Introduction to distributed speech enhancement algorithms for ad hoc microphone arrays and wireless acoustic sensor networks

Size: px
Start display at page:

Download "Introduction to distributed speech enhancement algorithms for ad hoc microphone arrays and wireless acoustic sensor networks"

Transcription

1 Introduction to distributed speech enhancement algorithms for ad hoc microphone arrays and wireless acoustic sensor networks Part IV: Random Microphone Deployment Sharon Gannot 1 and Alexander Bertrand 2 1 Faculty of Engineering, Bar-Ilan University, Israel 2 KU Leuven, E.E. Department ESAT-STADIUS, Belgium EUSIPCO 2013, Marrakesh, Morocco S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

2 Special Thanks This presentation is partly based on the Ph.D. dissertation by Shmulik Markovich-Golan S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

3 Synchronization Problem formulation Blind Sampling Rate Offset Estimation and Compensation [Markovich-Golan et al., 2012] Scenario Fully connected N nodes network with M n microphones at the nth node. Nominal sampling rate f s. Sampling rate f s,n = (1 + ɛ n ) f s, sampling period T s,n with Sampling rate offset ɛ n. TF-GSC [Gannot et al., 2001] with Sampling Rate Offsets RTF is constantly changing: signal distortion. ANC is constantly updating: increased noise level. Microphone signals are less coherent: degraded performance. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

4 Synchronization Solution sketch Block Diagram of Synchronized TF-GSC Input Offsets compensation TF-GSC Output Offsets estimation Synchronized TF-GSC Sampling rate estimation: based on the phase drift of the coherence between microphones in stationary noise-only segments (in coherent frequency bands). Resampling with Lagrange polynomials interpolation [Erup et al., 1993]. Other beamforming sync. methods: [Wehr et al., 2004]; [Ono et al., 2009]; [Liu, 2008]. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

5 Synchronization Offset estimation at the nth node Continuous Microphone Signals Received noise component at microphone s, 1st node: v 1,s (t). Received noise component at microphone r, nth node: v n,r (t). Noise only time-segment z 1,s (t), f s =1 z n,r (t), f s,n = t S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

6 Synchronization Offset estimation at the nth node Statistics of Noise Components v 1,s (t) and v n,r (t) Cross-covariance: R s,r (τ) = E {v 1,s (t) v n,r (t τ)}. Cross-spectrum: θ s,r (ξ) = R s,r (τ) exp ( jξτ) dτ θ s,r (ξ) 2 1 R s,r (τ) θ s,r (ξ) [rad] τ Cross-covariance ξ Cross-spectrum S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

7 Synchronization Offset estimation at the nth node Sampled Microphone Signals v 1,s [l] = v 1,s (lt s ). v n,r [l] = v n,r (lt s,n ) z 1,s (t), f s =1 z n,r (t), f s,n = t S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

8 Synchronization Offset estimation at the nth node Statistics of Microphones v 1,s (t) and v n,r (t ) Cross-covariance: R s,r (τ + ). Cross-spectrum: θ s,r (ξ) = exp (jξ ) θ s,r (ξ). Time difference at the lth sample: = lt s lt s,n lt s ɛ n (using first-order Taylor series approximation) R s,r (τ) 3 θ s,r (ξ) 0.3 R s,r (τ+δ) 2 θ Δ (ξ) s,r ( ) [rad] τ Cross-covariance ξ Cross-spectrum S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

9 Synchronization Offset estimation at the nth node Statistics of Sampled Microphones v 1,s [l] and v n,r [l] Cross-spectrum is band-limited by fs 2 Assume small offset. θs,r l ( [k] = θs,r ltsɛn 2πkfs ) K. ( Let, θ s,r [k] = θ 2πkfs ) s,r K, then: ( θs,r l [k] = exp j 2πklɛ ) n θ s,r [k] K Coherence between microphones s and r at the lth sample Define γ l s,r [k] = θ l s,r [k] θs,s [k]θ r,r [k] and γ s,r [k] = k = 0, 1,..., K 1. Then γs,r l [k] = αnγ l s,r [k] with α n = exp ( ) j 2πkɛn K. ɛ n can be extracted from α n. θ s,r [k] θs,s [k]θ r,r [k] ; for S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

10 Synchronization Offset estimation at the nth node Offset Estimation at the nth Node Given a noise-only time segment of P s samples. Partition into I frames of P samples: l = i P; i = 0, 1,..., I 1. estimate M 1 M n coherence matrix ˆΓ ip 1,n [k] between microphones of the 1st and the nth nodes P 1 P 2P 1 ( I 2) P ( I 1) P 1 l 0 1 I 1 ˆ 0 1, n Γ [ k] ˆ P 1, n Γ [ k ] ˆ I Γ ( 1) P 1, n [ k] ɛ n < ɛ max no 2π ambiguity for k k max = K 2Pɛ max. Estimate the nth node sampling rate offset: ( s, r pair: ˆɛ n,s,r = avg K k 2πPk avg ˆγ s,r ip [k]/γ (i 1)P ) s,r [k]. i }{{} Average all microphone pairs: ˆɛ n = 1 M 1M n M1 s=1 ˆα P n Mn r=1 ˆɛ n,s,r. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

11 Synchronization Offset compensation at the nth node Resampling with Lagrange Polynomials Interpolation [Pawig et al., 2010],[Erup et al., 1993] I Resample z n,r (pt s,n ) to z n,r (pt s ) Interpolate z n,r [p] by factor 4: z n,r [ p]. Denote: ṗ = 4pTs T s,n = 4p (1 + ˆɛ n ), the closest interpolated sample index from the left to time pt s. The resampled value of z n,r (pt s ) is ẑ n,r [p] = β p 1 z n,r [ṗ 1] + β p 0 z n,r [ṗ] + β p 1 z n,r [ṗ + 1] + β p 2 z n,r [ṗ + 2]. Input Lagrange Output 4 LPF interpolation znr, p z p ˆn, r S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

12 Synchronization Offset compensation at the nth node Resampling with Lagrange Polynomials Interpolation [Pawig et al., 2010],[Erup et al., 1993] II Calculate Weights η = 4p (1 + ˆɛ n ) ṗ. β p 1 = η(η 1)(η 2) 6. β p 0 = (η+1)(η 1)(η 2) 2. β p 1 = (η+1)η(η 2) 2. β p 2 = (η+1)η(η 1) , 1, S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

13 Synchronization Experimental study Experimental Study Q directional stationary interfering sources TF-GSC Algorithms W.o. offsets; Conventional TF-GSC; Synchronized TF-GSC Criteria Signal to Distortion ratio (SDR); Signal to Noise (SNR) Without offset With offset Conventional Conventional Synchronized Q SDR SNR Ex. Ex. Ex. Ex. Dist. Noise Dist. Noise Values in db, Ex. - excess values S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

