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IEEE SIGNAL PROCESSING LETTERS 1

Optimal FIR Subband Beamforming for Speech Enhancement in Multipath Environments

Nedelko Grbi´c

Sven Nordholm Antonio Cantoni

Abstract— This paper provides an analysis of optimal fi- nite impulse response subband beamforming for speech en- hancement in multipath environments. A modification of the direct path standard Wiener formulation is shown to give near-optimal performance when it comes to signal-to- noise plus interference ratio, by including coherent multi- path propagation into the criterion.

Keywords— Acoustic, Subband, Beamforming, Spread Sources, Multipath

Proposed SPL EDICS: SAM - Sensor Array and Multi- channel Signal Processing

corresponding address:

Nedelko Grbi´c

Western Australian Telecommunications Research Insti- tute

The University of Western Australia 35 Stirling Highway, Crawley, WA 6009 Australia

Tel: +61 8 9380 8017 Fax: +61 8 9380 7254

E-mail: grbic@watri.uwa.edu.au signature: Nedelko Grbi´c

I. Introduction

T

HE realisation of the flexibility, safety and conve- nience afforded by handsfree communication systems is dependent on successful acoustic speech acquisition [1].

Speech enhancement by means of spectral subtraction [2], [3], [4], temporal filtering such as Wiener filtering, noise cancellation and multi-microphone methods using array techniques [5], [6], [7] may be used to achieve successful speech acquisition. Room reverberation in this context is most effectively handled with arrays or by appropriate mi- crophone design and placement.

A suitable signal and room model is required in order to develop effective solutions for a handsfree communica- tion scenario. Room modeling based on the image method is described in [8], [9], and the problem of equalisation in acoustic reverberant environments based on statistical room modeling is studied in [10]. In the context of mul- tiple sensors, it is shown in [11] that an improved far-to- near field suppression may be achieved by adding a con- straint on the white noise gain across frequency. This ob- servation may be used to reduce reverberation effects in

The authors are with Western Australian Telecommunication Research Institute, which is a joint venture of The Univer- sity of Western Australia and Curtin University. The work has also been sponsored by ARC under grant no. A00105530.

(e-mail: grbic@watri.uwa.edu.au, sven@watri.uwa.edu.au, can- toni@watri.uwa.edu.au)

small rooms since wall reflections appears as far-field repli- cas of the source, [12]. The impact of room reverberation on speech intelligibility is studied experimentally in [13], for multi-sensor hearing aid applications. A significant in- crease is reported in speech intelligibility by incorporating beamforming algorithms.

This paper analyses optimal FIR subband beamforming and presents an extended Wiener formulation for a new signal model with spread sources in enclosed environments.

The model assumes spherically spread sources and coher- ent reflections of signal propagation based on the image method. The spherical spreading of the source is based on the fact that human speech gives rise to a vibration of the skull, creating a frequency dependent directivity pat- tern, [14].

II. Signal and Propagation Model

A scenario where a desired speech source and interfering speech sources are located in an enclosure with reflecting walls is considered, e.g. modeling an indoor teleconferenc- ing environment. A linear array with N omnidirectional microphones is situated in the center of the enclosure. The received signals are sampled with sampling period T and digital finite impulse response (FIR) filtering of received array signals is used to filter the desired source while at- tenuating or canceling unwanted disturbing sources. The received signals caused by directional reflections are mod- eled as source coherent signals. The surrounding noise field is assumed to be the result of a large number of broad band noise components at different far-field locations, and it is modeled as a diffuse field [15]. Since the desired source is assumed to be a human speaker, the propagating sound field is modeled as the result of a vibrating spherical object with a frequency dependent directivity pattern [14].

The spatio-temporal covariance matrix contained in the spectral frequency band [Ωab], received from M sources originating from regions in space Sm, m = 1, 2 · · · , M , with power spectral density (PSD) Sm(−→s , Ω) at a spatial point with position vector −→s is given by

Rx= XM m=1

b

Z

a

Z Z Z

Sm

Sm(−→s , Ω)P(−→s , Ω)PH(−→s , Ω)d−→s dΩ+Rn. (1) The dimension of the matrix is N L × N L, where L is the length of the FIR filters and N is the number of spatial sensors. The element at position (i · k; p · l) of the diffuse

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IEEE SIGNAL PROCESSING LETTERS 2

noise covariance matrix Rn is given by, [16]

