce the very low frequencies. This means the cone has to move as
much as one inch in and out. A typical guitar amp speaker only moves about
1/8" in comparison. So, in order for the voice coil in the small speaker
to move that much, there has to be alot of room between the voice
coil and the back plate. Unfortunately, stacking ceramic magnets, as
impressive as it looks, only yields about 20-25% more magnetism than a
single magnet. This is due in part to the extended pole required and the
loss of flux associated with it. The bottom line, then, is that the
main purpose for stacking magnets is to facilitate extended cone movement.
From: John Zwern
I want to build a cabinet for an amp head I bought and was wondering about
the various speaker configurations, etc. For instance, if I use two
ten inch speakers, is that the equivalent of one twenty inch speaker?
I guess what I'm asking is what is the relationship between one
speaker and several speakers connected in parallel?
John, historically, musical instrument amps have used a single full range
speaker rather than multiple, specific range speakers with crossover networks
such as that used in HIFI installations. In order to reproduce the
high frequencies, the musical instrument speaker must have a fairly
low mass (weight) cone system so it can vibrate fast enough and with
sufficient movement to reproduce the high frequencies with adequate volume.
Connecting two of these speakers in parallel will increase the output at the
low frequencies thereby emulating a larger speaker, while retaining
the low mass properties of each speaker. Since the cone is round or
circular in shape, the math used to describe the effective area of
the speaker involves pi (3.14) and the square of the radius of the
cone. At low frequencies, where the entire cone is moving in and out
as one piece, the effective area of the two speakers in parallel is
approximately equal to the square root of the number of speakers times
the diameter of one of the speakers. So, two ten inch speakers in parallel
would be approximately equivalent to 10 x 1.414 (the square root of two)
or a single 14 inch speaker for the low frequencies. So, in theory, you would
get the low frequency response of what you would expect from a big 14 inch
speaker, while retaining the high frequency response of the lower mass ten
inch speakers. An important point we need to make here is that the
low frequency response is determined by both the size of the cone
and the movement of the cone. So, it requires more power to reproduce
the signal as the frequency goes down. This is particularly true below
the resonant frequency of the speaker, and the output falls quite rapidly
below that resonant frequency. If the 14" speaker we calculated had a
lower resonant frequency than the 10" speakers, which in all
probability would, it will have a better bass response than the two 10"
inch speakers, despite what we calculated. As far as comparing relative
or calculated equivalent sizes, I emphasized that this equivalency
would occur at low frequencies where the cone is moving in and out as one
piece, because at higher frequencies, say above 1Khz, the cone
vibrates in sections. At low frequencies, the sound from the speaker
covers a very broad, even pattern in front of the speaker cabinet. At
higher frequencies, this pattern is reduced to smaller, narrow
patterns call lobes. The collision of these smaller lobes from
one speaker with the other one add texture and a certain amount of
comb filtering to the overall sound coming from the cabinet. This is
why some players hear what they describe as a kind of reverb, phase
shifting, almost stereo sound coming from a multiple speaker cabinet
as opposed to a single speaker cabinet.
From: Chris
I have picked up some Trapezoidal enclosures (made out of Birch
plywood). for sound reinforcement. I am interested in reloading them
with JBL speakers. The bottom end boxes had two 18"speakers (HxWxD -
48"x24"x24")
The top boxes had two 15" speakers and a horn. (HxWxD - 42"x24"x24")
(Cabinets are 12" across back). What is the best way to ensure that the
cabinet has the correct number of ports and size? Any suggestions on
design of trapezoid enclosures would be appreciated. Thanks
Chris, that's quite a setup you have there! Let's go through some preliminary
information, then I'll point you to a website where you can get all
the math and other information you'll need. Trapezoidal or
any shape of cabinet other than square and rectangle really does serve
a purpose other than looking cool. In theory, an unusual shaped cabinet
like that has fewer parallel walls, so fewer standing waves are set
up that can affect the sound in an adverse way. As far as port and cabinet
dimensions go, you really have to know the technical parameters of
the loudspeaker(s) you are going to use so that you can tune the cabinet
and port(s) properly. While it is possible to experiment endlessly until
you are happy with the performance, it would be much better to calculate
it as close as you can, then make minor tweaks, just like an electronics
experimenter does. In guitar work, as opposed to HIFI work, we often
have the unusual arrangement of connecting several drivers or speakers
in parallel rather than using specific range speakers with
crossovers, etc. While this may seem to complicate matters, it really
doesn't. For the port, you can simply calculate for one speaker, then
add a port of the same dimensions for each of the other speakers,
or you can use only one port, increase its diameter by the square root
of the number of speakers connected in parallel and adjust the length
accordingly. Probably the easiest and fastest method would be to just
calculate for one and add one for each speaker. 4 speakers, 4 ports.
