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Summary. The adaptive ltering techniques have plenty of applications in any ar-
eas where the modeled signals or systems are constantly changing. An adaptive lter
is a system whose structure is alterable or adjustable in such a way that its behavior
or performance improves through contact with its environment [16]. This chapter
focuses on adaptive ltering techniques for forensic audio applications. Multichan-
nel multirate specialized structures are presented as general cases. Five approaches
are studied: spectral equalization, adaptive linear prediction (ALP), adaptive noise
cancellation (ANC), beamforming and deconvolution or derreverberation. Objec-
tive and subjective measurements for the evaluation of intelligibility after speech
enhancement are revised.
1 Introduction
the recording is intelligible [5]. Forensic ltering can be used in the laboratory
to reject the noise and interference, as well as to restore, clarify or enhance the
audio information to assist law enforcement agencies criminal investigation,
civil investigation and the court process.
There is a number of possible degradations that can be found in a speech
recording and that can aect its quality. On one hand, the signal arriving
the microphone usually incorporates multiple sources: the desired signal plus
other unwanted signals generally termed as noise. On the other hand, there are
dierent sources of distortion that can reduce the clarity of the desired signal:
amplitude distortion caused by the electronics; frequency distortion caused by
either the electronics or the acoustic environment; and time-domain distortion
due to reection and reverberation in the acoustic environment.
Adaptive lters have traditionally found a eld of application in noise and
reverberation reduction, thanks to their ability to cope with changes in the
signals or the sound propagation conditions in the room where the recording
takes place. This chapter is an advanced tutorial about multichannel adaptive
ltering techniques suitables for forensic audio to provide relevant theoretical
foundation in this regard. The employment of more than one microphone is
useful for audio surveillance purposes. This is possible when the room where
the recording will be made can be prepared in advance. However, monochan-
nel adaptive ltering can be seen as a particular case of the more complex
and general multichannel adaptive ltering. The dierent adaptive ltering
techniques are presented in a common foundation useful in other forensic dis-
ciplines.
This chapter is organized as follows: in Sect. 1.1, we introduce a formal
denition of the forensic audio scenario from the multiple-input and multiple-
output (MIMO) perspective and the terminology that is used throughout the
chapter. In Sect. 1.2, signals and systems related to forensic audio are briey
summarized. Section 1.3 discusses quality measurements. Section 2 is dedi-
cated to the theoretical foundation of the adaptive lters. In Sects. 2.1 and 2.2
the lters structure and the adaptations algorithms are discussed. The dif-
ferent cost functions, stochastic estimations and optimization strategies over
transversal and lattice structures are shown. In Sect. 3 specialized structures
based on multirate techniques are presented. These schemes allow computa-
tionally ecient algorithms to be suitable for very large impulse responses
involved in forensic audio applications and real time implementations. Two
approaches are considered in Sects. 3.1 and 3.2: the subband and frequency-
domain partitioned adaptive ltering respectly. In Sect. 3.3 the partitioned
convolution is described and in Sect. 3.4 a delayless approach for real-time
applications are commented. Section 4 focuses on the adaptive ltering tech-
niques for forensic audio application: spectral equalization, linear prediction,
noise cancellation, beamforming and deconvolution.
Adaptive Filtering Techniques for Forensic Audio 3
V x2 (n) W y2 (n)
s2 (n)
xP (n) yO (n)
sI (n)
r(n)
The box, on the left, represents a room where the evidence is being
recorded. V is a P LI matrix that contains the acoustic impulse responses
(AIR) between the I sources and P microphones 1
v11 v12 v1I
v21 v22 v2I
V= . .. . . .. ,
.. . . .
vP 1 vP 2 vP I
vpi = vpi1 vpi2 vpiL . (1)
The involved signals in forensic audio science are very particular. In general, it
is possible to group them in three big classes: speech, noise, and the acoustic
impulse responses of the involved room.
