A Multichannel Acoustic Echo Canceler Double-Talk Detector Based on a Normalized Cross-Correlation Matrix

Size: px
Start display at page:

Download "A Multichannel Acoustic Echo Canceler Double-Talk Detector Based on a Normalized Cross-Correlation Matrix"

Transcription

1 Acoustic Echo and Noise Control A Multichannel Acoustic Echo Canceler Double-Talk Detector Based on a Normalized Cross-Correlation Matrix JACOB BENESTY Bell Laboratories, Lucent Technologies, 700 Mountain Avenue, Murray Hill, NJ, USA jbenesty@bell-labs.com TOMAS GÄNSLER Agere Systems gaensler@agere.com Abstract. Multichannel acoustic echo cancellation is basically composed of two parts. One part is a multichannel system identification problem which is nontrivial to solve. The other part, which is also nontrivial, is the so-called double-talk detection problem. Near-end speech detection is based on a test statistic. Recently, a new double-talk test statistic based on the normalized cross-correlation vector was proposed for the single-channel case. Obviously, in the multichannel case, there are several solutions, but what should be the optimal one is not yet known. Then, a fundamental question arises: how do we deal optimally with multiple channels? In this paper, we generalize the idea of normalized cross-correlation vector (single-channel) to the matrix case (multichannel), derive a frequency-domain version, and show how to combine both the multichannel frequency-domain adaptive filter and the multichannel double-talk detector. INTRODUCTION Multichannel acoustic echo cancellation is basically composed of two parts. One part is a multichannel system identification problem which is nontrivial to solve; see, 2 for more details. The other part, which is also nontrivial, is the so-called double-talk detection problem 3. When the multiple echo paths are identified by a multichannel adaptive filter, a function should be included to freeze the adaptation whenever a near-end signal is detected, and thereby avoid the divergence of the adaptive algorithm. This is the role of a double-talk detector (DTD) 4. Consequently, a suitable DTD decision variable has to be found. An optimum decision variable ξ for double-talk detection should behave as follows: (i) if double-talk is not present, ξ T ; (ii) if double-talk is present, ξ<t. The threshold T must be a constant (in this application), independent of the data. Moreover, ξ must be insensitive to echo path variations when there is no double-talk. A A shorter version of this paper was presented at the International Workshop on Acoustic Echo and Noise Control, IWAENC 200. Optimum in the sense that for a given probability of false alarm (of double-talk), the probability of miss is minimized. decision variable that exhibits this behavior was proposed in 5 for the single-channel case, using a new normalized cross-correlation vector. In many acoustic signal processing problems such as voice activity detection, time delay estimation, source localization, double-talk detection, etc, multiple microphone signals are available. However, it is not obvious as to how all of this information should be taken into account for best performance. In the case of (multichannel) double-talk detection, there are at least three options. The first one is to consider only a single microphone and build a test statistic based on the normalized cross-correlation vector between the loudspeaker signals and the chosen microphone signal. This is the approach that was taken in 6, 7. However, this approach may increase the probability of miss since the microphones pick the near-end speech up with different levels and different SNRs, depending on the position of the talker and microphones. Furthermore, not all of this information is taken into account in the decision variable. The second option is to use a test statistic for each microphone, so that the decision made on one microphone is independent of the others. But a large amount of tests may not be easy to handle and we may have some contradictory results on the Vol. 3, No. 2, March-April 2002

2 J. Benesty, T. Gänsler presence of a near-end talker. In this paper, a third option is proposed which consists of using one test statistic that combines all the (spatially sampled) information of the microphone signals. This approach leads to a nice generalization of the normalized cross-correlation vector to the normalized cross-correlation matrix. By its nature, this method considers all the microphone signals equally. Other possibilities based on this approach can be derived but what will be the optimal choice is not yet known. Thus, multichannel double-talk detection is a very new topic of research as well as a challenging and an interesting problem. The organization of this paper is as follows. In section 2, a brief outline is given on multichannel acoustic echo cancellation and the multichannel Wiener-Hopf equation. In section 3, the normalized cross-correlation matrix is defined and it is shown how to apply it to double-talk detection. Section 4 gives a frequency-domain approach. In section 5, it is shown how to combine, in the frequency domain, echo cancellation and double-talk detection with the new approach. Finally, conclusions are given in section 6. 2 MULTICHANNEL ACOUSTIC ECHO CAN- CELLATION First, we assume that we have Q loudspeakers and microphones. We also assume that the system (room) is linear and time-invariant. Acoustic echo cancellation consists of identifying Q echo paths at each microphone so that in total, Q echo paths need to be estimated. We have output (microphone) signals (see figure for notation): y p (n) hqp T x q(n) + v p (n), () q p, 2,...,, superscript T denotes transpose of a vector or a matrix, h qp T h qp,0 h qp, h qp,l is the echo path of length L between loudspeaker q and microphone p, x q (n) xq (n) x q (n ) x q (n L + ) T, q, 2,..., Q, is the qth reference (loudspeaker) signal (also called the far-end speech), and v p is the near-end speech added at microphone p, assumed to be uncorrelated with the far-end speech. We define the error signal at time n for microphone p as e p (n) y p (n) ŷ p (n) y p (n) ĥ T qp x q(n), (2) q ĥ qp ĥ qp,0 ĥ qp, ĥ qp,l T are the model filters. It is more convenient to define an error signal vector for all the microphones: x (n) x 2 (n) x q (n) e p (n) e(n) y(n) ŷ(n) y(n) Ĥ T x(n), (3) y(n) H T x(n) + v(n), h h 2 h h 2 h 22 h 2 H , h Q h Q2 h Q v(n) v (n) v 2 (n) v (n) T, e(n) e (n) e 2 (n) e (n) T, y(n) y (n) y 2 (n) y (n) T, ŷ(n) ŷ (n) ŷ 2 (n) ŷ (n) T, ĥ ĥ 2 ĥ ĥ 2 ĥ 22 ĥ 2 Ĥ......, ĥ Q ĥ Q2 ĥ Q T x(n) x T (n) xt 2 (n) xt Q (n). Ad. FIR NL NL NL + ŷ p (n) y p (n) h p h 2p h qp Figure : Multichannel acoustic echo cancellation. (Ad. and NL stand respectively for adaptive and nonlinearity.) Having written the error signal, we now define the cost 96 ETT

3 A Multichannel Acoustic Echo Canceler Double-Talk Detector Based on a Normalized Cross-Correlation Matrix function J E{e T (n)e(n)} (4) E{e 2 p (n)} p J p. p E{ } denotes the statistical expectation operator. The minimization of (4) leads to the multichannel Wiener-Hopf equation: R xx Ĥ R xy, (5) R xx E{x(n)x T (n)} (6) is the covariance matrix of size (QL QL) of the reference signals x, and R xy E{x(n)y T (n)} (7) is the cross-correlation matrix of size (QL ) between x and y. It can easily be seen that the multichannel Wiener-Hopf equation (5) can be decomposed in independent Wiener- Hopf equations, each one corresponding to a microphone signal: R xx ĥ p r xyp, p, 2,...,, (8) ĥ p (resp. r xyp )isthepth column of matrix Ĥ (resp. R xy ). This result implies that minimizing J or minimizing each J p independently gives the same results. This makes sense from an identification point of view, since the identification of the impulse responses for one microphone is completely independent of the others. However, as far as double-talk is concerned, it is preferable to have a global and unique test statistic that takes into account the information of all the microphone signals, since the near-end speech is picked-up by all the microphones with different levels. Choosing one microphone signal and using a single test statistic based on this signal is not enough. On the other hand, using independent decision variables will be much harder to handle (computational complexity, inconsistency among the different tests,...). Thus, the approach taken here is to develop a unique test statistic by looking at the covariance matrix R yy of the microphone signals y. In many situations, the signals x q are generated from a unique source s, so that: x q (n) gq T s(n), q, 2,..., Q, (9) g q g q,0 g q, g q,l T is the impulse response between the source and microphone q in the transmission room in the case of a teleconferencing system 2. Therefore the signals x q are linearly related and we have the following Q(Q )/2 relations: xq T (n)g i xi T (n)g q, (0) i, q, 2,..., Q, i q. Now, consider the following vector: Q u q2 ζ qgq T ζ 2 g T ζ Q g T T, ζ q are arbitrary factors. We can verify using (0) that R xx u 0 QL,soR xx is not invertible. Vector u represents the nullspace of matrix R xx. The dimension of this nullspace depends of the number of channels and is equal to (Q 2)L + (for Q 2). So the problem becomes worse as Q increases. Thus, there is no unique solution to the problem and an adaptive algorithm will drift to any one of many possible solutions, which can be very different from the true desired solution ĥ qp h qp. These nonunique solutions are dependent on the impulse responses in the transmission room: ĥ p h p + β ζ q g q, () q2 ĥ qp h qp βζ q g, q 2,..., Q, (2) β is an arbitrary factor. This, of course, is intolerable because g q can change instantaneously, for example, as one person stops talking and another starts, 2. In the following, we suppose that the covariance matrix R xx is invertible. To almost guarantee that, we may add (for Q > ) a non-linear (NL) transformation (or add perceptually acceptable uncorrelated noise) to each input signal x q in order to reduce the coherence of the signals two-by-two 2. Other existing methods to reduce the coherence can be used 8, 9, 0,, 2, 3. 3 A NORMALIZED CROSS-CORRELATION MATRIX FOR MULTICHANNEL DOUBLE- TALK DETECTION For the single-channel case, it has been shown 5 that the so-called normalized cross-correlation vector is well suitable for DTD. In this section, we derive a decision variable based on a normalized cross-correlation matrix for the multichannel situation. Suppose that v 0 (no near-end speech). In this case: R yy E{y(n)y T (n)} H T R xx H. (3) Vol. 3, No. 2, March-April 2002