Here then, is a website you can go to that will lead you to more
information on cabinet building than you could ever imagine, including
pictures, the math, design philosophy, suggestions, parts sources,
etc.
Do It Yourself Audio Links
Best of luck to you with your cabinet project.
From: Larry Weissenborn
I'm interested in the differences between edgewound versus regular voice coils
and how this factor effects the sound of a speaker. I'm particularly
interested in this subject with regard to bass guitar speakers. I've heard
that edgewound coils generally handle more power but tend to sound more harsh
than conventionally wound coils. What are the pros and cons of each and
what's are the trade offs between power handling, frequency response, and
tone?
Larry, speaker design involves alot of physics theory because we are trying to
accelerate a mass (weight) in a short amount of time. This is
necessary to move the speaker cone in and out to reproduce high frequencies.
In the early days of speaker design, many attempts were made to build
full range speakers, rather than specific range speakers that are combined
with other speakers and networks to achieve the full frequency range.
Again, in order to get the full range, we have to make all of the components
in the speaker system very lightweight. Figures 1 and 4 below are examples of
strands of copper wire as they would look when wound on a voice coil former.
Notice the wasted spaces between the windings, due to the curvature of the
round copper wire. Figure 2 shows how we can eliminate most of the wasted
space by using square wire. If we take it one step further and flatten
the wire into a ribbon shape, we reduce the wasted space and also
add more current (and power) capability, as shown in Figures 3 and 5.
This also allows us to have more windings for a given coil length. Most
voice coils use copper wire, however, aluminum wire was used in
some ribbon or edgewound coils to save even more weight. Many of the
original edgewound coils were short coils, usually just the length of
the width of the front plate. These are called 'even hung' voice coils
since the ends of the coil are even with the edge of the front plate.
Although even hung coils exhibit more sensitivity, they run out of
coil in the gap during big excursions of the cone. The result is
square-waving of the mechanical signal (force) to the cone. In other words,
it just stops moving. If the cone is very lightweight and has high
top note efficiency, this square-waving can cause ringing and ghost
notes in the cone and it can sound very harsh.

From: Chuck
I have heard various views on impedance mismatches between the amp and
the speakers. One is that you should always match the impedance (4 ohm
amp = 4 ohm speaker or two 8 ohm speakers in parallel), or you can blow
your transformer. The other is that it is fine to mismatch, but you may
lose power. Should the impedence match? If yes, then how quickly could
you damage your amp when you have a mismatched impedence?
Chuck, technically, you should always provide a load that is recommended
by the manufacturer of the amp. The designer of the amp chose a particular
output device (tube) and specified all of the operating voltages for the
output stage so the tube would work at its optimum efficiency while
delivering maximum power to the load with minimum distortion. Ok, so
let's discuss the problems associated with mismatches.
When you use a load that is lower than the intended load, the output
has to drive the load (speaker) with more current because it is a lower
impedance than is expected. Two inherent problems associated with
transformers are flux leakage and regulation. Flux leakage is also referred
to as leakage inductance. It is related to the current in the secondary,
and these problems increase as the current increases. As the current
draw in the secondary increases, the primary has a more difficult time
transferring the signal to the secondary, so the secondary signal to the
load gets squashed, or 'soft-clipped'. This soft clipping is called
regulation. While regulation is desireable in a power supply, it is
undesireable in a transformer. In other words, in a power supply, if the
input voltage or the output load current changes, we don't want the
output voltage to change. In a transformer, we want the output voltage
to follow the input voltage and not regulate at all. When you put a heavier
load on the output than was intended, it will pull the output voltage down,
hence regulation. The leakage inductance problem arises because the
current from the heavier load causing the regulation to occur reduces
the efficiency of the transformer by not allowing the output to follow the
input. Transformer designers simulate or view this problem as having extra
inductance in series with the primary. The extension of this idea then,
is that with the heavier load, you could affect the efficiency of the
transformer, alter the frequency response (due to the extra leakage inductance
in series with the primary), and cause other distortions to occur.
OK, on to mismatching the other way. A speaker is a current operated device
in that it responds to the current through it to generate a magnetic field
that works against the magnetic field of the speaker magnet to make the
cone move in and out. Thinking in very short amounts of time, when the
output charges up the voice coil with current, then the signal goes away
or gets reduced, the cone system moves the voice coil back to its home
or resting position. As it is moving back, it generates a voltage that
is fed back up the line into the transformer and appears in the output
circuit of the amp. This generated voltage is often referred to as
flyback voltage, because we are charging up an inductor, then when
we disconnect or stop charging the inductor, the magnetic field in the
inductor collapses and induces this big voltage into itself. This big
voltage then 'flies back' to the source of the charging current.
