Loudspeakers and Rooms - Working Together

by
Floyd E. Toole, Ph.D.
Vice President Engineering, Harman International Industries, Inc.
Stereophonic and multichannel surround sound systems are encode/decode processes in which the loudspeakers and the listening
room are dominant factors. Together, they influence timbre, dynamic range, and directional and spatial effects - in other words, almost
everything that matters to critical listeners. The sounds arriving at the ears are the only information that the auditory system has to work
with. If the sounds are different in different circumstances, the perceptions will be different.
The only real solution is to control these variations and, ultimately, to standardize the important factors. Problem is that there
are no industry standards for either loudspeakers or rooms. In this paper we will attempt to make the best of this imperfect system by
identifying the major variables in the loudspeaker/room system, and discussing methods for their measurement and control.
1 THE OBJECTIVES
Normally, we strive for some semblance of realism, whatever that may be. The relationship between the perceived sound
qualities in recordings and those in live performances is close, but imperfect. In a domestic room, practical factors make it very difficult
to create truly convincing impressions of being, for example, in a concert hall.
Some of the difficulty stems from the restrictions of traditional two channel stereo systems. Only those listeners on the axis of
symmetry of the loudspeakers can hear the full benefits of stereo, and the effects diminish with distance. Conventional stereo recording
techniques are the result of trial and error, attempting to make the best of a system which cannot recreate all of the directional
impressions that may have been a part of an original live performance. The directional diversity of sounds arriving at listeners' ears is
much reduced compared to any live experience.
As a result, strict realism is an impossible objective, so we try to approach it as closely as possible, within the limitations of
our hardware. For the bulk of recorded music, however, realism is an inappropriate objective. In popular music, for example, the
original "performance" occurs in a recording control room at the time of the final mixdown. Since recording studios are not
standardized, unless you happened to be there, you dont know what the original was.
Attempting to bring a little more spatial flavor to the reproduction process, loudspeakers are available in directivities ranging
from conventional front-firing, through bipole (bidirectional in phase), dipole (bidirectional out-of-phase), predominantly-reflecting, to
omnidirectional. These present listeners with very different combinations of direct and reflected sounds, and in most of them the room
is a major determinant. Stereo, therefore, is not really a system at all but, rather, a basis for individual experimentation.
Multichannel systems offer a partial solution in that there are more channels, and therefore more directions from which sounds
can be made to appear. This also brings a certain independence from room acoustics, since there are more real, as opposed to
reflected sources. Still, multi-directional, including dipole, speakers rely on reflections, so there are instances in which one must
attend to the details.
The standardization of at least some of the more obvious variables by the film industry has helped enormously in achieving
some consistency in multichannel film sound tracks. Consequently, what one hears in a cinema is similar to what was heard in the
dubbing stage where the sound track was assembled. Good engineering practices and the Home THX program attempt to continue that
relationship into the home.
It will be interesting to see how multichannel music develops.
1.1 Cause and Effect in Loudspeaker/Room Combinations
Accurate sound reproduction from a loudspeaker is largely dependent on the absence of linear (frequency and phase response) and
non-linear (harmonic and intermodulation) distortions, and on maintaining a constant directivity across the frequency range. In a
reflection-free environment, directivity would be unimportant, as one would listen to the sound radiated along a single axis, preferably
the best one. In real rooms, sounds radiated in most directions from loudspeakers eventually reach the listeners.
The directional properties of the loudspeaker, the physical arrangement and the acoustical properties of the listening room
determine the spectrum, amplitude, directional and temporal factors of the multitude of sounds arriving at listeners' ears. All of these
sounds combine and interact physically, at the entrance to the ears, and perceptually, in the auditory systems and minds of listeners. As
a result, virtually every perceptual aspect of stereo reproduction can be affected.
Variations in the perceived spatial representation, or imaging:
(a) the size of specific images of voices or instruments are changed by reflected sounds, especially those in the horizontal plane.
(b) the positions (lateral or in depth) of specific images are changed by reflected sound, and
(c) the sense of spaciousness or envelopment are influenced by reflected sounds, again mainly those occurring in the horizontal plane.
Variations in sound quality, or timbre caused by:
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(d) the acoustical coupling of sound through the standing-wave system (the room resonances or modes that are related to its
dimensions),
(e) listeners positioned on different direct-sound axes from the loudspeakers receiving different initial sounds,
(f) acoustical interference (comb filtering) occurring when the direct sound and one or more strong early reflections combine at the
listener's ears,
(g) frequency-dependent variations in sound absorption in room boundaries and furnishings modifying the spectra of reflected sounds,
and thereby of the integrated sound field at the listener's ears,
(h) strong reflections of inferior-quality off-axis sound from loudspeakers degrading the spectrum of the integrated sound field ( a
variation of the previous item),
(i) perceptual "amplification" of non-delayed resonances by reflections and reverberation (some sounds are more audible in a complex
sound field), and,
(j) perceptual "attenuation" of delayed sounds by reflections and reverberation (some sounds are less audible in a complex sound field).
In short, the physical characteristics of loudspeakers and listening rooms can alter virtually all of the perceived qualities of
sound that are held as being fundamental to satisfactory sound reproduction.
2. THE PHYSICAL VARIABLES
Although there are substantial areas of overlap, as will become clear, for purposes of explanation it is simpler if we separate the
room variables into the following categories:
1. Dimensions and proportions.
2. Loudspeaker and listener positions.
3. Sound absorption and reflection.
2.1 DIMENSIONS AND PROPORTIONS
Rooms have acoustic resonances, or modes. The proportions of a room, length to width to height, determine the distribution of
room modes in the frequency domain, whether there are clusters or gaps in the distribution. The dimensions themselves determine the
frequencies at which the resonances occur, whether specific musically important frequencies will be reinforced, or not. In perfectly
rectangular rooms, with perfectly flat and perfectly reflecting boundaries, these resonances can be easily calculated, using the well-known
formula:

f= c
2

nx 2

ny 2

nz 2

Lx

Ly

Lz

where: f = frequency of the nth mode

nx, ny, nz = integers with independently chosen values between 0 and, say, 4.
lx, ly, lz = dimensions of the enclosure (length, width and height) in meters (feet)
c = speed of sound = 345 m/s (1130 feet/s) at room temperature.
To calculate all of the modes of a room it is necessary to work through all possible combinations on integers (whole numbers)
for nx, ny and nz. In practice only the low-frequency modes need to be calculated; up to n = 4 is usually sufficient.
Individual modes are identified by the integers nx, ny, and nz. For example, (1,0,0) identifies the first order mode along the x
dimension of the room, (0,2,0) identifies the second order mode along the y dimension of the room, and so on. If two of the three
numbers are zeros, the equation reduces to a simple calculation of the modal frequency for the standing waves set up between a pair of
opposing reflecting surfaces along one of the room dimensions. These are the axial modes of the room, normally the most energetic of
all room resonances, and the easiest to calculate. If you do nothing else, have a look at these axial modes for each of the length, width
and height of the room.