14 Statistical Beamformer Problem formulation & definitions WASNs with Random Node Deployment [Markovich-Golan et al., 2011]; [Markovich-Golan et al., 2013]; general reading [Lo, 1964] Scenarios Ad hoc sensor networks. Large volume (and many microphones). High fault percentage. Arbitrary microphone deployment. Questions How many microphones are required? What is the expected performance? Is there an optimal deployment? [Kodrasi et al., 2011] S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

15 Statistical Beamformer Problem formulation & definitions Outline Array of randomly located microphones in a reverberant enclosure. Single desired speaker. Utilizing the statistical model of the ATFs, statistical models for the SIR and WNG are derived. The reliability of the SDW-MWF is computed for: Multiple coherent noise sources. Diffuse sound field. The reliability functions can be used to determine the number of microphones required to assure a desired performance level (with a controlled level of uncertainty). S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

16 Statistical Beamformer Problem formulation & definitions Notations I Signals Let s d (l, k) be a desired speaker signal located at r d. Microphone signals: z(l, k) h d (l, k) s d (l, k) + v(l, k). Microphone signals PSD: Φ zz (l, k) E{z(l, k)z H (l, k)} = σd 2 (l, k)h d(l, k)h H d (l, k) + Φ vv(l, k) Noise PSD: Φ vv (l, k) E{v(l, k)v H (l, k)}. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

17 Statistical Beamformer Problem formulation & definitions Notations II Room Constellation Room volume and surface area: V D x D y D z Reverberation time: T 60. A 2 (D x D y + D x D z + D y D z ) M microphones randomly deployed with a uniform distribution at coordinates r m [ rx m ry m rz m ] T ; m = 1,..., M. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

18 Statistical Beamformer Problem formulation & definitions Criterion SDW-MWF ( (w w argmin 1 ) ) H hd 2 σd 2 + µ ( w ) H Φvv w = w SINR and WNG are Random Variables Signal to Interference and Noise (SINR): Φ 1 vv h d h H d Φ 1 vv h d + µ σ 2 d κ σ2 d wh h d 2 w H Φ vv w = σ2 d hh d Φ 1 vv h d White noise gain (WNG): ( ξ wh h d 2 h H w 2 = d Φ 1 ) 2 vv h d h H d Φ 2 vv h d S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

19 Statistical Beamformer Statistical ATF modelling Statistical ATF Modelling ATF relating a coherent source at r d, and the mth microphone at r m h h + ĥ h the direct arrival. ĥ the reverberant component. The direct arrival and the reverberant tail assumed uncorrelated. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

20 Statistical Beamformer Statistical ATF modelling Reverberant Tail Model [Schroeder, 1987],[Kuttruff, 2000] Under the Assumptions: The signal wavelength is much smaller than the room dimensions. The microphones and sources are at least half wavelength away from the walls. The signal frequency is above the Schroeder frequency, f Schroeder 2000 (typically few hundred Hz). The Tail Statistics: with ˆα 1 ε πεa T 60 V ĥ CN (0, ˆα) 0.161V and ε AT 60, the exponential decay rate of the RIR tail. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

21 Statistical Beamformer Statistical ATF modelling The Direct Arrival (Spherical Wave Propagation) The Direct Arrival Model h ā exp ( j φ ) where: { 1 ; rd 1 ā = 4π 1 4π r d r m ; 1 4π < r d φ = 2π r d r m λ k S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

22 Statistical Beamformer Statistical ATF modelling Second-Order Statistics of single ATF Arbitrary Sensor Location within r Under the Assumptions: For r r c, where r c path is negligible [Kuttruff, 2000]. V 100πT 60 is the critical distance, the direct For r λ k multiple 2π phase cycles are repeated while sound wave propagates. r arbitrarily chosen (the results are not sensitive to the exact value). Approximations: E { h } 0 E {h} = 0. E { h 2} α = 4π r 3 6π r 1 3V ᾱ + ˆα with ᾱ = 32π 3 r. 3 The ATFs h m ; m = 1, dots, M relating the source and randomly deployed microphones are i.i.d. (for large sphere and few microphones). S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

23 Statistical Beamformer Statistical ATF modelling Covariance of ATFs Relating 2 Sources and Randomly Located Microphone ATFs covariance: E {h 1 h 2} = E { h 1 h 2} + E {ĥ1 ĥ 2 } Reverberant tail is diffused [Jacobsen and Roisin, 2000]: } ( ) E {ĥ1 ĥ2 2π r1 r 2 = ˆα sinc Assuming r 1 r 2 λ k : } E {ĥ1 ĥ2 0, since the sinc is decaying. E { h1 h } 2 0, since multiple 2π phase cycles are repeated while sound wave propagates. λ k S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

24 Statistical Beamformer Statistical ATF modelling Reliability Measures SIR Reliability The reliability of an SIR level of κ 0 is defined as the probability that the output SIR will exceed κ 0 : R κ (κ 0 ) Pr (κ κ 0 ). White Noise Gain (WNG) Reliability The reliability of a WNG level of ξ 0 is defined as the probability that the WNG will exceed ξ 0 : R ξ (ξ 0 ) Pr (ξ ξ 0 ). S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

25 Statistical Beamformer Statistical ATF modelling P Directional Noise Sources For high INR and P M ( 2 σ 2 ) u R κ,c (κ 0 ) =1 F η,c α σd 2 κ 0 ( ) 2 R ξ,c (ξ 0 ) =1 F η,c α ξ 0. σu 2 - sensor noise variance and σd 2 - desired source varaince. η c χ 2 (2 (M P)) Chi-square RV with 2 (M P) degrees of freedom. F η,c (η 0 ) Pr (η c η 0 ) = γ f (M P, η 0 2 ) Γ f (M P) is the respective CDF. Γ f is the Gamma function. γ f is the lower incomplete Gamma function. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

26 Statistical Beamformer Statistical ATF modelling Diffused Noise Source Noise Field σ 2 dif ( Φ vv m, m ) ( ) = σdif 2 sinc 2π rm r m σdif 2 λ I. k variance of the diffuse field. For enclosures larger than λ k. Reliability ( 2 σ 2 ) dif R κ,dif (κ 0 ) =1 F η,dif κ 0 α R ξ,dif (ξ 0 ) =1 F η,dif ( 2 α ξ 0 σ 2 d ) η dif χ 2 (2M) Chi-square RV with 2M degrees of freedom. F η,dif (η 0 ) Pr (η dif η 0 ) = γ f (M, η 0 2 ) Γ f (M) is the respective CDF. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