Speech
Speech is produced by inhaling air into the lungs and exhaling it through a
vibrating glottis and the vocal tract. The random noise-like, air ow from the
lungs is spectrally shaped and amplied by the vibrations of the glottal cords
and the resonance of the vocal tract. The eect of the glottal cords and the
vocal tract is to introduce a measure of correlation and predictability on the
random variations of the air from the lungs [29]. The speech signal (voice)
is formed by silence, noisy and fricative segments and harmonic or occlusive
segments and is highly modulated.
Speech signal
1
1
0 0.1 0.2 0.3 0.4 0.5
Seconds
Autocorrelation
400 0.8
200 0.2
0.4
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4
Seconds
Timefrequency analysis
2000 0
20
Hz
1000 40
60
80
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4
Seconds
Fig. 2. Example of speech word. In the upper part a 0.5-seconds segment of speech
signal corresponding to the word sota [so ta] is depicted in the time-domain. The
signal was sampled at 8192 Hz. The superimposed solid light gray line corresponds
to the normalized energy of the segment and shows the low-frequency spectrum
that denes the rate at which we utter phonemes while the dotted dark gray line
corresponds to the normalized zero crossing. In the middle part, an autocorrelation
analysis of the speech signal is depicted. In the lower part a time-frequency analysis
of the same segment is depicted. Dark colors represent areas with high energy, light
colors display areas with low energy.
At the bottom of the Fig. 2 each one of these parts can be seen: rst a
fricative segment corresponding to s is a noisy segment, relatively low-pass;
although there is not a particular spectral region with preponderate energy.
6 Lino Garca et. al.
Noise
The types of noises and interferences which can be present in the evidence
recording can be strong, subtle, and/or time varying and these may occur in
the acoustic environment where the microphones are located in. Some common
examples of acoustic noises and interferences include: air conditioning and fan
hum, reverberation and echoes, engines and other machinery, wind and rain,
radio and TV, live music, background speech in public places, other talkers,
vehicular trac and road noise, etc. Noise and unwanted sounds may lead to
listener fatigue and confusion for untrained listeners. Another class of noise
is related with a distortion that can be introduced by the recording devices
(bandwidth distortion). Both cases have a dierent approach. In the rst case
the noise and interference signal can be considered as any input signal si (n).
In the second case the noise r(n) is added equally to all the channels. Figure
3 shows an example of speech aected by an additive noise.
AIR
1
0 0.1 0.2 0.3 0.4 0.5
Seconds
Autocorrelation
400 0.8
200 0.2
Hz
1000
40
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4
Seconds
Fig. 3. Example of a contaminated speech word. The same segment speech signal
depicted in the Fig. 2 is contaminated with a broad band noise signal with a -3 dB of
signal noise rate (SNR). It is dicult to recognize the occlusive segments, however,
in spite of the high level of the noise the speech signal is intelligible and perfectly
recognized (even until for SNR = -10 dB).
1
0 0.1 0.2 0.3 0.4 0.5
Seconds
Autocorrelation
400 0.8
200 0.2
0.4
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4
Seconds
Timefrequency analysis
2000 0
20
40
Hz
1000
60
80
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4
Seconds
Fig. 4. Example of reverberating speech word. The same segment speech signal
depicted in the Fig. 2 is ltered simulating a recording of the evidence in the AIR
room. Note this the harmonic distortion.
8 Lino Garca et. al.
plication elds. Among them, the only one which accounts correctly for band-
limiting noise, reverberation, echoes and non-linear distortion is the speech
transmission index (STI), standardized by IEC [18]. The STI is based on the
generation and analysis of an articial probe signal that replaces the speech
signal, on which it is easier to measure the eects of noise and distortion.
Under some circumstances, it is possible to mathematically derive the STI
from the impulse response of the transmission system, thus avoiding the use
of the probe signal. The applicability of STI to evaluate the performance of
speech enhancement algorithms will be discussed in Sect. 5.
(a)
x(n) Adaptive
lter
Parameters
(b)
(c) d(n)
In the estimator application [see Fig. 5(a)], the internal parameters of the
adaptive lter are used as estimate.