4 J. Benesty, T. Gänsler Since y(n) H T x(n),wehave: and (3) may be re-written as R xy E{x(n)y T (n)} R xx H (4) R yy R T xy R xx R xy. (5) Now, in general for v 0, R yy R T xy R xx R xy + R vv, (6) R vv E{v(n)v T (n)} (7) is the covariance matrix of the near-end speech v. Then from (5) and (6), the following decision statistic is proposed, ξ C xy E tr(c T xy C xy) tr(r T xy R xx R xy R yy ), (8) C xy R /2 xx R xy R /2 yy (9) is the normalized cross-correlation matrix between the two vectors x and y. Substituting (4) and (6) into (8), ξ trh T R xx H(H T R xx H + R vv ). (20) It is easily deduced from (20) that for v 0, ξ and for v 0, ξ<. So that, in the noiseless case, we can set the threshold T. For the single-channel case (Q ), (9) becomes the normalized cross-correlation vector between vector x and scalar y : c x y (σ 2 y R x x ) /2 r x y, (2) with σy 2 E{y 2 (n)}. The test statistic is now: ξ c x y E tr(c T x y c x y ) c x y 2 r T x y (σy 2 R x x ) r x y. (22) It can easily be seen that (22) can also be written as: h T ξ R x x h, (23) h T R x x h + σ 2 v so that for v 0, ξ and for v 0, ξ<. Thus, the normalized cross-correlation matrix is a natural and elegant extension of the normalized crosscorrelation vector 5 to the multichannel case. 4 A FREQUENCY-DOMAIN AROACH In this section, a frequency-domain DTD is derived that will be more useful in practice than its time-domain counterpart, because it is much more efficient from a computational complexity point of view. We define the block error signal (of length L) for microphone p as: e p (m) y p (m) ŷ p (m) y p (m) X q (m)ĥ qp, (24) p, 2,...,, q m is the block time index, X q (m) is a Toeplitz matrix, and e p (m) e p (ml) e p (ml + L ) T, y p (m) y p (ml) y p (ml + L ) T, X q (m) x q (ml) x q (ml + L ) T. It is well known that a Toeplitz matrix X q (m) can be transformed, by doubling its size, to a circulant matrix X C q (m) q (m) X q (m) X q (m) X q (m), X q (m) is also a Toeplitz matrix obtained by shifting rows of X q (m); the diagonal of C q (m) is arbitrary, but it is natural and customary to set it equal to the first sample in the previous block. It is also well known that a circulant matrix is easily decomposed as follows: C q (m) F D q (m)f, F is the Fourier matrix of size (2L 2L) and D q is a diagonal matrix whose elements are the discrete Fourier transform of the first column of C q (m). Before going to the frequency domain, we need to rewrite (24) in a different way: 0L e p (m) 0L y p (m) W q C q (m) ĥqp 0 L, (25) 98 ETT

5 A Multichannel Acoustic Echo Canceler Double-Talk Detector Based on a Normalized Cross-Correlation Matrix W 0L L 0 L L. 0 L L I L L Now, multiplying both sides of (25) by F, we obtain the frequency-domain error signal: Q e p (m) y p (m) G D q (m)ĥ qp, (26) q 0L e p (m) F, e p (m) 0L y p (m) F, y p (m) G FW F, ĥ qp F ĥqp 0 L. Minimizing the frequency-domain criterion J fd E{e H p (m)e p (m)} (27) p leads to the multichannel Wiener-Hopf equation in the frequency domain: S xx Ĥ S xy, (28) H denotes conjugate transpose and S xx E{D H (m)g D(m)}, D(m) D (m) D 2 (m) D Q (m), ĥ ĥ 2 ĥ ĥ 2 ĥ 22 ĥ 2 Ĥ......, ĥ Q ĥ Q2 ĥ Q S xy E{D H (m)y(m)}, Y(m) y (m) y 2 (m) y (m). Following the same philosophy as in section 3, we define the pseudo-coherence matrix: Ɣ xy S /2 xx S xy R /2 yy with R yy E{YH (m)y(m)}. For Q, Ɣ xy becomes: (29) γ x y S /2 x x s x y E /2 {y H (m)y (m)}, (30) which is also called the pseudo-coherence vector 6, 7. The difference between the true coherence and the pseudocoherence is a different normalization with respect to the signal y. We define the multichannel frequency-domain decision variable as: ξ fd Ɣ xy E tr(ɣ H xy Ɣ xy) tr(sxy H S xx S xy R yy ), (3) and it can be checked that for v 0, ξ fd and for v 0,0 ξ fd <. 5 COMBINATION OF MULTICHANNEL IDENTIFICATION AND DOUBLE-TALK DE- TECTION IN THE FREQUENCY DOMAIN In this part, it is shown how to combine both the adaptive identification of the echo paths and the near-end speech detection. Since the DTD will also be adaptive, two different multichannel model filters (one foreground and one background) will be used, like the two-path model 4. The background filter will be updated permanently to estimate the test statistic. Each time double-talk is detected, the adaptation of the foreground filter (for identification) will be halted. A multichannel frequency-domain adaptive algorithm can be easily derived from the Wiener-Hopf equation. In this section, only the algorithm is given 5, which is: Ŝ xx (m) λ f Ŝ xx (m ) + ( λ f )D H (m)d(m), (32) e f,p (m) y p (m) G D q (m)ĥ f,qp (m ) q y p (m) G D(m)ĥ f,p (m ), (33) ĥ f,p (m) (34) ĥ f,p (m ) + µg 2 Ŝ xx (m)dh (m)e f,p (m), p, 2,...,, the subscript f stands for foreground, λ f, 0 < λ f <, is an exponential forgetting factor, µ µ ( λ f ),0< µ 2, is the adaptation step size, and G 2 FW 2 F, IL L 0 W 2 L L. 0 L L 0 L L The decision variable should be estimated as follows: ˆξ 2 fd (m) trŝh xy (m)ŝ xx (m)ŝ xy (m) ˆR yy (m) trŝh xy (m)ĥ b (m) ˆR yy (m), (35) Vol. 3, No. 2, March-April 2002

