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MATLAB program for loudspeaker equalization and crossover design

Conference Paper · September 1998

DOI: 10.13140/RG.2.2.23696.20486

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 I

ENGINEERING
REPORTS

MATLAB Programfor LoudspeakerEqualizatiOnand

CrossoverDesign*

M. O. J. HAWKSFORD, AES Fellow

Centre for Audio Research and Engineering, University of Essex, UK C04 3SQ

A digital design filter program is presented written in the MATLAB environment. Impulse
response time editing is implemented together with various options for spectral domain
processing. The filter inputs time-domain impulse response data and outputs filter coefficients
for both FIR and ILR implementations. Comprehensive display options are incorporated,
including minimum-phase processing and CDS, and consideration is given to both specular
and diffuse loudspeakers. Applications include the design of digital equalization filters and
digital crossover filters for loudspeaker systems.

0 INTRODUCTION observe the effect on the overallequalization error.

Digital filters enable accurate loudspeaker equaliza-
tion and the implementation of near-theoretic crossover
alignments [1], [2]. As such it is possible to implement
loudspeaker systems which yield extremely accurate
overall frequency responses by taking full account of
the drive units' inherent response irregularities. This
technique is not only attractive for digital and active
loudspeaker systems, but it also applies to passive loud-
speakers that are preceded by a digital filter to fine-tune
the frequency response,
To develop crossover and equalizer filters, design
tools are required to compute the digital filter coeffi-
cients that can accommodate a rangeof target responses,
including those of the crossover. This engineering report
describes a filter design program that is written within
the MATLAB 1environment (version 5.2). The program
inputs data in the form of a loudspeaker's impulse re-
sponse and can embed specific equalization target func-
tions, including frequency response shaping and cross-
over design. A flexible design is offered, where either
an FIR or an IIR filter structure can be selected and the
number of coefficients in the numerator and the denomi-
nator specified. An excess-phase equalizer is also corn-
puted to enable correction of phase distortion. This allows
the user to change the number of coefficients and then to
* Presented at the 105th Convention of the Audio Engi-
neering Society, San Francisco, CA, 1998 September 26-29;
revised 1999 June 29.
i MATLAB is the trade name of a commercial matrix-based
processinglanguage.
The filter design can take account of a drive unit's
individual response (which is entered as an impulse re-
sponse) and matches this to a user-specified target fre-
quency response. In its simplest form this target is con-
stant with frequency, but can be adapted to include a
number of equalization functions. The target function
can also include a crossover filter, and a procedure is
described in Section 2.3 (see [3] for more detail).
The program incorporates a range of data display op-
tions including time-domain data, frequency domain and
cumulative decay spectrum (CDS) together with the cu-
mulative decay root (CDR) spectrum variant reported
earlier [3], and the energy-time curve.(ETC).
Also, a modified linear-phase Butterworth crossover
filter is incorporated as well as "tilt" and psychoacoustic
filters [4]. Pivotal to the design procedure is minimum-
phase signal processing, which is used in both the filter
synthesis and the data display options, including impulse
and CDS. As such the program can be used to design digital
filters for two-way digital and active loudspeakers [5].
The background to the signal processing techniques
was described in a recent paper [3], and it is suggested
that this be read as a companion to the present text. The
program listing is available on request, 2while this report
presents an overview of the computation procedures. To
illustrate the capabilities of the program two loudspeaker
examples are presented. The first is a conventional small
two-way system using moving-coil drive units, whereas
the second is a distributed-mode loudspeaker (DML) [6].
2E-mailaddress:mjh@essex.ac.uk.