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Calculating the Axial Modes

Calculate the modes along each of the
room dimensions, length, width and
height, using the following simple
equation:
1130 2 x room dimension = frequency
For example: a room 20 feet long has a
first length mode at 1130 40 = 28 Hz.
The other modes are 2x, 3x, 4x this. So
a 20 foot long room has resonances at
28 Hz, 56 Hz, 84 Hz, 112 Hz, and so on.

Tangential modes occur when sound is reflected among four

surfaces, traveling parallel to the remaining two. These modes are
calculated by combining two integers and one zero. For example, (1,1,0)
identifies the first order mode in the x-y plane. These standing waves
involve the four walls and run parallel to the ceiling and floor.
Oblique modes interact with all room boundaries. Numerous
reflections occur in each "circuit" of the room, and because some sound
energy is lost at each one, these are the least energetic of all room modes.
They are calculated by combining all possible combinations of three
integers.

2.1.1 THE "IDEAL' ROOM

It has long been assumed that a uniform distribution of room modes in the frequency domain is a good thing. Concentrations
of modes could cause artificial accentuation of certain frequencies, and gaps in the modal distribution could cause some frequencies to be
less audible.
Over the years, there have been suggestions of various room dimension ratios, length-to width-to height, that offer superior
modal distributions [1,2]. These and other studies have not always acknowledged three problems of real listening spaces that render the
predictions unreliable.

First - the calculations assume that the room is perfectly rectangular, and constructed of perfectly flat, perfectly reflecting surfaces.
Reality is rarely so simple, since rooms commonly have irregular shapes, large surfaces that absorb sound (they vibrate),
furnishings, etc. These departures from the theoretical ideal result in errors in the calculated frequencies.

Second - All of the modes are not equally important. In general, the axial modes are the dominant factors. Assessments of rooms
should, therefore, incorporate a weighting in which the axial, tangential, and oblique modes are considered in that order of
importance
Third - The positions of the sound sources and listeners in practical stereo listening arrangements do not result in uniform acoustical
coupling to the room modes. As a result, the loudspeakers do not supply energy uniformly to all of the existing modes, and the
listeners are not in positions to hear the effects of even those modes that are energized.

These complications mean that, in practical situations,

predictive schemes may be helpful, but not perfectly satisfying. Insitu measurements may be the only method of determining exactly
what is happening.

MICROPHONE

Figure 1 Placing a loudspeaker on the floor in a corner will ensure

that all of the low-order modes are energized (all modes have a
high-pressure region at any intersection of the three room surfaces
- wall, floor and ceiling). Placing a microphone at an opposite
corner (floor or ceiling) will ensure that it responds to all modes.
Obviously, the loudspeaker must be a closed-box or vented-box
design (a pressure source) and the microphone must be an
omnidirectional design (a pressure detector) with good low-

frequency response.

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020
002
300

200

010

100

Figure 2. This plot of sound pressure

vs. frequency is really only useful for
identifying the frequencies of the
strongest modes in the room. It is not
uncommon to find that the measured
frequencies are not exactly the same as
those that can be calculated. This is
because the room is a real one, not an
idealized one.
Note the substantial
acoustical gain at the resonant
frequencies.

001

30
20

THE MOST OBVIOUS MODES

ARE USUALLY AXIAL MODES

dB
10
0

-10
As a means of determining how
20 50 100
500 1K
5K 10K 20K
a room might perform in a realistic stereo
FREQUENCY (Hz)
or multichannel application, however, this
measurement is useless.
Moving the
loudspeaker away from the corner will change the way it energizes the modes and moving the microphone will change the modes it
responds to.
However, if the room is basically rectangular, it is still useful to calculate the modal frequencies in order to avoid obviously
troublesome dimensional combinations in new constructions, and to identify the problematic modes in existing rooms. Figure 3 shows
the modal distributions for a room which is likely to present problems, and another one that might be more successful. Notice that, in
the "bad" room, there are really two problems, the frequency distribution of the modes is uneven, and the same combination of modes
appear in repeated patterns. The second room is an improvement in both respects.
These simple back of an envelope calculations are an essential beginning to analyzing a room setup. Of course, if the room is
dramatically non-rectangular, this cannot be done, and life just became much more complicated.

H:W:L = 1:1.5:2

H
W
L

H
L W L W L
0

100

H
W
L

H
L W L W L
200

H
L W L

300

400

FREQUENCY (Hz)

H:W:L = 1:1.5:2.4
L W H LW HWL
0

100

H
W L

WH L WH L W HLW

200

300

400

FREQUENCY (Hz)

Figure 3 (top) Axial modes calculate for a rectangular room. "L" identifies a length mode, "W" a width mode, and "H" a height mode.
The dimensions are simple multiples of each other and, as a consequence, there are regularly repeated combinations of the same
axial modes, and similarly regular gaps. This room is likely to have some problems.
Figure 3 (bottom) Axial modes for a room in which the dimensions have been adjusted to produce a more favorable distribution.
Because it is slightly larger, the axial modes begin at a lower frequency and are closely spaced. Because of the irregular ratio of
length-to width-to -height, the modes do not coincide at all at very low frequencies and, at higher frequencies, they combine in
differing combinations.

Some have argued that there are advantages to using non-rectangular rooms. The thinking behind this is that, by reflecting
sounds in directions other that directly back towards a parallel surface, the development of standing waves will be inhibited, and
diffusion will be increased. The reality is that sloping the room boundaries has a substantial effect on the modal structure, but the
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modes do not disappear. In fact, the magnitudes of the variations in sound pressure throughout the room remain much the same, but the
frequencies of the various modes are changed in an unsystematic manner, and the nodal lines are repositioned in a manner that is not
always obvious [3,4]. Simple predictions of the kind just discussed are no longer possible, and one must resort to finite-element
analysis or models in order to predict what might happen in practice. This is a serious disadvantage in some situations, as will be seen,
for example, in Section 2.2.
In other applications, like reverberation rooms for acoustical measurements, the advantages outweigh the disadvantages. If the
modal mixing generated by non-parallel surfaces is judged to be desirable, it is interesting to note that it may not be necessary to angle
all of the surfaces. There appear to be substantial gains from the sloping of just one room boundary [4].
2.2 Loudspeaker and Listener Positions
Two factors are involved in determining the effects of loudspeaker and listener positions at low frequencies:
1. interactions with adjacent room boundaries, and
2. interactions with room modes.
Although such considerations are frequently included in discussions of loudspeaker placement, it is less common, but just as
important, that they be considered in choosing listener locations.