6 J. Benesty, T. Gänsler Ŝ xy (m) λ b Ŝ xy (m ) + ( λ b )D H (m)y(m) (36) e b,p (m) y p (m) G D(m)ĥ b,p (m ) (37) ĥ b,p (m) (38) ĥ b,p (m ) + ( λ b )G 2 Ŝ xx (m)dh (m)e b,p (m) p, 2,...,, ˆR yy (m) λ b ˆR yy (m ) + ( λ b)y H (m)y(m), (39) subscript b stands for background, and λ b,0<λ b <, is an exponential forgetting factor. We must choose λ b <λ f (for a faster tracking of the background filter used in the DTD) in order that the DTD alerts the foreground filter before it diverges. Table summarizes the combination of both the multichannel acoustic echo canceler and the multichannel DTD in the frequency domain. Table : A frequency-domain multichannel acoustic echo canceler combined with a multichannel double-talk detector., number of microphones 0 <µ 2, normalized adaptation step size λ f,λ b, exponential windows T, threshold G FW F 0L L 0 W L L 0 L L I L L G 2 FW 2 F IL L 0 W 2 L L 0 L L 0 L L Spectral and correlation estimation Ŝ xx (m) λ f Ŝ xx (m ) + ( λ f )D H (m)d(m) Ŝ xy (m) λ b Ŝ xy (m ) + ( λ b )D H (m)y(m) ˆR yy (m) λ b ˆR yy (m ) + ( λ b)y H (m)y(m) 6 CONCLUSIONS In the literature, a lot of attention has been given to the identification part of multichannel acoustic echo cancellation but little has been done for near-end speech detection. Multichannel double-talk detection is not trivial and has to be investigated more deeply. In this paper, some possibilities to handle this problem have been proposed, centered on the use of the normalized cross-correlation matrix as the test statistic. This is a natural extension of the normalized cross-correlation vector used in the single-channel case which has been previously discussed 5. The advantage of the proposed approach is that all the microphone signals are taken into account in one decision variable. We also applied this method in the frequency domain and obtained a pseudo-coherence matrix. Finally, an efficient way to combine acoustic echo cancellation and double-talk detection in the frequency domain was proposed. Obviously, for future research there are still lots of things to study. One important aspect of this work that has to be carefully and judiciously investigated is an objective way to evaluate the different multichannel DTD algorithms and compare their performance. Multichannel double-talk detector (background filter) e b,p (m) y p (m) G D(m)ĥ b,p (m ) ĥ b,p (m) ĥ b,p (m ) + µ ( λ b )G 2 Ŝ xx (m)dh (m)e b,p (m) p, 2,..., ˆξ fd 2 (m) trŝh xy (m)ĥ b (m) ˆR yy (m) ˆξ fd (m) < T, double-talk, µ 0 ˆξ fd (m) T, no double-talk, µ µ ( λ f ) Multichannel acoustic echo canceler (foreground filter) e f,p (m) y p (m) G D(m)ĥ f,p (m ) ĥ f,p (m) ĥ f,p (m ) + µg 2 Ŝ xx (m)dh (m)e f,p (m) p, 2,..., ACKNOWLEDGMENT The authors would like to thank Dennis R. Morgan for his helpful suggestions. Manuscript received on December 9, 200. REFERENCES M. M. Sondhi, D. R. Morgan, and J. L. Hall. Stereophonic acoustic echo cancellation An overview of the fundamental problem. IEEE Signal rocessing Lett., Vol. 2, pages 48-5, Aug ETT

7 A Multichannel Acoustic Echo Canceler Double-Talk Detector Based on a Normalized Cross-Correlation Matrix 2 J. Benesty, D. R. Morgan, and M. M. Sondhi. A better understanding and an improved solution to the specific problems of stereophonic acoustic echo cancellation. IEEE Trans. Speech Audio rocessing, Vol. 6, pages 56-65, Mar J. H. Cho, D. R. Morgan, and J. Benesty. An objective technique for evaluating doubletalk detectors in acoustic echo cancelers. IEEE Trans. Speech Audio rocessing, Vol. 7, pages , Nov C. Breining,. Dreiseitel, E. Hänsler, A. Mader, B. Nitsch, H. uder, T. Schertler, G. Schmidt, and J. Tilp. Acoustic echo control, an application of very-high-order adaptive filters. IEEE Signal rocessing Mag., Vol. 6, pages 42-69, July J. Benesty, D. R. Morgan, and J. H. Cho. A new class of doubletalk detectors based on cross-correlation. IEEE Trans. Speech Audio rocessing, Vol. 8, pages 68-72, Mar J. Benesty, T. Gänsler, D. R. Morgan, M. M. Sondhi, and S. L. Gay. Advances in Network and Acoustic Echo Cancellation. Springer-Verlag, Berlin, Germany, T. Gänsler and J. Benesty. A frequency-domain double-talk detector based on a normalized cross-correlation vector. Signal rocessing, Vol. 8, pages , Aug M. Ali. Stereophonic echo cancellation system using timevarying all-pass filtering for signal decorrelation. In roc. IEEE ICASS, pages , A. Gilloire and V. Turbin. Using auditory properties to improve the behavior of stereophonic acoustic echo cancellers. In roc. IEEE ICASS, pages , T. Gänsler and. Eneroth. Influence of audio coding on stereophonic acoustic echo cancellation. In roc. IEEE ICASS, pages , 998. Y. Joncour and A. Sugiyama. A stereo echo canceller with pre-processing for correct echo path identification. In roc. IEEE ICASS, pages , S. Shimauchi and S. Makino. Stereo projection echo canceller with true echo path estimation. In roc. IEEE ICASS, pages , S. Shimauchi, Y. Haneda, S. Makino, and Y. Kaneda. New configuration for a stereo echo canceller with nonlinear preprocessing. In roc. IEEE ICASS, pages , K. Ochiai, T. Araseki, and T. Ogihara. Echo canceler with two echo path models. IEEE Trans. Commun., Vol. 25, pages , June J. Benesty and D. R. Morgan. Multi-channel frequencydomain adaptive filtering. In Acoustic Signal rocessing for Telecommunication, S. L. Gay and J. Benesty, Eds., Kluwer Academic ublishers, ch. 7, Vol. 3, No. 2, March-April 2002

8 J. Benesty, T. Gänsler 02 ETT

New Insights Into the Stereophonic Acoustic Echo Cancellation Problem and an Adaptive Nonlinearity Solution

New Insights Into the Stereophonic Acoustic Echo Cancellation Problem and an Adaptive Nonlinearity Solution IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, VOL 10, NO 5, JULY 2002 257 New Insights Into the Stereophonic Acoustic Echo Cancellation Problem and an Adaptive Nonlinearity Solution Tomas Gänsler,

More information

Samira A. Mahdi University of Babylon/College of Science/Physics Dept. Iraq/Babylon

Samira A. Mahdi University of Babylon/College of Science/Physics Dept. Iraq/Babylon Echo Cancelation Using Least Mean Square (LMS) Algorithm Samira A. Mahdi University of Babylon/College of Science/Physics Dept. Iraq/Babylon Abstract The aim of this work is to investigate methods for

More information

Chapter 2 Fundamentals of Adaptive Filter Theory

Chapter 2 Fundamentals of Adaptive Filter Theory Chapter 2 Fundamentals of Adaptive Filter Theory In this chapter we will treat some fundamentals of the adaptive filtering theory highlighting the system identification problem We will introduce a signal

More information

Acoustic MIMO Signal Processing

Acoustic MIMO Signal Processing Yiteng Huang Jacob Benesty Jingdong Chen Acoustic MIMO Signal Processing With 71 Figures Ö Springer Contents 1 Introduction 1 1.1 Acoustic MIMO Signal Processing 1 1.2 Organization of the Book 4 Part I

More information

BLOCK-BASED MULTICHANNEL TRANSFORM-DOMAIN ADAPTIVE FILTERING

BLOCK-BASED MULTICHANNEL TRANSFORM-DOMAIN ADAPTIVE FILTERING BLOCK-BASED MULTICHANNEL TRANSFORM-DOMAIN ADAPTIVE FILTERING Sascha Spors, Herbert Buchner, and Karim Helwani Deutsche Telekom Laboratories, Technische Universität Berlin, Ernst-Reuter-Platz 7, 10587 Berlin,

More information

FAST IMPLEMENTATION OF A SUBBAND ADAPTIVE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION

FAST IMPLEMENTATION OF A SUBBAND ADAPTIVE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION Journal of ELECTRICAL ENGINEERING, VOL. 55, NO. 5-6, 24, 113 121 FAST IMPLEMENTATION OF A SUBBAND ADAPTIVE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION Khaled Mayyas The block subband adaptive algorithm in