706 J.AudioEng.Soc.iVol.47,No.9,1999September
ENGINEERING REPORTS
These widelydiffering designs were chosen specifically
to enable a deeper appreciation of the characteristics of
a DML when compared against conventional specular
radiating devices. However, it should be observed that
a DML is a spatiotemporal dispersive radiator. As such
the example presented here is specific to one measure-
ment location. In practice there is stochastic variation
in the polar response,
1 OVERVIEW OF DIGITAL EQUALIZATION
FILTER DESIGN
The equalization filter design procedure is reviewed
briefly in this section,
1) Impulse response measurement. A direct measure
of the loudspeaker impulse response is captured using,
for example, a maximum-length sequence (MLS)
excitation,
2) Time-domain editing of measured impulse response
a) Preresponse editing. Redundant samples prior
to the main impulse response can be removed using on-
screen editing. However, any prering due to antialiasing
filters must be considered in this truncation process.
b) Postresponse editing. First and subsequent
boundary reflections from within the measurement envi-
ronment can be removed together with noise that often
contaminates the low-level "tail" of an extended impulse
response measurement. This truncation includes a short
raised-cosine window to smooth the endpoint data.
3) Frequency domain editing of the transformed, time-
edited impulse response. Because the loudspeaker im-
pulse response has been captured over a finite time win-
dow and at a finite sampling rate, the high-frequency and
low-frequency measured responses are in error. Also, in
designing an equalization filter the extrema of the high-
frequency and low-frequency responses may be defined
so that the equalizer is not required to provide excessive
and inappropriate gain. A degree of modification can be
performed again by on-screen editing of the displayed
amplitude-frequency response,
a) Low-frequency editing of amplitude-frequency
response. Two edit points are selected on screen (using
the mouse), and a curve is computed using a quadratic
approximation. Consequently the inverse Fourier trans-
form of the impulse response now extends in length
beyond the original truncated response. For example,
where an impulse response is truncated in 2) to 256
samples and subsequently represented .for processing by
4096 samples, the additional samples computed to match
the edited frequency response are virtually noiseless,
b)High-frequency editing ofarnplitude-frequency
response. A similar two-point selection editing pro-
cess is performed at high frequency, allowing the se-
lected high-frequency curve to be replaced by a qua-
dratic approximation.
4) Linear-phase target amplitude-frequency re-
sponse. The following four options have been included
in the program for modifying the overall frequency re-
sponse. They are performed only on the amplitude re-
sponse. Consequently their native form results in a
J. Audio Eng. Soc., Vol. 47, No. 9, 1999 September
LOUDSPEAKER EQUALIZATION AND CROSSOVER DESIGN
linear-phase response.
a) Constant gain with frequency (no additional
equalization).
b) Constant slope filter (dB gain against logarithmic
frequency scale) of N dB per octave (both positive and
negative slopes enabled). This modification has been
found to be a useful tool in matching loudspeakers to
different room acoustics.
c) Linear-phase loudspeaker crossover response us-
ing a modified Butterworth amplitude alignment. Both
low-pass and high-pass options are available, and cross-
over frequency is specified at -6-dB gain. H°Wever,
the Butterworth response is modified to guarantee that
high-pass filter and low-pass filter sum to a unity-
amplitude composite response. (See Section 2.3 for
more detail, as well as [3].)
d) Psychoacoustic weighted frequency response,
available for subjective tailoring in the midrange fre-
quency region where the ear is most sensitive. This char-
acteristic has found favor in Germany [4].
The program allows these options to be selected individ-
ually and mixed if required.
5) Calculation of equalizer amplitude-frequency re-
sponse. The equalizer amplitude response is calculated
by inverting the product of the loudspeaker amplitude-
frequency response [which has been both time-domain
and frequency-domain edited, see 2) and 3)] and the
target equalizer amplitude frequency response. This is
performed only on the magnitude response.
6) Calculation of minimum-phase impulse response of
equalizer. Using the Hilbert transform, the minimum-
phase impulse response of the equalizer is evaluated.
7) Digitalfilter design. The equalizer's impulse re-
sponse is divided into two regions within the program.
The first region defines an FIR filter, whereas an IIR
filter represents the second region where coefficients are
calculated using either the Prony method 3 [7] or a least-
mean-square (LMS) technique. The method that yields
the lower error can then be selected. A general digital
filter is then defined with both numerator and denomina-
tor coefficients.
8) Excess-phase equalization of loudspeaker. Using
the Hilbert transform, the truncated loudspeaker impulse
response is decomposed into minimum- and excess-
phase components and then described in the time domain
[3]. Using truncation and time reversal of the excess-
phase impulse response, the filter coefficients required
for phase correction are estimated. The phase equalizer
can then be implemented either as an additional cascaded
filter or convolved with the numerator coefficients de-
rived in 7) to form a single filter.
2 PROGRAM FUNCTIONALITY
This section highlights some of the mathematical pro-
cedures used within the main program anddescribes the
functionality in pseudo-MATLAB code.
3 The Prony function was authored by L. Shure and is written
into the MATLAB signal processing suite as a function. Details
can be found in [7].
707
HAWKSFORD ENGINEERING
REPORTS

1 and that the first derivative is zero. Also, the curve

2.1 Minimum-Phase Processing must match in level at two adjacent samples x = m with
The minimum-phase signal-processing algorithm used a corresponding amplitude y(m) and x = m + 1 with a
in the program has been described previously [3]. The corresponding amplitude y(m + 1).
algorithm enables a straightforward method of comput- Now for a cubic curve,
ing the peak-level normalized minimum-phase impulse
response eit from a magnitude-frequency response xefa y = ax 3 + bff + cx + d .
(which is a symmetrical Fourier transform about fs/2,
where fs is the sampling frequency), that is, Then

eit = real(ifft(exp(conj(hilbert)log(xefa)))))); dy = 3ax2 + 2bx + c

eit = eit./max(abs(eit)); dx

2.2 Low-Frequency and High-Frequency Editing where for x = 1, dy/dx = O,

of Amplitude-Frequency Response
This editing routine modifies the amplitude-fre- 0 = 3a + 2b + c.
quency response at both low frequency and high fre-
quency. Two options are analyzed, a cubic fit and a In addition, matching levels at x = m and x = m + 1
quadratic fit. generates four equations,

o '_ 3 2 i o]
y(m+ 1) (m + 1)3 (m + 1)2 (m + 1) 1

y(m) m3 m2 m 1

y(1) 1 1 1 1]

Hence matrix inversion allows the coefficients

2.2,_ Cubic Fit Approximation [a b cd] to be calculated, and the cubic curve is
In each of these frequency regions two edit points are fully defined. A similar calculation procedure can be
marked and a cubic-approximation curve is generated to performed at high frequency.
replace the actual response. As shown in Fig. 1 the curve
2.2.2 Quadratic Fit Approximation
is matched in both level and first derivative by forcing
a fit to two adjacent samples, x = m and x = m + 1, A similar procedure can be followed for a quadratic
to smoothly link with the amplitude response. For low approximation, except that there is one less degree of
frequency, the first derivative is also set to zero for freedom. Thus here, as shown in Fig. 2, the derivative
sample x = 1, whereas at high frequency this is per- can be matched only at one of the extreme ends of the
formed at f/2. By way of example, the procedure is spectrum.
shown for the low-frequency cubic fit. Now for a quadratic curve,
Assume that the cubic curve has a value y(1) at x = y = axa + bx + c .