H:W:L = 1:1.5:2.4
L
0

WH

H L

100

W L

WH L WH L W HLW

200

300

400

FREQUENCY (Hz)
Figure 4 In this diagram the size of the mode identifiers has been adjusted to reflect their audible importance to a listener in a
realistic stereo system setup.

The discussion of the subject is in two parts, but it should be emphasized that they are by no means independent. Perhaps the
simplest way to conceptually distinguish between the two, is to think of "adjacent boundary" problems as those involving only the
direct sound and the first reflections from nearby room surfaces. Room modes develop as a result of multiple reflections which include
these surfaces, as well as others in the room.
2.2.1 Interactions with Adjacent Room Boundaries
This is a topic that has been carefully examined by Allison [5,6], Waterhouse [7,8] and Waterhouse and Cook [9]. The effects
are a major determinant in the low-frequency performance of loudspeakers in listening rooms. Allison's papers illustrate well the
magnitudes of the effects, as does Figure 5, in a
30
dramatic fashion.
Figure 5.
The bottom curve shows a
loudspeaker measured in a true free field, with
no reflecting surfaces, such as a very good
20
anechoic chamber, or outdoors far away from
dB
any large object, including the ground.
In
such a situation the sound radiates into a full
10
sphere solid angle, identified as 4 steradians.
Placing the loudspeaker on a floor reduces the
solid angle by half, and the sound pressure at
0
low frequencies will be increased by
approximately 6 dB because sound that would
have propagated away will be reflected back.
Leaving it on the floor and backing it up
-10
20 50 100
500 1K
5K 10K 20K against a wall reduces the solid angle by two
FREQUENCY (Hz)
factors of two, to steradians, and it can be
seen at the left of the curves that the sound
pressure is elevated by about 12 dB. Sliding the loudspeaker into a corner reduces the solid angle into which the sound radiates by

4 STERADIANS
STERADIANS
STERADIANS

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a further factor of two, to /2 steradians, and the pressure goes up by another 6 dB for a total low-frequency gain of about 18 dB!
With respect to amplifier power and stress and strain on the loudspeaker, these acoustic gains are absolutely free, so wise people
take advantage of them. In rooms with normally flexible boundaries, the gains will be slightly less, but still substantial.

An unfortunate side effect is that some irregularity is introduced at upper bass and lower mid frequencies. If one is using a
separate subwoofer, this is not likely to be a problem. However, for free-standing full-way loudspeakers finding a suitable compromise
between good imaging and good bass can be a frustrating, and frequently impossible, task. There is no solution to this but to
experiment with position [11,12]. This is probably the single most persuasive argument for subwoofers.
As a means of reducing the variables, a few manufacturers have integrated the floor and/or rear wall into the loudspeaker design.
This restricts the choice for loudspeaker locations in the room, but it reduces the chance of a serious degradation of sound quality due to
unfortunate placement.
Treatments of this subject have tended to use either sound power or sound pressure level as the measured dimension. It is
correct to use either, of course, and both are useful in different situations. A good explanation of the relationship between sound power
and sound pressure (as well as sound intensity) can be found in Section 1.4.3 of reference 13. In assessing the audible effects of these
quantities, it is most directly relevant to use sound pressure levels, as the ears respond to sound pressure, and the psychoacoustical
relationships are expressed in those terms.
Reducing by a factor of two the solid angle into which a loudspeaker radiates, can result in an increase in sound pressure of a
factor of two, or 6 dB, measured at a particular location. This is entirely consistent with the notion of the sound power radiated by the
loudspeaker into the reduced solid angle being increased by a
factor of two, or 3 dB.

Wave Effects
ACOUSTICAL
INTERFERENCE

STANDING WAVES

Figure 6. An illustration of wave effects, those phenomena

that exist because sound propagates as a pressure wave. In
this simplistic sketch, the upper illustration shows the direct
and early reflected sounds arriving at a listeners ears. The
lower portion of the sketch shows stylized sound pressure
standing waves between the front and back walls of the room.
Mode 1,0,0 has one pressure minimum at the half-way point
along the room, and mode 2,0,0 has two minima.

In the preceding discussion of solid angles, it was shown that at long wavelengths (low frequencies) sounds reflected from
adjacent boundaries added constructively because they arrived at the measuring/listening point substantially in step with each other. At
higher frequencies, this will not always happen and, as a result, there will be frequencies at which the sounds add (constructive
interference) and frequencies at which they subtract (destructive interference), depending on the path lengths involved. In order for this to
happen, of course, both the direct and the reflected sounds must be simultaneously present. In such situations, we can measure a familiar
effect, known as comb filtering, because of the tooth-like regularity of the repeating destructive interference cancellations.

Comb Filtering - a Steady-State Effect

MEASUREMENT

TIME

FREQUENCY

Figure 7. A sequence of two transient acoustical events,

e.g. a direct and reflected sound, viewed from the
perspective of a measuring system which assumes steadystate conditions, and from the perspective of the ear, which
knows the difference and, in addition, has the advantage of
forward temporal masking to attenuate the apparent
loudness of the second transient. Perceptually the events
may not be as audibly disturbing as the rather nasty-looking
measurement would indicate. For long-duration sounds, the
combing is, of course, real, and the perception and the
measurement are more similar.

PERCEPTION

The standing waves in Figure 6 show the pressure

distributions across the length of the room at those
frequencies at which the room is precisely one-half
wavelength (mode 1,0,0) and precisely one wavelength (mode 2,0,0) long. Note that the first minimum from each wall is always onequarter wavelength from the reflecting surface. Note also that the instantaneous pressure on opposite sides of a pressure minimum (null)
has the opposite polarity. That means that as the pressure on one side is increasing, the pressure on the other side is decreasing.
Remember this, it will be useful.
TIME

2.2.2 Interaction with Room Modes

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Rooms in homes are usually rectangular. However, that is where the similarity ends in most cases. Nothing is standardized,
and the differences in detailed shape, placement of doors, archways, windows, large pieces of furniture, etc. ensure that every situation
will have its own distinctive problems.
At low frequencies it is frequently possible to be quite accurately analytical about what is happening in a room, but the standing-wave
structure at higher frequencies is usually lost in confusion. The
35 4
following simple examples should be helpful.
AXIAL (LENGTH)
MODE 1,0,0

Figure 8. A view of a listening room showing the distribution

of sound pressure and particle velocity for the first-order
mode, 1,0,0, along the length of the room.