27 Statistical Beamformer Model verification SIR Reliability R,c P=1, empirical P=1. theory P=2, empirical P=2, theory P=3, empirical P=3, theory P=4, empirical P=4, theory c improvement [db] R κ,c M=5, empirical M=5, theoretical M=10, empirical M=10, theoretical M=15, empirical M=15, theoretical M=20, empirical M=20, theoretical M=25, empirical M=25, theoretical κ c improvement (a) M = 5 (b) P = 1 SINR out SNR in for coherent noise sources. T 60 = 0.4sec, room dimensions 4 4 3m. Similar trends for diffused noise field. S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

28 Bibliography References and Further Reading I Erup, L., Gardner, F., and Harris, R. (1993). Interpolation in digital modems, II- implementation and performance. IEEE Transactions on Communications, 41(6): Gannot, S., Burshtein, D., and Weinstein, E. (2001). Signal enhancement using beamforming and nonstationarity with applications to speech. IEEE Transactions on Signal Processing, 49(8): Jacobsen, F. and Roisin, T. (2000). The cohrerence of reverberant sound fields. Journal of the Acoustical Society of America, 108(1): Kodrasi, I., Rohdenburg, T., and Doclo, S. (2011). Microphone position optimization for planar superdirective beamforming. In IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages , Prague, Czech Republic. Kuttruff, H. (2000). Room acoustics. Taylor & Francis. Liu, Z. (2008). Sound source separation with distributed microphone arrays in the presence of clock synchronization errors. In The International Workshop on Acoustic Echo and Noise Control (IWAENC), Seattle, WA, USA. Lo, Y. (1964). A mathematical theory of antenna arrays with randomly spaced elements. IEEE Transactions on Antennas and Propagation, 12(3): S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

29 Bibliography References and Further Reading II Markovich-Golan, S., Gannot, S., and Cohen, I. (2011). Performance analysis of a randomly spaced wireless microphone array. In The IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages , Prague, Czech Republic. Markovich-Golan, S., Gannot, S., and Cohen, I. (2012). Blind sampling rate offset estimation and compensation in wireless acoustic sensor networks with application to beamforming. In The International Workshop on Acoustic Signal Enhancement (IWAENC), Aachen, Germany. Final list for best student paper award. Markovich-Golan, S., Gannot, S., and Cohen, I. (2013). Performance of the SDW-MWF with randomly located microphones in a reverberant enclosure. IEEE Trans. Audio, Speech and Language Processing. Accepted for publication. Ono, N., Kohno, H., Ito, N., and Sagayama, S. (2009). Blind alignment of asynchronously recorded signals for distributed microphone array. In IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), pages , New Paltz, NY, USA. Pawig, M., Enzner, G., and Vary, P. (2010). Adaptive sampling rate correction for acoustic echo control in voice-over-ip. IEEE Transactions on Signal Processing, 58(1): Schroeder, M. R. (1987). Statistical parameters of the frequency response curves of large rooms. Journal of the Audio Engineering Society, 35(5): S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

30 Bibliography References and Further Reading III Wehr, S., Kozintsev, I., Lienhart, R., and Kellermann, W. (2004). Synchronization of acoustic sensors for distributed ad-hoc audio networks and its use for blind source separation. In IEEE Sixth International Symposium on Multimedia Software Engineering, pages S. Gannot (BIU) and A. Bertrand (KUL) Distributed speech enhancement EUSIPCO / 30

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

ESTIMATION OF RELATIVE TRANSFER FUNCTION IN THE PRESENCE OF STATIONARY NOISE BASED ON SEGMENTAL POWER SPECTRAL DENSITY MATRIX SUBTRACTION

ESTIMATION OF RELATIVE TRANSFER FUNCTION IN THE PRESENCE OF STATIONARY NOISE BASED ON SEGMENTAL POWER SPECTRAL DENSITY MATRIX SUBTRACTION ESTIMATION OF RELATIVE TRANSFER FUNCTION IN THE PRESENCE OF STATIONARY NOISE BASED ON SEGMENTAL POWER SPECTRAL DENSITY MATRIX SUBTRACTION Xiaofei Li 1, Laurent Girin 1,, Radu Horaud 1 1 INRIA Grenoble

More information

Diffuse noise suppression with asynchronous microphone array based on amplitude additivity model

Diffuse noise suppression with asynchronous microphone array based on amplitude additivity model Diffuse noise suppression with asynchronous microphone array based on amplitude additivity model Yoshikazu Murase, Hironobu Chiba, Nobutaka Ono, Shigeki Miyabe, Takeshi Yamada, and Shoji Makino University

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

Distributed Speech Processing for Microphone Networks

Distributed Speech Processing for Microphone Networks Distributed Speech Processing for Microphone Networks Shmulik Markovich-Golan Faculty of Engineering Ph.D. Thesis Submitted to the Senate of Bar-Ilan University Ramat-Gan, Israel March 2013 This work

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

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

AN APPROACH TO PREVENT ADAPTIVE BEAMFORMERS FROM CANCELLING THE DESIRED SIGNAL. Tofigh Naghibi and Beat Pfister

AN APPROACH TO PREVENT ADAPTIVE BEAMFORMERS FROM CANCELLING THE DESIRED SIGNAL. Tofigh Naghibi and Beat Pfister AN APPROACH TO PREVENT ADAPTIVE BEAMFORMERS FROM CANCELLING THE DESIRED SIGNAL Tofigh Naghibi and Beat Pfister Speech Processing Group, Computer Engineering and Networks Lab., ETH Zurich, Switzerland {naghibi,pfister}@tik.ee.ethz.ch

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

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

arxiv: v1 [eess.as] 12 Jul 2018

arxiv: v1 [eess.as] 12 Jul 2018 OPTIMAL BINAURAL LCMV BEAMFORMING IN COMPLEX ACOUSTIC SCENARIOS: TEORETICAL AND PRACTICAL INSIGTS Nico Gößling 1 Daniel Marquardt 1 Ivo Merks Tao Zhang Simon Doclo 1 1 University of Oldenburg Department

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

arxiv: v1 [cs.sd] 30 Oct 2015

arxiv: v1 [cs.sd] 30 Oct 2015 ACE Challenge Workshop, a satellite event of IEEE-WASPAA 15 October 18-1, 15, New Paltz, NY ESTIMATION OF THE DIRECT-TO-REVERBERANT ENERGY RATIO USING A SPHERICAL MICROPHONE ARRAY Hanchi Chen, Prasanga