More information

Efficient Use Of Sparse Adaptive Filters

Efficient Use Of Sparse Adaptive Filters Efficient Use Of Sparse Adaptive Filters Andy W.H. Khong and Patrick A. Naylor Department of Electrical and Electronic Engineering, Imperial College ondon Email: {andy.khong, p.naylor}@imperial.ac.uk Abstract

More information

1 Introduction. Bell Laboratories TECHNICAL MEMORANDUM

1 Introduction. Bell Laboratories TECHNICAL MEMORANDUM Lucent Technologies Bell Labs Innovations L Bell Laboratories subject: General Derivation of Frequency-Domain Adaptive Filtering Work Project No. MA30033002 File Case 38794-43 date: May 16, 2000 from:

More information

Simple and efficient solutions to the problems associated with acoustic echo cancellation

Simple and efficient solutions to the problems associated with acoustic echo cancellation Scholars' Mine Doctoral Dissertations Student Research & Creative Works Summer 2007 Simple and efficient solutions to the problems associated with acoustic echo cancellation Asif Iqbal Mohammad Follow

More information

A SPARSENESS CONTROLLED PROPORTIONATE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION

A SPARSENESS CONTROLLED PROPORTIONATE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION 6th European Signal Processing Conference (EUSIPCO 28), Lausanne, Switzerland, August 25-29, 28, copyright by EURASIP A SPARSENESS CONTROLLED PROPORTIONATE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION Pradeep

More information

ADAPTIVE ALGORITHMS FOR MIMO ACOUSTIC ECHO CANCELLATION

ADAPTIVE ALGORITHMS FOR MIMO ACOUSTIC ECHO CANCELLATION Chapter 1 ADAPTIVE ALGORITHMS FOR MIMO ACOUSTIC ECHO CANCELLATION Jacob Benesty Université duquébec, INRS-EMT benesty@inrs-emtuquebecca Tomas Gänsler Agere Systems gaensler@agerecom Yiteng (Arden) Huang

More information

EE482: Digital Signal Processing Applications

EE482: Digital Signal Processing Applications Professor Brendan Morris, SEB 3216, brendan.morris@unlv.edu EE482: Digital Signal Processing Applications Spring 2014 TTh 14:30-15:45 CBC C222 Lecture 11 Adaptive Filtering 14/03/04 http://www.ee.unlv.edu/~b1morris/ee482/

More information

NOISE ROBUST RELATIVE TRANSFER FUNCTION ESTIMATION. M. Schwab, P. Noll, and T. Sikora. Technical University Berlin, Germany Communication System Group

NOISE ROBUST RELATIVE TRANSFER FUNCTION ESTIMATION. M. Schwab, P. Noll, and T. Sikora. Technical University Berlin, Germany Communication System Group NOISE ROBUST RELATIVE TRANSFER FUNCTION ESTIMATION M. Schwab, P. Noll, and T. Sikora Technical University Berlin, Germany Communication System Group Einsteinufer 17, 1557 Berlin (Germany) {schwab noll

More information

COMPARISON OF MULTIPLE-MICROPHONE AND SINGLE-LOUDSPEAKER ADAPTIVE FEEDBACK/ECHO CANCELLATION SYSTEMS

COMPARISON OF MULTIPLE-MICROPHONE AND SINGLE-LOUDSPEAKER ADAPTIVE FEEDBACK/ECHO CANCELLATION SYSTEMS 19th European Signal rocessing Conference EUSICO 211) Barcelona, Spain, August 29 - September 2, 211 COMARISON OF MULTILE-MICROHONE AND SINGLE-LOUDSEAKER ADATIVE FEEDBACK/ECHO CANCELLATION SYSTEMS Meng

More information

Sparseness-Controlled Affine Projection Algorithm for Echo Cancelation

Sparseness-Controlled Affine Projection Algorithm for Echo Cancelation Sparseness-Controlled Affine Projection Algorithm for Echo Cancelation ei iao and Andy W. H. Khong E-mail: liao38@e.ntu.edu.sg E-mail: andykhong@ntu.edu.sg Nanyang Technological University, Singapore Abstract

More information

A Flexible ICA-Based Method for AEC Without Requiring Double-Talk Detection

A Flexible ICA-Based Method for AEC Without Requiring Double-Talk Detection APSIPA ASC 2011 Xi an A Flexible ICA-Based Method for AEC Without Requiring Double-Talk Detection Marko Kanadi, Muhammad Tahir Akhtar, Wataru Mitsuhashi Department of Information and Communication Engineering,

More information

Adaptive Stereo Acoustic Echo Cancelation in reverberant environments. Amos Schreibman

Adaptive Stereo Acoustic Echo Cancelation in reverberant environments. Amos Schreibman Adaptive Stereo Acoustic Echo Cancelation in reverberant environments Amos Schreibman Adaptive Stereo Acoustic Echo Cancelation in reverberant environments Research Thesis As Partial Fulfillment of the

More information

STATISTICAL MODELLING OF MULTICHANNEL BLIND SYSTEM IDENTIFICATION ERRORS. Felicia Lim, Patrick A. Naylor

STATISTICAL MODELLING OF MULTICHANNEL BLIND SYSTEM IDENTIFICATION ERRORS. Felicia Lim, Patrick A. Naylor STTISTICL MODELLING OF MULTICHNNEL BLIND SYSTEM IDENTIFICTION ERRORS Felicia Lim, Patrick. Naylor Dept. of Electrical and Electronic Engineering, Imperial College London, UK {felicia.lim6, p.naylor}@imperial.ac.uk

More information

MITIGATING UNCORRELATED PERIODIC DISTURBANCE IN NARROWBAND ACTIVE NOISE CONTROL SYSTEMS

MITIGATING UNCORRELATED PERIODIC DISTURBANCE IN NARROWBAND ACTIVE NOISE CONTROL SYSTEMS 17th European Signal Processing Conference (EUSIPCO 29) Glasgow, Scotland, August 24-28, 29 MITIGATING UNCORRELATED PERIODIC DISTURBANCE IN NARROWBAND ACTIVE NOISE CONTROL SYSTEMS Muhammad Tahir AKHTAR

More information

A. Ideal model. where i is an attenuation factor due to propagation effects,

A. Ideal model. where i is an attenuation factor due to propagation effects, Adaptive eigenvalue decomposition algorithm for passive acoustic source localization Jacob Benesty Bell Laboratories, Lucent Technologies, 700 Mountain Avenue, Murray Hill, New Jersey 07974-0636 Received

More information

EE482: Digital Signal Processing Applications

EE482: Digital Signal Processing Applications Professor Brendan Morris, SEB 3216, brendan.morris@unlv.edu EE482: Digital Signal Processing Applications Spring 2014 TTh 14:30-15:45 CBC C222 Lecture 11 Adaptive Filtering 14/03/04 http://www.ee.unlv.edu/~b1morris/ee482/

More information

An overview on optimized NLMS algorithms for acoustic echo cancellation

An overview on optimized NLMS algorithms for acoustic echo cancellation Paleologu et al. EURASIP Journal on Advances in Signal Processing (015) 015:97 DOI 10.1186/s13634-015-083-1 REVIEW Open Access An overview on optimized NLMS algorithms for acoustic echo cancellation Constantin

More information

Alternative Learning Algorithm for Stereophonic Acoustic Echo Canceller without Pre-Processing

Alternative Learning Algorithm for Stereophonic Acoustic Echo Canceller without Pre-Processing 1958 PAPER Special Section on Digital Signal Processing Alternative Learning Algorithm or Stereophonic Acoustic Echo Canceller without Pre-Processing Akihiro HIRANO a),kenjinakayama, Memers, Daisuke SOMEDA,

More information

Sound Listener s perception

Sound Listener s perception Inversion of Loudspeaker Dynamics by Polynomial LQ Feedforward Control Mikael Sternad, Mathias Johansson and Jonas Rutstrom Abstract- Loudspeakers always introduce linear and nonlinear distortions in a

More information

SINGLE-CHANNEL SPEECH PRESENCE PROBABILITY ESTIMATION USING INTER-FRAME AND INTER-BAND CORRELATIONS

SINGLE-CHANNEL SPEECH PRESENCE PROBABILITY ESTIMATION USING INTER-FRAME AND INTER-BAND CORRELATIONS 204 IEEE International Conference on Acoustic, Speech and Signal Processing (ICASSP) SINGLE-CHANNEL SPEECH PRESENCE PROBABILITY ESTIMATION USING INTER-FRAME AND INTER-BAND CORRELATIONS Hajar Momeni,2,,