f = 1130 = 16 Hz
2 x 35.3

Notice in this figure that the pressure is highest at the

reflecting surfaces - the point at which the change in direction
occurs - and that the particle velocity is at a minimum at the
+
SOUND PRESSURE
reflecting surface - where the air molecules are literally up
against the wall. The loudspeaker shown is a conventional
closed or vented box device, and therefore is a pressure source.
Such a device will acoustically couple to a mode when it is
PARTICLE VELOCITY
positioned in a high pressure region of the standing wave
pattern. In this figure, the loudspeaker will couple with near
maximum effectiveness. The listeners ears are also well
positioned to hear this mode, but they are not at the highest pressure position, which is probably a good thing, given the acoustical gain
that a resonance provides. If the listener were moved forward into the pressure minimum (null), the coupling would be minimal, and the
listener would simply not hear this frequency, even though the loudspeaker is functioning properly and there is an abundance of 16 Hz
energy in the room.
As a matter of interest, if the loudspeaker were a dipole design, it would be a velocity source, not a pressure source, and
therefore would couple most effectively to the mode if it were located at the velocity maximum at the center of the room. This means
that any room arrangement that is found to be flattering to this type of loudspeaker will probably not work at all well for conventional
35 4
loudspeakers - at least at low frequencies.
AXIAL (LENGTH)
MODE 2,0,0
f = f 1,0,0 x 2 = 32 Hz

SOUND PRESSURE

PARTICLE VELOCITY

20 0

AXIAL (WIDTH)
MODE 0,1,0
f = 1130 = 28 Hz
2 x 20

SOUND PRESSURE

PARTICLE VELOCITY

Figure 9. A view of a listening room showing the distribution

of sound pressure and particle velocity for the second-order
mode, 2,0,0, along the length of the room

At this frequency, the listener is in a pressure

minimum, and therefore will not hear 32 Hz. Obviously, if the
customer wants to hear the rumbling organ pedal note at the
beginning of 2001 (R. Strauss: Also Sprach Zarathustra), he
or she will be seriously disappointed. Fortunately, a moment
of analysis reveals that a small movement of the chair, forward
or back, can solve the problem. In fact, it is generally a good
idea to calculate and plot out the pressure distributions for the
first two or three modes, and locate chairs to avoid any of the
nulls.
Figure 10.
A view of a listening room showing the
distribution of sound pressure and particle velocity for the
first-order mode, 0,1,0, across the width of the room

In this sketch, we can see a problem and a solution at

the same time. The problem is that the listener is at a null in
this, and every odd-order modal frequency along this room
axis (i.e.28 Hz, 84 Hz, 140 Hz, etc.). The solution is that the
two woofers of the loudspeakers are each located in a different
lobe of the standing wave pattern.
Now, at low frequencies it is usual for the signals to
be mono (the same in both channels). This is absolutely true
of LPs, where failure to combine the bass would cause the
stylus to be thrown vertically out of the record groove. In

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general it is good practice because of mono compatibility issues. In home theater, there is a dedicated subwoofer channel, so there is no
option. In such a case, so long as there is one woofer in each of the left- and right-hand lobes, this particular mode will not be
energized. The reason is that the woofers are operating in-phase, and the lobes need to be driven out-of-phase because of their opposite
20 0
polarity.
Figure 11. A view of a listening room showing the distribution
of sound pressure and particle velocity for the second-order
mode, 0,2,0 across the width of the room.

AXIAL (WIDTH)
MODE 0,2,0
f = f 0,1,0 x 2 = 56 Hz

SOUND PRESSURE

PARTICLE VELOCITY

10

AXIAL (HEIGHT)
MODE 0,0,1

Just as we were feeling comfortable, here comes a

situation to spoil our fun. At the second-order mode frequency,
the listener is in a high pressure region, and the woofers are
both in lobes of the same polarity. This mode will therefore be
very effectively driven and heard. A solution would be to move
the woofers closer together, towards the nulls. This will, of
course reduce the width of the stereo soundstage, which may
not be welcomed. Here is another example where having
separate subwoofer(s) would be advantageous - they can be
positioned for best bass, and the satellites can be positioned for
best imaging.
Figure 12.
A view of a listening room showing the
distribution of sound pressure and particle velocity for the
first-order mode, 0,0,1, across the height of the room.

f = 1130 = 56 Hz
2 x 10

In two-channel stereo there is a stereo seat, a sweet spot where the soundstage comes into focus. Multichannel systems can
entertain several listeners, meaning that, at least at low frequencies, everyone hears something different because of the way they couple
into the horizontal room modes. However, in the vertical direction, everyone is treated equally. If there is a choice, it is probably wise
to choose a ceiling height that does not place the ears at the half-way point.

The Damping of Room Modes

Depends mainly on the acoustical absorption of
the major room surfaces: floor, walls, ceiling.

Figure 13 An illustration of how room boundaries that have some

low-frequency absorption, i.e. flexure, can improve the uniformity
of sound distribution at the modal frequencies. This is especially
useful in home theater applications where several listeners need to
have a similar auditory experience.

REFLECTING, i.e.
MASSIVE AND RIGID,
WALLS

Acoustical absorption by the room boundaries remove

low-frequency energy from the sound field. As was mentioned in
discussions of solid angles, this loss of energy to room structure,
and thus to adjacent rooms, reduces sound power that is contained
WALLS WITH SOME
LOW-FREQUENCY
in the listening room.
While this could be viewed as a
SOUND ABSORPTION,
P
disadvantage, making woofers work harder, it can also be viewed as
i.e. FLEXURE
a benevolent acoustical factor. This absorption reduces the Q of the
room modes, resulting in lower maxima and higher minimal tones are described by the stylized curves in Figure 13. In rooms, the nulls
are not perfect. Absorption also controls reverberation at low frequencies, providing damping for room "booms". Diaphragmatic, or
membrane absorption in room boundaries is one of the few practical mechanisms acoustical absorption at very low frequencies. Most
popular add-on acoustical devices cease to be effective absorbers at the very low frequencies where room resonances can be problematic.
P

2.2.3 Dealing with the Problem

Clearly, preferences in the bass balance of loudspeakers can be influenced by the choice of listening and loudspeaker positions in
the same listening room. Strong timbral variations are also consequences of these differences. The listening room can even dominate
the overall impression [14,15]. Since the basic physical mechanisms can, in many cases, be traced to combinations of room resonances
and adjacent-boundary interactions, some amount of control is possible.
There are few choices:
Modify the acoustical coupling of the loudspeakers to the room boundaries and/or the room modes. i.e. move the loudspeakers.
Modify the acoustical coupling of the listener to the room boundaries and/or the room modes, i.e. move the listener.
Move both the loudspeakers and the listener.
Modify the signal on the way to the loudspeakers, in the loudspeakers themselves, i.e. equalize.
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Acoustically modify the room, i.e. get out the hammers and saws.
Each of these options has restrictions and negative side effects, visual, acoustical and economical.