More information

BLIND SOURCE EXTRACTION FOR A COMBINED FIXED AND WIRELESS SENSOR NETWORK

BLIND SOURCE EXTRACTION FOR A COMBINED FIXED AND WIRELESS SENSOR NETWORK 2th European Signal Processing Conference (EUSIPCO 212) Bucharest, Romania, August 27-31, 212 BLIND SOURCE EXTRACTION FOR A COMBINED FIXED AND WIRELESS SENSOR NETWORK Brian Bloemendal Jakob van de Laar

More information

Source localization and separation for binaural hearing aids

Source localization and separation for binaural hearing aids Source localization and separation for binaural hearing aids Mehdi Zohourian, Gerald Enzner, Rainer Martin Listen Workshop, July 218 Institute of Communication Acoustics Outline 1 Introduction 2 Binaural

More information

Applications of Robust Optimization in Signal Processing: Beamforming and Power Control Fall 2012

Applications of Robust Optimization in Signal Processing: Beamforming and Power Control Fall 2012 Applications of Robust Optimization in Signal Processing: Beamforg and Power Control Fall 2012 Instructor: Farid Alizadeh Scribe: Shunqiao Sun 12/09/2012 1 Overview In this presentation, we study the applications

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

Relative Transfer Function Identification Using Convolutive Transfer Function Approximation

Relative Transfer Function Identification Using Convolutive Transfer Function Approximation IEEE TRANSACTIONS ON AUDIO, SPEECH AND LANGUAGE PROCESSING, VOL. XX, NO. Y, MONTH 28 1 Relative Transfer Function Identification Using Convolutive Transfer Function Approximation Ronen Talmon, Israel Cohen,

More information

Musical noise reduction in time-frequency-binary-masking-based blind source separation systems

Musical noise reduction in time-frequency-binary-masking-based blind source separation systems Musical noise reduction in time-frequency-binary-masing-based blind source separation systems, 3, a J. Čermá, 1 S. Arai, 1. Sawada and 1 S. Maino 1 Communication Science Laboratories, Corporation, Kyoto,

More information

New Statistical Model for the Enhancement of Noisy Speech

New Statistical Model for the Enhancement of Noisy Speech New Statistical Model for the Enhancement of Noisy Speech Electrical Engineering Department Technion - Israel Institute of Technology February 22, 27 Outline Problem Formulation and Motivation 1 Problem

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

Support recovery in compressive sensing for estimation of Direction-Of-Arrival

Support recovery in compressive sensing for estimation of Direction-Of-Arrival Support recovery in compressive sensing for estimation of Direction-Of-Arrival Zhiyuan Weng and Xin Wang Department of Electrical and Computer Engineering Stony Brook University Abstract In the estimation

More information

Adaptive beamforming for uniform linear arrays with unknown mutual coupling. IEEE Antennas and Wireless Propagation Letters.

Adaptive beamforming for uniform linear arrays with unknown mutual coupling. IEEE Antennas and Wireless Propagation Letters. Title Adaptive beamforming for uniform linear arrays with unknown mutual coupling Author(s) Liao, B; Chan, SC Citation IEEE Antennas And Wireless Propagation Letters, 2012, v. 11, p. 464-467 Issued Date

More information

Binaural Beamforming Using Pre-Determined Relative Acoustic Transfer Functions

Binaural Beamforming Using Pre-Determined Relative Acoustic Transfer Functions Binaural Beamforming Using Pre-Determined Relative Acoustic Transfer Functions Andreas I. Koutrouvelis, Richard C. Hendriks, Richard Heusdens, Jesper Jensen and Meng Guo e-mails: {a.koutrouvelis, r.c.hendriks,

More information

Spatial Diffuseness Features for DNN-Based Speech Recognition in Noisy and Reverberant Environments

Spatial Diffuseness Features for DNN-Based Speech Recognition in Noisy and Reverberant Environments Spatial Diffuseness Features for DNN-Based Speech Recognition in Noisy and Reverberant Environments Andreas Schwarz, Christian Huemmer, Roland Maas, Walter Kellermann Lehrstuhl für Multimediakommunikation

More information

J. Liang School of Automation & Information Engineering Xi an University of Technology, China

J. Liang School of Automation & Information Engineering Xi an University of Technology, China Progress In Electromagnetics Research C, Vol. 18, 245 255, 211 A NOVEL DIAGONAL LOADING METHOD FOR ROBUST ADAPTIVE BEAMFORMING W. Wang and R. Wu Tianjin Key Lab for Advanced Signal Processing Civil Aviation

More information

APPLICATION OF MVDR BEAMFORMING TO SPHERICAL ARRAYS

APPLICATION OF MVDR BEAMFORMING TO SPHERICAL ARRAYS AMBISONICS SYMPOSIUM 29 June 2-27, Graz APPLICATION OF MVDR BEAMFORMING TO SPHERICAL ARRAYS Anton Schlesinger 1, Marinus M. Boone 2 1 University of Technology Delft, The Netherlands (a.schlesinger@tudelft.nl)

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

Overview of Beamforming

Overview of Beamforming Overview of Beamforming Arye Nehorai Preston M. Green Department of Electrical and Systems Engineering Washington University in St. Louis March 14, 2012 CSSIP Lab 1 Outline Introduction Spatial and temporal

More information

Minimum Mean Squared Error Interference Alignment

Minimum Mean Squared Error Interference Alignment Minimum Mean Squared Error Interference Alignment David A. Schmidt, Changxin Shi, Randall A. Berry, Michael L. Honig and Wolfgang Utschick Associate Institute for Signal Processing Technische Universität

More information

Comparison of RTF Estimation Methods between a Head-Mounted Binaural Hearing Device and an External Microphone

Comparison of RTF Estimation Methods between a Head-Mounted Binaural Hearing Device and an External Microphone Comparison of RTF Estimation Methods between a Head-Mounted Binaural Hearing Device and an External Microphone Nico Gößling, Daniel Marquardt and Simon Doclo Department of Medical Physics and Acoustics

More information

Acoustic Signal Processing. Algorithms for Reverberant. Environments

Acoustic Signal Processing. Algorithms for Reverberant. Environments Acoustic Signal Processing Algorithms for Reverberant Environments Terence Betlehem B.Sc. B.E.(Hons) ANU November 2005 A thesis submitted for the degree of Doctor of Philosophy of The Australian National

More information

Independent Component Analysis and Unsupervised Learning

Independent Component Analysis and Unsupervised Learning Independent Component Analysis and Unsupervised Learning Jen-Tzung Chien National Cheng Kung University TABLE OF CONTENTS 1. Independent Component Analysis 2. Case Study I: Speech Recognition Independent

