M-S SPEAKERSYSTEM

BY REIDWOODBURY

Published
inSpeaker Builder  magazine ONE/89.

Youcan
add this design to your “gee whiz” stack of novelty speakers (Photo1). It is based on
the M-S (middle-side) stereo mic’ing technique, which is popular formany
reasons, but few people seem to understand how this methodrealizes stereo
signals.

The secret with thesingular stereo
speaker I designed is that the front driver receives the sum of the left and
right channels, and the difference is sent tothe rear drivers, which are wired
out-of-phase, like the M-S microphone (see “Theory"). My original belief was
that the leftand right channel program material would hover to the left and
right of the speaker, but no such luck. I’ve discovered it’s similar to
Lauridsen’s method (Fig.1) for producing
pseudo-stereofrom a mono source1which my speaker
does with some success. However, my single speakerdoes have an incredible sense
of depth.

The M-S is my first and only speaker
construction project and I continue to startle friends with the sound.

ORIGINAL DESIGN. I came up with thedesign in
1983, reflecting that I was rarely positioned front and center while listening
to my stereo system, but either working ons ome project, homework, cleaning or
cooking, and not positioned optimally to hear the stereo image, often being in
the next room. My design produces the same acoustic energy as standard stereo,
and since I am generally economical, I figured I shouldn’t pay formore speakers
than I absolutely needed. For normal stereo I’d just wear headphones.

I started this project just prior to
subscribing to Speaker Builder. I’d never built any kind of furniture and
most of the electronic projects I’d ever finished were kits. I modified my
original design as my reading and building experience progressed.

DRIVER SELECTION. I chose driversonly by specs,
reputation, and by listening to similar drivers. These weren’t my first choices,
and in some cases the third or fourth choice, due to availability at the
time.

The “front”drivers use a standard
three-way design with a Thiele/Small aligned enclosure. The recommended
enclosure specifications were supplied by the retailer (McGee Radio &
Electronics), which I found very useful and fit closely with Bullock and
White’s BOXRESPONSE program.2,3

I chose a 10-inch woofer (Eminence EM40WOAE) for my
design because Ithought it was the best compromise between full-sounding and
tight bass. I chose a dome midrange (Peerless PMT51) for its smooth sound and
wide dispersion, and a leaf tweeter (Foster E110T08) for itsclarity.4

The crossover is a stock
second-order design, rated at 150W, with crossover frequencies of 500Hz and
7kHz. I chose a wide midrange bandwidth because the PMT51 has a much greater
power rating (120W) than the other drivers do (85W and 50W,
respectively).

For the rear drivers, I tried
full-range Philips AD5061/M8, but they were not of high enough quality, power,
or sensitivity. An album of Carmina Burana, with its lush chorus, has so
much out-of-phase material I was getting distortion from the amp trying to drive
them, and the small speakers were bottoming out. I replaced them with two of the
same dome midrange drivers PMT51) I used for the front.

In my enthusiasm while choosing the
drivers, I’d somehow lost the idea of keeping the cost down. I was learning how
much is involved to make a good sounding speaker an excellent sounding speaker.
Changing the rear drivers and adding adjustable L-pads to the more efficient
front midrange and tweeter were two major changes.

ELECTRONIC CIRCUIT. I naively tried to come up
with a passive design to achieve the sum and difference, but this would have
caused a tremendous signal loss. The active circuit I designed (Fig.2) is simple and
anyone with alittle knowledge of op amps could design one. The op amp can be any
good quality device that is unity-gain stable or stabilized with proper
compensation. The design works well and parts are readily available (all can be
found at a Radio Shack).

Basic Circuit. I added my original
circuit to a parametric equalizer kit I’d built some time before. I chose the
TL084 quad op amp because it’s similar to the TL074s that are used in therest of
the circuit. Not very scientific, but this saved me the trouble of building a
power supply and enclosure for such a simple circuit, and I knew this supply
would work. I’ve since discovered many types of IC op amps with faster and
quieterspecs.5

I used a pre-made RadioShack IC
experimenter’s board (Photo2 and
Fig.3) for the project. The“difference” circuit in the Photo has
about 10dB of gain (330k½ feedback resistor on IC1c) because the rear drivers
on my speaker are wired in series. I suggest you wire in parallel and keep the
gain at unity as in the schematic, depending on the load your own power amp can
handle. This is one likely cause of the amp’s clipping distortion when I was
using the AD5061s.