2.2.4 A "Real-world" Example

There is nothing more real than one's own personal world and I shall use as an example a serious problem that I had to deal
with in one of my own sound systems.
The example relates to a large living/dining room designed to be a fitting acoustical complement to mainly classical music
(another room was equipped for home theater and rock n roll). To that end this room had a large volume (about 220 m3), a high
cathedral ceiling, and irregular surfaces for good diffusion. The loudspeakers were at one end of the room, and the listeners sat about 7.5
m away, close to the other end. With good symphonic recordings, the effect was not unlike sitting well back in a concert hall. The
listening room seemed to become an extension to the recorded ambiance, without being such a contrast in scale that it competed with it.
With close miked, or multi-track recordings, the room provided much of the acoustical setting. It was very enjoyable . . . but.
30

MODE 2,0,0

Figure 14. A measurement from loudspeakers to listening

position in a very large listening room.

0.5 m

20
dB
10

14 dB!

0
-10

20

50 100
500 1K
5K
FREQUENCY (Hz)

10K 20K

RESOLUTION:
25 Hz, 40 ms

Figure 15.
A
In this waterfall
- i.e. each point
In
the
time

The problem in the large room was excessive low bass

- too much of a good thing. The 14 dB excess at 40 Hz had
absolutely nothing to do with the loudspeakers, which were
known to be very well behaved. Subjectively, the bass was
wrong from any point of view. Sustained organ pedal and bass
guitar notes were truly overwhelming when they moved into the
range of the resonance. Impulsive sounds, like kick drums,
were indistinct and overly resonant.
three-dimensional view of the situation depicted by Figure 14.
display resolution is limited to 25 Hz in the frequency domain
on each curve represents an average over a spread of 25 Hz.
domain, the resolution is 40 ms.

The curve
of Figure 14 tells us that there is a powerful resonance
around 40 Hz.
The height and width of the bump reveal that it is a medium-Q
resonance, and this
means that it will ring significantly. It is not necessary to look
into
the
time
domain to confirm this. However, since we can, lets do it.
Figure 15
shows the very impressive and pretty waterfall
diagram
of
amplitude vs. both time and frequency. This looks only at the
low frequencies of ORIGINAL LISTENER POSITION
Figure 14, and it shows what would be expected: a resonant
peak around 40 Hz
and a decaying resonant tail at that frequency. It doesnt show
it in full detail, because it is a characteristic of such displays that there are compromises in resolution. For example, the steady-state
measurement in Figure 14 shows the amplitude vs. frequency response very accurately. In the display of Figure 15, the peak shown at
the back of the family of curves should be identical but, due to the limited resolution, the peak is very considerably blunted. Likewise
the time sequence of events is blurred. Nevertheless, the point is made that, if there is evidence of a resonant peak in the frequency
response, then there will be ringing in the time domain. This is a law.
A quick calculation suggested that the problem was an overly active second-order mode along the length of the room. For
practical reasons it was not possible to relocate the loudspeakers, so a change in listening position was explored. Originally, the listener
was about 0.5 m from the back wall. Moving the listener forward progressively reduced the bass excess until, at about 2 m from the rear
wall, the problem was acceptably subdued. At 2 m the ears are closer to the 1/4 wavelength null in the standing wave pattern at 40 Hz.
The solid angle that the listeners' ears "see" is also increased,
+
+
and the low bass is correspondingly attenuated.

2.0 m / 80

Figure 16. A plan view of the room showing the pressure

distribution of the second-order length mode (2,0,0), and the
relocated listening position.

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Harman International Industries, Inc. 8500 Balboa Blvd., Northridge, CA 91329 (818) 893 8411

PAGE 9

30
20

Figure 17. A comparison of before and after measurements

showing the dramatic improvement resulting from moving the
listener out of the powerful high-pressure region of the second
order length mode in the room.

0.5 m

dB
10

2.0 m

0
-10

20

50 100
500 1K
5K
FREQUENCY (Hz)

10K 20K

Figure 18. A waterfall plot of the situation after listener

relocation.

In Figure 17 it can clearly be seen that immediately

after TIME=0, moving towards the front of the diagram, the
sound level in the vicinity of 40 Hz drops by about 12 dB.
After this, the ringing continues as before, but at a much
reduced level. All of this is logical because, in moving the
listener, we have not eliminated the resonance, but we have
reduced the strength of the acoustical coupling to it.
Subjectively, the improvement was nothing less than dramatic. Organ pedal notes walking down the scale were wonderfully
even and powerful. Kick drums were distinct and conveyed dramatic punch. String-bass notes were discrete and tuneful, having been
relieved of the one-note bias of the powerful 40 Hz resonance.
All, it seems, was well. However, we live in a practical world, and the constraints of living and interior decor dictated that the
listening chair could not live in the middle of the living room floor. Moving the loudspeakers would have equally solved the problem,
but for similar reasons, that was not an option. What do we do? We reach for our trusty parametric equalizer, dial in the correct center
frequency, Bandwidth (Q), and attenuation and, behold, the
30
frequency response problem is again fixed.
Original Condition
After Parametric
20
Equalization: One
Figure 19. Measurements of the system with the listener in
dB
Filter Only
the original location, but with one filter of parametric
10
equalization used to eliminate the objectionable resonance.

0
-10

20

50 100
500 1K
5K
FREQUENCY (Hz)

10K 20K

Although this method appears to eliminate the resonance as well as did the listener relocation, it should be noted that it did
nothing else. Relocating the listener to eliminate the problem with one room mode, put the listener in a different relationship with other
modes - compare the previous curves (Figure 17) in the 80 - 200 Hz region. Thus the two solutions will not sound exactly the same,
although there should be a similar improvement in terms of taming the 40 Hz room resonance. Let us check a waterfall diagram to
confirm this.

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Figure 20. A waterfall plot of the situation after parametric

equalization.

A comparison of the corrected frequency responses in

Figures 17 and 19, and those of the corrected waterfall plots
in Figures 18 and 20, indicate more similarities than
differences.
Both methods appear to have successfully
addressed the problem of the run-away room mode. But . . .
how does it sound?
Which one wins - natural acoustics
(relocation) or electronics?
After extensive back and forth switching between the listener relocation and equalization cures, the conclusion was that they
both worked, and they worked about equally well. Much of the time it was difficult to differentiate one from the other. However, there
were times when a preference was evident. In these cases, it was the equalized system that won. Why? Because with the equalizer in
circuit the woofer distortion was reduced. It was, after all, working far less, since the amplifier drive had been reduced by more that a
factor of ten at low frequencies. In general, though, listeners were struck more by similarities than by differences. This is good,
because now we have several ways to solve problems of this kind.
Both of these solutions addressed the specific problem and the changes outside of the frequency range of interest were either
beneficial, or negligible. Other methods could have been used, absorption being one - damp the offensive mode. The problem with this
is that most kinds of acoustical absorbers tend to operate over quite wide bandwidths and, in addressing a problem in one narrow range
of frequencies, the devices will be removing energy at frequencies where there is no problem. In alleviating one difficulty there is a risk
of creating another.
2.3 ABSORPTION AND REFLECTION OF SOUND
It is impossible to talk about this subject without bringing into the conversation important details about the loudspeakers
themselves, specifically their directivities, and the nature of the system itself, specifically whether it is a stereo system or a multichannel
system, or both. If you are really picky, one can also take into account the musical tastes of the customer, classical or popular, in
designing the listening experience.
Some of these will come up in the following sections, but for now it is important to understand some fundamentals about the
materials and devices that are used in our business. A good book on the basics of this subject is Alton Everests The Master Handbook
of Acoustics 3rd Edition [16].