Cubic fit c ontinu ous derivative

- .... 7(m+l)
cubic curve

first derivative zero /_ .... '- -'y(m)

/
y(1)- x

x=l x=-m x=m +1

Fig. 1. Low-frequency cubic substitution (continuous derivative at x = m).

708 d.AudioEng.Soc.,Vol.47,No.9, 1999September

ENGINEERING REPORTS LOUDSPEAKER EQUALIZATION AND CROSSOVER DESIGN

Then, characteristic aj to enable a two-way digital crossover

filter to be designed and made compatible with aj.
dy The frequency range over which the computation is
dx = 2ax + b performed is defined by a frequency vector mx with an
element range {1:m2}, where
where for x = 1, dy/dx = 0,

0 = 2a + b. mx(l:m2) = ml*(l:m2);

Two options now exist. Either the derivative at the ex-
treme frequency range can be equated to zero, or the
function can be matched to the derivative of the signal
by again forcing a fit to two adjacent points at x = m
and x = m + 1. This gives rise to two matrix equations,
which can be solved for [a b c] by matrix inversion,
that is, either for a zero first derivative at low-frequency,
Here mi is the lowest frequency in the vector whereas
m2*ml = fs/2, where fs is the sampling rate.
The process commences by calculating an amplitude-
frequency response based on a Butterworth magnitude
filter template.
The crossover frequency xof (scaled by ml to form a
corresponding element number in the vector mx) and
the asymptotic attenuation slope of the filter N dB per

design. The order ord of the Butterworth filter is then

y(rn) = 2 m 1 determined,

[ y(1)
0 ] Ii 1 01 ord = abs(N/(20*logl0(2)));
restrictions on the filter) are specified as input to the
octave (N need not be an integer as there are no "analog"

or for a matching first derivative of the actual fre- from which the 3-dB break frequencies xofl and xofh of
quency response, the low-pass and high-pass filters are calculated,

[yml[m
mil]II
xomxof
y(m)

y(1)ora / =
1
]
rn2

1
m

Matching at high or low frequency is similar, although

there are detailed changes of variables in the equations.
1

1
xofh = ml*xof*(.5 A( -- 2/ord) - 1) A .5;

The amplitude-frequency responses of the low-pass and

high-pass filters ail and ajh based on the Butterworth
magnitude frequency response then follow as

In practice the quadratic fit with a forced zero first deriv-
ative at the frequency extremes was found to give the
best overall results and has been adopted in the current
version of the program.
2.3 Pseudo-Butterworth Linear-Phase
Crossover Alignment
The program includes a routine to embed either a low-
pass or a high-pass filter in the target equalization
ail = aj./(1 + (mx/xofl). A2). A(orr/2);
ajh = aj.*(((mx/xofh), n 2)./(1 + (mx/xofh). n 2)). ^ (orr/2);
However, the composite zero-phase summation (ail +
ajh) does not generally sum to the target response aj,
which itself may not be fiat due to other frequency shap-
ing characteristics selected in the program. Co^se-
quently a symmetrical (about the crossover frequency

Quadraticl-i

y(m+l)
disc on t/nu ous de_hvative _-_----_
quadratic curve _ / ',
first derivative zero / ._ .... '_-'-3'(m)

y(1)- x I

x= 1 x=m x=m +1
Fig. 2. Low-frequency quadratic substitution (discontinuous derivative at x = m).

J. Audio Eng. Soc., Vol. 47, No. 9, 1999 September 709

HAWKSFORD ENGINEERING
REPORTS

xof) compensation to ajl and ajh must be made. The impulse response eit can form a digital filter di-
For ajl, the response is modified in the frequency rectly, although in general it represents an excessive
range 1 to xof, number of coefficients. In the present program the im-
pulse response is subdivided into two regions. The first
ajl(l:xof) = ajl(l:xof) + aj(l:xof)- ajl(l:xof region of length en forms the first-stage FIR filter di-

-ajh(l:xof); rectly, whereas the remainder of the impulse response

is represented by a second-stage FIR-IIR filter and is
while for ajh, the response is modified in the frequency designed using either the Prony method or an LMS
range xof to m2, method. Both processes are supported in MATLAB. In-
tuitively this appears a logical subdivision as the FIR
ajh(xof:m2) = ajh(xof:m2) + aj(xof:m2) response carries much of the fine detail required of the
equalizer. However, the low-level tail of the impulse
- ajl(xof:m2) - ajh(xof:m2); response is primarily a result of bandwidth constraints

