Click to edit Master text styles Second level Third level Fourth level Fifth level Click to edit Master title style Blind Source Separation: Finding Needles.

Slides:



Advertisements
Similar presentations
DFT & FFT Computation.
Advertisements

Contrasting Log Sine Sweep method and MLS for room acoustics measurements Angelo Farina  Industrial Engineering Dept., University of Parma, Via delle Scienze.
Subspace Embeddings for the L1 norm with Applications Christian Sohler David Woodruff TU Dortmund IBM Almaden.
Cognitive Radio Communications and Networks: Principles and Practice By A. M. Wyglinski, M. Nekovee, Y. T. Hou (Elsevier, December 2009) 1 Chapter 12 Cross-Layer.
Learning Introductory Signal Processing Using Multimedia 1 Outline Overview of Information and Communications Some signal processing concepts Tools available.
Environmental Remote Sensing GEOG 2021
1 OFDM Synchronization Speaker:. Wireless Access Tech. Lab. CCU Wireless Access Tech. Lab. 2 Outline OFDM System Description Synchronization What is Synchronization?
Copyright © Cengage Learning. All rights reserved.
Detection Chia-Hsin Cheng. Wireless Access Tech. Lab. CCU Wireless Access Tech. Lab. 2 Outlines Detection Theory Simple Binary Hypothesis Tests Bayes.
A Subspace Method for MIMO Radar Space-Time Adaptive Processing
Independent Component Analysis
Introduction[1] •Three techniques are used independently or in tandem to improve receiver signal quality •Equalization compensates for.
16. Mean Square Estimation
Multiuser Detection for CDMA Systems
Discriminative Training in Speech Processing Filipp Korkmazsky LORIA.
Math Review with Matlab:
Chapter 5 The Mathematics of Diversification
Probabilistic Reasoning over Time
ECE 8443 – Pattern Recognition ECE 8423 – Adaptive Signal Processing Objectives: The Linear Prediction Model The Autocorrelation Method Levinson and Durbin.
(t,x) domain, pattern-based ground roll removal Morgan P. Brown* and Robert G. Clapp Stanford Exploration Project Stanford University.
Speech Enhancement through Noise Reduction By Yating & Kundan.
Comparison of different MIMO-OFDM signal detectors for LTE
2008 SIAM Conference on Imaging Science July 7, 2008 Jason A. Palmer
ISWCS’06, Valencia, Spain 1 Blind Adaptive Channel Shortening by Unconstrained Optimization for Simplified UWB Receiver Design Authors: Syed Imtiaz Husain.
Blind Source Separation of Acoustic Signals Based on Multistage Independent Component Analysis Hiroshi SARUWATARI, Tsuyoki NISHIKAWA, and Kiyohiro SHIKANO.
Model-Based Fusion of Bone and Air Sensors for Speech Enhancement and Robust Speech Recognition John Hershey, Trausti Kristjansson, Zhengyou Zhang, Alex.
Independent Component Analysis (ICA)
Top Level System Block Diagram BSS Block Diagram Abstract In today's expanding business environment, conference call technology has become an integral.
Speech Enhancement Based on a Combination of Spectral Subtraction and MMSE Log-STSA Estimator in Wavelet Domain LATSI laboratory, Department of Electronic,
3/24/2006Lecture notes for Speech Communications Multi-channel speech enhancement Chunjian Li DICOM, Aalborg University.
Subband-based Independent Component Analysis Y. Qi, P.S. Krishnaprasad, and S.A. Shamma ECE Department University of Maryland, College Park.
Independent Component Analysis (ICA) and Factor Analysis (FA)
Project Presentation: March 9, 2006
Audio Source Separation And ICA by Mike Davies & Nikolaos Mitianoudis Digital Signal Processing Lab Queen Mary, University of London.
ICA Alphan Altinok. Outline  PCA  ICA  Foundation  Ambiguities  Algorithms  Examples  Papers.
1 Blind Separation of Audio Mixtures Using Direct Estimation of Delays Arie Yeredor Dept. of Elect. Eng. – Systems School of Electrical Engineering Tel-Aviv.
EE513 Audio Signals and Systems Statistical Pattern Classification Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