The circuit can be powered by any
power supply giving ±6–18v, or by two orfour 9v batteries. D’Appolito’s power
supply is suitable also (SB 4/88, p. 20).6Be sure not to
connect power to the ICs backwards, or this will fry them.

Use metal film resistorsfor lowest
noise. I originally used carbon resistors, but I did hand match the values of
the resistors around each op-amp to 0.1%. The circuit adds noise to my system,
but it is below all my program material sources (LPs, FM stereo, my own
reel-to-reel, and digital masters).

The circuit associated with IC2 can
be eliminated altogether if you know the output of the device feeding it, and
the connecting lines are short enough (lessthan 2 feet). Inverting amplifiers
should be isolated from the “outside world” by a non-inverting buffer (IC2c,
IC2d; Fig.2). In the schematic, the section within the dotted line is shown
in the photo, the rest of thecircuit is on the original EQ board. The input
buffers are identicalto the ones on the EQ board. I picked up the signal after
the inputbuffer on the EQ board and returned it to the same point after thecut
in the trace. If you’re unsure where the output of the sum/difference circuit
will be connected, include the 100½ resistorsand 1µF capacitor.

BASSIC CONCERNS. An advantage of this
speaker is the summed version of the bass is the actual program material (see
“LP Mastering’"). Turntable rumble tends to be out-of-phase, which cancels
itself in the front driver and remains in the rear where it can’t be heard. This
does eat up your system headroom, though, and should be filtered out. I use the
matrix circuit in a tape loop, instead of between the pre-amp and power amp,
with the left channel as the front and the right as the rear. This allows me to
use the right channel bass control to cut the out-of-phase turntable rumble and
control the balance between the front and rear drivers.

In the original circuit,I added two
parallel .03µF capacitors to the output of the difference amp (0.06µF to help
roll off the bass. Calculations show the circuit to be 3dB down at 27Hz when
feeding a 100k½ load.The resistance and capacitance can be slightly smaller,
depending on the desired low rolloff.

CONSTRUCTION. The speaker cabinet isconstructed
of 3/4-inch particleboard (Fig.4). I’ve included
the entire blueprint so you can compare my actual design with my description of
the sound. I attached the corners with Elmer’s glue and counter-sunk cabinet
screws. I added cleats later, after tests with the woofer installed indicated
air leaks. I lined the interiors with fiberglass. I mounted all the drivers with
bathroom silicone sealant.

I designed a square port because the
area required would have been an unusual diameter.I’ve since learned the port
can be almost any diameter with a corresponding change in port length.7

The upper chambers were for the
original 5-inch full-range drivers.

Initially I thought the speaker
would sound best in a corner. This improved the bass response but gave the
ambient rear signal a constrained character. The“stereo” image is much wider
against a flat wall. I chose a30° angle for the drivers so they wouldn’t be
pointed directly into the walls of a corner or, conversely, be too close tothe
line of sight on the sides mounted perpendicular to the front drivers. I
soldered all the electrical connections within the enclosure with 16-gauge
stranded wire.

LISTENING IN. This speaker seems to work best
with any source material that has major program material coming equally and
in-phase from the left and right channels (panned center). This includes M-S
mic’ing, stereo trio mic’ing, or pop music where leads and bass are panned hard
center, and backgrounds and effects are left and right. This creates a halo of
sound around the center, letting the more expensive drivers handle the
important part of the sound. Changing the rear drivers to PMT51s subjectively
improves the overall frequency response and seems to give a more even sense of
depth, possibly because the front and rear drivers are more closely
matched.

I suggest you use a dome tweeter rather than
the E110T08. The ribbon’s dispersion is so narrow you must be positioned
directly in front to really hear it. Otherwise it is a very nice driver with
aclean, clear sound.

One problem with playback of
recordings made with wider spaced microphones: more out-of-phasematerial is
present (see “Theory"). When the level is balanced between the front and rear,
the front drivers will sound weak. If thefront level is increased, the image
sounds restrained.

Even though the direct sound from
the rear drivers wraps around directly to the front, a greater spaciousness is
apparent when the speaker sits farther from a wall or corner. Besides the
initial direct sound, you hear reflections from more distant walls and surfaces
in the room.