In the predictor application [see Fig. 5(b)], the lter is used to lter an
input signal, x(n), in order to minimize the size of the output signal, e(n) =
x(n) y(n), within the constrains of the lter structure. A predictor structure
is a linear weighting of some nite number of past input samples used to
estimate or predict the current input sample.
Adaptive Filtering Techniques for Forensic Audio 11
In the joint-process estimator application [see Fig. 5(c)] there are two
inputs, x(n) and d(n). The objective is usually to minimize the size of the
output signal, e(n) = d(n) y(n), in which case the objective of the adaptive
lter itself is to generate an estimate of d(n), based on a ltered version of
x(n), y(n) [17].
Transversal
T T
with w = w1 w2 wL and x(n) = x(n) x(n 1) x(n L + 1) .
Equation (8) is called nite convolution sum 4 .
x(n) z 1 z 1 z 1
w1 w2 wL
y(n)
d(n)
xP (n) yP (n)
wP
Lattice
The lattice lter is an alternative to the transversal lter structure for the
realization of a predictor [10]. Figure 8 shows a Lattice-ladder joint-process
estimation consisting of L 1 stages; the number L 1 is refered to as the
predictor order. The coecient of the lattice structure, kl , l = 1 L 1, are
commonly called the reection or PARCOR coecients. In this framework,
instead of applying the input signal to a tapped-delay line, a prediction linear
lattice structure is used between both of them. The L observations in the x(n)
vector are replaced by the set of backward prediction errors b(n).
Stage 1 Stage L 1
x(n) f1 (n) f2 (n) fL1 (n) fL (n)
...
k1 kL1
b1 (n) k1 b2 (n)
...
bL1 (n) kL1 bL (n)
1 1
z z
w1 w2
wL1
wL
y(n)
...
Fig. 8. Multistage lattice lter.
Applying the projection theorem and knowing that the optimal backward
2 2
and forward prediction errors have the same norm f l (n) = bl (n) , 1
l L, the reection coecients can be obtained as
y(n) e(n)
yP (n)
T
where w = wT1 wT2 wTL is a LP 1 vector of the joint-process
T
estimator coecients, wl = w1l w2l wP l .
T
b(n) = bT1 (n) bT2 (n) bTL is a LP 1 backward predictor coecients
vector. A is a LP LP matrix obtained with a recursive development of (18)
and (19)
0P P 0P P 0P P 0P P 0P P
IP P 0P P 0P P 0P P 0P P
K1 K2 I 0 0P P 0P P
P P P P
K1 K3 K2 K3 0P P 0P P 0P P
A= .. .. .. .. .. .. . (23)
. . . . . .
K KL3 K KL3 0P P 0P P 0P P
1 2
K1 KL2 K2 KL2 IP P 0P P 0P P
K1 KL1 K2 KL1 KL2 KL1 IP P 0P P
Once a lter structure has been selected, an adaptation algorithm must also be
chosen. From control engineering point of view, the forensic ltering is a sys-
tem identication problem that can be solved by choosing an optimum criteria
or cost function J(w) in a block or recursive approach. Several alternatives
are available, and they generally exchange increased complexity for improved
performance (speed of adaptation and accuracy of the transfer function after
adaption or misalignment5 ).
Cost Functions
Cost functions are related to the statistics of the involved signals and depends
on some error signal
The error signal e(n) depends on the specic structure and the adaptive
ltering strategy but it is usually some kind of similarity measure between
the target signal si (n) and the estimated signal yo (n) si (n) (for I = O).
The most habitual cost functions are listed in Table 1.
Stochastic Estimation
J(w) Comments
2
e (n) Mean squared error (MSE). Statistic mean operator
N1
1
N n=0
e2 (n) MSE estimator. MSE is normally unknown
e2 (n) Instantaneous squared error
|e(n)|
Absolute error. Instantaneous module error
n nm 2
m=0
e (m) Least squares (Weighted sum of the squared error)
E{f l (n)2 + bl (n)2 } Mean squared predictor errors (for a lattice structure)
where N represents the memory size. The input signal matrix to the mul-
tichannel adaptive ltering has the form
T
X(n) = XT1 (n) XT2 (n) XTP (n) . (29)
In the most general case (with order memory N ), the input signal X(n) is
a matrix of size LP N . For N = 1 (memoryless) and P = 1 (single channel)
(29) is reduced to (5).