There is a mathematical formula to determine how big
the voltage is and it is related to the inductance of the voice coil, the
amount of time it was fed current, and how much current it was charged with.
The bottom line is that the voltage fed back to the output circuit is
oftentimes much higher than the voltage that was used to drive or charge
up the voice coil initially. This voltage gets transformed up by the
turns ratio of the output transformer, and in many cases can be over 1,000
volts. What happens then is that arcing can occur between the pins on
the output tube socket. Once this has occured, a carbon path forms on the
tube socket between the pins. The carbon path allows a steady current
to flow between the pins and eventually burns up the socket due to the
heat that is generated. For example, it wouldn't be too uncommon to see
a transformer turns ratio of 30:1. If we had a voltage fed back from the
voice coil that was around 50 volts, 30 times 50 would be a 1,500 volt
spike at the plate of the output tube. This is why you often see designers
connect diodes in a string between the output tube plates and ground.
They are trying to suppress these spikes and dissipate the energy in the diodes
rather than allowing an arc to occur at the tube socket. So, when you
use a higher impedance load on a lower impedance tap, the turns ratio is
higher and resulting fed-back (flyback) voltage gets multiplied up higher than
what it would have been with the correct impedance load.
It's just about impossible for me to answer how long an amp would last under
these conditions. It all depends on how the designer took these potential
problems into account in his or her design with regards to the quality of the
tube sockets, the use of stringed diodes, the output circuit operating
voltages, etc.
From: Thomas Sherill
I hear alot of talk about the magnet cover making a big difference in
the overall tone of a speaker. Is that true, and if so, how much of
an effect does it have? Should I try to make magnet covers for
my speakers?
Tom, I'm walking on thin ice answering this one because it involves some beliefs that are closely held by many. A speaker is a dipole device, meaning the same amount of air is moved when the speaker cone moves back as when it is moved forward. So, since a stream of air is generated towards the back of the speaker as the cone moves back, it stands to reason that we would want the path to be as smooth as possible with the least amount of interference to the airflow as possible. Any interference would cause reflections of air back to the speaker which would affect the overall tone and character of the speaker tone. OK, now that we have the technical description of what's happening out of the way, is it relevant? No, not really. The undulations or variations in the surface of the magnet circuit would be the concern, but the aggregate of these undulations is insignificant compared to the slots and struts in the basket. Most magnet circuits are smaller in diameter than the plateau where the magnet circuit is mounted, further evidence that the effect is negligible. This kind of belief reminds me of the beliefs concerning paper versus other voice coil formers, and special kinds of wire compared to high grade standard copper wire. Technically, the mathematics is there to substantiate a claim, but the measured differences are out 5 and 6 decimal places, rendering the issue a nonissue to the average ear. Magnet covers, or pot covers are and always have been used for cosmetic reasons. The one exception is as I pointed out in an earlier Q&A question. Pot covers have recently been ressurected for use in surround sound systems to provide magnetic shielding.
From: Jim Chaverel
I've always wondered how the ports in a speaker cabinet work. Also,
how do they design them or decide how many to use?
Jim, the purpose of the port is to replace the bass that is lost from
the main speaker as its effeciency drops off at lower frequencies.
As mentioned in an earlier Q&A answer, in order to reproduce strong bass, we
either need a larger speaker, more movement of the cone, or both. Since
the bass output of the speaker is determined by both the movement of the cone and
the size of the speaker, you can see why the bass drops off quite rapidly as
the frequency goes down. OK, on to the design of the port. What we do is
size the speaker cabinet and the port so that the port will resonate at the
frequency below the resonant frequency of the speaker where the speaker output
power or loudness drops to half. So, we are taking the energy (in airflow)
from the back of the speaker cone, since it moves just as much as the front side,
and blowing it into the port. This is no different than blowing across the top of
a coke bottle. The idea then, is that this 'free energy' is put to use
by replacing the sound that is decreasing from the speaker. Of course, nothing's
free or perfect. Since the port is only tuned to one specific frequency, it has
a resonant frequency where it works best and provides the most help to the main speaker.
A problem associated with ports is due to the fact that we are blowing air through
the port. The result is wind noise due to the tumbling of the air through the port.
Designers refer to this as port viscosity. What it means is that the walls of the port
are a different resistance to the air movement as the air in the middle of the port.