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

A Simulation Study on Binaural Dereverberation and Noise Reduction based on Diffuse Power Spectral Density Estimators

A Simulation Study on Binaural Dereverberation and Noise Reduction based on Diffuse Power Spectral Density Estimators A Simulation Study on Binaural Dereverberation and Noise Reduction based on Diffuse Power Spectral Density Estimators Ina Kodrasi, Daniel Marquardt, Simon Doclo University of Oldenburg, Department of Medical

More information

Enhancement of Noisy Speech. State-of-the-Art and Perspectives

Enhancement of Noisy Speech. State-of-the-Art and Perspectives Enhancement of Noisy Speech State-of-the-Art and Perspectives Rainer Martin Institute of Communications Technology (IFN) Technical University of Braunschweig July, 2003 Applications of Noise Reduction

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

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

ENGR352 Problem Set 02

ENGR352 Problem Set 02 engr352/engr352p02 September 13, 2018) ENGR352 Problem Set 02 Transfer function of an estimator 1. Using Eq. (1.1.4-27) from the text, find the correct value of r ss (the result given in the text is incorrect).

More information

A multichannel diffuse power estimator for dereverberation in the presence of multiple sources

A multichannel diffuse power estimator for dereverberation in the presence of multiple sources Braun and Habets EURASIP Journal on Audio, Speech, and Music Processing (2015) 2015:34 DOI 10.1186/s13636-015-0077-2 RESEARCH Open Access A multichannel diffuse power estimator for dereverberation in the

More information

Estimation of Reflections from Impulse Responses

Estimation of Reflections from Impulse Responses Proceedings of the International Symposium on Room Acoustics, ISRA 2 29 3 August 2, Melbourne, Australia Estimation of Reflections from Impulse Responses PACS: 43.55 Sakari Tervo, Teemu Korhonen 2, and

More information

Speaker Tracking and Beamforming

Speaker Tracking and Beamforming Speaker Tracking and Beamforming Dr. John McDonough Spoken Language Systems Saarland University January 13, 2010 Introduction Many problems in science and engineering can be formulated in terms of estimating

More information

MAXIMUM LIKELIHOOD BASED NOISE COVARIANCE MATRIX ESTIMATION FOR MULTI-MICROPHONE SPEECH ENHANCEMENT. Ulrik Kjems and Jesper Jensen

MAXIMUM LIKELIHOOD BASED NOISE COVARIANCE MATRIX ESTIMATION FOR MULTI-MICROPHONE SPEECH ENHANCEMENT. Ulrik Kjems and Jesper Jensen 20th European Signal Processing Conference (EUSIPCO 202) Bucharest, Romania, August 27-3, 202 MAXIMUM LIKELIHOOD BASED NOISE COVARIANCE MATRIX ESTIMATION FOR MULTI-MICROPHONE SPEECH ENHANCEMENT Ulrik Kjems

More information

ROBUSTNESS OF PARAMETRIC SOURCE DEMIXING IN ECHOIC ENVIRONMENTS. Radu Balan, Justinian Rosca, Scott Rickard

ROBUSTNESS OF PARAMETRIC SOURCE DEMIXING IN ECHOIC ENVIRONMENTS. Radu Balan, Justinian Rosca, Scott Rickard ROBUSTNESS OF PARAMETRIC SOURCE DEMIXING IN ECHOIC ENVIRONMENTS Radu Balan, Justinian Rosca, Scott Rickard Siemens Corporate Research Multimedia and Video Technology Princeton, NJ 5 fradu.balan,justinian.rosca,scott.rickardg@scr.siemens.com

More information

GLOBALLY OPTIMIZED LEAST-SQUARES POST-FILTERING FOR MICROPHONE ARRAY SPEECH ENHANCEMENT

GLOBALLY OPTIMIZED LEAST-SQUARES POST-FILTERING FOR MICROPHONE ARRAY SPEECH ENHANCEMENT GLOBALLY OPTIMIZED LEAST-SQUARES POST-FILTERING FOR MICROPHONE ARRAY SPEECH ENHANCEMENT Yiteng (Arden) Huang, Alejandro Luebs, Jan Skoglund, W. Bastiaan Kleijn Google Inc., USA {ardenhuang,aluebs,jks,kleijnb}@google.com

More information

Non-Stationary Noise Power Spectral Density Estimation Based on Regional Statistics

Non-Stationary Noise Power Spectral Density Estimation Based on Regional Statistics Non-Stationary Noise Power Spectral Density Estimation Based on Regional Statistics Xiaofei Li, Laurent Girin, Sharon Gannot, Radu Horaud To cite this version: Xiaofei Li, Laurent Girin, Sharon Gannot,

More information

Experimental investigation of multiple-input multiple-output systems for sound-field analysis

Experimental investigation of multiple-input multiple-output systems for sound-field analysis Acoustic Array Systems: Paper ICA2016-158 Experimental investigation of multiple-input multiple-output systems for sound-field analysis Hai Morgenstern (a), Johannes Klein (b), Boaz Rafaely (a), Markus

More information

Virtual Array Processing for Active Radar and Sonar Sensing

Virtual Array Processing for Active Radar and Sonar Sensing SCHARF AND PEZESHKI: VIRTUAL ARRAY PROCESSING FOR ACTIVE SENSING Virtual Array Processing for Active Radar and Sonar Sensing Louis L. Scharf and Ali Pezeshki Abstract In this paper, we describe how an

More information

Laboratory synthesis of turbulent boundary layer wall-pressures and the induced vibro-acoustic response

Laboratory synthesis of turbulent boundary layer wall-pressures and the induced vibro-acoustic response Proceedings of the Acoustics 22 Nantes Conference 23-27 April 22, Nantes, France Laboratory synthesis of turbulent boundary layer wall-pressures and the induced vibro-acoustic response C. Maury a and T.