There are adaptive algorithms that use memory N > 1 to modify the
coecients of the lter, not only in the direction of the input signal x(n),
but within the hyperplane spanned by the input vector x(n)
and its N 1
immediate predecessors x(n) x(n 1) x(n N + 1) per channel. The
block adaptation algorithm updates its coecients once every N samples as
The new estimator w(n+ 1) is updated from the previous estimation w(n)
plus adapting-step or gradient obtained from the cost function minimization
J(w). These algorithms have an innite memory. The trade-o between con-
vergence speed and the accuracy is intimately tied to the length of memory
of the algorithm.
The error of the joint-process estimator using a transversal lter with
memory can be rewritten like a vector as
The unknown system solution, applying the MSE like the cost function,
leads to the normal or Wiener-Hopf equation. The energy of the error vector
(sum of the squared elements of the error vector) is given by the inner vector
product as6
2
J(w) = e = eH e = (d XH w)H (d XH w), (33)
J(w)
= 2Xd + 2XXH w. (34)
w
The Wiener lter coecients are obtained by setting the gradient of the
square error function to zero, this yields
1
w = XXH Xd = R1 r. (35)
R is not invertible and no unique problem solution exists. The adaptive al-
gorithm leads to one of many possible solutions which can be very dierent
from the target v. This is known as a non-unicity problem.
For a prediction application, the cross-correlation vector r must be slightly
modied assuming a particular form r = Xx(n 1), P = 1 and x(n 1) =
T
x(n 1) x(n 2) x(n N ) .
The optimal Wiener-Hopf solution wopt = R1 r requires the knowledge
of both magnitudes: the correlation matrix R of the input matrix X and the
cross-correlation vector r between the input vector and desired answer d. That
is the reason why it has little practical value. So that the linear system given
by (35) has solution, the correlation matrix R must be nonsingular.
It is possible to estimate both magnitudes according to the windowing
type of the input vector.
The sliding windowing method uses the sample data within a window of
nite length N . Correlation matrix and cross-correlation vector are estimated
averaging in time,
The method that estimates the autocorrelation matrix like in (38) with
samples organized as in (27) is known as the covariance method. The matrix
that results is positive semidenite but it is not Toeplitz.
The exponential windowed method uses a recursive estimation according
to certain forgetfulness factor in the rank 0 < < 1,
R1 (n) = 1 R1 (n 1) (41)
X(n)XH (n)R1 (n 1)
2 R1 (n 1) .
I + 1 XH (n)R1 (n)X(n)
When the excitation signal to the adaptive system is not stationary and
the unknown system is time-varying, the exponential and sliding windowed
methods allow the lter to forget or to eliminate errors happened farther in
time. The price of this forgetfulness is a deterioration in the delity of the
lter estimation [12].
20 Lino Garca et. al.
A recursive estimator has the form dened in (31). In each iteration, the
update of the estimator is made in a w(n) direction. For all the optimiza-
tion deterministic iterative schemes, a stochastic algorithm approach exists.
It is enough to replace the terms related to the cost function and calculate
the approximate values by each new set of input/output samples. In general,
most of adaptive algorithms turn an optimization stochastic problem into a
deterministic one7 and the obtained solution is an approximation to the one
of the original problem. The stochastic approximation method to the system
identication problem yields to squared minimum estimation as
Optimization strategies
7
Sampled data of the random variable are used.
Adaptive Filtering Techniques for Forensic Audio 21
The vector g(n) = J(w) is the gradient evaluated at w(n), and the
matrix H(n) = 2 J(w) is the Hessian of the cost function evaluated at
w(n).