Now when the summation (ajl + ajh) is formed over
{l:m2}, the composite amplitude response is aj, which
is the target response of the equalizer.
2.4 Tilt Filter
A simple frequency tilt filter is included
target equalization response aj can be tilted at a constant
slope of Nx dB per octave (as described in a logarithmic
space). The response is normalized to unity gain at the
geometric frequency mean fj and computed as
such as the natural low-frequency rolloff of a loud-
speaker. As such it may be surmised to have a simpler
form more amenable to a low coefficient representation.
However, because en can be user selected together with
the second-stage filter numerator coefficients nx and de-
nominator coefficients nd required to represent the tail,
where the
a wide range of filters can be chosen extending from
pure FIR to pure IIR. As the second-stage filter is an
FIR-IIR form, the total number of numerator coeffi-
cie^ts in the overall filter is (en + rix). However, it
turns out that when the tail response is calculated, it is

aj = (mx/fj). ^ (Nx/(20*log 10(2))); not generally minimum phase. Consequently additional

2.5 Psychoacoustic Equalizer
A psychoacoustic filter was described earlier [3], [4].
The filter adds a mild reduction in the midband frequency
region where the ear is most sensitive. This can prove
a subjectively useful modification for some (poorer) re-
cordings. This filter is included in the present program
as an option and is defined by the routine
numerator coefficients nx can be specified. However,
the number of coefficients in each of the three categories
remains under user control, and through the use of time
and frequency displays the impact of a given design
selection can be observed. Also, after en is specified,
the program can be selected to interrogate the tail re-
sponse and to recalculate en so that the highest value in
the initially selected tail impulse becomes the new first
sample of the tail. The stage-one filter length en is then

np = ml*fix(3000/ml); reassigned. An autoselect procedure can be selected to

ajj = [ - 2.5*logl0((ml:ml:np)/3000) 6.4*logl0(np + mi:mi:rah)/3000)];

ajj ajj - (ajj(m2) + min(ajj))/2; facilitate this process. However, if only an FIR filter is
required, then en remains fixed and a short raised-cosine
aJj = (10*ones(size(1 :m2))). n (ajj/20); window is applied to the end of the FIR filter.

aj = aj.*ajJi The tail response is selected according to

tail = eit(en + l:m2);
where again aj is the target amplitude-frequency response.
to which both a Prony [7] and an LMS filter design
2.6 Two-Stage FIR-IIR Filter Using Prony's procedure (the latter using the MATLAB invfreq func-
Method tion 4) are applied to determine an FIR-IIR filter with
Once the amplitude-frequency responses of the target nx numerator and nd denominator coefficients. In the
aj and of the loudspeaker c are determined, the exact LMS procedure an option is included to both bandlimit
equalizer amplitude-frequency response xefa can be and power-weight the tail spectrum, where appropriate
calculated, parameterscan be user selected. The MATLABfunc-
tions have a form
xefa = aj./c;
from which the normalized minimum-phase equalizer [tnp tdp] = prony(tail(l:tmx),nx,nd);

impulse response eit follows, [tnl tdl] = invfreqz(fft(tail) ,rix - 1,nd,);

eit = real(ifft(exp(conj(hilbert(log(xefa))))));
4 Written as a function in MATLAB, coauthored by J. N.
eit = eit./max(abs(eit)); Little, J. O. Smith, Lennart Ljung, and T. Krauss.

710 J.AudioEng.Soc.,VoL47,No.9, 1999September

 ENGINEERING REPORTS
After selecting the desired tail approximation using a
normalized LMS error calculation, the FIR and IIR filter
sections can be spliced to form a complete filter. In the
program up to three vectors are outputted to describe
the overall filter. The first filter Nl(z) of length en repre-
sents the first-stage FIR filter while the second recursive
filter has a numerator polynomial N2(z) of length nx and
a denominator polynomial D2(z) of length nd (the first
coefficient being unity and not included in nd). The
complete z-domain equalizer response EQ(z) is then cal-
culated as
EQ(z) = Nl(z) + N2(z)z -(eh+ ])
D2(z)
where a tap delay of length {eh + 1} is included in the
second-stage filter to locate the approximated tail in its
correct position within the overall impulse response.
2.7 CDS Display
The generation of a CDS has been described [3] in
MATLAB where either a linear or a logarithmic magni-
tude display can be formed. The option of applying a
two-dimensional Gaussian filter mask is also included
where fine detail may require smoothing for presenta-
tional reasons. In the program, options exist for in-
eluding minimum-phase processing based on the algo-
rithm described in Section 2.1. This is a useful tool as
it gives often a better description of a loudspeaker's
performance stripped of the excess-phase distortion
which generally has lower subjective significance yet
may introduce dominant features in the impulse response
displayed in the CDS. The CDS is described by a two-
dimensional matrix cd, where
cd = 20*logl0(abs(fft(hankel(e(l:cm)))));
The MATLAB "mesh" function then translates this ma-
trix into a corresponding three-dimensional display
forming the CDS.
2.8 CDR Display
The CDR [3] is used here to display the equalizer
impulse response ei. First the roots rt of ei are calculated
and sorted into rank order. Then they are edited and
converted into a corresponding magnitude-frequency
response. Finally minimum-phase impulse responses are
calculated from the set of magnitude responses and con-
verted into corresponding ETC (see Section 2.9), from
which a two-dimensional matrix is assembled to describe
the CDR. During processing a frequency response ap-
proximation is displayed to check the validity of the
CDR after the roots are sorted and edited,
2.9 ETC Display with Excess-Phase Response
Correction
The ETC et is calculated directly from the magnitude
envelope of the Hilbert transform of an impulse response
function a, where
et = abs(hilbert(a));
· J. Audio Eng. Soc., Vol. 47, No. 9, 1999 September
LOUDSPEAKER EQUALIZATION AND CROSSOVER DESIGN
However, in the program the practice reported earlier
[3] of eliminating the noncausal distortion by minimum-
phase processing is taken. That is, a vector is formed
representing the magnitude Fourier transform of et,
fa = abs(fft(abs(et)));
from which a minimum-phase impulse response is corn-
puted which has the same amplitude spectrum as fa,
ett = real(ifft(exp(conj(hilbert(log(fa))))));
The option for displaying the ETC and the minimum-
phase ETC is provided together with an excess-phase
corrected ETC [3], where the magnitude of ett is con-
volved with the excess-phase impulse response f, that is,
ere = abs(conv(abs(ett),f));
The function ete then has a similar precursive response
to the envelope of the impulse response by including
both minimum-phase and excess-phase attributes.
3 FILTER DESIGN EXAMPLES FOR
LOUDSPEAKERS
This section presents example filter designs for two
classes of loudspeaker, the broad-band passive system
and the digital and active loudspeaker system [5]. Dis-
cussion as to the advantages of digital and active loud-
speaker technology is also included.
3.1 Broad-Band Equalization Filters for FUll-
Range Passive Loudspeakers
This section compares two filter examples with the
results are shown in Figs. 3-12. The (a) parts are for
a small two-way conventional drive unit loudspeaker
whereas the (b) parts are for a single DML. All curves
were generated from within the MATLAB design pro-
gram and for comparison 150 coefficients are force se-
leered in the first FIR stage with 50 denominator and 30
numerator coefficients in the second-stage tail filter. The
same filter size was used for both two-way and DML
loudspeakers.
The target response for filter set (a) includes a tilt of
0.5 dB per octave, whereas in set (b) it is flat. For the
DML its native response persists for well over 1000
samples, including diffuse room reflections. However,
this was truncated to about 250 samples for processing,
which is not strictly representative of this class of loud-
speaker. For proper analysis of a DML, large-space
measurements are required to extend 'the first reflection
arrival time together with the option for spatial averaging
data derived from the long impulse response and spa-
tially diffuse character of these transducers.
3.9 Digital Equalization Filters Including
Crossover Alignment
A two-way digital and active system is shown in Fig.
13. Here the digital-to-analog converters (DACs) are
711·
HAWKSFORD ENGINEERING REPORTS