THEORY. I prefer minimalmic’ing, like a stereo
pair with no highlight mikes, forrecording classical ensembles. I’ve also heard
of jazz bandsthat are recorded with just a stereo trio (left-center-right,
LCR).Even with minimal stereo mic’ing, differences in sound can begreat,
particularly recordings of classical music in a concert hail,caused by different
stereo mic’ing techniques, but differentmusicians and halls add variables,
also.

In designing sound for live theater
(sound effects, music and live mic’ing), I know speakers and their placement
must fit the mic’ing techniques used to record the effects or music. I’ve seen
SBarticles on driver and speaker design, but nothing on the program
material they reproduce. Let’s review the many different ways sound is recorded,
and presented in the home; you can adapt my speaker design, or any other design,
accordingly.

STEREO MIC’ING. Two audio cues help us perceive
direction between two loudspeakers: relative intensity, which is the most
significant and relative arrival time. The sound will appear to come completely
from the right speaker when the intensity of the right channel is 20dB greater
or the arrival time is 2msec sooner than the left. This time difference is also
known as the precedence or Haas effect.8
Lowering the level in the right speaker about 8–10dB, while delaying the
left up to about 25ms will keep the stereo image centered, but will broaden the
image between the speakers proportional to the amount of delay.9 Beyond 25ms you
start to hear a distinct echo.

Different stereomic’ing techniques,
such as X-Y, ORTF,10
Blumlein,11 and
M-S, use coincident (one microphone directly on top of the other), slightly
spaced, and widely spaced pairs; and trios.

Slightly spaced pairs use both time
delay (the Haas effect) and level differences to give localization. Widely
spaced pairs generally use omni-directional (equal pickup in all directions)
mikes, therefore only the Haas effect produces localization. The time delay is
created by the distance between the two mikes, most apparent with sound coming
from the sides.

Coincident mic’inguses only level
differences for localization because there is no delay between mikes for sound
traveling from the sides.

X-Y generally refers to a coincident or
near-coincident stereo pair of mikes, with one panned hard left and the other
panned hard right, which can be applied to all these techniques except M-S
(which uses a simple electronic matrix to derive the left and right
channels).

Music in a studio is generally
recorded with a large number of single mikes and each is assigned a position in
the stereo image by panning or adjusting only the relative level between the two
channels. Signal delay is use donly as an effect, that is, reverberation, echo,
flanging, and chorusing.

Middle-Side. TheM-S method
uses a coincident pair, however one is a cardioid pick uppattern mike pointed
directly toward the ensemble (middle), and the other is a perpendicular
(side-to-side) figure-eight pickup pattern mike (Figs. 5a and 5b); the
positive lobe is usually pointed to the left.

The electronic matrix isa simple sum
and difference, unity gain mixer. The sum of the twomikes is panned hard left
and the difference is panned hard right. Varying the level between the two
microphones allows you to adjustthe width of the stereo image (Figs.5c–f). The points where the
microphone pickups are equal, or where their pattern lines cross, indicate where
a musician would be positioned to be heard from theleft or right speaker. The
negative rear lobes which result from combining the two mike signals, put some
reverberation out-of-phase in the stereo signal, so the reverberation seems to
come from outside the normal stereo stage between the speakers.

The closer the microphone elements
are matched, the clearer the stereo image; the best resultsare obtained with
stereo microphone where the elements are factory matched.

Inverting the phase of one
microphone will cause the left and right signals to exchange outputs. This is
handy if you accidentally place the figure-eight mike backward or hang the
stereo mike upside down.

M-S stereo (or other coincident
techniques) is also perfectly mono compatible. When you add left and right
channels together, the figure-eight sides cancel,leaving only the middle
cardioid mike signal, as if only one mike was used. Commercial FM radio also
uses this matrix to broadcast in stereo and be mono compatible. M-S recordings
are more likely to broadcast unscathed.

My M-S speaker matrix circuit can
also be used for the M-S microphone matrix. The signal from the microphones must
go through an appropriate mike pre-amp before going through this matrix.
This design is too noisy for professional applications, though. If there’s a
demand I’ll make a more esoteric design available, but the corresponding parts
may be hard to find. In addition, I suggest you read Walt Jung’sAudio IC
Op-Amp Applications.12

X-Y mic’ing can sometimes suffer
from a “hole in the middle” effect. Performers situated at the center of the
microphones are off-axis from both, and the recording sounds different than the
live performance. One cure for this hole is to add a third, center mike with its
signal going equally to both the left and right channels.