More information

A comparative study of time-delay estimation techniques for convolutive speech mixtures

A comparative study of time-delay estimation techniques for convolutive speech mixtures A comparative study of time-delay estimation techniques for convolutive speech mixtures COSME LLERENA AGUILAR University of Alcala Signal Theory and Communications 28805 Alcalá de Henares SPAIN cosme.llerena@uah.es

More information

EE 5407 Part II: Spatial Based Wireless Communications

EE 5407 Part II: Spatial Based Wireless Communications EE 5407 Part II: Spatial Based Wireless Communications Instructor: Prof. Rui Zhang E-mail: rzhang@i2r.a-star.edu.sg Website: http://www.ece.nus.edu.sg/stfpage/elezhang/ Lecture II: Receive Beamforming

More information

Acoustic Vector Sensor based Speech Source Separation with Mixed Gaussian-Laplacian Distributions

Acoustic Vector Sensor based Speech Source Separation with Mixed Gaussian-Laplacian Distributions Acoustic Vector Sensor based Speech Source Separation with Mixed Gaussian-Laplacian Distributions Xiaoyi Chen, Atiyeh Alinaghi, Xionghu Zhong and Wenwu Wang Department of Acoustic Engineering, School of

More information

A Study of the LCMV and MVDR Noise Reduction Filters

A Study of the LCMV and MVDR Noise Reduction Filters IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 58, NO. 9, SEPTEMBER 2010 4925 [6] W. S. Cleveland, Robust locally weighted regression and smoothing scatterplots, J. Amer. Stat. Assoc., vol. 74, pp. 829 836,

More information

MANY digital speech communication applications, e.g.,

MANY digital speech communication applications, e.g., 406 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 15, NO. 2, FEBRUARY 2007 An MMSE Estimator for Speech Enhancement Under a Combined Stochastic Deterministic Speech Model Richard C.

More information

PROPAGATION PHASE REPRESENTATION IN 3D SPACE USING POLES AND ZEROS IN COMPLEX FREQUENCY PLANE. ( address of lead author)

PROPAGATION PHASE REPRESENTATION IN 3D SPACE USING POLES AND ZEROS IN COMPLEX FREQUENCY PLANE. ( address of lead author) ICSV14 Cairns Australia 9-12 July, 2007 PROPAGATION PHASE REPRESENTATION IN 3D SPACE USING POLES AND ZEROS IN COMPLEX FREQUENCY PLANE Yoshinori Takahashi 1, Mikio Tohyama 2, and Kazunori Miyoshi 1 1 Kogakuin

More information

Physical Layer Binary Consensus. Study

Physical Layer Binary Consensus. Study Protocols: A Study Venugopalakrishna Y. R. and Chandra R. Murthy ECE Dept., IISc, Bangalore 23 rd Nov. 2013 Outline Introduction to Consensus Problem Setup Protocols LMMSE-based scheme Co-phased combining

More information

Robust Adaptive Beamforming Based on Interference Covariance Matrix Reconstruction and Steering Vector Estimation

Robust Adaptive Beamforming Based on Interference Covariance Matrix Reconstruction and Steering Vector Estimation IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 60, NO. 7, JULY 0 388 Robust Adaptive Beamforming Based on Interference Covariance Matrix Reconstruction and Steering Vector Estimation Yujie Gu and Amir Leshem

More information

DIRECTION ESTIMATION BASED ON SOUND INTENSITY VECTORS. Sakari Tervo

DIRECTION ESTIMATION BASED ON SOUND INTENSITY VECTORS. Sakari Tervo 7th European Signal Processing Conference (EUSIPCO 9) Glasgow, Scotland, August 4-8, 9 DIRECTION ESTIMATION BASED ON SOUND INTENSITY VECTORS Sakari Tervo Helsinki University of Technology Department of

More information

Spatial Source Subtraction Based on Incomplete Measurements of Relative Transfer Function

Spatial Source Subtraction Based on Incomplete Measurements of Relative Transfer Function Spatial Source Subtraction Based on Incomplete Measurements of Relative Transfer Function Zbyněk Koldovský a, Jiří Málek a, and Sharon Gannot b a Faculty of Mechatronics, Informatics, and Interdisciplinary

More information

DIFFUSION-BASED DISTRIBUTED MVDR BEAMFORMER

DIFFUSION-BASED DISTRIBUTED MVDR BEAMFORMER 14 IEEE International Conference on Acoustic, Speech and Signal Processing (ICASSP) DIFFUSION-BASED DISTRIBUTED MVDR BEAMFORMER Matt O Connor 1 and W. Bastiaan Kleijn 1,2 1 School of Engineering and Computer

More information

IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 24, NO. 3, MARCH

IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 24, NO. 3, MARCH IEEE/ACM TRANSACTIONS ON AUDIO SPEECH AND ANGUAGE PROCESSING VO. 24 NO. 3 MARCH 2016 543 The Binaural CMV Beamformer and its Performance Analysis Elior Hadad Student Member IEEE Simon Doclo Senior Member

More information

Design Criteria for the Quadratically Interpolated FFT Method (I): Bias due to Interpolation

Design Criteria for the Quadratically Interpolated FFT Method (I): Bias due to Interpolation CENTER FOR COMPUTER RESEARCH IN MUSIC AND ACOUSTICS DEPARTMENT OF MUSIC, STANFORD UNIVERSITY REPORT NO. STAN-M-4 Design Criteria for the Quadratically Interpolated FFT Method (I): Bias due to Interpolation

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

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

Albenzio Cirillo INFOCOM Dpt. Università degli Studi di Roma, Sapienza

Albenzio Cirillo INFOCOM Dpt. Università degli Studi di Roma, Sapienza Albenzio Cirillo INFOCOM Dpt. Università degli Studi di Roma, Sapienza albenzio.cirillo@uniroma1.it http://ispac.ing.uniroma1.it/albenzio/index.htm ET2010 XXVI Riunione Annuale dei Ricercatori di Elettrotecnica

More information

JOINT SOURCE LOCALIZATION AND DEREVERBERATION BY SOUND FIELD INTERPOLATION USING SPARSE REGULARIZATION

JOINT SOURCE LOCALIZATION AND DEREVERBERATION BY SOUND FIELD INTERPOLATION USING SPARSE REGULARIZATION JOINT SOURCE LOCALIZATION AND DEREVERBERATION BY SOUND FIELD INTERPOLATION USING SPARSE REGULARIZATION Niccolò Antonello 1, Enzo De Sena, Marc Moonen 1, Patrick A. Naylor 3 and Toon van Waterschoot 1,

More information

Blind Source Separation with a Time-Varying Mixing Matrix

Blind Source Separation with a Time-Varying Mixing Matrix Blind Source Separation with a Time-Varying Mixing Matrix Marcus R DeYoung and Brian L Evans Department of Electrical and Computer Engineering The University of Texas at Austin 1 University Station, Austin,

More information

940 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 25, NO. 5, MAY 2017

940 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 25, NO. 5, MAY 2017 940 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 25, NO. 5, MAY 2017 Multispeaker LCMV Beamformer and Postfilter for Source Separation and Noise Reduction Ofer Schwartz, Student

More information

A Second-Order-Statistics-based Solution for Online Multichannel Noise Tracking and Reduction