GCT731 Fall 2014 Topics in Music Technology - Music Information Retrieval Overview of MIR Systems Audio and Music Representations (Part 1) 1.
Microphone Integration – Can Improve ARS Accuracy? Tom Houy
Eigenstructure Methods for Noise Covariance Estimation Olawoye Oyeyele AICIP Group Presentation April 29th, 2003.
INDEPENDENT COMPONENT ANALYSIS OF TEXTURES based on the article R.Manduchi, J. Portilla, ICA of Textures, The Proc. of the 7 th IEEE Int. Conf. On Comp.
Heart Sound Background Noise Removal Haim Appleboim Biomedical Seminar February 2007.
REVISED CONTEXTUAL LRT FOR VOICE ACTIVITY DETECTION Javier Ram’ırez, Jos’e C. Segura and J.M. G’orriz Dept. of Signal Theory Networking and Communications.
IT-Master Thesis Themes 2008 Discrete Systems Lab Prof. Dr.-Ing. Volker Lohweg Contact:
Independent Component Analysis Zhen Wei, Li Jin, Yuxue Jin Department of Statistics Stanford University An Introduction.
2010/12/11 Frequency Domain Blind Source Separation Based Noise Suppression to Hearing Aids (Part 1) Presenter: Cian-Bei Hong Advisor: Dr. Yeou-Jiunn Chen.
Multiuser Detection (MUD) Combined with array signal processing in current wireless communication environments Wed. 박사 3학기 구 정 회.
Blind Separation of Speech Mixtures Vaninirappuputhenpurayil Gopalan REJU School of Electrical and Electronic Engineering Nanyang Technological University.
Adaptive Methods for Speaker Separation in Cars DaimlerChrysler Research and Technology Julien Bourgeois
Communication Group Course Multidimensional DSP DoA Estimation Methods Pejman Taslimi – Spring 2009 Course Presentation – Amirkabir Univ Title: Acoustic.
Authors: Sriram Ganapathy, Samuel Thomas, and Hynek Hermansky Temporal envelope compensation for robust phoneme recognition using modulation spectrum.
Signal Processing Algorithms for Wireless Acoustic Sensor Networks Alexander Bertrand Electrical Engineering Department (ESAT) Katholieke Universiteit.
Doc.: IEEE /0553r1 Submission May 2009 Alexander Maltsev, Intel Corp.Slide 1 Path Loss Model Development for TGad Channel Models Date:
Independent Component Analysis Algorithm for Adaptive Noise Cancelling 적응 잡음 제거를 위한 독립 성분 분석 알고리즘 Hyung-Min Park, Sang-Hoon Oh, and Soo-Young Lee Brain.
An Introduction to Blind Source Separation Kenny Hild Sept. 19, 2001.
A Passive Approach to Sensor Network Localization Rahul Biswas and Sebastian Thrun International Conference on Intelligent Robots and Systems 2004 Presented.
2010/12/11 Frequency Domain Blind Source Separation Based Noise Suppression to Hearing Aids (Part 2) Presenter: Cian-Bei Hong Advisor: Dr. Yeou-Jiunn Chen.
Professors: Eng. Diego Barral Eng. Mariano Llamedo Soria Julian Bruno
Independent Component Analysis Independent Component Analysis.
Single Correlator Based UWB Receiver Implementation through Channel Shortening Equalizer By Syed Imtiaz Husain and Jinho Choi School of Electrical Engineering.
Yi Jiang MS Thesis 1 Yi Jiang Dept. Of Electrical and Computer Engineering University of Florida, Gainesville, FL 32611, USA Array Signal Processing in.
Spatial Covariance Models For Under- Determined Reverberant Audio Source Separation N. Duong, E. Vincent and R. Gribonval METISS project team, IRISA/INRIA,
Spatial vs. Blind Approaches for Speaker Separation: Structural Differences and Beyond Julien Bourgeois RIC/AD.
Variable Step-Size Adaptive Filters for Acoustic Echo Cancellation Constantin Paleologu Department of Telecommunications
Motorola presents in collaboration with CNEL Introduction  Motivation: The limitation of traditional narrowband transmission channel  Advantage: Phone.
Introduction to Audio Watermarking Schemes N. Lazic and P
Outline Introduction Signal, random variable, random process and spectra Analog modulation Analog to digital conversion Digital transmission through baseband.
INTRODUCTION TO ADVANCED DIGITAL SIGNAL PROCESSING
Presenter: Shih-Hsiang(士翔)
Combination of Feature and Channel Compensation (1/2)
Presentation transcript:

Click to edit Master text styles Second level Third level Fourth level Fifth level Click to edit Master title style Blind Source Separation: Finding Needles in Haystacks Scott C. Douglas Department of Electrical Engineering Southern Methodist University

Signal Mixtures are Everywhere Cell Phones Radio Astronomy Brain Activity Speech/Music How do we make sense of it all?

Example: Speech Enhancement

Example: Wireless Signal Separation

Outline of Talk Blind Source Separation General concepts and approaches Convolutive Blind Source Separation Application to multi-microphone speech recordings Complex Blind Source Separation What differentiates the complex-valued case Conclusions

Blind Source Separation (BSS) - A Simple Math Example Let s 1 (k), s 2 (k),…, s m (k) be signals of interest Measurements: For 1 i m, x i (k) = a i1 s 1 (k) + a i2 s 2 (k) + … + a im s m (k) Sensor noise is neglected Dispersion (echo/reverberation) is absent AB s(k)s(k)x(k)x(k) y(k)y(k)

Blind Source Separation Example (continued) AB s(k)s(k)x(k)x(k) y(k)y(k) Can Show: The s i (k)s can be recovered as y i (k) = b i1 x 1 (k) + b i2 x 2 (k) + … + b im x m (k) up to permutation and scaling factors (the matrix B is like the inverse of matrix A) Problem: How do you find the demixing b ij s when you dont know the mixing a ij s or s j (k)s?

Why Blind Source Separation? (Why not Traditional Beamforming?) BSS requires no knowledge of sensor geometry. The system can be uncalibrated, with unmatched sensors. BSS does not need knowledge of source positions relative to the sensor array. BSS requires little to no knowledge of signal types - can push decisions/ detections to the end of the processing chain.

What Properties Are Necessary for BSS to Work? Separation can be achieved when (# sensors) (# of sources) The talker signals {s j (t)} are statistically-independent of each other and are non-Gaussian in amplitude OR have spectra that differ from each other OR are non-stationary Statistical independence is the critical assumption.

Entropy is the Key to Source Separation Entropy: A measure of regularity In BSS, separated signals are demixed and, have more order as a group. First used in 1996 for speech separation. - In physics, entropy increases (less order) - In biology, entropy decreases (more order)

Convolutive Blind Source Separation Mixing system is dispersive: Separation System B(z) is a multichannel filter

Goal of Convolutive BSS Key idea: For convolutive BSS, sources are arbitrarily filtered and arbitrarily shuffled

Non-Gaussian-Based Blind Source Separation Basic Goal: Make the output signals look non- Gaussian, because mixtures look more Gaussian (from the Central Limit Theorem) Criteria Based On This Goal: Density Modeling Contrast Functions Property Restoral [e.g. (Non-)Constant Modulus Algorithm] Implications: Separating capability of the criteria will be similar Implementation details (e.g. optimization strategy) will yield performance differences

BSS for Convolutive Mixtures Idea: Translate separation task into frequency domain and apply multiple independent instantaneous BSS procedures Does not work due to permutation problems A Better Idea: Reformulate separation tasks in the context of multichannel filtering Separation criterion stays in the time domain – no implied permutation problem Can still employ fast convolution methods for efficient implementation

Natural Gradient Convolutive BSS Alg. [Amari/Douglas/Cichocki/Yang 1997] where f(y) is a simple vector-valued nonlinearity. Criterion: Density-based (Maximum Likelihood) Complexity: about four multiply/adds per tap =

Blind Source Separation Toolbox A MATLAB toolbox of robust source separation algorithms for noisy convolutive mixtures (developed under govt. contract) Allows us to evaluate relationships and tradeoffs between different approaches easily and rapidly Used to determine when a particular algorithm or approach is appropriate for a particular (acoustic) measurement scenario

Speech Enhancement Methods Classic (frequency selective) linear filtering Only useful for the simplest of situations Single-microphone spectral subtraction: Only useful if the signal is reasonably well- separated to begin with ( > 5dB SINR ) Tends to introduce musical artifacts Research Focus: How to leverage multiple microphones to achieve robust signal enhancement with minimal knowledge.

Novel Techniques for Speech Enhancement Blind Source Separation: Find all the talker signals in the room - loud and soft, high and low-pitched, near and far away … without knowledge of any of these characteristics. Multi-Microphone Signal Enhancement: Using only the knowledge of target present or target absent labels on the data, pull out the target signal from the noisy background.

SMU Multimedia Systems Lab Acoustic Facility Room (Nominal Configuration) Acoustically-treated RT = 300 ms Non-parallel walls to prevent flutter echo Sources Loudspeakers playing Recordings as well as live talkers. Distance to mics: 50 cm Angles: -30 o, 0 o, 27.5 o Sensors Omnidirectional Micro- phones (AT803b) Linear array (4cm spacing) Data collection and processing entirely within MATLAB. Allows for careful characterization, fast evaluation, and experimentation with artificial and human talkers.