M-S mic’ingdoesn’t suffer from this
problem; the middle element is pointed toward the center. Both M-S and stereo
trio mic’ing have the advantage of three on-axis directions, giving a much more
pleasant recording of the concert hall sound. While M-S requires only two
microphones or one stereo mike, it does require the electronic matrix. Even
though Blumlein’s method has four on-axis directions, a soloist is still
off-axis from both microphones. Interestingly, performers tend to prefer M-S,
while engineers prefer ORTF.13

LPMASTERING. I believe you can’t detect the direction of
asignal source below about 200Hz, except by moving around andlistening closer
and farther away from the source. The wavelength is long enough to go around the
head and affect both ears equally. The same effect occurs when recording with
coincident or near-coincident microphones. Also, excessive out-of-phase bass,
which on LPs could cause the stylus to skip out of the groove, is filtered out
during disc mastering, using an elliptical equalizer and leaving only
thein-phase bass.

Left and right (lateral) stylus
motion is center program material, and up and down (vertical) motion is
out-of-phase or ambient material. The popular Neumann system has switchable
turnover frequencies of 150Hz and 300Hz.14, 15

Even on CDs, because of the mic’ing
techniques and LP engineering habits, the low-frequency information tends to be
the same in both channels. In regular stereo (or P.A. setups, for that matter),
out-of-phase bass will sound weak for the same amount of program power. Still, I
prefer stereo using two woofers. Recordings are likely to be monitored and
adjusted with two equally full-range speakers and then checked on one
full-range speaker for mono compatibility, rather than on an odd stereo setup
with different frequencies getting a different number of speakers. Boosting the
bass of the front driver of my M-S speaker at all frequencies below that of the
rear drivers (about 300Hz with the PMT51s) takes care of this consideration, but
is not a perfect solution.

TV STEREO. Many movies are broadcastor released
on video tape with the original surround encoding of a Dolby stereo optical
print. The two-channel recording is decoded to four channels (Dolby started by
using a technique similar to the Sansui QS quadraphonic matrix16) but, instead of
speakers in the four corners of a room, the new format specifies three speakers
across the front-left, center, and right; the surround channel is the
fourth.

Soundtracks must pinpoint
information across the front and keep the center material in the center because
phantom center material of two-channel stereo drifts toward you if you sit at
the sides of a listening space. The closest speaker is louder and its sound
arrives sooner. This is not as big a problem in the home, though dialogue and
effects in stereo television shows are kept fairly close to center in the stereo
field to keep the sound of the on-screen action from running off the small
screen. The 70mm, six-track stereo format does not use this
encode/decodeprocess.

When a theatrical show is mixed, the
dialogue program material is always given to the center channel; never panned
from speaker to speaker. Through psychoacoustics (the study of how people
perceive sound), movement, even following the characters across a screen, was
found to be distracting. Panning is only used for an effect, On the set, a
shotgun or other narrow pattern microphone records the action and dialogue. If
a stereo recorder is used, one track gets the main mix and the other is used for
backup-rarely a stereo mix. Replaced dialogue is recorded mono and effects are
recorded mono and stereo. Room tone and other atmospherics may be recorded in
stereo after the main “take.” On the simplest stereo shows, only the musicis
stereo.

The surround channel alsohas a very
limited bandwidth, 500Hz–7kHz.17 (Matching these
frequencies with the front crossover of my speakerwas unintentional.) Surround
information is kept to a minimum for compatibility with mono-only theater
systems. Any bass information comes only from the front speakers. The 7kHz
low-pass filter keeps distracting optical noise out of the surrounds. You can
design your own home surround speakers with a limited bandwidth, also.

The simplest surround decoder takes
the left and right recorded signals (left total and right total) and sends them
to the left and right speakers; makes a sum and sends it to the center speaker;
and takes the difference signal and sends that to the surrounds. At the other
end of the spectrum, professional theater decoders use a complex phase
detection, steering circuitry and surround-channel delay to increase separation
between speakers. Home units fall somewhere in-between. Interestingly, M-S
recorded material is easily encoded and decoded through the Dolby process. The
middle mike is sent to the center speaker, matrixed left and right to the left
and right speakers, and the surrounds are fed by the figure-eight mike (which
picks up mostly reverberation).