Times 10 1

0.2 0.8

0.6
0
0.4

-0.2 0.2

-0.4 0

-0.2 J ......

-0,4 Times I0 .......................

-0.8 -0.6

-1 -08
50 100 150 200 250 50 100 150 200
(a) (b)

Fig. 3. Time-edited impulse response derived using MLS measurement. (a) Two-way dynamic loudspeaker. (b) DML.

09 ............................................................................................................
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o.1.........i...........
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i_ Tree i........
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_' i i i i i m i i i
50 _00 180 200 250 300 360 80 100 180 =00 280 308 380 400 480 800
(a) (b)

Fig. 4. Minimum-phase impulse response. (a) Two-way dynamic loudspeaker. (b) DML.

0.4

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-04 .-4..........i.........i..........i..........i..........i..........i.........._.........i...........

-o.8 ................................................................................
i_........................... -o.8 .....i ..........i.........i..........:._.........i..........i..........i..........i .........i...........
_86..........................................................................................................
-1 m
80 100
m
150
_
200
_
250
m
300
m
350 400
-1[!!!t; _ i i ! i i i
200 400 600 800 1000 1200 1400 1600 1800

(a) (b)
Fig. 5. Excess-phase impulse response. (a) Two-way dynamic loudspeaker. (b) DML.

712 J. Audio Eng. Soc., Vol. 47, No. 9, 1999 September

 ENGINEERING REPORTS LOUDSPEAKER EQUALIZATIONAND CROSSOVER DESIGN

interfaced directly to the power amplifiers, resulting in rate drive unit equalization. Although analog crossover
a welcome reduction of analog signal processing. Using filters can be synthesized, a more precise strategy is to
digital signal processing (DSP) it is straightforward to define the low-pass filter AL(z) and the high-pass filter
integrate near-ideal crossover filters together with accu- Au(z) as delay derived, as this eliminates the nonlinear

l iiii
i
-'°
.......
_1_
-20
''...........
ii......
.......

101
.............
i
....
:.........
........
[....................................................... ii=''''''"'"=_'',
i ; i;;ill

102
i i ilJiiii

103
; ;
_ '_,,
'_;'.:il i iiii
,_'i
;ii;i;;

104
: : ::::

105 101
: : :::fill

102
i i iiiiiii

103
i :::;:iii

104
i i ;;;[

105

(a) (b)
Fig. 6. Target, loudspeaker, and equalized frequency responses. (a) Two-way dynamic loudspeaker. (b) DML.

-lO -lO

-2o 20

-3o -3o

-4o -4o

-5o -50
o o

SO 50

100 60 80 _
100 60 80
40 40
20 20
1500 1500

(a) (b)
Fig. 7. CDS. (a) Two-way dynamic loudspeaker. (b) DML.

0. 0_

-10. -10.

-20, -20.

-30. -30,

-40 -40,

-50 -50.
0 0

50 50
80 80
60 100 60
100 40 40
20 20
1500 1500
(a) (b)

Fig. 8. Minimum-phase CDS. (a) Two-way dynamic loudspeaker. (b) DML.