More information

System Identification and Adaptive Filtering in the Short-Time Fourier Transform Domain

System Identification and Adaptive Filtering in the Short-Time Fourier Transform Domain System Identification and Adaptive Filtering in the Short-Time Fourier Transform Domain Electrical Engineering Department Technion - Israel Institute of Technology Supervised by: Prof. Israel Cohen Outline

More information

Title without the persistently exciting c. works must be obtained from the IEE

Title without the persistently exciting c.   works must be obtained from the IEE Title Exact convergence analysis of adapt without the persistently exciting c Author(s) Sakai, H; Yang, JM; Oka, T Citation IEEE TRANSACTIONS ON SIGNAL 55(5): 2077-2083 PROCESS Issue Date 2007-05 URL http://hdl.handle.net/2433/50544

More information

A Low-Distortion Noise Canceller and Its Learning Algorithm in Presence of Crosstalk

A Low-Distortion Noise Canceller and Its Learning Algorithm in Presence of Crosstalk 414 IEICE TRANS. FUNDAMENTALS, VOL.E84 A, NO.2 FEBRUARY 2001 PAPER Special Section on Noise Cancellation Reduction Techniques A Low-Distortion Noise Canceller Its Learning Algorithm in Presence of Crosstalk

More information

A new method for a nonlinear acoustic echo cancellation system

A new method for a nonlinear acoustic echo cancellation system A new method for a nonlinear acoustic echo cancellation system Tuan Van Huynh Department of Physics and Computer Science, Faculty of Physics and Engineering Physics, University of Science, Vietnam National

More information

A SPARSENESS CONTROLLED PROPORTIONATE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION

A SPARSENESS CONTROLLED PROPORTIONATE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION A SPARSENESS CONTROLLED PROPORTIONATE ALGORITHM FOR ACOUSTIC ECHO CANCELLATION Pradeep Loganathan, Andy W.H. Khong, Patrick A. Naylor pradeep.loganathan@ic.ac.uk, andykhong@ntu.edu.sg, p.naylorg@ic.ac.uk

More information

Wiener Filtering. EE264: Lecture 12

Wiener Filtering. EE264: Lecture 12 EE264: Lecture 2 Wiener Filtering In this lecture we will take a different view of filtering. Previously, we have depended on frequency-domain specifications to make some sort of LP/ BP/ HP/ BS filter,

More information

A Local Model of Relative Transfer Functions Involving Sparsity

A Local Model of Relative Transfer Functions Involving Sparsity A Local Model of Relative Transfer Functions Involving Sparsity Zbyněk Koldovský 1, Jakub Janský 1, and Francesco Nesta 2 1 Faculty of Mechatronics, Informatics and Interdisciplinary Studies Technical

More information

EIGENFILTERS FOR SIGNAL CANCELLATION. Sunil Bharitkar and Chris Kyriakakis

EIGENFILTERS FOR SIGNAL CANCELLATION. Sunil Bharitkar and Chris Kyriakakis EIGENFILTERS FOR SIGNAL CANCELLATION Sunil Bharitkar and Chris Kyriakakis Immersive Audio Laboratory University of Southern California Los Angeles. CA 9. USA Phone:+1-13-7- Fax:+1-13-7-51, Email:ckyriak@imsc.edu.edu,bharitka@sipi.usc.edu

More information

Dominant Pole Localization of FxLMS Adaptation Process in Active Noise Control

Dominant Pole Localization of FxLMS Adaptation Process in Active Noise Control APSIPA ASC 20 Xi an Dominant Pole Localization of FxLMS Adaptation Process in Active Noise Control Iman Tabatabaei Ardekani, Waleed H. Abdulla The University of Auckland, Private Bag 9209, Auckland, New

More information

FUNDAMENTAL LIMITATION OF FREQUENCY DOMAIN BLIND SOURCE SEPARATION FOR CONVOLVED MIXTURE OF SPEECH

FUNDAMENTAL LIMITATION OF FREQUENCY DOMAIN BLIND SOURCE SEPARATION FOR CONVOLVED MIXTURE OF SPEECH FUNDAMENTAL LIMITATION OF FREQUENCY DOMAIN BLIND SOURCE SEPARATION FOR CONVOLVED MIXTURE OF SPEECH Shoko Araki Shoji Makino Ryo Mukai Tsuyoki Nishikawa Hiroshi Saruwatari NTT Communication Science Laboratories

More information

Nonlinear Echo Cancellation using Generalized Power Filters

Nonlinear Echo Cancellation using Generalized Power Filters Nonlinear Echo Cancellation using Generalized Power Filters Jiri Malek and Zbyněk Koldovský Faculty of Mechatronics, Informatics, and Interdisciplinary Studies, Technical University of Liberec, Studentská

More information

Research Letter An Algorithm to Generate Representations of System Identification Errors

Research Letter An Algorithm to Generate Representations of System Identification Errors Research Letters in Signal Processing Volume 008, Article ID 5991, 4 pages doi:10.1155/008/5991 Research Letter An Algorithm to Generate Representations of System Identification Errors Wancheng Zhang and

More information

3. ESTIMATION OF SIGNALS USING A LEAST SQUARES TECHNIQUE

3. ESTIMATION OF SIGNALS USING A LEAST SQUARES TECHNIQUE 3. ESTIMATION OF SIGNALS USING A LEAST SQUARES TECHNIQUE 3.0 INTRODUCTION The purpose of this chapter is to introduce estimators shortly. More elaborated courses on System Identification, which are given

More information

Generalized Sidelobe Canceller and MVDR Power Spectrum Estimation. Bhaskar D Rao University of California, San Diego

Generalized Sidelobe Canceller and MVDR Power Spectrum Estimation. Bhaskar D Rao University of California, San Diego Generalized Sidelobe Canceller and MVDR Power Spectrum Estimation Bhaskar D Rao University of California, San Diego Email: brao@ucsd.edu Reference Books 1. Optimum Array Processing, H. L. Van Trees 2.

More information

ON THE NOISE REDUCTION PERFORMANCE OF THE MVDR BEAMFORMER IN NOISY AND REVERBERANT ENVIRONMENTS

ON THE NOISE REDUCTION PERFORMANCE OF THE MVDR BEAMFORMER IN NOISY AND REVERBERANT ENVIRONMENTS 0 IEEE International Conference on Acoustic, Speech and Signal Processing (ICASSP) ON THE NOISE REDUCTION PERFORANCE OF THE VDR BEAFORER IN NOISY AND REVERBERANT ENVIRONENTS Chao Pan, Jingdong Chen, and

More information

MULTIPLE-CHANNEL DETECTION IN ACTIVE SENSING. Kaitlyn Beaudet and Douglas Cochran

MULTIPLE-CHANNEL DETECTION IN ACTIVE SENSING. Kaitlyn Beaudet and Douglas Cochran MULTIPLE-CHANNEL DETECTION IN ACTIVE SENSING Kaitlyn Beaudet and Douglas Cochran School of Electrical, Computer and Energy Engineering Arizona State University, Tempe AZ 85287-576 USA ABSTRACT The problem

More information

Reduced-cost combination of adaptive filters for acoustic echo cancellation

Reduced-cost combination of adaptive filters for acoustic echo cancellation Reduced-cost combination of adaptive filters for acoustic echo cancellation Luis A. Azpicueta-Ruiz and Jerónimo Arenas-García Dept. Signal Theory and Communications, Universidad Carlos III de Madrid Leganés,

More information

Convolutive Transfer Function Generalized Sidelobe Canceler

Convolutive Transfer Function Generalized Sidelobe Canceler IEEE TRANSACTIONS ON AUDIO, SPEECH AND LANGUAGE PROCESSING, VOL. XX, NO. Y, MONTH 8 Convolutive Transfer Function Generalized Sidelobe Canceler Ronen Talmon, Israel Cohen, Senior Member, IEEE, and Sharon

More information

INTRODUCTION Noise is present in many situations of daily life for ex: Microphones will record noise and speech. Goal: Reconstruct original signal Wie