This mix type works well with my M-S
speaker. It’s great for television or stereo video, since sound effects tend to
run off the picture with a normal stereo setup. For movie theater systems,
recordings are mixed with the left and right speaker behind the screen, not on
each side. With the M-S speaker, the action takes place in the center, and the
music and atmospherics take place all around. As with the M-S microphone, the
M-S speaker has a direct center element so center program material has an
actual driver and not a phantom combination of other elements to produce the
sound.

My 13-inch color television sits
comfortably on my speaker, but the magnets in the speaker cause the color to
change. The sky turns red, grass turns yellow, and faces turn green. I stacked
three large phone books under the television and this works fine. If you’re
going to add this speaker to your television, be sure to include magnetic
shielding.

OTHER IDEAS. The left and right signals don’t
recombine on the left and right sides of the speaker because of many
inaccuracies, such as diffraction problems from the recessed front panel, and
drivers not vertically aligned. I’ve come up with a few possible cures you might
try, also, the best application I’ve thought of for this
design—a“ghetto-blaster."’

Front Delay. Mark Rumreich’s electronic
time delay project (SB 3/88) would be perfect to try on this
speaker.18 I
agree with his philosophy that sound from each driver must reach the listener at
the same time for the best imaging. Try to delay the front signal so it will
reach you at the same time as the rear signal, for my design. The amount of
delay may depend on the distance between your front and rear drivers or on the
length of the sound’s path from the rear drivers, reflecting off a wall, to the
plane of the front drivers. If you align the front signal to match the
reflections of a speaker sitting far from the wall, the delay time must stretch
into milliseconds.

To calculate rough delay times for
most situations, I use as a rule of thumb one foot per millisecond; thus
estimating sound travel at 1000ft./sec. Sound actually travels a little faster:
in dry air at 32°F (0°C), 1087ft/s (331m/s); and at 68°F (20°C), 1130ft/s
(343m/s).19 This is about
73.7–76.3 µsec/in.

Driver Considerations. You
could vertically align the front drivers bymoving the midrange, high-frequency
and rear drivers to the top of awoofer cabinet in a triangle configuration, or
three sides of asquare. This brings them to a near coincident position without
adding delay.

Wall reflections from therear
drivers may be more desirable than sound that wraps around from the sides by
diffraction. Cone midrange drivers rather than dome types may work better for
this. Also, cabinet designs to isolate the direct sound from the rear drivers
may be necessary.

I thought about using electrostatic
panels in a “T” configuration (an ES-T?) The top or bottom of the “T” could be
pointed toward the listener. Adding delay to the front driver would make the
drivers effectively coincident. Could changing the delay of whichever panel is
in front affect how the phantom left and right channels are beamed into the
listening room?20 Should the back
be covered by sound absorbing material?21

I no longer have access to a wood
shop to try these ideas so I’m turning my many questions over to you. I’ve
started with my design and the theory behind it, and you can experiment to your
heart’s delight. Have fun.

NOTE. Please contact the author concerning
licensing arrangements for commercial manufacture.(How arrogant! RAW 00/02/21)

ABOUT THE AUTHOR

Reid Woodbury is a freelance audio-for-video
(sweetening) engineer in Los Angeles, working with sound for TV and film, and
has done sound designs for various live theaters in the US. He graduated from
the University of Missouri at Kansas City Conservatory of Music and also studied
two years of electrical engineering. He recorded recitals, concerts and radio
programs for the Missouri Repertory Theater and the Conservatory’s recording
department, and currently runs his own location classical recording
service.

FIGURE1: Lauridsen’s
method for pseudo-stereo (from Sound Recording, p. 74).

PHOTO1: Beauty shot of
finished speaker.

PARTS LIST

2  TL084/74s
12 100k½
resistors
4  100½ resistors
4  0.1µF ceramic
capacitors
3  1.0µF nonpolar electrolytic capacitors

2  0.03µF ceramic capacitors
1  switch, DPDT

1  circuit board, Radio Shack #276-159

Misc.: hardware, screws, nuts, spacers,
bipolar power supply or batteries, box, RCA input and output connectors.