So, tumbling of the air occurs. One way to cure this is to use multiple ports to
split up the airflow and divide it among the ports. Many designers of multi-speaker
cabinets use one port for each speaker. Some use as many as four ports for a large
base woofer. Although porting has shown up in guitar amp speaker cabinets over the
years, it has been used mainly in HIFI systems. Lately, some of the more
popular specialty guitar amp manufacturers are offering ported cabinets and
are receiving good reports from customers.
Alot of information about ports and port design can be found at:
Do It Yourself Audio Page
From: Mitch Cornell
I have several old Fender Champ amps from the 50's and 60's. I thought I had
the speaker impedance thing all figured out until I saw that you offer
8" speakers in both 3.2 and 4 ohm. I always thought they were all 4 ohm,
and the 3.2 was either a misprint or someone put the DC resistance on the label instead
of the impedance rating. What's the deal, and what's the difference between the two?
Mitch, pretty confusing, eh? I'm not sure where the best starting point is,
so I'll just lay out the facts and hope it makes sense. First and foremost,
speakers for guitar amp applications have always been designed by subjective
analysis rather than objective. In other words, tone. A typical guitar amp
speaker has a short voice coil so it will be very sensitive, and a thin, paper
cone so it will be very lightweight and have a wide frequency response.
The efficiency (loudness) is both for marketing of the amp and the fact that
we need high sensitivity for it to be a wide-range speaker since it is all by
itself, i.e. no help from bass woofers and high frequency tweeters like the setups used in
HIFI speaker systems. So, we typically use what's referred to as an even hung
voice coil. This means that the length of the voice coil is the same as the width
of the front plate of the magnet circuit as shown in the diagram below:

From Tim Nickels:
Why aren't more speaker frames made out of Aluminum?
Tim, there are several reasons why stamped aluminum frames, or baskets, are not used.
1. Aluminum is softer than steel, so a stamped aluminum basket could bend or twist causing a voice coil rub.
2. Steel is much less expensive than aluminum.
3. The weight of a typical steel basket, or frame, is insignificant compared to the weight of the magnet circuit. Therefore, there wouldn't really be a weight advantage using aluminum for a stamped basket.
Of course, there always seems to be an exception to the rule. Cast aluminum frames have been used over the years in
some of the higher priced speakers where lightweight but very rigid baskets were necessary because large and heavy magnet circuits were used with very tight voice coil gaps.
The advantage here is that the aluminum casting can be quite complex and beefy for the added strength where a similar steel casting would be too heavy. Note these are the high end speakers such as EV, JBL, and others.
The implementation ratio of stamped steel baskets to cast aluminum baskets is huge.
From Ralph Nichols:
This may seem like a "non question" to one looking for good sound but, I
am really curious to know if a standard has ever been established regarding the
polarity of the magnet in a loudspeaker. In other words, does the "north pole" of
the magnet always face the front or cone part of the speaker while the "south pole"
always faces the rear of the speaker housing, or does this vary with manufacturers
or, perhaps even models and sizes of speakers?
Ralph, that's a good question. I suspect at some point in the early years of speaker development there was a standard, but it has never stuck. Virtually all custom speaker manufactures conform to the customers wishes rather than their own, so for whatever reason, the customers have flipped and flopped over the years both in the polarity and the actual marking of the plus terminal. This could have something to do with the customers assembly line installation of the speaker harness, or just about any reason, really. Speakers that are/were intended to be used in a single speaker installation, such as in an elevator, etc., for background music may not be marked at all. In fact, I saw some identical speakers going down two separate lines at a speaker company once, and noticing the line equipment was slightly different, I checked out the magnetizers and final test stations. Sure enough, the two identical speakers were magnetized oppositely. I asked my host about it, and learned about the doesn't matter application. Of course, I already understood that in a single speaker application such as a guitar amp using a single 10 or 12" speaker, the polarity doesn't matter as far as hooking it up, but I always thought manufacturers would at least magnetize everything in one direction. I also asked, as you have, and looked in several books that I have found on the subject and have never been able to find a specific agreed-upon standard for polarity, either at the magnet N/S position, or at the terminals with the polarity indication. Out of curiousity, I have run a bunch of tests with different speakers and amps to determine if I could hear a difference, knowing that the real physics of it says there shouldn't be. I couldn't hear a difference in any of the setups, and I even had a person who I couldn't see flipping the switch so I couldn't know which way it was while I was listening. I must admit, though, I was biased by the fact that the physics of it says there should not be a difference. Some people say they can definitely hear a difference, though, particulary when they have their favorite guitar, effects, and amp in the equation.