Several rst order adaptation strategies are: to choose a starting initial
point w(0), to increment election w(n) = (n)g(n); two decisions are due
to take: movement direction g(n) in which the cost function decreases fastest
and the step-size in that direction (n). The iteration stops when a certain
level of error is reached w(n) < ,
w(n + 1) = w(n) + (n)g(n). (47)
Both parameters (n), g(n) are determined by a cost function. The second
order methods generate values close to the solution in a minimum number of
steps but, unlike the rst order methods, the second order derivatives are
very expensive computationally. The adaptive lters and its performance are
characterized by a selection criteria of (n) and g(n) parameters.
gl 2
l = 2, (50)
gl1
wl+1 (n) = wl (n) + l (n)ql . (51)
K2 K1 K2 K2 K2 KL
2 J(K) = .. .. .. . (53)
..
.
2 . . .
J(K) 2 J(K) 2 J(K)
KL K1 KL K2 KL KL
GAL is a NLMS extension for a lattice structure that uses two cost func-
tions: instantaneous squared error for the tranversal part and prediction MSE
2 2
for the lattice-ladder part, Bl (n) = Bl (n1)+(1)(|f l (n))| +|bl (n 1))| ,
where and are relaxation factors. For CGAL, the same algorithm described
in (48-51) is used but it is necessary to rearrange the gradient matrices of
the lattice system in a column vector. It is possible to arrange the gradi-
T
ents of all lattice structures in matrices. U(n) = gT1 (n) gT2 (n) gTP (n)
is the P L gradient matrix with respect to the transversal coecients,
T T
gp = gp1 gp2 gpL , p = 1 P . V(n) = G1 (n) G2 (n) GP (n) is
a P (L 1)P gradient matrix with respect to the reection coecients; and
T
rearranging these matrices in one single column vector, uT vT is obtained
T
u = g11 g1L g21 g2L gP 1 gP L ,
T
v = G111 G1P 1 GP 11 GP P 1 G112 GP P (L1) .
gl l = 1,
ql = (56)
gl + l ql1 l > 1
T
uTl vTl l = 1,
gl = T (57)
gl1 + (1 ) uTl vTl l>1
2
gl
l = 2, (58)
gl1
wl+1 = wl + ul , (59)
Kl+1 = Kl + l Vl . (60)
The time index n has been removed by simplicity. 0 < < 1 is a forget-
fulness factor which weights the innovation importance specied in a low-pass
ltering in (57). The gradient selection is very important. A mean value that
uses more recent coecients is needed for gradient estimation and to generate
more than one conjugate directions vector (57).
Adaptive Filtering Techniques for Forensic Audio 25
from the subband adaptive lters per each channel wpm , p = 1 P ,m =
1 M/2 [25]. The subband lters are very short, of length C = L+N 1
N K
K + 1, which allows to use much more complex algorithms. Although the
input signal vector per channel xp (n) has size L 1, it acts as a delay line
which, for each iteration k, updates K samples. K is an operator that means
26 Lino Garca et. al.
z 1 z 1 z 1 d(n)
K K K
H
x1 (n) y1 (n) y(n) e(n)
w1
e1 (k)
K w11
z 1
e2 (k)
K H w12
1
z
eM/2 (k)
z 1 K w1(M/2)
xP (n) yP (n)
wP
K
wP 1
z 1
K H wP 2
z 1
z 1 wP (M/2)
K
Fig. 10. Subband adaptive ltering. This conguration is known as open-loop be-
cause the error is in the time-domain. An alternative closed-loop can be used where
the error is in the subband-domain. Gray boxes corresponds to ecient polyphase
implementations. See detail in [25].