INTRODUCTION Noise is present in many situations of daily life for ex: Microphones will record noise and speech. Goal: Reconstruct original signal Wie WIENER FILTERING Presented by N.Srikanth(Y8104060), M.Manikanta PhaniKumar(Y8104031). INDIAN INSTITUTE OF TECHNOLOGY KANPUR Electrical Engineering dept. INTRODUCTION Noise is present in many situations

More information

Implementation of the LMS and NLMS algorithms for Acoustic Echo Cancellation in teleconference system using MATLAB

Implementation of the LMS and NLMS algorithms for Acoustic Echo Cancellation in teleconference system using MATLAB School of Mathematics and Systems Engineering Reports from MSI - Rapporter från MSI Implementation of the LMS and NLMS algorithms for Acoustic Echo Cancellation in teleconference system using MALAB Hung

More information

A low intricacy variable step-size partial update adaptive algorithm for Acoustic Echo Cancellation USNRao

A low intricacy variable step-size partial update adaptive algorithm for Acoustic Echo Cancellation USNRao ISSN: 77-3754 International Journal of Engineering and Innovative echnology (IJEI Volume 1, Issue, February 1 A low intricacy variable step-size partial update adaptive algorithm for Acoustic Echo Cancellation

More information

Adaptive Filtering Part II

Adaptive Filtering Part II Adaptive Filtering Part II In previous Lecture we saw that: Setting the gradient of cost function equal to zero, we obtain the optimum values of filter coefficients: (Wiener-Hopf equation) Adaptive Filtering,

More information

Adaptive Filter Theory

Adaptive Filter Theory 0 Adaptive Filter heory Sung Ho Cho Hanyang University Seoul, Korea (Office) +8--0-0390 (Mobile) +8-10-541-5178 dragon@hanyang.ac.kr able of Contents 1 Wiener Filters Gradient Search by Steepest Descent

More information

Adaptive MMSE Equalizer with Optimum Tap-length and Decision Delay

Adaptive MMSE Equalizer with Optimum Tap-length and Decision Delay Adaptive MMSE Equalizer with Optimum Tap-length and Decision Delay Yu Gong, Xia Hong and Khalid F. Abu-Salim School of Systems Engineering The University of Reading, Reading RG6 6AY, UK E-mail: {y.gong,x.hong,k.f.abusalem}@reading.ac.uk

More information

Estimating Correlation Coefficient Between Two Complex Signals Without Phase Observation

Estimating Correlation Coefficient Between Two Complex Signals Without Phase Observation Estimating Correlation Coefficient Between Two Complex Signals Without Phase Observation Shigeki Miyabe 1B, Notubaka Ono 2, and Shoji Makino 1 1 University of Tsukuba, 1-1-1 Tennodai, Tsukuba, Ibaraki

More information

THE INSTRUMENTAL VARIABLE MULTICHANNEL FAP-RLS AND FAP ALGORITHMS. Mahdi Triki, Dirk T.M. Slock

THE INSTRUMENTAL VARIABLE MULTICHANNEL FAP-RLS AND FAP ALGORITHMS. Mahdi Triki, Dirk T.M. Slock THE INSTRUMENTAL VARIABLE MULTICHANNEL FAP-RLS AND FAP ALGORITHMS Mahdi Trii, Dir T.M. Sloc Eurecom Institute 2229 route des Crêtes, B.P. 93, 06904 Sophia Antipolis Cedex, FRANCE Email: {trii,sloc}@eurecom.fr

More information

A METHOD OF ICA IN TIME-FREQUENCY DOMAIN

A METHOD OF ICA IN TIME-FREQUENCY DOMAIN A METHOD OF ICA IN TIME-FREQUENCY DOMAIN Shiro Ikeda PRESTO, JST Hirosawa 2-, Wako, 35-98, Japan Shiro.Ikeda@brain.riken.go.jp Noboru Murata RIKEN BSI Hirosawa 2-, Wako, 35-98, Japan Noboru.Murata@brain.riken.go.jp

More information

Department of Electrical and Electronic Engineering

Department of Electrical and Electronic Engineering Imperial College London Department of Electrical and Electronic Engineering Final Year Project Report 27 Project Title: Student: Course: Adaptive Echo Cancellation Pradeep Loganathan ISE4 Project Supervisor:

More information

DETECTION theory deals primarily with techniques for

DETECTION theory deals primarily with techniques for ADVANCED SIGNAL PROCESSING SE Optimum Detection of Deterministic and Random Signals Stefan Tertinek Graz University of Technology turtle@sbox.tugraz.at Abstract This paper introduces various methods for

More information

Massachusetts Institute of Technology

Massachusetts Institute of Technology Massachusetts Institute of Technology Department of Electrical Engineering and Computer Science 6.011: Introduction to Communication, Control and Signal Processing QUIZ, April 1, 010 QUESTION BOOKLET Your

More information

IS NEGATIVE STEP SIZE LMS ALGORITHM STABLE OPERATION POSSIBLE?

IS NEGATIVE STEP SIZE LMS ALGORITHM STABLE OPERATION POSSIBLE? IS NEGATIVE STEP SIZE LMS ALGORITHM STABLE OPERATION POSSIBLE? Dariusz Bismor Institute of Automatic Control, Silesian University of Technology, ul. Akademicka 16, 44-100 Gliwice, Poland, e-mail: Dariusz.Bismor@polsl.pl

More information

An Adaptive Sensor Array Using an Affine Combination of Two Filters

An Adaptive Sensor Array Using an Affine Combination of Two Filters An Adaptive Sensor Array Using an Affine Combination of Two Filters Tõnu Trump Tallinn University of Technology Department of Radio and Telecommunication Engineering Ehitajate tee 5, 19086 Tallinn Estonia

More information

REAL-TIME TIME-FREQUENCY BASED BLIND SOURCE SEPARATION. Scott Rickard, Radu Balan, Justinian Rosca. Siemens Corporate Research Princeton, NJ 08540

REAL-TIME TIME-FREQUENCY BASED BLIND SOURCE SEPARATION. Scott Rickard, Radu Balan, Justinian Rosca. Siemens Corporate Research Princeton, NJ 08540 REAL-TIME TIME-FREQUENCY BASED BLIND SOURCE SEPARATION Scott Rickard, Radu Balan, Justinian Rosca Siemens Corporate Research Princeton, NJ 84 fscott.rickard,radu.balan,justinian.roscag@scr.siemens.com

More information

Advanced Digital Signal Processing -Introduction

Advanced Digital Signal Processing -Introduction Advanced Digital Signal Processing -Introduction LECTURE-2 1 AP9211- ADVANCED DIGITAL SIGNAL PROCESSING UNIT I DISCRETE RANDOM SIGNAL PROCESSING Discrete Random Processes- Ensemble Averages, Stationary

More information

TECHNICAL MEMORANDUM

TECHNICAL MEMORANDUM Bell Laboratories subject: Proposals for Dynamic Resource Allocation for Network Echo Cancellation Work Project No. 31152-273 File Case 38794-43 date: March 9, 1999 from: Tomas Gaensler Lund University

More information

IMPROVED MULTI-MICROPHONE NOISE REDUCTION PRESERVING BINAURAL CUES

IMPROVED MULTI-MICROPHONE NOISE REDUCTION PRESERVING BINAURAL CUES IMPROVED MULTI-MICROPHONE NOISE REDUCTION PRESERVING BINAURAL CUES Andreas I. Koutrouvelis Richard C. Hendriks Jesper Jensen Richard Heusdens Circuits and Systems (CAS) Group, Delft University of Technology,

More information

Least Squares with Examples in Signal Processing 1. 2 Overdetermined equations. 1 Notation. The sum of squares of x is denoted by x 2 2, i.e.