J. Audio Eng. Soc., Vol. 47, No. 9, 1999 September 713

I
HAWKSFORO ENGINEERING
REPORTS

phase distortion of an all-pass crossover alignment, synchronization and exact replication.

where 3) Delay compensation to align on-axis acoustic cen-
ters of drive units is straightforward to implement using
AL(z) + AH(z) = z-_ a memory-based digital delay.
4) Simple adjustment for individual drive unit sensi-
where z- x is an appropriate delay, tivities without wasting power in passive elements of
Section 2.3 described a pseudo-Butterworth design a crossover, thus enabling amplifier power to be used
that matches this criterion and is included within the efficiently. Also, use of digital gain control (with dither)
filter design Program as an integral part of the target guarantees exact tracking of each filter channel.
function generator. 5) Because individual power amplifiers are directly
Some performance and system attributes of a digital coupled to each drive unit, there is a corresponding divi-
and active loudspeaker are summarized as follows, sion of signal power. This reduces the voltage and cur-
l) DSP enables the crossover target transfer functions, rent demand placed on an individual amplifier, where
including amplitude and phase compensation of each usually an individual drive unit has a benign terminal
drive unit together with enclosure characteristics, to be impedance compared with passive systems with compli-
specified to a high degree of accuracy. Thus the polar cated crossovers.
response can be optimized and the overall on-axis fre-: 6) Momentary clipping of a single power amplifier can
quency response can approach a constant-amplitude, be softened by preprocessing and has only a localized, in
linear-phase characteristic, the sense of frequency, impact on subjective perform-
2) Crossover filters can include options for either ance. Hence the system is more overload tolerant.
sharp frequency transition bands or more gentle fre- 7) Close coupling of amplifiers and drive units (com-
quency response slopes, including all-pass analog align- pared to passive systems) results in tighter control of
ments. The new class of interleave crossover alignments the speech coil and -lower dependence on nonlinearity
can be incorporated [3]. Multiple filters offer guaranteed in the drive unit impedance.

-20. -20,

-40. -40.

-60. -80.
0 0
20 20
40 1O0 40 1O0

60 SO 60 50
80 80
lOO0 1000

(a) (b)

Fig. 9. CDR of equalization filter. (a) Two-way dynamic loudspeaker. (b) DML.

-10 -10_

-20. -20.

-30. -30..

-40.

-50. -50
0 0

50 50
80 80
60 100 60
100 40 40
20 20
1500 150 0

(a) (b)
Fig. 10. CDS, minimum-phasecorrectiononly. (a) EqualiZedtwo-way dynamicloudspeaker. (b) Equalized DML.

714 J. Audio Eng. Soc.,Vol. 47, No. 9, 1999September

ENGINEERING
REPORTS LOUDSPEAKER
EQUALIZATION
ANDCROSSOVER
DESIGN

8) Use of current-drive technology or mixed current-
drive-voltage-drive systems can enhance system per-
formance,
9) Each DAC only handles a band-limited audio sig-
nal, thus reducing intermodulation distortion and low-
ering the probability overload in drive-unit frequency-
response correction processes,
10) Signals can be routed to an active loudspeaker
system via either an optical or an electrical digital inter-
face with central commands such as standby mode, vol-
ume level, equalization programs being remotely down-
loaded from a central (or indeed distributed) control
center. There is the option here to define a local-area
network for digital or video distribution systems,
In Fig. 14 an example crossover target filter design
is presented based on the pseudo-Butterworth algorithm
of Section 2.3, whereas Figs. 15 and 16 show the corres-
ponding low-pass and high-pass filter responses, in-
cluding drive-unit response equalization. Each filter
used a total of 285 coefficients.
Finally, the new class of interleave-crossover align-
ment [3] is illustrated by way of example in Fig. 17,
which is also incorporated in the program. The purpose
of this alignment is to distribute and randomize in fre-
quency the interference patterns that result in the off-
axis polar response [8]. Both low-pass and high-pass
filters for a stochastic-interleave function are shown that
includes a mild equalization tilt function. Fig. 18 shows
the resulting composite frequency response when there
is a time offset between drive units. A complicated inter-
ference spectrum is formed in the crossover region that
is in some respects analogous to the behavior of a DML
[6]. Also, by comparing the inverted connection com-
posite responses shown in the top curves of Figs. 14 and

2.5 t_' ETC of loudspeakerimpulseresponse

q,
2

1.5 [_ Minimum-phaseof loudspeakerETC

1 -- "-"-"'"-'"-_- ---- -

[3.5 Loudspeakerimpulseresponse

-0.5 ' ' ' ' ' ' '

0 20 40 80 80 100 120 140
(a)

2.5 _ [ ETC of loudspeakerimpulseresponse ]

,
1.5 [_ I Minimum-phase
of loudspeakerETC

0.5 [ Loudspeakerimpulseresponse }

-0.5 ' ' ' ' ' ' '

0 20 40 80 80 100 120 140
(b)
Fig. 11. Minimum-phaseETC. (a) Two-way dynamic loudspeaker. (b) DML.