Rn(i·k;p·l)= 1 3

Z b

a

Sn(Ω)sin

³Ωdip

c

´

Ωdip

c

e−jΩ(k−l)TdΩ (2) where i and p denotes sensor indices, and k and l denotes time lag indices, for row and column, respectively. Param- eter c is the speed of wave propagation, Sn(Ω) is the noise PSD and dip is the distance between sensors i and p. The response vector P(−→s , Ω) at the point with position vector

s , is given by the image method as the sum of direct path and all paths of coherent reflective images [8],

P(−→s , Ω) = X o=−∞

X s=−∞

X t=−∞

α|o·s·t|E(Ω) ⊗ G(−−→sost, Ω) (3)

where ⊗ denotes the Kronecker matrix multiplication, 0 <

α < 1 is a reflections coefficient determining the gain in magnitude for each reflection. The spatial response vector in (3) is normalised to unit gain at origin of coordinates, given by

G(−−→sost, Ω) = ejΩRmc

h

β1e−jΩτ1(−−→sost), β2e−jΩτ2(−−→sost), . . . , βNe−jΩτN(−−→sost) iT (4) where −−→sostis the image position vector, Rmis the distance to the center point of the spatial region Sm and τn(−−→sost) denotes the signal propagation time delay from the image position to sensor n. Parameter βn= kdnk

k

dn−−−→sostk, where −→ dn is the position vector of sensor n. The temporal response vector in (3) is given by

E(Ω) = ejL2ΩT h

1, e−jΩT, . . . , e−j(L−1)ΩT iT

(5) which is normalised to the center lag by the constant ejL2ΩT, where L is the FIR filter length. The explicit po- sition in Cartesian coordinates of an image source indexed (o, s, t), caused by a room enclosure of size (X × Y × Z) m, can readily be shown to be

−−→sost= diag¡£

(−1)o, (−1)s, (−1)t¤¢ −→s + [oX, sY, tZ]T (6) where −−→s000

def= −→s is the position vector of the source, con- tained inside the enclosure. The origin of coordinates is defined at the center of the enclosure.

III. Optimal Beamformers

In order to characterise optimal beamforming the re- ceived spatio-temporal covariance matrix in (1) is split up as

Rx= Rs+ Rj+ Rn (7) where Rs, Rjand Rndenotes covariance matrices received by contributions of the target source, undesired (jammer) sources and noise, respectively. Without loss of generality it is assumed the first source is the target source.

A. The optimal SNIR beamformer

The set of FIR filters optimising the signal-to-noise plus interference ratio (SNIR) is given by [15]

wSN IB= arg max

w

µ wHRsw wH(Rj+ Rn)w

(8) where w is a vector of stacked FIR filter parameters. The optimal solution is given by, [16], [15]

wSN IB= (Rj+ Rn)−1/2˜vmax (9) where ˜vmaxis an eigenvector, corresponding to the largest eigenvalue λ, complying with

³

(Rj+ Rn)−1/2´H

Rs(Rj+ Rn)−1/2max= λ˜vmax. (10) The measure of SNIR is scale invariant and any constant scaling of the weight vector given in (8) also maximizes the SNIR.

B. The Wiener beamformer

The Wiener filter is the solution to the problem of linear mean-square waveform estimation of signals corrupted by noise, provided the noise and the signals are stationary random variables, [17], given by

wW iener = arg min

w

¡σ2d+ wHRsw − 2<© wHrs

ª¢ (11)

where < {·} denotes real part, and rs = E {x(t)d(t)}

where E {·} is the expectation operator. The vector x(t) denotes the recieved array signals, d(t) is a desired output signal with variance σd2. In the standard Wiener formula- tion a delayed version of the source signal is defined as the desired signal. The corresponding cross-covariance vector is then given by,

rs=

b

Z

a

Z Z Z

S1

S1(−→s , Ω)E(Ω) ⊗ G(−→s , Ω)d−→s dΩ (12)

and the solution to (11) is given by

wW iener= Rx−1rs= [Rs+ Rj+ Rn]−1rs. (13) The coherent signals contained in reflections are not uti- lized in (12). By simply including coherent reflections to form the desired signal, we arrive at the coherent multipath Wiener solution given by (13), with the cross-covariance vector replaced by

rsc=

b

Z

a

Z Z Z

S1

S1(−→s , Ω)P(−→s , Ω)d−→s dΩ. (14)

A practical algorithm implementing (13) is given in [7], where the source covariance matrix is calculated analyt- ically and the jammer and the noise covariance matrices are estimated recursively from received data.