Performance improvement: Between 10 dB and 15 dB for equal-level mixtures, and even higher for unequal-level ones. Blind Source Separation Example Convolutive Mixing (Room) Separation System (Code) Talker 1 (MG) Talker 2 (SCD)

Unequal Power Scenario Results Time-domain CBSS methods provide the greatest SIR improvements for weak sources; no significant improvement in SIR if the initial SIR is already large

Noise Source Speech Source Linear Processing Adaptive Algorithm Multi-Microphone Speech Enhancement Contains most speech Contains most noise y1y1 y2y2 y3y3 ynyn z1z1 z2z2 z3z3 znzn

Speech Enhancement via Iterative Multichannel Filtering System output at time k: a linear adaptive filter is a sequence of (n x n) matrices at iteration k. Goal: Adapt, over time such that the multichannel output contains signals with maximum speech energy in the first output.

Multichannel Speech Enhancement Algorithm A novel* technique for enhancing target speech in noise using two or more microphones via joint decorrelation Requires rough target identifier (i.e. when talker speech is present) Is adaptive to changing noise characteristics Knowledge of source locations, microphone positions, other characteristics not needed. Details in [Gupta and Douglas, IEEE Trans. Audio, Speech, Lang. Proc., May 2009] *Patent pending

28 Performance Evaluations Room –Acoustically-treated, RT = 300 ms –Non-parallel walls to prevent flutter echo Sources –Loudspeakers playing BBC Recordings (Fs = 8kHz), 1 male/1-2 noise sources –Distance to mics: 1.3 m –Angles: -30 o, 0 o, 27.5 o Sensors –Linear array adjustable (4cm spacing) Room –Ordinary conference room (RT = 600ms) Sources –Loudspeakers playing BBC Recordings (Fs = 8kHz), 1 male/1-2 noise sources –Angles: -15 o, 15 o, 30 o Sensors –Omnidirectional Microphones (AT803b) –Linear array adjustable (4cm nominal spacing)

Audio Examples Acoustic Lab: Initial SIR = -10dB, 3-Mic System Before:After: Acoustic Lab: Initial SIR = 0dB, 2-Mic System Before:After: Conference Room: Initial SIR = -10dB, 3-Mic System Before:After: Conference Room: Initial SIR = 5dB, 2-Mic System Before:After:

Effect of Noise Segment Length on Overall Performance

31 Diffuse Noise Source Example Noise Source: SMU Campus-Wide Air Handling System Data was recorded using a simple two- channel portable M- Audio recorder (16-bit, 48kHz) with it associated T-shaped omnidirectional stereo array at arms length, then downsampled to 8kHz.

32 Air Handler Data Processing Step 1: Spatio-Temporal GEVD Processing on a frame-by-frame basis with L = 256, where Rv(k) = Ry(k-1); that is, data was whitened to the previous frame. Step 2: Least-squares multichannel linear prediction was used to remove tones. Step 3: Log-STSA spectral subtraction was applied to the first output channel.

Complex Blind Source Separation AB s(k)s(k)x(k)x(k) y(k)y(k) Signal Model: x(k) = A s(k) Both the s i (k)s in s(k) and the elements of A are complex-valued. Separating matrix B is complex-valued as well. It appears that there is little difference from the real-valued case…

Complex Circular vs. Complex Non- Circular Sources (Second-Order) Circular Source: The energies of the real and imaginary parts of s i (k) are the same. (Second-Order) Non-Circular Source: The energies of the real and imaginary parts of s i (k) are not the same. Non-CircularCircular

Why Complex Circularity Matters in Blind Source Separation Fact #1: It is possible to separate non-circular sources by decorrelation alone if their non- circularities differ [Eriksson and Koivunen, IEEE Trans. IT, 2006] Fact #2: The strong-uncorrelating transform is a unique linear transformation for identifying non- circular source subspaces using only covariance matrices. Fact #3: Knowledge of source non-circularity is required to obtain the best performance of a complex BSS procedure.

Complex Fixed Point Algorithm [Douglas 2007] NOTE: The MATLAB code involves both transposes and Hermitian transposes… and no, those arent mistakes!

Performance Comparisons

Complex BSS Example Original Sources Sensor Signals 16-elem ULA, /4 Spacing 3000 Snapshots SINRs/elem: -17,-12,-5,-12,-12 (dB). DOAs( o ): -45,20,-15,49,35 CFPA1 Outputs Output SINRs (dB): 7,24,18,15,23 Complexity: ~3500 FLOPS per output sample

Conclusions Blind Source Separation provides unique capabilities for extracting useful signals from multiple sensor measurements corrupted by noise. Little to no knowledge of the sensor array geometry, the source positions, or the source statistics or characteristics is required. Algorithm design can be tricky. Opportunities for applications in speech enhancement, wireless communications, other areas.

For Further Reading My publications page at SMU: It has available for download 82% of my published journal papers 75% of my published conference papers