Signal Separation

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Full List of Titles
1: Speech Processing
CELP Coding
Large Vocabulary Recognition
Speech Analysis and Enhancement
Acoustic Modeling I
ASR Systems and Applications
Topics in Speech Coding
Speech Analysis
Low Bit Rate Speech Coding I
Robust Speech Recognition in Noisy Environments
Speaker Recognition
Acoustic Modeling II
Speech Production and Synthesis
Feature Extraction
Robust Speech Recognition and Adaptation
Low Bit Rate Speech Coding II
Speech Understanding
Language Modeling I
2: Speech Processing, Audio and Electroacoustics, and Neural Networks
Acoustic Modeling III
Lexical Issues/Search
Speech Understanding and Systems
Speech Analysis and Quantization
Utterance Verification/Acoustic Modeling
Language Modeling II
Adaptation /Normalization
Speech Enhancement
Topics in Speaker and Language Recognition
Echo Cancellation and Noise Control
Coding
Auditory Modeling, Hearing Aids and Applications of Signal Processing to Audio and Acoustics
Spatial Audio
Music Applications
Application - Pattern Recognition & Speech Processing
Theory & Neural Architecture
Signal Separation
Application - Image & Nonlinear Signal Processing
3: Signal Processing Theory & Methods I
Filter Design and Structures
Detection
Wavelets
Adaptive Filtering: Applications and Implementation
Nonlinear Signals and Systems
Time/Frequency and Time/Scale Analysis
Signal Modeling and Representation
Filterbank and Wavelet Applications
Source and Signal Separation
Filterbanks
Emerging Applications and Fast Algorithms
Frequency and Phase Estimation
Spectral Analysis and Higher Order Statistics
Signal Reconstruction
Adaptive Filter Analysis
Transforms and Statistical Estimation
Markov and Bayesian Estimation and Classification
4: Signal Processing Theory & Methods II, Design and Implementation of Signal Processing Systems, Special Sessions, and Industry Technology Tracks
System Identification, Equalization, and Noise Suppression
Parameter Estimation
Adaptive Filters: Algorithms and Performance
DSP Development Tools
VLSI Building Blocks
DSP Architectures
DSP System Design
Education
Recent Advances in Sampling Theory and Applications
Steganography: Information Embedding, Digital Watermarking, and Data Hiding
Speech Under Stress
Physics-Based Signal Processing
DSP Chips, Architectures and Implementations
DSP Tools and Rapid Prototyping
Communication Technologies
Image and Video Technologies
Automotive Applications / Industrial Signal Processing
Speech and Audio Technologies
Defense and Security Applications
Biomedical Applications
Voice and Media Processing
Adaptive Interference Cancellation
5: Communications, Sensor Array and Multichannel
Source Coding and Compression
Compression and Modulation
Channel Estimation and Equalization
Blind Multiuser Communications
Signal Processing for Communications I
CDMA and Space-Time Processing
Time-Varying Channels and Self-Recovering Receivers
Signal Processing for Communications II
Blind CDMA and Multi-Channel Equalization
Multicarrier Communications
Detection, Classification, Localization, and Tracking
Radar and Sonar Signal Processing
Array Processing: Direction Finding
Array Processing Applications I
Blind Identification, Separation, and Equalization
Antenna Arrays for Communications
Array Processing Applications II
6: Multimedia Signal Processing, Image and Multidimensional Signal Processing, Digital Signal Processing Education
Multimedia Analysis and Retrieval
Audio and Video Processing for Multimedia Applications
Advanced Techniques in Multimedia
Video Compression and Processing
Image Coding
Transform Techniques
Restoration and Estimation
Image Analysis
Object Identification and Tracking
Motion Estimation
Medical Imaging
Image and Multidimensional Signal Processing Applications I
Segmentation
Image and Multidimensional Signal Processing Applications II
Facial Recognition and Analysis
Digital Signal Processing Education

Author Index
A B C D E F G H I
J K L M N O P Q R
S T U V W X Y Z

Blind Separation of Temporomandibular Joint Sound Signals

Authors:

Yinchao Guo, Time-Frequency Lab, c/o Chris Koh, Division of SEM, School of Mechanical and Production Engineering, Nanyang Technological University, Singapore 639798 (Singapore)
Farook Sattar, Division of Information Engineering, School of EEE, Nanyang Technological University, Nanyang Technological University, Singapore 639798 (Singapore)
Christopher Koh, School of Mechanical and Production Engineering, Nanyang Technological University, Singapore 639798 (Singapore)

Page (NA) Paper number 1226

Abstract:

In order to develop a cheap, efficient and reliable diagnostic tool for the detection of temporomandibular joint disorders (TMD), sounds from the temporomandibular joint (TMJ) are recorded using a pair of microphone inserted in the auditory canals. However, the TMJ sounds originating from one side of head can also be picked up by microphone at the other side. Blind source separation (BSS) is thus proposed as a method to recover the original sound. The authors propose to use non-casual filters for the separation of TMJ signals. The algorithm is based on information theory and is an extension of early work by Torkkola. The separation was successful and the output can now be used for subsequent analysis of TMJ sounds.