Adaptive Filtering Techniques for Forensic Audio 27
d(n)
Where the total lter length L, for each channel, is a multiple of the length
of each segment L = QK, K L. Thus, using the appropriate data section-
ing procedure, the Q linear convolutions (per channel) of the lter can be
independently carried out in the frequency-domain with a total delay of K
samples instead of the QK samples needed by standard FDAF implementa-
tions. Figure 11 shows the block diagram of the algorithm using the overlap-
save method. In the frequency-domain with matricial notation, (62) can be
expressed as
Y = X W, (63)
e = d y, (64)
T
with d = d(mK) d(mK + 1) d((m + 1)K 1) . The error in the
frequency-domain (for the actualization of the lter coecients) can be ob-
tained as
0
e = F K1 . (65)
e
G = X E. (67)
This is the unconstrained version of the algorithm which saves two FFTs
from the computational burden at the cost of decreasing the convergence
speed. The constrained version basically makes a gradient projection. The
gradient matrix is transformed into the time-domain and is transformed back
into the frequency-domain using only the rst K elements of G as
G
G=F . (68)
0KQP
The ltering operation can be made delayless by operating the rst block in
the time-domain (direct convolution) while the rest of the blocks continue to
operate in the frequency domain [22]. The fast convolution starts after the
samples have been processed for direct convolution. The direct convolution
allows giving samples to the output while data is incomming. This approach
is applicable to the multirate frameworks described.
blocks of L samples
1st 2nd J 1 Jth
T T
v1 v2 vQ v1
11stbloque
seg. 22nd seg.
bloque Qth seg.
Q bloque 1st data block
1st seg. 2nd seg. Qth seg. 2nd data block
T1 T1 T1 T1
Fig. 12. Partitioned convolution. Each output signal block is produced taking only
the L K last samples of the block.
The adaptive spectral equalization is widely used for noise suppression and
corresponds to the single-input and single-output (SISO) estimator applica-
tion (class a, Fig. 5); a single microphone, P = 1, is employed. This approach
estimates a noiseprint spectra and subtracts it from the whole signal in the
frequency-domain.
The Wiener lter estimator is the result of estimating y(n) from s(n) that
minimizes the MSE y(n) s(n)2 given by y = Qx, x = s + r, and that
results
|x|2 |d|2
q
= 2 , (69)
|x|
Q = diag q1 q2 qM is a diagonal matrix which contains the spec-
tral gain in the frequency-domain; normally T is a short-time Fourier trans-
form (STFT), suitable for not stationary signals, and T1 its inverse. In this
case this algorithm is known as short-time spectral attenuation (STSA). The
M 1 vector q contains the main diagonal components of Q. d is the noise
32 Lino Garca et. al.
T T1
r(n)
w
d(n) Wiener lter
estimator
Power Subtraction
and the phase of the noisy signal x can be used, if its SNR is reasonably
high, in place of s. is an exponent and is a parameter introduced to
control the amount of noise to be subtracted ( = 1 for full subtraction and
> 1 for over subtraction). A paramount issue in spectral subtraction is to
obtain a good noise estimate; its accuracy greatly aects the noise reduction
performance [3].
r(n)
Fig. 14. Adaptive linear predictor.
Most signals, such as speech and music, are partially predictable and par-
tially random. The random input models the unpredictable part of the signal,
whereas the lter models the predictable structure of the signal. The aim of
linear prediction is to model the mechanism that introduces the correlation
in a signal [29]. The solution to this system corresponds to a Wiener solu-
tion (35) with the cross-correlation vector, r, slighty modied. The delay z D
in the ALP lter should be selected in such a way that d(n) = x(n) + r(n)
and d(n D) are still correlated. If D is too long, the correlation in d(n)
and d(n D) is weak and unpredictable for the ALP lter; for that reason
it cannot be canceled suitably. If D is too short, the deterministic part of
signal in d(n) and d(n D) remains correlated after D; for that reason it
can be predictable and cancelled by the ALP lter. D = 1 causes that the
voice in d(n) and d(n D) is strongly correlated. A cascade of ALP lters of
lower order independently adapted improves the modeling of the general ALP
lter. In this case, the prediction is performed in successive renements, the
adaptation steps can be greater, and thus each stage is less aected by the
disparity of eigenvalues which results in a faster convergence.