Least Squares with Examples in Signal Processing 1. 2 Overdetermined equations. 1 Notation. The sum of squares of x is denoted by x 2 2, i.e. Least Squares with Eamples in Signal Processing Ivan Selesnick March 7, 3 NYU-Poly These notes address (approimate) solutions to linear equations by least squares We deal with the easy case wherein the

More information

A STATE-SPACE PARTITIONED-BLOCK ADAPTIVE FILTER FOR ECHO CANCELLATION USING INTER-BAND CORRELATIONS IN THE KALMAN GAIN COMPUTATION

A STATE-SPACE PARTITIONED-BLOCK ADAPTIVE FILTER FOR ECHO CANCELLATION USING INTER-BAND CORRELATIONS IN THE KALMAN GAIN COMPUTATION A STATE-SPACE PARTITIONED-BLOCK ADAPTIVE FILTER FOR ECHO CANCELLATION USING INTER-BAND CORRELATIONS IN THE KALAN GAIN COPUTATION aría Luis Valero, Edwin abande and Emanuël A. P. Habets International Audio

More information

Assesment of the efficiency of the LMS algorithm based on spectral information

Assesment of the efficiency of the LMS algorithm based on spectral information Assesment of the efficiency of the algorithm based on spectral information (Invited Paper) Aaron Flores and Bernard Widrow ISL, Department of Electrical Engineering, Stanford University, Stanford CA, USA

More information

Machine Learning. A Bayesian and Optimization Perspective. Academic Press, Sergios Theodoridis 1. of Athens, Athens, Greece.

Machine Learning. A Bayesian and Optimization Perspective. Academic Press, Sergios Theodoridis 1. of Athens, Athens, Greece. Machine Learning A Bayesian and Optimization Perspective Academic Press, 2015 Sergios Theodoridis 1 1 Dept. of Informatics and Telecommunications, National and Kapodistrian University of Athens, Athens,

More information

Reduced-Rank Multi-Antenna Cyclic Wiener Filtering for Interference Cancellation

Reduced-Rank Multi-Antenna Cyclic Wiener Filtering for Interference Cancellation Reduced-Rank Multi-Antenna Cyclic Wiener Filtering for Interference Cancellation Hong Zhang, Ali Abdi and Alexander Haimovich Center for Wireless Communications and Signal Processing Research Department

More information

USING STATISTICAL ROOM ACOUSTICS FOR ANALYSING THE OUTPUT SNR OF THE MWF IN ACOUSTIC SENSOR NETWORKS. Toby Christian Lawin-Ore, Simon Doclo

USING STATISTICAL ROOM ACOUSTICS FOR ANALYSING THE OUTPUT SNR OF THE MWF IN ACOUSTIC SENSOR NETWORKS. Toby Christian Lawin-Ore, Simon Doclo th European Signal Processing Conference (EUSIPCO 1 Bucharest, Romania, August 7-31, 1 USING STATISTICAL ROOM ACOUSTICS FOR ANALYSING THE OUTPUT SNR OF THE MWF IN ACOUSTIC SENSOR NETWORKS Toby Christian

More information

Optimal Time-Domain Noise Reduction Filters

Optimal Time-Domain Noise Reduction Filters SpringerBriefs in Electrical and Computer Engineering Optimal Time-Domain Noise Reduction Filters A Theoretical Study Bearbeitet von Jacob Benesty, Jingdong Chen 1. Auflage 2011. Taschenbuch. vii, 79 S.

More information

GAUSSIANIZATION METHOD FOR IDENTIFICATION OF MEMORYLESS NONLINEAR AUDIO SYSTEMS

GAUSSIANIZATION METHOD FOR IDENTIFICATION OF MEMORYLESS NONLINEAR AUDIO SYSTEMS GAUSSIANIATION METHOD FOR IDENTIFICATION OF MEMORYLESS NONLINEAR AUDIO SYSTEMS I. Marrakchi-Mezghani (1),G. Mahé (2), M. Jaïdane-Saïdane (1), S. Djaziri-Larbi (1), M. Turki-Hadj Alouane (1) (1) Unité Signaux

More information

JOINT DEREVERBERATION AND NOISE REDUCTION BASED ON ACOUSTIC MULTICHANNEL EQUALIZATION. Ina Kodrasi, Simon Doclo

JOINT DEREVERBERATION AND NOISE REDUCTION BASED ON ACOUSTIC MULTICHANNEL EQUALIZATION. Ina Kodrasi, Simon Doclo JOINT DEREVERBERATION AND NOISE REDUCTION BASED ON ACOUSTIC MULTICHANNEL EQUALIZATION Ina Kodrasi, Simon Doclo University of Oldenburg, Department of Medical Physics and Acoustics, and Cluster of Excellence

More information

NONLINEAR ACOUSTIC ECHO CANCELLATION

NONLINEAR ACOUSTIC ECHO CANCELLATION NONLINEAR ACOUSTIC ECHO CANCELLATION A Thesis Presented to The Academic Faculty by Kun Shi In Partial Fulfillment of the Requirements for the Degree Doctor of Philosophy in the School of Electrical and

More information

IN real-world adaptive filtering applications, severe impairments

IN real-world adaptive filtering applications, severe impairments 1878 IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 56, NO. 5, MAY 2008 A New Robust Variable Step-Size NLMS Algorithm Leonardo Rey Vega, Hernán Rey, Jacob Benesty, Senior Member, IEEE, and Sara Tressens

More information

System Identification in the Short-Time Fourier Transform Domain

System Identification in the Short-Time Fourier Transform Domain System Identification in the Short-Time Fourier Transform Domain Yekutiel Avargel System Identification in the Short-Time Fourier Transform Domain Research Thesis As Partial Fulfillment of the Requirements

More information

GENERALIZED DEFLATION ALGORITHMS FOR THE BLIND SOURCE-FACTOR SEPARATION OF MIMO-FIR CHANNELS. Mitsuru Kawamoto 1,2 and Yujiro Inouye 1

GENERALIZED DEFLATION ALGORITHMS FOR THE BLIND SOURCE-FACTOR SEPARATION OF MIMO-FIR CHANNELS. Mitsuru Kawamoto 1,2 and Yujiro Inouye 1 GENERALIZED DEFLATION ALGORITHMS FOR THE BLIND SOURCE-FACTOR SEPARATION OF MIMO-FIR CHANNELS Mitsuru Kawamoto,2 and Yuiro Inouye. Dept. of Electronic and Control Systems Engineering, Shimane University,

More information

Symmetry Methods for Differential and Difference Equations. Peter Hydon

Symmetry Methods for Differential and Difference Equations. Peter Hydon Lecture 2: How to find Lie symmetries Symmetry Methods for Differential and Difference Equations Peter Hydon University of Kent Outline 1 Reduction of order for ODEs and O Es 2 The infinitesimal generator

More information

Performance Analysis and Enhancements of Adaptive Algorithms and Their Applications

Performance Analysis and Enhancements of Adaptive Algorithms and Their Applications Performance Analysis and Enhancements of Adaptive Algorithms and Their Applications SHENGKUI ZHAO School of Computer Engineering A thesis submitted to the Nanyang Technological University in partial fulfillment

More information

KNOWN approaches for improving the performance of

KNOWN approaches for improving the performance of IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS II: EXPRESS BRIEFS, VOL. 58, NO. 8, AUGUST 2011 537 Robust Quasi-Newton Adaptive Filtering Algorithms Md. Zulfiquar Ali Bhotto, Student Member, IEEE, and Andreas

More information

On Information Maximization and Blind Signal Deconvolution

On Information Maximization and Blind Signal Deconvolution On Information Maximization and Blind Signal Deconvolution A Röbel Technical University of Berlin, Institute of Communication Sciences email: roebel@kgwtu-berlinde Abstract: In the following paper we investigate

More information

Chapter 2 Notes, Linear Algebra 5e Lay

Chapter 2 Notes, Linear Algebra 5e Lay Contents.1 Operations with Matrices..................................1.1 Addition and Subtraction.............................1. Multiplication by a scalar............................ 3.1.3 Multiplication

More information

Optimal and Adaptive Filtering

Optimal and Adaptive Filtering Optimal and Adaptive Filtering Murat Üney M.Uney@ed.ac.uk Institute for Digital Communications (IDCOM) 26/06/2017 Murat Üney (IDCOM) Optimal and Adaptive Filtering 26/06/2017 1 / 69 Table of Contents 1

More information

BLIND DECONVOLUTION ALGORITHMS FOR MIMO-FIR SYSTEMS DRIVEN BY FOURTH-ORDER COLORED SIGNALS

BLIND DECONVOLUTION ALGORITHMS FOR MIMO-FIR SYSTEMS DRIVEN BY FOURTH-ORDER COLORED SIGNALS BLIND DECONVOLUTION ALGORITHMS FOR MIMO-FIR SYSTEMS DRIVEN BY FOURTH-ORDER COLORED SIGNALS M. Kawamoto 1,2, Y. Inouye 1, A. Mansour 2, and R.-W. Liu 3 1. Department of Electronic and Control Systems Engineering,

More information

Problem Set 3 Due Oct, 5

Problem Set 3 Due Oct, 5 EE6: Random Processes in Systems Lecturer: Jean C. Walrand Problem Set 3 Due Oct, 5 Fall 6 GSI: Assane Gueye This problem set essentially reviews detection theory. Not all eercises are to be turned in.