LATE REFLECTIONS

REVERBERATION
EARLY REFLECTIONS

Figure 27. Reflected sound in a room can be divided into three

general classes based on how long it takes to arrive at the listener
after a single reflection (early and late reflections), and the spatial
and temporal pattern of arrival after multiple reflections
(reverberation). Flutter echo is a special case in which sounds
reflect back and forth between two parallel opposing surfaces.
With transient sounds this causes a fluttering sound.

FLUTTER

2.3.1 RESISTIVE ABSORPTION

Resistive absorbers remove energy from sound by making it do work in moving the air molecules back and forth between and
among the closely-spaced fibers of fabrics and fiberglass, or through the openings of open-cell acoustic foam. Fabrics that you cannot
blow air through (e.g. good cotton sheeting) are useless, as are those (e.g. polyester double-knit - grille cloth) that are so open that you
can easily see through them. Rigid fiberglass board is an excellent absorber, until the surface is closed by painting or covering with the
wrong kind of fabric. Thus we arrive at the fundamental concept of flow resistance: to maximize the acoustic absorption, there is an
optimum resistance to air flow.
The resistive material must also be in the right location. Since the mechanism is flow resistance, these materials will obviously
be most effective when located where molecular motion is at a maximum. The Figures in Section 2.2.2 show sound pressure and
particle velocity distributions for standing waves in rooms. Suppose we wanted to try to damp an overly energetic axial mode using a
resistive absorber. Using the example shown in Figure 8, where would you place a 2-inch thick fiberglass panel to be most effective?
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PAGE 11

Attaching it to the wall would be useless, because the particle velocity - i.e. molecular motion - is minimal. Nothing is
moving much so nothing much happens. Moving it away from the wall improves things, until we get to the middle of the room, and
some serious damping occurs. Looking at the other low-order / low-frequency modes in these figures lead to similar conclusions. At
low frequencies it is simply not practical to use resistive absorbers. As wavelengths get shorter, the regions of maximum particle
velocity get sufficiently close to the reflecting surfaces that materials of practical thickness, or drapes hung at normal distances, will
work. Thus we come to a rule that resistive absorbers are the devices of choice for absorbing mid- and high-frequency sounds.
2.3.2 DIAPHRAGMATIC or MEMBRANE ABSORPTION
The most common membrane absorbers we encounter are the walls, floors and ceilings in rooms. The proof that they are
absorbers is that they vibrate in response to bass sounds; acoustical energy is converted into mechanical energy. As luck would have it,
conventional home construction, gypsum board on 2 x 4 studs, is an effective low-frequency absorber - play some loud music and feel
the walls. Double-glazed windows are quite similar in their absorption properties, so we are off to a good start. The worst possible
rooms are those built in basements, with concrete floor and walls. Such rooms need to have false walls built inside them. To improve
on normal construction, I suggest two layers of gypsum board on some (not necessarily all) surfaces, possibly with a layer of 1/2-inch
sound board between (glue the three layers together, using nails or screws only to hold while the adhesive sets).
Concrete floors are a problem for two reasons. First, they dont absorb sound. Second, because they dont absorb sound, they
dont vibrate and supply the tactile sensation of bass through the soles of the feet. In these cases, a false floor is a useful addition. Oh
yes, leather upholstered furniture is also a membrane absorber, and also provides vibratory sensations to stimulate the audience. H-m-mm.
Obviously, it is possible to construct customized absorbers to address problems at specific frequencies. Reference 16 has design
aids for diaphragmatic absorbers (p.172). Remember to place them at high-pressure points for the mode that is being damped.
2.3 DIFFUSORS, DIFFUSION AND CONFUSION
Diffusion refers to the degree of randomness in the directions of arrival of sounds at a point in space. Diffusion in the sound
field surrounding a listener in a concert hall is an absolutely essential requirement. It is of some use in home theater rooms. It is of
little use in stereo rooms.
In a concert hall, diffusion helps communicate all of the sound from all of the instruments on stage, to all of the listeners in the
audience. If there were total diffusion, listeners would not know where the sounds are coming from, so there must be a balance among
direct, reflected and diffuse sounds.
In a surround sound application, Dolby ProLogic requires low correlation in the sounds arriving at the ears from the surround
channel speakers in order to generate a sense of ambiguous spaciousness. In THX systems, electronic decorrelation in the signals sent to
the left and right surrounds helps, and bidirectional out-of-phase dipole surround speakers can be of further assistance. The latter is an
attempt to increase the diffusion in the sound field. Acoustically dead rooms work against that effort, and irregular reflective surfaces
and objects in the room work with it. Unfortunately, the front channels really need to see a relatively dead room, which creates a
dilemma that has not been satisfactorily solved. Dolby Digital / AC-3 and DTS multichannel music call for five identical channels and
loudspeakers, thus presumably calling for a relatively dead room. On the other hand, multichannel music might well be better in a
normally live room. Then there are systems like Lexicons Logic-7 and Citation 6-axis that go directly to the core issue - five channels
are not enough. They add channels in the rear of the room. And so it goes. This issue is not settled yet.
Conventional two-channel stereo is another real dilemma. Here loudspeakers range from highly unidirectional through
multidirectional to omnidirectional. Each category requires different room treatment and positions. Listener preferences cannot be
ignored. Some like a large spacious illusion (a relatively live room), others like pinpoint imaging (a relatively dead room).
A common compromise is to avoid the decision of whether to absorb or to reflect, and opt to diffuse the sound with any one of
the several available irregular reflecting devices. A diffuser is an elaborate reflector, in that it sends any incoming sound off in many
different directions. This is not a bad idea, and diffusers have an important place in the repertoire of acoustical devices. Just remember
that they are still reflectors, and the sound energy is merely redirected. Stereo is a system in which fragile phantom images are an
essential part of the illusion. I have been in rooms where many dollars were spent on diffusers which did little more than destroy the
stereo image that every other device in the signal path attempted to preserve. One can have too much of a good thing.
A simple test is to reproduce monophonic pink noise at equal levels through both loudspeakers. For a listener on the axis of
symmetry, the result should be a compact auditory image midway between the loudspeakers. Moving the head slightly to the left and
right should reveal a symmetrical brightening, as the acoustical-cross talk interference is changed, and the stereo axis should "lock in"
with great precision.
3 LOUDSPEAKERS
Obviously the design of a state-of-the-art audio system must include the listening room and the manner in which it is set up.
The loudspeaker is an essential part of that equation and it is a part which is frequently misunderstood. In the discussions up to this
point we have focused on low frequency communication between the loudspeakers and listeners. At middle and high frequencies
loudspeakers and rooms interact in different ways, and these need to be understood. Choosing the right loudspeaker can make life a lot
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Harman International Industries, Inc. 8500 Balboa Blvd., Northridge, CA 91329 (818) 893 8411