The adaptive noise cancellation (ANC) cancels the primary unwanted noise
r(n) by introducing a canceling antinoise of equal amplitude but opposite
phase using a reference signal. This reference signal is derived from one or
more sensors located at points near the noise and interference sources where
the interest signal is weak or undetectable. A typical ANC conguration is
depicted in Fig. 15. Two microphones are used, P = 2. The primary input
d(n) = s(n) + r(n) collects the sum of unwanted noise r(n) and speech signal
s(n), and the auxiliary or reference input measures the noise signal x(n) =
r(n).
ANC corresponds to multiple-input and single-output (MISO) joint-process
estimator application (class c, Fig. 5) with at least two microphones, P = 2.
34 Lino Garca et. al.
d(n)
s(n)
r(n)
x(n) y(n) e(n)
w
4.4 Beamforming
x1 (n)
h1
d(n) e(n)
i
cos
hP
xP (n) AIC
i
b1 bP
FB
g1
y(n)
gP
ABM
4.5 Deconvolution
Both blind signal separation (BSS), also known as blind source separation,
and multichannel blind deconvolution (MBD) problems are a type of inverse
problems with similarities and subtle dierences between them: in the MBD
36 Lino Garca et. al.
only one source is considered, and thus the system is single-input single-output
(SISO), while in BSS there are always multiple independent sources and the
mixing system is MIMO; the interest of MBD is to deconvolve the source from
the AIRs, while the task of BSS is double: on the one hand the sources must
be separated, on the other hand the sources must be deconvolved from the
multiple AIRs since each sensor collects a combination of every original source
convolved by diferent lters (AIRs) according to (2) [13].
z D d(n) = si (n)
xP (n)
vP wP
r(n)
x1 (n) y(n)
v1 w1
si (n)
xP (n)
vP wP
BSFS
r(n)
to achieve the minimum norm solution. In the rst, the right generalized in-
verse of V is estimated and then applied to the set of microphone signals
x(n). Another class of algorithms employ the sparseness of speech signal to
design better inversion strategies and identify the minimum norm solution.
Many techniques of convolutive BSS have been developed by extending meth-
ods originally designed for blind deconvolution of just one channel. A usual
practice is to use blind source factor separation (BSFS)Blind source factor sep-
aration technique, where one source (factor) is separated from the mixtures,
and combine it with a deationary approach, where the sources are extracted
one by one after deating, i.e. removing, them from the mixed signals. The
MIMO FIR lter W used for BSS becomes a multiple-input single-output
(MISO) depicted in Fig. 18. The output y(n) corresponds to (11) and the
tap-stacked column vector containing all demixing lter weights dened in
(7) is obtained as
u = Rp, (71)
u
w= ,
H
u Ru
where R is a block matrix where its blocks are the correlation matrices Rpq be-
tween the p-th channel and q-th channel dened in (36) and p is a block vector
where its blocks are the cross-cumulant vector p = cum{x(n), y(n) y(n)}
[13]. The second step in (71) is just the normalization of the output signal
y(n). This is apparent left multiplying by x(n).
The deationary BSS algorithm for i = 1 I sources can be summa-
rized as following: one source is extracted with the BSFS iterative scheme
till convergence (71) and the ltering of the microphone signals with the
estimated lters from the BSFS method (11) is performed; the contribu-
tion of the extracted source into the mixtures xp , p = 1 P , is estimated
(with the LS criterion) and the contribution of the o-th source into i-th
mixture is computed by using the estimated
lter b, c(n) = b, y(n) with
y(n) = y(n) y(n 1) y(n B + 1) , B << L; deate the contribution
c(n) from the p-th mixture, xp (n) = xp (n) c(n), p = 1 P . This method is
very suitable for audio forensic application where only one source should be
extracted, i.e. speech.
It is possible to consider the deationay BSFS (DBSFS) structure as a
GSC. ABM exactly corresponds to the deating lters of the deationary ap-
proach. By comparing the dierent parts, i.e. the BSFS block and the xed
beamformer, it is concluded that it may be possible to construct similar algo-
rithms to those of GSC.
5 Conclusions
This chapter is an advanced tutorial about multichannel adaptive ltering for
forensic audio. Dierent techniques have been examined in a common foun-
38 Lino Garca et. al.
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