FIGURE2: Sum and
difference circuit with low rolloff. Labels in parentheses mark connections for
M-S microphone.

PHOTO2: Finished board
installed in EQ enclosure with an unstuffed board, foil side. Signal hook-up
wires are two-conductor shielded (left, right, and ground).

FIGURE3: M-S circuit
pattern and component layout. Resistors are rectangular and capacitors areoval.
All resistors are 100k except whore marked.

FIGURE4: Blueprint of
enclosure. Panels were cut from 3/4-inch particle board.

FIGURE5: Cardioid (M)
andfigure-eight (S) microphone pickup patterns and their resulting matrixed
patterns (L and R). Drawings are on a linear scale. The dotted arrow is the
“on-axis” direction of the right matrix output. The solid arrow is the
performer’s physical direction from the mikes, to be heard only from the right
speaker in standard stereo.

1. Eargle, John, Sound
Recording, Van Nostrand Reinhold Co., New York, 1980, pp. 73–79.

2. Bullock, Robert M., and
Bob White, “BoxResponse,” SB 1/84,pp. 13–18, 42.

3. I took the Bullock and
White program and typed it into a powerful spreadsheet program that has built-in
graphing (Excel for the Macintosh). I’ve spent many hours subtly changing a
speaker enclosure’s parameters with a particular driver and studying the numbers
and frequency response graphs. I modified the input of the program, i.e.,
instead of entering the compliance/box volume ratio, I enter just the box
volume and the spreadsheet calculates the ratio. I enlarged my enclosure
slightly and I always use the 30Hz rumble filter in my pre-amp as the program
suggests.

4. McGee Radio&
Electronics Corp. Catalog, Kansas City, MO, 1983.

5. Pennington, Terry, and
Larry Winter, “Understanding Circuit Principles,” Recording
Engineer/Producer, March 1987, pp. 60–65.

6. D’Appolito, Joseph and
James W. Bock, “The Swan IV SpeakerSystem,” SB 4/88, pp. 9–21.

7. Bullock, Robert M., and
Bob White, “BoxResponse,” SB 1/84, pp. 13–18, 42.

8. db: The Sound
Engineering Magazine Dec. 1979, pp. 42–46.

9. Eargle, John, Sound
Recording, Van Nostrand Reinhold Co., New York, 1980, pp. 41–42.

10. ORTF is a microphone
setup specified by the French Broadcasting Organization where two cardioid mikes
are at a 110° angle and 17cm apart.

11. Blumlein or
“stereosonic” mic’ing uses two coincident figure-eight mikes at a 90°
angle.

12. Jung, Walt, Audio IC
Op-Amp Applications, 3rd ed., Howard W. Sams & Co., Indianapolis, IN,
1987.

13. Cross, Lowell,
“…Assessments of Studio Microphones,” Recording Engineer/Producer April
1984, pp. 120-129, and Dec. 1985, pp. 70–85.

14. Runstein, Robert E.,
Modern Recording Techniques, Howard W. Sams& Co., Inc., Indianapolis,
IN, 1974, p. 300.

15. Eargle, John, Sound
Recording, Van Nostrand Reinhold Co., New York, 1980, p. 298.

16. Blake, Larry, “Mixing
Dolby Stereo Film Sound,” Recording Engineer/Producer, Feb. 1981.

17. Blake, Larry, “Mixing
Techniques for Dolby Stereo Film and Video Releases,” Recording
Engineer/Producer, June 1985, pp. 94–107.

18. Rumreich, Mark,
“Electronic Time Delay Line for Speakers,” SB3/88, pp. 15–21, 67.

19. Halliday, David, and
Robert Resnick, Fundamentals of Physics, John Wiley& Sons Inc., New
York, 1981, p. 319.

20. Williamson, Reg, “The
Quad 63,” SB 1/82, pp. 10–18.

21. Cox, Thomas, “Spot
Sound Absorption,” SB 3/88, pp. 40–43.

ACKNOWLEDGMENTS

Thanks to Ray Taylor for the photography, my
mom for makin’ me do good English, and the University of Missouri at Kansas City
Conservatory of Music Recording Department for the space to try out my crazy
ideas.