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easier when it comes to positioning it and treating the interior of the room. Understanding some key facts about loudspeakers in general
makes mistakes much less likely.
3.1 WHAT IS GOOD? LOUDSPEAKERS / REFLECTIONS / EQUALIZATION
Subjective evaluation is a massive subject in itself. Fortunately a lot of research work has been done and published, much of it
by the author [17,18,19]. Let me start by stating simply that, when listening evaluations are done under controlled circumstances - i.e.
identities concealed, loudness levels equalized, room, listening arrangements, and program material known and constant - strong
differences of opinion among people tend to disappear. Most people, most of the time tell us that they like the same sounds, and dislike
the same sounds, for mostly the same reasons.
Obviously, there are some individual differences. The most serious is that those listeners with hearing abnormalities are not
good listeners. Hearing loss results in inconsistent and aberrant opinions. Fortunately, at least 80 % of the population has hearing that
is essentially normal, in this context. Otherwise the main differentiating factor is experience. People who have never listened critically
have problems knowing what to listen for, and they may bring with them a lot of miscellaneous opinions - the baggage of having
listened to little else than mediocre televisions, boom boxes and car radios. Given this, it is amazing how little practice these people
require, in a reasonable context, before they settle down to having opinions that make sense. Knowing this, nowadays we try to work
with listeners we have trained, who are practiced in knowing what to listen for and how to evaluate and comment on what they hear [20].
Wine tasters train themselves to recognize tastes and smells, and rely on blind tasting to form trustworthy opinions. Why should audio
be different?
An important confirmation of why this is a reasonable thing to do comes from studies in which listener opinions have been
correlated
with
+10
physical
measurements.
The
10
dB
relationships
make
sense.
+10
9
-10
20

FIDELITY

8
7

+10

dB

50

100

200

500
1K
2K
5K
FREQUENCY (HZ)

10K

20K

dB

-10
20

50

100

200

500
1K
2K
5K
FREQUENCY (HZ)

10K

20K

5
-10
20

50

100

200

500
1K
2K
5K
FREQUENCY (HZ)

10K

20K

+10

dB

-10

Figure
21.
Representative
onaxis
frequency
response
measurements
of
loudspeakers
that
received
different
fidelity ratings in
scientificallycontrolled
listening
tests.

The
implications of this
kind of result are
0
enormous.
Firstly,
large
numbers
of
people agree on what they like. Secondly, the loudspeakers they like are the ones that have the most ideal , i.e. smooth and flat onaxis frequency response. However, there is more. Having only a good axial response is not enough. That this should be so, should not
be surprising, because we do not listen in anechoic chambers. Real rooms have reflecting boundaries and furnishings, which redirect offaxis sound radiated from loudspeakers back towards the listener, sometimes after several reflections. Since the total amount of energy
arriving at the ears is dominated by these reflected and reverberant sounds, it is reasonable that they should be timbrally similar to the
direct (axial) sound. This means that the loudspeaker must also have smooth off-axis behavior; in other words the loudspeaker should
have constant directivity.
20

50

100

200

500 1K
2K
5K
FREQUENCY (HZ)

10K

20K

10

8 woofer, 1 tweeter

Figure 22. Stylized on- and off-axis measurements on an 8-inch

two-way loudspeaker

0
dB
-10
-20

20

ON AXIS
30 DEGREES OFF AXIS
60 DEGREES OFF AXIS

50 100
500 1K
5K
FREQUENCY (Hz)

10K 20K

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In Figure 22 it can be seen that the 8-inch woofer becomes quite directional before it is crossed over to the 1-inch tweeter in the
region of 2-3 kHz which, in turn, becomes directional at high frequencies. The directivity of the loudspeaker is not constant meaning
that, even when the loudspeaker is aimed directly at the listener, the perfect performance on the design axis will be diluted by the
timbrally colored off-axis sounds heard after reflection within the room. This configuration is fundamentally flawed. It can never be as
good as well-designed systems employing a dedicated mid-range driver.
45 DEG. LEFT
DESIGN AXIS

30 DEG. RIGHT

10

8 woofer, 1 tweeter - parallel with rear wall

Figure 24. Two of the first sounds to arrive at a listener from

the loudspeaker described in Figure 23. Knowing this, we can
conclude that most of the remaining reflected sounds, arriving
later, will be even less good. This is a loudspeaker which
cannot sound neutral in a normal room, and which will take on
different timbral personalities, depending on position and the
acoustics of the listening space.

0
dB
-10
-20

10

Figure 23. A room layout in which the loudspeakers are facing

directly forward, as happens in many home stereos and custom
installations. In this case the best sound component misses
the listener completely and is lost in the general reverberant
sound field in the room. If the loudspeaker does not have
constant directivity, the system will take on a coloration related
to how the loudspeakers misbehave off axis.

20

0
dB
-10
-20
20

30 DEGREES OFF AXIS

45 DEGREES OFF AXIS

50 100

500 1K

5K

10K 20K

(Hz)
10 woofer,FREQUENCY
3 midrange, 0.5
tweeter

Figure 25.
The design objective for a room friendly
loudspeaker. Practical examples can come remarkably close
to these targets.

ON AXIS
30 DEGREES OFF AXIS
60 DEGREES OFF AXIS

50 100
500 1K
5K
FREQUENCY (Hz)

10K 20K

10
Figure 26. An expensive three-way loudspeaker that does
not meet the design objective. Obviously a lot of effort went
into obtaining a smooth and almost flat on-axis response.
However, it is equally evident that the designers thought little
about off axis behavior. The heavy curve at the top is a room
curve, measured at the listening location in a typical room. It
is obviously dominated by the off-axis misbehavior of the
loudspeaker. The loudspeaker sounds moderately colored.