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IEEE SIGNAL PROCESSING LETTERS 3

C. Equivalence between the optimal SNIR and the coherent multipath Wiener beamformer

In the theoretical case of monochromatic input sig- nals with temporal frequency Ωc, or when the signals are assumed to be temporally filtered into narrow non- overlapping subbands with center frequency Ωc, the inte- grand in (1) includes the delta function δ(Ω − Ωc), and the temporal domain vanishes [16]. Further, if the spatial spread of the source is much smaller than the array aper- ture, the properties of the source becomes close to that of a point source with position vector −→s1. In this case the region of spatial integration in (1) vanishes and each source adds a single rank contribution to the spatial covariance matrix, [15]. The solution to (8) may then be expressed in closed form

wSN IB= [Rj+ Rn]−1P(−→s1, Ωc) (15) and the solution given by the coherent Wiener formulation is given by

wW iener

P(−→s1, Ωc)PH(−→s1, Ωc) + Rj+ Rn¤−1

P(−→s1, Ωc).

(16) We may rewrite (16) by applying the matrix-inversion- lemma, [16], as

wW iener = [Rj+ Rn]−1P(−→s1, Ωc



1 − 1

1 + 1

PH(−→s1, Ωc) [Rj+ Rn]−1P(−→s1, Ωc)



 . (17)

Since the measure of SNIR is scale invariant and the solu- tion given in (15) only differs in scale to that of (17), the coherent Wiener solution coincides with the optimum SNIR solution, when considering monochromatic point sources.

IV. Evaluation

In the numerical studies, a room of size (X × Y × Z) = (6 × 4 × 4) m is considered. An eight sensor linear array is located at the center of the room, receiving speech from persons located along a line in front of the array. The per- sons are modeled as spheres representing vibrating human skulls, i.e. Sm in (1) are spheres with 15 cm diameter, see Fig. 1. One desired source speaker and one interference speaker are assumed to be active simultaneously.

Fig. 1. Source and array set-up illustrated in 2-D space.

The spheres are assumed to emit constant power across frequency along the surface of the semi-spheres facing the array. A linear decrease in power across subbands is as- sumed for the opposite semi-sphere, where the power drops

from 0 dB at 250 Hz to −12 dB at 3400 Hz. The model is an approximation of the model presented in [14].

The FIR beamforming solutions are analysed by sub- dividing the traditional telephone bandwidth into regions with linearly increasing logarithmic bandwidth where each subband operates at Nyquist sampling rate. Since the spatial resolution increases with increasing frequency, the number of FIR filter taps needed becomes approximately constant across the subbands. The optimal solutions are calculated by truncating the sums in (3) such that 99 % of the power contained in all reflections are captured. Fig- ure 2 shows the output SNIR as a function of FIR filter length for the different solutions when using six subbands with the reflection coefficient α = 0.5. The number of FIR parameters needed to reach optimal performance is approx- imately constant, around 10-15 taps. Figure 3 shows the output SNIR for three different values of the reflection co- efficient, when 12 bands with the same linearly increasing logarithmic bandwidth are used and with 10 FIR parame- ters in each subband. It can be seen that the optimal SNIR beamformer and the coherent Wiener beamformer reaches almost the same level of performance, which supports the results presented in sec. III-C for monochromatic sources.

It can also be seen in Fig. 3, that performance in low fre- quency bands is reduced when the reflection coefficient is increased for these two solutions. The standard Wiener so- lution does not exhibit such simple relation to the reflection coefficient.

Fig. 2. Output SNIR vs. FIR length for six non-uniformly dis- tributed frequency bands. Solid line represents the maximal SNIB (eq. (8)), dashed line is the multipath Wiener (eq. (13) & (14)) and the dashed-dotted line is the standard Wiener (eq. (13) &

(12)). Linear array with N = 8, sensor distance is 0.05 m, source SIR=0 dB, source SNR=30 dB, source in position 3, in- terference in position 4, reflection coefficient α = 0.5.

Table I gives the resulting total SNIR performance for some different locations of the source and the interferer, as given in Fig. 1. It can be seen that the standard Wiener solution gives a substantial increase in performance as the angular separation between the source and the interferer in- creases. When the separation between the source and the interference increases so does the separation between their

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IEEE SIGNAL PROCESSING LETTERS 4

Fig. 3. Output SNIR vs. subband index for 12 non-uniformly dis- tributed frequency bands, with 10 FIR filter parameters in each subband. Left figure - reflections coefficient α=0.1, center figure - α=0.5, right figure - α=0.9. Same setting and notation as in Fig. 2.

respective image reflections. Since all source reflections are regarded as interferers in the standard Wiener formula- tion, the resulting performance improves as the separation increases. The performance of the optimal SNIR beam- former and the coherent Wiener beamformer can be seen to be less dependent on the spatial separation between the source and the interferer.