More information

THEORY OF MIMO BIORTHOGONAL PARTNERS AND THEIR APPLICATION IN CHANNEL EQUALIZATION. Bojan Vrcelj and P. P. Vaidyanathan

THEORY OF MIMO BIORTHOGONAL PARTNERS AND THEIR APPLICATION IN CHANNEL EQUALIZATION. Bojan Vrcelj and P. P. Vaidyanathan THEORY OF MIMO BIORTHOGONAL PARTNERS AND THEIR APPLICATION IN CHANNEL EQUALIZATION Bojan Vrcelj and P P Vaidyanathan Dept of Electrical Engr 136-93, Caltech, Pasadena, CA 91125, USA E-mail: bojan@systemscaltechedu,

More information

SPEECH ENHANCEMENT USING PCA AND VARIANCE OF THE RECONSTRUCTION ERROR IN DISTRIBUTED SPEECH RECOGNITION

SPEECH ENHANCEMENT USING PCA AND VARIANCE OF THE RECONSTRUCTION ERROR IN DISTRIBUTED SPEECH RECOGNITION SPEECH ENHANCEMENT USING PCA AND VARIANCE OF THE RECONSTRUCTION ERROR IN DISTRIBUTED SPEECH RECOGNITION Amin Haji Abolhassani 1, Sid-Ahmed Selouani 2, Douglas O Shaughnessy 1 1 INRS-Energie-Matériaux-Télécommunications,

More information

CCA BASED ALGORITHMS FOR BLIND EQUALIZATION OF FIR MIMO SYSTEMS

CCA BASED ALGORITHMS FOR BLIND EQUALIZATION OF FIR MIMO SYSTEMS CCA BASED ALGORITHMS FOR BLID EQUALIZATIO OF FIR MIMO SYSTEMS Javier Vía and Ignacio Santamaría Dept of Communications Engineering University of Cantabria 395 Santander, Cantabria, Spain E-mail: {jvia,nacho}@gtasdicomunicanes

More information

Improved Detected Data Processing for Decision-Directed Tracking of MIMO Channels

Improved Detected Data Processing for Decision-Directed Tracking of MIMO Channels Improved Detected Data Processing for Decision-Directed Tracking of MIMO Channels Emna Eitel and Joachim Speidel Institute of Telecommunications, University of Stuttgart, Germany Abstract This paper addresses

More information

ELEG 5633 Detection and Estimation Signal Detection: Deterministic Signals

ELEG 5633 Detection and Estimation Signal Detection: Deterministic Signals ELEG 5633 Detection and Estimation Signal Detection: Deterministic Signals Jingxian Wu Department of Electrical Engineering University of Arkansas Outline Matched Filter Generalized Matched Filter Signal

More information

Signal Modeling Techniques in Speech Recognition. Hassan A. Kingravi

Signal Modeling Techniques in Speech Recognition. Hassan A. Kingravi Signal Modeling Techniques in Speech Recognition Hassan A. Kingravi Outline Introduction Spectral Shaping Spectral Analysis Parameter Transforms Statistical Modeling Discussion Conclusions 1: Introduction

More information

System Identification in the Short-Time Fourier Transform Domain

System Identification in the Short-Time Fourier Transform Domain System Identification in the Short-Time Fourier Transform Domain Electrical Engineering Department Technion - Israel Institute of Technology Supervised by: Prof. Israel Cohen Outline Representation Identification

More information

Research Article Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation

Research Article Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation Hindawi Publishing Corporation EURASIP Journal on Audio, Speech, and Music Processing Volume 27, Article ID 78439, 6 pages doi:1.1155/27/78439 Research Article Efficient Multichannel NLMS Implementation

More information

NOISE reduction is an important fundamental signal

NOISE reduction is an important fundamental signal 1526 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 20, NO. 5, JULY 2012 Non-Causal Time-Domain Filters for Single-Channel Noise Reduction Jesper Rindom Jensen, Student Member, IEEE,

More information

A METHOD OF ADAPTATION BETWEEN STEEPEST- DESCENT AND NEWTON S ALGORITHM FOR MULTI- CHANNEL ACTIVE CONTROL OF TONAL NOISE AND VIBRATION

A METHOD OF ADAPTATION BETWEEN STEEPEST- DESCENT AND NEWTON S ALGORITHM FOR MULTI- CHANNEL ACTIVE CONTROL OF TONAL NOISE AND VIBRATION A METHOD OF ADAPTATION BETWEEN STEEPEST- DESCENT AND NEWTON S ALGORITHM FOR MULTI- CHANNEL ACTIVE CONTROL OF TONAL NOISE AND VIBRATION Jordan Cheer and Stephen Daley Institute of Sound and Vibration Research,

More information

TRINICON: A Versatile Framework for Multichannel Blind Signal Processing

TRINICON: A Versatile Framework for Multichannel Blind Signal Processing TRINICON: A Versatile Framework for Multichannel Blind Signal Processing Herbert Buchner, Robert Aichner, Walter Kellermann {buchner,aichner,wk}@lnt.de Telecommunications Laboratory University of Erlangen-Nuremberg

More information

Inverse of a Square Matrix. For an N N square matrix A, the inverse of A, 1

Inverse of a Square Matrix. For an N N square matrix A, the inverse of A, 1 Inverse of a Square Matrix For an N N square matrix A, the inverse of A, 1 A, exists if and only if A is of full rank, i.e., if and only if no column of A is a linear combination 1 of the others. A is

More information

Alpha-Stable Distributions in Signal Processing of Audio Signals

Alpha-Stable Distributions in Signal Processing of Audio Signals Alpha-Stable Distributions in Signal Processing of Audio Signals Preben Kidmose, Department of Mathematical Modelling, Section for Digital Signal Processing, Technical University of Denmark, Building 3,

More information

Acoustic Research Institute ARI

Acoustic Research Institute ARI Austrian Academy of Sciences Acoustic Research Institute ARI System Identification in Audio Engineering P. Majdak piotr@majdak.com Institut für Schallforschung, Österreichische Akademie der Wissenschaften;

More information

X n = c n + c n,k Y k, (4.2)

X n = c n + c n,k Y k, (4.2) 4. Linear filtering. Wiener filter Assume (X n, Y n is a pair of random sequences in which the first component X n is referred a signal and the second an observation. The paths of the signal are unobservable

More information

Lesson 1. Optimal signalbehandling LTH. September Statistical Digital Signal Processing and Modeling, Hayes, M:

Lesson 1. Optimal signalbehandling LTH. September Statistical Digital Signal Processing and Modeling, Hayes, M: Lesson 1 Optimal Signal Processing Optimal signalbehandling LTH September 2013 Statistical Digital Signal Processing and Modeling, Hayes, M: John Wiley & Sons, 1996. ISBN 0471594318 Nedelko Grbic Mtrl

More information

Optimal Speech Enhancement Under Signal Presence Uncertainty Using Log-Spectral Amplitude Estimator

Optimal Speech Enhancement Under Signal Presence Uncertainty Using Log-Spectral Amplitude Estimator 1 Optimal Speech Enhancement Under Signal Presence Uncertainty Using Log-Spectral Amplitude Estimator Israel Cohen Lamar Signal Processing Ltd. P.O.Box 573, Yokneam Ilit 20692, Israel E-mail: icohen@lamar.co.il

More information

Analysis of Communication Systems Using Iterative Methods Based on Banach s Contraction Principle

Analysis of Communication Systems Using Iterative Methods Based on Banach s Contraction Principle Analysis of Communication Systems Using Iterative Methods Based on Banach s Contraction Principle H. Azari Soufiani, M. J. Saberian, M. A. Akhaee, R. Nasiri Mahallati, F. Marvasti Multimedia Signal, Sound

More information