J. AudioEng.Soc.,Vol.47,No.9,1999September 715
HAWKSFORD ENGINEERING REPORTS

3
t

2.5
ETC of loudspeaker impulse response

2 _ '/_Wlx'x''_'_''_

1.5 convolved with excess phase impulse response

t Minimum-phaseof loudspeakerETC

0.5 Loudspeakerimpulseresponse.

-0.5

'1

O 20 40
I

60

(a)
80
I

I00
I -- I

120
I

140

2.5 ETC of loudspeaker impulse response Ii

[

Minimum-phase of loudspeaker ETC

1.5 convolved with excess phase impulse response

0.5 Loudspeakerimpulse response

0 _

-0.5

-1 I I I I I I l

0 20 40 60 80 1O0 120 140

(b)
Fig. 12. Excess-phase corrected ETC. (a) Two-way dynamic loudspeaker. (b) DML.

-- Driveunit compensation FIR phase IjrV

and crossover network
+ digital gain control
equalization ' sta

Digital
input

-- Driveunit compensation FIR phase I/V

end
+ crossover
digital network
gain control equalization stag

Fig. 13. Two-way loudspeaker system using digital crossover filters with amplitude and phase correction.

716 J. Audio Eng. Soc., Vol. 47, No. 9, 1999 September

 ENGINEERING REPORTS LOUDSPEAKER EQUALIZATIONAND CROSSOVER DESIGN

17 (with and without interleave function, respectively),

the effect of a stochastic interleave function is dramati-
FT(linear
phase):
yIIow+highl,
bIlowl-Jhighl,
gIlowl.
r lhighl cally illustrated. In studying these results it should be
50 _ . recalled [3] that the interference patterns are bounded
· ]
/ bytheenvelopesIA[+ A.IandIA,- A.I.

i ..........

A MATLAB program to facilitate the design of loud-

speaker equalization and crossover FIR-IIR filters has
been described. Two equalization examples were pre-
sented without crossover filters and one example with a
I. 4 CONCLUSION
-s.............. crossoverfiltertogetherwith examplesof computedout-
put data. In particular, the minimum-phase and excess-
phase data of a DML reveal interesting detail about this
class of drive unit. The main differences observed are
-los ; in the impulse response, which has an extended duration
10' l0s l02 10' l0_ and "noiselike" character. This makes equalization less
Fig. 14. Low-pass and high-pass pseudo-Butterworth target straightforward. Because of the spatially diffuse form
responses, of a DML's frequency response, it is better to perform

FT(linearphase):y Ilow+highl,b Ilowl-Ihighl,

g Ilowl,rIhighl
10, L i I 60 ,, ,, ,,
/I Drive unit response _--_o',-_. _'[

...........
-'°F
.......
JJ-'i .................
i............
\i .....
I........... :o _
-20 ...........
J

-40

'BO ................. -80 ................................................. : ..............

-7o , i i 400 '/

101 102 103 104 105 101 102 lO 2 104 105

Fig. 15. Low-pass filter response including drive-unit equal- Fig. 17. Example target response of stochastic interleave align-
ization, merit. (Compare with noninterleaved response of Fig. 14.)

Composite frequency response (linear-phase targets) with polar offset

50

Drive unit response

o __.,,',_................ _,__,,
-5o................. :.,................. .,, ...........
_.................
-100
......... 15
............................
_ -10

15o ---- _-/- ......................................

/ ii
2QO
I_ ', response ', -30-
/
-250 / i i i -35 , , i
101 102 103 104 105 101 102 10a 104 105

Fig. 16. High-pass filter response including drive-unit equal- Fig. 18. Family of four composite responses time offsets be-
ization, tweed drive units.