Source ⇔ Interference position 3 ⇔ 4 2 ⇔ 5 1 ⇔ 6 Max. SNIR, eq. (8)

Std. Wiener, eq. (12) & (13) Mod. Wiener, eq. (14) & (13)

28.9 11.7 28.0

29.9 14.5 28.6

30.5 19.2 28.9

SNIR dB dB dB

TABLE I

Total output SNIR with 12 non-uniformly distributed frequency bands, for some combinations of source and interference positions

along a speaker line-up, as given in Fig 1. Linear array with N = 8, sensor distance is 0.05 m, source SIR=0 dB, source

SNR=30 dB, reflection coefficient α = 0.5.

V. Summary and Conclusions

Optimal FIR subband beamforming in multipath envi- ronments are analysed, and an extension of the standard Wiener solution is presented. While this formulation is shown to be equivalent to the optimal SNIR beamformer in the case of monochromatic point sources, it is also shown to be nearly optimal in SNIR sense in the case of spatially spread subband decomposed wideband sources.

References

[1] M. S. Brandstein and D. B. Ward, Microphone Arrays: Tech- niques and Applications, Springer Verlag, 2001.

[2] J. R. Deller, J. G. Proakis, and J. H. L. Hansen, Discrete-Time Processing of Speech Signals, Macmillan, 1993.

[3] S. F. Boll, “Suppression of acoustic noise in speech using spectral subtraction,” IEEE Trans. Speech and Audio Processing, vol.

ASSP-27, no. 2, pp. 113–120, Apr. 1979.

[4] J. Yang, “Frequency domain noise suppression approaches in mobile telephone systems,” in IEEE International Conference on Acoustics, Speech and Signal Processing, Apr. 1993, vol. 2, pp. 363–366.

[5] S. Nordholm, I. Claesson, and B. Bengtsson, “Adaptive array noise suppression of handsfree speaker input in cars,” IEEE Trans. Vehicular Tech., vol. 42, no. 4, pp. 514–518, Nov. 1993.

[6] N. Grbi´c, X. J. Tao, S. E. Nordholm, and I. Claesson, “Blind signal separation using overcomplete subband representation,”

IEEE Trans. Speech and Audio Processing, vol. 9, no. 5, pp. 524 –533, Jul. 2001.

[7] N. Grbi´c and S. Nordholm, “Soft constrained subband beam- forming for hands-free speech enhancement,” in IEEE Interna- tional Conference on Acoustics, Speech and Signal Processing, May 2002, vol. 1, pp. 885–888.

[8] F. Jabloun and B. Champagne, “A fast subband room response simulator,” in IEEE International Conference on Acoustics, Speech and Signal Processing, 2000, vol. 2, pp. 925–928.

[9] P. M. Peterson, “Simulating the response of multiple micro- phones to a single acoustic source in a reverberant room,” J.

Acoust. Soc. Am., vol. 80, no. 5, pp. 1527–1529, 1986.

[10] B. D. Radlovi´c, R. C. Williamson, and R. A. Kennedy, “Equal- ization in an acoustic reverberant environment: Robustness re- sults,” IEEE Trans. Speech and Audio Processing, vol. 8, no. 3, pp. 311–319, May 2000.

[11] J. G. Ryan and R. A. Goubran, “Array optimization applied in the near field of a microphone array,” IEEE Trans. Speech and Audio Processing, vol. 8, no. 2, pp. 173–176, Mar. 2000.

[12] J. G. Ryan and R. A. Goubran, “Near-field beamforming for mi- crophone arrays,” in IEEE International Conference on Acous- tics, Speech and Signal Processing, 1997, vol. 1, pp. 363–366.

[13] P. W. Shields and D. R. Campbell, “Intelligibility improvements obtained by an enhancement method applied to speech corrupted by noise and reverberation,” Speech Commun., vol. 25, pp. 75–

79, 1998.

[14] J. Huopaniemi, K. Kettunen, and J. Rahkonen, “Measurement and modeling techniques for directional sound radiation from the mouth,” in Proc. 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 1999, pp. 183–186.

[15] J. E. Hudson, Adaptive Array Principles, Peter Peregrinus, 1991.

[16] D. Johnson and D. Dudgeon, Array Signal Processing - Concepts and Techniques, Prentice Hall, 1993.

[17] B. Widrow and S. D. Stearns, Adaptive Signal Processing, Pren- tice Hall, 1985.

References

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