A Second-Order-Statistics-based Solution for Online Multichannel Noise Tracking and Reduction A Second-Order-Statistics-based Solution for Online Multichannel Noise Tracking and Reduction Mehrez Souden, Jingdong Chen, Jacob Benesty, and Sofiène Affes Abstract We propose a second-order-statistics-based

More information

MAXIMUM LIKELIHOOD BASED MULTI-CHANNEL ISOTROPIC REVERBERATION REDUCTION FOR HEARING AIDS

MAXIMUM LIKELIHOOD BASED MULTI-CHANNEL ISOTROPIC REVERBERATION REDUCTION FOR HEARING AIDS MAXIMUM LIKELIHOOD BASED MULTI-CHANNEL ISOTROPIC REVERBERATION REDUCTION FOR HEARING AIDS Adam Kuklasiński, Simon Doclo, Søren Holdt Jensen, Jesper Jensen, Oticon A/S, 765 Smørum, Denmark University of

More information

Reverberation Impulse Response Analysis

Reverberation Impulse Response Analysis MUS424: Signal Processing Techniques for Digital Audio Effects Handout #18 Jonathan Abel, David Berners April 27, 24 Lecture #9: April 27, 24 Lecture Notes 7 Reverberation Impulse Response Analysis 1 CCRMA

More information

SUPERVISED INDEPENDENT VECTOR ANALYSIS THROUGH PILOT DEPENDENT COMPONENTS

SUPERVISED INDEPENDENT VECTOR ANALYSIS THROUGH PILOT DEPENDENT COMPONENTS SUPERVISED INDEPENDENT VECTOR ANALYSIS THROUGH PILOT DEPENDENT COMPONENTS 1 Francesco Nesta, 2 Zbyněk Koldovský 1 Conexant System, 1901 Main Street, Irvine, CA (USA) 2 Technical University of Liberec,

More information

Efficient Target Activity Detection Based on Recurrent Neural Networks

Efficient Target Activity Detection Based on Recurrent Neural Networks Efficient Target Activity Detection Based on Recurrent Neural Networks D. Gerber, S. Meier, and W. Kellermann Friedrich-Alexander-Universität Erlangen-Nürnberg (FAU) Motivation Target φ tar oise 1 / 15

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

Last time: small acoustics

Last time: small acoustics Last time: small acoustics Voice, many instruments, modeled by tubes Traveling waves in both directions yield standing waves Standing waves correspond to resonances Variations from the idealization give

More information

The Effect of Spatial Correlations on MIMO Capacity: A (not so) Large N Analytical Approach: Aris Moustakas 1, Steven Simon 1 & Anirvan Sengupta 1,2

The Effect of Spatial Correlations on MIMO Capacity: A (not so) Large N Analytical Approach: Aris Moustakas 1, Steven Simon 1 & Anirvan Sengupta 1,2 The Effect of Spatial Correlations on MIMO Capacity: A (not so) Large N Analytical Approach: Aris Moustakas 1, Steven Simon 1 & Anirvan Sengupta 1, 1, Rutgers University Outline Aim: Calculate statistics

More information

DIFFERENTIAL microphone arrays (DMAs) refer to arrays

DIFFERENTIAL microphone arrays (DMAs) refer to arrays 760 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 6, NO. 4, APRIL 018 Frequency-Domain Design of Asymmetric Circular Differential Microphone Arrays Yaakov Buchris, Member, IEEE,

More information

Sound Source Tracking Using Microphone Arrays

Sound Source Tracking Using Microphone Arrays Sound Source Tracking Using Microphone Arrays WANG PENG and WEE SER Center for Signal Processing School of Electrical & Electronic Engineering Nanayang Technological Univerisy SINGAPORE, 639798 Abstract:

More information

A POSTERIORI SPEECH PRESENCE PROBABILITY ESTIMATION BASED ON AVERAGED OBSERVATIONS AND A SUPER-GAUSSIAN SPEECH MODEL

A POSTERIORI SPEECH PRESENCE PROBABILITY ESTIMATION BASED ON AVERAGED OBSERVATIONS AND A SUPER-GAUSSIAN SPEECH MODEL A POSTERIORI SPEECH PRESENCE PROBABILITY ESTIMATION BASED ON AVERAGED OBSERVATIONS AND A SUPER-GAUSSIAN SPEECH MODEL Balázs Fodor Institute for Communications Technology Technische Universität Braunschweig

More information

Novel spectrum sensing schemes for Cognitive Radio Networks

Novel spectrum sensing schemes for Cognitive Radio Networks Novel spectrum sensing schemes for Cognitive Radio Networks Cantabria University Santander, May, 2015 Supélec, SCEE Rennes, France 1 The Advanced Signal Processing Group http://gtas.unican.es The Advanced

More information

THE PROBLEMS OF ROBUST LPC PARAMETRIZATION FOR. Petr Pollak & Pavel Sovka. Czech Technical University of Prague

THE PROBLEMS OF ROBUST LPC PARAMETRIZATION FOR. Petr Pollak & Pavel Sovka. Czech Technical University of Prague THE PROBLEMS OF ROBUST LPC PARAMETRIZATION FOR SPEECH CODING Petr Polla & Pavel Sova Czech Technical University of Prague CVUT FEL K, 66 7 Praha 6, Czech Republic E-mail: polla@noel.feld.cvut.cz Abstract

More information

Active noise control in a pure tone diffuse sound field. using virtual sensing. School of Mechanical Engineering, The University of Adelaide, SA 5005,

Active noise control in a pure tone diffuse sound field. using virtual sensing. School of Mechanical Engineering, The University of Adelaide, SA 5005, Active noise control in a pure tone diffuse sound field using virtual sensing D. J. Moreau, a) J. Ghan, B. S. Cazzolato, and A. C. Zander School of Mechanical Engineering, The University of Adelaide, SA

More information

Robust Capon Beamforming

Robust Capon Beamforming Robust Capon Beamforming Yi Jiang Petre Stoica Zhisong Wang Jian Li University of Florida Uppsala University University of Florida University of Florida March 11, 2003 ASAP Workshop 2003 1 Outline Standard

More information

ASYMPTOTIC PERFORMANCE ANALYSIS OF DOA ESTIMATION METHOD FOR AN INCOHERENTLY DISTRIBUTED SOURCE. 2πd λ. E[ϱ(θ, t)ϱ (θ,τ)] = γ(θ; µ)δ(θ θ )δ t,τ, (2)