0
dB
-10
-20

20

ROOM CURVE

50 100
500 1K
5K
FREQUENCY (Hz)

10K 20K

With respect to Figure 26, the question arises: can

we smooth out this room curve with an equalizer and end up
with better sound.? The answer is Probably not. For the reason, we must look back at Figure 21, showing the on-axis responses
favored by listeners - they were all quite definitely in the direction of smooth and flat. If we equalize the loudspeaker in Figure 26, we
will have destroyed the only good attribute it had - a smooth and flatish on-axis response. The cure, regrettably for owners of these
products, is a better loudspeaker.
3.2 SOME NOTES ABOUT MEASUREMENTS AND EQUALIZATION
The example in Figure 26 tells us that, without some key information about the loudspeaker, equalization at middle and high
frequencies is a high-risk venture. Sadly, the information one needs is not commonly available. In fact, some manufacturers dont even
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have it themselves and cant measure it accurately. If the loudspeaker is from a reputable source, it may simply be best to assume that
they did their job properly, and to avoid any form of equalization at frequencies above a few hundred Hz.
In fact, if the loudspeaker manufacturer has done a really competent job, there is nothing that can be measured in the listening
room that will enable any improvements to be made at middle and high frequencies. To some this may appear to be a bold assertion,
now that clever time-gated measurement systems (like
10
MLSSA and TEF) the are now available. To them I offer
True Level
this small example.

dB

Figure 27. Measurements made in a very large room with

the loudspeaker and microphone suspended close to the
middle of the space, giving an anechoic interval of 17 ms
CANNOT MEASURE WHAT WE HEAR
before the first reflection. Given the need to measure at a
20 50 100
500 1K
5K 10K 20K
distance of at least 6 feet, such a large interval is not
FREQUENCY (Hz)
likely to be possible in any domestic room. This time
Measurement of a Q=50 resonance at the threshold of window translates into a frequency resolution of about 60
Hz, and this in turn results in the inability of the measuring
audibility using a time window of 17 ms (60Hz)
system to reveal the presence of audible high-Q problems
at frequencies below about 3 kHz. In the test, the high-Q resonance was present at intervals all the way down to 20 Hz, but the
measurement system cannot see them all. Many manufacturers and most product reviewers function with measurements less
capable than this. This is one reason why anechoic chambers are not obsolete.

At low frequencies we know that we can, and sometimes must, equalize and, since we are looking for steady-state
measurements, time windowing is not necessary. In fact we want a long measurement interval. The tale told in Section 2.2.4 indicates
clearly that equalization can work very well. However, there are some cautions. Attenuation of excessive levels appears to be very safe,
but avoid trying to fill deep holes. A narrow dip is probably caused by a null in a standing wave or interference pattern. As such it is
the acoustical equivalent of a bottomless pit - it cannot be filled. Narrow dips are difficult to hear in any event, and all that will happen
if you dial in a lot of gain is that the amplifiers will have reduced headroom, and the loudspeakers will be working harder to no avail.
The result will be increased distortion.
Obviously, measurements should be made using proper instrumentation. This does not mean the classic fixed-frequency 1/3octave real-time analyzer with dancing lights and overlapping (i.e. cheap) filters. These things may be entertaining, but they are not
useful measuring instruments in this context. Nowadays there are several computer-based alternatives, such as MLSSA, LMS, JBL
SMAART, TEF which can do remarkably better things. Even the inexpensive systems are quite adequate.
At the basis of this is the need to be able to measure what we are able to hear. Studies show that listeners are very sensitive to
resonances - in loudspeakers themselves as well as in rooms [21]. It is therefore important to be able to identify the presence and
significance of resonances of all kinds. Since we are able to hear high-Q (narrow bandwidth) resonances, measurements must have
adequate resolution or they will not be revealed. It can be said, flatly, that 1/3-octave resolution is not enough.
Make sure that it is possible to do spatial averaging, wherein you can make measurements at a number of locations, say four or
five, within the listening area and then average the results. This useful method helps to identify the resonances, and to avoid being
misled by the effects of acoustical interference. It also gets rid of a lot of the grass one sees in high-resolution room measurements. If
you want to smooth the curve, do so, carefully, after the measurements are made and averaged - not before.
3.4 SUBWOOFERS AND CROSSOVERS
Conventional woofers are either closed-box or vented-box (bass reflex or coupled-cavity) designs, and they can safely be
assumed to be omnidirectional at frequencies below about 100 Hz. This means that they can be pointed in any direction, and the sound
will reach the listeners equally well, but it does not mean that their position cannot be localized by listeners. It is really annoying to be
aware of the bass frequencies coming from a location that is disassociated from the rest of the sound.
Fortunately, this is easily avoided. The easy way is to place the subwoofer(s) in the same plane as, and close to the front
loudspeakers. If the subwoofers must be located far away from the front loudspeakers, it will be necessary to ensure that the sound
output above 70 to 80 Hz is attenuated very rapidly. If not, our keen ears will zero in on the location of the woofer. This will require an
electronic crossover with steep slopes (18 to 24 dB/octave).
The first requirement is to have enough bass. This will determine the number and size of subwoofers. The second requirement
is to see that all of the key listeners hear similarly good bass. Play repeating loops of various kinds of bass instrumentation and pink
noise, and move around the listening area. Listen for hot spots and gaps in the coverage. Since the performance is so intimately bound
up with room acoustics this is a situation demanding experimentation.
Always start with the subs as close to the corners as possible to take advantage of solid-angle gains. If you get too much bass
and you have an equalizer, make the necessary measurements and dial in some attenuation. The result will be lower distortion and
higher reliability. If the excessive bass is at one or more discrete frequencies, then it might be necessary to consider dealing with room
modes. And then the serious fun begins.
4 CONCLUSION
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This paper was inspired by, and partly based on, one that I wrote in 1990 called Loudspeakers and Rooms for Stereophonic
Sound Reproduction [22]. All of that paper is still valid, and it is probably worth a read for the content that is not included here.
Still, it is interesting to reflect on what has changed in our industry since then. Digital multichannel audio is a reality, and there is even
a choice of systems. Storage devices, like DVD, are revolutionizing our view of data storage and recovery. High resolution video
sources and computer convergence are looming. It is remarkable.
Still, the big problems in getting good sound then are still the same now - loudspeakers and rooms. Our tools are better
though. There are really powerful, affordable, computer-based measuring systems allowing us to more easily identify problems in the
field. This is true back in the lab too and, as a result, loudspeakers are getting better. DSP chips can generate time delays and complex
filters that can address residual problems in the loudspeakers themselves, as well as in the room after they are installed.
We have more scientific knowledge about the relationships between what we measure and what we hear so that more of our
effort can be focused on getting results than on guessing what the target might be. The dream of the psychoacoustically optimized
adaptive system is within reach.
April 1997.rev 8/19/99
5 REFERENCES
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