J. Audio Eng. Soc., Vol. 47, No. 9, 1999 September 717

HAWKSFORD
equalization to a spatially averaged frequency response,
In practice it is recommended that the spatial average
be performed using measurements taken over a sampled
hemisphere. Failing this a truncated impulse can be used
together with a more limited number of coefficients in
the equalization filter. It is important to appreciate that
spatial averaging is an' intrinsic requirement of DML
measurement, especially when considering equalization,
as these transducers offer an almost omnidirectional po-
larresponse together withdiffuseproperties. (One could
argue that this new class of polar response is omnidif-
fuse.) However, the contrary is true for most conven-
tional loudspeakers, where these generally show much
wider variations in frequency response over a measure-
ment sphere. Consequently any spatial averaging win-
dows have to be applied over a more restricted measure-
ment area with corresponding limitations on listening
postion and associated room interaction problems. Expe-
rience suggests that the on-axis response forms the best
data in well-designed, conventional loudspeakers.
The method of filter design has proved efficient in
terms of computational time, and using anIIRimplementa-
tion to describe the impulse response tail would appear
intuitively to offer advantage. However, in practice,
comparing a pure FIR design with a pure HR design showed
little overall advantage where for an N-coefficient filter
a good fit to the minimum-phase impulse response is
achieved over about N samples in each case. It appears
that the total number of coefficients, irrespective of their
distribution between the two filter sections, is the more
critical factor. As reported in earlier work [1], [2], the
results demonstrate that by using a digital equalization
filter, extremely accurate system performance is possi-
hie, and that a wide range of target frequency responses
can be easily accommodated.
The two loudspeaker examples selected attempt to
highlight some performance differences between a spec-
ular radiator and a DML. In particular, the DML
minimum-phase impulse response showed good form,
although the excess-phase impulse response revealed an
extended noiselike structure in addition to a well-
focused central response, confirming its temporally dif-
fuse nature. It is therefore suggested that DMLs probably
exhibit low spatial variation in both the initial period of
the minimum-phase impulse response and the central
period of the excess-phase impulse response (where
these responses also show low noise structure). How-
THE AUTHOR
Malcolm Hawksford is director of the Centre for
Audio Research and Engineering, a professor in the De-
partment of Electronic Systems Engineering at the Uni-
versity of Essex, and Postgraduate Scheme director,
where his interests encompass audio engineering, elec-
tronic circuit design, and signal processing. Professor
Hawksford studied electrical engineering at the Univer-
sity of Aston in Birmingham where he gained a First
Class Honours B.Sc. and Ph.D. His Ph.D. program,
which was sponsored by a BBC Research Scholarship,
718
ENGINEERING
REPORTS
ever, a much greater intersample variation in the pre-
and postresponse of the excess-phase impulse response
would be anticipated as the measured window encircles
the transducer. This is implied by the transducer being
spatially diffuse. It is in these areas where fundamental
performance differences between specular radiators and
DMLs are observed, from which a number of attributes
can be deduced, for example, those relating to imaging
when multiple arrays are employed.
The embedded crossover design function was also
demonstrated and a brief discussion given on the merits
of digital and active loudspeaker systems. This was used
to illustrate the stochastic interleave-crossover align-
ment which, it is suggested, disperses the interference
patterns in the off-axis frequency response, thus low-
ering polar-related coloration in the crossover region.
5 REFERENCES
[1] R: Greenfield and M. O. J. Hawksford, "Efficient
Filter Design for Loudspeaker Equalization," J. Audio
Eng. Soc., vol. 39, pp. 739-751 (1991 Oct.).
[2] M. O. J. Hawksford and R. G. Greenfield, "A
Comparative Study of FIR and IIR Digital Equalization
Techniques for Loudspeaker Systems," Proc. Inst.
Acoust., vol. 12, pt. 8, pp. 77-86 (1990).
[3] M. O. J. Hawksford, "Digital Signal Processing
Tools for Loudspeaker Evaluation and Discrete-Time
Crossover Design," J. Audio Eng. Soc., vol. 45, pp.
37-62 (1997 Jan./Feb.), Correction, ibid., vol. 45, p.
497) (1997 June).
[4] B. Stark, "Ein Fazit," in Das Lautsprecher Jahr-
buch 86/87, M. Gaedke, Ed. (Hifisound, Mfinster, Get-
many, 1986).
[5] M. O. J. Hawksford, "Digital and Active Loud-
speaker Systems for High-Quality Monitoring," in Proc.
Active 95 (1995 Int. Symp. on Active Control of Sound
and Vibration, Newport Beach, CA, 1995 July 6-8),
pp. 1247-1258.
[6] "NXT," white paper, Huntingdon, UK.
[7] T. W. Parks and C. S. Burrus, Digital FilterDe-
sign (Wiley, New York, 1987), p. 226.
[8] A. Rimell and M. O. J. Hawksford, "Reduction
of Loudspeaker Polar Response Aberrations through the
Application of Psychoacoustic Error Concealment,"IEE
Proc. Vision Image Signal Process., vol. 145, pp.
11-18 (1998 Feb.).
investigated delta modulation and delta-sigma modula-
tion (now commonly known as "bitstream" coding) for
color television and the development of a time-
compression/time-multiplex system for combining lumi-
nance and chrominance signals, a forerunner of the
MAC/DMAC video system.
While at Essex University, he has undertaken research
principally in the fields of analog amplifiers, digital sig-
hal processing, and loudspeaker systems. Since 1982
research on digital crossover networks and equalization
J. AudioEng.Soc.,Vol.47,No.9, 1999September
 ENGINEERING
REPORTS LOUDSPEAKER
EQUALIZATION
ANDCROSSOVER
DESIGN

for loudspeakers has culminated in an advanced digital Section of the AES.

and active loudspeaker system being designed within Professor Hawksford has published in the Journal
the university. This was one of the first systems of its of the Audio Engineering Society and at the Society's
type and in 1986 a prototype was demonstrated at the conventions on topics that include error correction in
Canon Research Centre in Tokyo. Research topics have amplifiers, oversampling techniques, jitter and MLS
also encompassed oversampling and noise shaping tech- techniques, and loudspeaker crossover systems. His
niques applied to analog-to-digital and digital-to-analog supplementary activities include writing contributions
conversion, the linearization of PWM encoders, and 3- for Hi-Fi News and Record Review (a magazine at
dimensional spatial audio and telepresence including which he is a technical adviser) and Stereophile maga-
multichannel sound reproduction, zinc as well as designing high-end analog and digital
Since 1971 and throughout his career at Essex Univer- audio equipment. He is a chartered engineer and is a
sity, Professor Hawksford has lectured at university fellow of the AES, the Institution of Electrical Engi-
level in electronics and in particular the undergraduate ricers, and the Institute of Acoustics. Professor
scheme in audio engineering. He has supervised numer- Hawksford is a member of the technical committee of
ous Ph.D. and M.Sc. students, many of whom are now Acoustic Renaissance for Audio (ARA), a group that
employed within the audio industry. Malcolm is particu- has been instrumental in promoting multichannel,
larly proud of the fact that with his encouragement, two high-definition audio signals on high-capacity DVD
former research students have established the Singapore optical disks.

J. AudioEng.Soc.,Vol. 47, No.9, 1999September 719

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