ASYMPTOTIC PERFORMANCE ANALYSIS OF DOA ESTIMATION METHOD FOR AN INCOHERENTLY DISTRIBUTED SOURCE. 2πd λ. E[ϱ(θ, t)ϱ (θ,τ)] = γ(θ; µ)δ(θ θ )δ t,τ, (2) ASYMPTOTIC PERFORMANCE ANALYSIS OF DOA ESTIMATION METHOD FOR AN INCOHERENTLY DISTRIBUTED SOURCE Jooshik Lee and Doo Whan Sang LG Electronics, Inc. Seoul, Korea Jingon Joung School of EECS, KAIST Daejeon,

More information

DOA Estimation of Quasi-Stationary Signals Using a Partly-Calibrated Uniform Linear Array with Fewer Sensors than Sources

DOA Estimation of Quasi-Stationary Signals Using a Partly-Calibrated Uniform Linear Array with Fewer Sensors than Sources Progress In Electromagnetics Research M, Vol. 63, 185 193, 218 DOA Estimation of Quasi-Stationary Signals Using a Partly-Calibrated Uniform Linear Array with Fewer Sensors than Sources Kai-Chieh Hsu and

More information

A LOCALIZATION METHOD FOR MULTIPLE SOUND SOURCES BY USING COHERENCE FUNCTION

A LOCALIZATION METHOD FOR MULTIPLE SOUND SOURCES BY USING COHERENCE FUNCTION 8th European Signal Processing Conference (EUSIPCO-2) Aalborg, Denmark, August 23-27, 2 A LOCALIZATION METHOD FOR MULTIPLE SOUND SOURCES BY USING COHERENCE FUNCTION Hiromichi NAKASHIMA, Mitsuru KAWAMOTO,

More information

Experimental investigation on varied degrees of sound field diffuseness in enclosed spaces

Experimental investigation on varied degrees of sound field diffuseness in enclosed spaces 22 nd International Congress on Acoustics ` Isotropy and Diffuseness in Room Acoustics: Paper ICA2016-551 Experimental investigation on varied degrees of sound field diffuseness in enclosed spaces Bidondo,

More information

(3) where the mixing vector is the Fourier transform of are the STFT coefficients of the sources I. INTRODUCTION

(3) where the mixing vector is the Fourier transform of are the STFT coefficients of the sources I. INTRODUCTION 1830 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 18, NO. 7, SEPTEMBER 2010 Under-Determined Reverberant Audio Source Separation Using a Full-Rank Spatial Covariance Model Ngoc Q.

More information

Energy Based Acoustic Source Localization

Energy Based Acoustic Source Localization Energy Based Acoustic Source Localization Xiaohong Sheng and Yu-Hen Hu University of Wisconsin Madison, Department of Electrical and Computer Engineering Madison WI 5376, USA, sheng@ece.wisc.edu, hu@engr.wisc.edu,

More information

Vast Volatility Matrix Estimation for High Frequency Data

Vast Volatility Matrix Estimation for High Frequency Data Vast Volatility Matrix Estimation for High Frequency Data Yazhen Wang National Science Foundation Yale Workshop, May 14-17, 2009 Disclaimer: My opinion, not the views of NSF Y. Wang (at NSF) 1 / 36 Outline

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

A LINEAR OPERATOR FOR THE COMPUTATION OF SOUNDFIELD MAPS

A LINEAR OPERATOR FOR THE COMPUTATION OF SOUNDFIELD MAPS A LINEAR OPERATOR FOR THE COMPUTATION OF SOUNDFIELD MAPS L Bianchi, V Baldini Anastasio, D Marković, F Antonacci, A Sarti, S Tubaro Dipartimento di Elettronica, Informazione e Bioingegneria - Politecnico

More information

LIMITATIONS AND ERROR ANALYSIS OF SPHERICAL MICROPHONE ARRAYS. Thushara D. Abhayapala 1, Michael C. T. Chan 2

LIMITATIONS AND ERROR ANALYSIS OF SPHERICAL MICROPHONE ARRAYS. Thushara D. Abhayapala 1, Michael C. T. Chan 2 ICSV4 Cairns Australia 9-2 July, 2007 LIMITATIONS AND ERROR ANALYSIS OF SPHERICAL MICROPHONE ARRAYS Thushara D. Abhayapala, Michael C. T. Chan 2 Research School of Information Sciences and Engineering

More information

SINGLE-CHANNEL BLIND ESTIMATION OF REVERBERATION PARAMETERS

SINGLE-CHANNEL BLIND ESTIMATION OF REVERBERATION PARAMETERS SINGLE-CHANNEL BLIND ESTIMATION OF REVERBERATION PARAMETERS Clement S. J. Doire, Mike Brookes, Patrick A. Naylor, Dave Betts, Christopher M. Hicks, Mohammad A. Dmour, Søren Holdt Jensen Electrical and

More information

Tracking of Spread Spectrum Signals

Tracking of Spread Spectrum Signals Chapter 7 Tracking of Spread Spectrum Signals 7. Introduction As discussed in the last chapter, there are two parts to the synchronization process. The first stage is often termed acquisition and typically

More information

The Probabilistic Instantaneous Matching Algorithm

The Probabilistic Instantaneous Matching Algorithm he Probabilistic Instantaneous Matching Algorithm Frederik Beutler and Uwe D. Hanebeck Abstract A new Bayesian filtering technique for estimating signal parameters directly from discrete-time sequences

More information

Multiple Sound Source Counting and Localization Based on Spatial Principal Eigenvector

Multiple Sound Source Counting and Localization Based on Spatial Principal Eigenvector INTERSPEECH 27 August 2 24, 27, Stockholm, Sweden Multiple Sound Source Counting and Localization Based on Spatial Principal Eigenvector Bing Yang, Hong Liu, Cheng Pang Key Laboratory of Machine Perception,

More information

IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL.?, NO.?,

IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL.?, NO.?, IEEE TANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE POCESSING, VOL.?, NO.?, 25 An Expectation-Maximization Algorithm for Multi-microphone Speech Dereverberation and Noise eduction with Coherence Matrix Estimation

More information

Spatial sound. Lecture 8: EE E6820: Speech & Audio Processing & Recognition. Columbia University Dept. of Electrical Engineering

Spatial sound. Lecture 8: EE E6820: Speech & Audio Processing & Recognition. Columbia University Dept. of Electrical Engineering EE E6820: Speech & Audio Processing & Recognition Lecture 8: Spatial sound 1 Spatial acoustics 2 Binaural perception 3 Synthesizing spatial audio 4 Extracting spatial sounds Dan Ellis

More information

International Journal of Advanced Research in Computer Science and Software Engineering

International Journal of Advanced Research in Computer Science and Software Engineering Volume 2, Issue 11, November 2012 ISSN: 2277 128X International Journal of Advanced Research in Computer Science and Software Engineering Research Paper Available online at: www.ijarcsse.com Acoustic Source

More information