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Generalized Anti-Hebbian Learning For Source Separation

Authors:

Hsiao-Chun Wu,
Jose C. Principe,

Page (NA) Paper number 2394

Abstract:

The information-theoretic framework for source separation is highly suitable. However the choice of the nonlinearity or the estimation of the multidimensional joint probability density function are nontrivial. We propose here a generalized Gaussian model to construct a generalized blind source separation network based on the minimum entropy principle. This new separation network can suppress the interference to a significant amount compared to the traditional LMS-echo-canceler. The simulation is given to show the disparity of the performance as a varies. Finally how to choose the appropriate a in our generalized anti-Hebbian rule is discussed.

IC992394.PDF (From Author) IC992394.PDF (Rasterized)

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Complex Independent Component Analysis By Nonlinear Generalized Hebbian Learning with Rayleigh Nonlinearity

Authors:

Eraldo Pomponi,
Simone Fiori,
Francesco Piazza,

Page (NA) Paper number 1898

Abstract:

The aim of this paper is to present a non-linear Extension of the Sanger's Generalized Hebbian Algorithm to the processing of complex-valued data. A possible choice of the involved non-linearity is discussed recalling the Sudjianto-Hassoun nterpretation of the non-linear Hebbian learning. Extension of this interpretation to the complex case leads to a nonlinearity called Rayleigh function, which allows for separating mixed independent complex-valued source signals.

IC991898.PDF (Scanned)

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Sparse Basis Selection, ICA, and Majorization: Towards a Unified Perspective

Authors:

Ken Kreutz-Delgado,
Bhaskar D Rao,

Page (NA) Paper number 2411

Abstract:

Sparse solutions to the linear inverse problem Ax = y and the determination of an environmentally adapted overcomplete dictionary (the columns of A) depend upon the choice of a ``regularizing function'' d(x) in several recently proposed procedures. We discuss the interpretation of d(x) within a Bayesian framework, and the desirable properties that ``good'' (i.e., sparsity ensuring) regularizing functions, d(x) might have. These properties are: Schur-concavity (d(x) is consistent with majorization); concavity (d(x) has sparse minima); parameterizability (d(x) is drawn from a large, parameterizable class); and factorizability of the gradient of d(x) in a certain manner. The last property (which naturally leads one to consider separable regularizing functions) allows d(x) to be efficiently minimized subject to Ax=y using an Affine Scaling Transformation (AST)-like algorithm ``adapted'' to the choice of d(x). A Bayesian framework allows the algorithm to be interpreted as an Independent Component Analysis (ICA) procedure.

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Two Spatio-temporal Decorrelation Learning Algorithms and their Application to Multichannel Blind Deconvolution

Authors:

Seungjin Choi, School of Electrical and Electronics Engineering, Chungbuk National University, KOREA (Korea)
Andrzej Cichocki, Brain-style Information Systems Research Group, BSI, RIKEN, JAPAN (Japan)
Shun-ichi Amari, Brain-style Information Systems Research Group, BSI, RIKEN, JAPAN (Japan)

Page (NA) Paper number 1819

Abstract:

In this paper we present and compare two different spatio-temporal decorrelation learning algorithms for updating the weights of a linear feedforward network with FIR synapses (MIMO FIR filter). Both standard gradient and the natural gradient are employed to derive the spatio-temporal decorrelation algorithms. These two algorithms are applied to multichannel blind deconvolution task and their performance is compared. The rigorous derivation of algorithms and computer simulation results are presented.

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Adaptive Paraunitary Filter Banks for Principal and Minor Subspace Analysis

Authors:

Scott C Douglas, Dept. of Electrical Engr., Southern Methodist Univ., Dallas, TX USA (USA)
Shun-ichi Amari, RIKEN Brain Science Institute, Saitama, JAPAN (Japan)
S.-Y. Kung, Dept. of Electrical Engineering, Princeton Univ., Princeton, NJ USA (USA)

Page (NA) Paper number 1786

Abstract:

Paraunitary filter banks are important for several signal processing tasks. In this paper, we consider the task of adapting the coefficients of a multichannel FIR paraunitary filter bank via gradient ascent or descent on a chosen cost function. The proposed generalized algorithms inherently adapt the system's parameters in the space of paraunitary filters. Modifications and simplifications of the techniques for spatio-temporal principal and minor subspace analysis are described. Simulations verify one algorithm's useful behavior in this task.

IC991786.PDF (From Author) IC991786.PDF (Rasterized)

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