IWAENC 2003 Online Proceedings
  • Table of contents, author index and covers of the paper proceedings(PDF)
  • bibtex database file
  • IWAENC 2003 webpage is http://www.kecl.ntt.co.jp/icl/signal/iwaenc03/
  • "*" indicate the papers which were highly eveluated by the reviewers.
    Table of Contents

    Plenary Talks    

      [T-1] Statistical Methods for the Enhancement of Noisy Speech
        Rainer Martin, pp.1-6, [PDF]
      [T-2] The Automatic DJ: An Appealing and Instructive Signal Processing Education Project
        Piet Sommen, Harrie van Meer, Leo Vogten, Jean Ritzerfeld, pp.7-14, [PDF] [PostScript]
      [T-3] Independent Component Analysis and its Applications to Sound Signal Separation
        Kiyotoshi Matsuoka, pp.15-18, [PDF]

    Poster 1: Acoustic Echo Control and Adaptive Filtering Algorithm

    *[P1-01] An Outlier-Robust Extended Multidelay Filter with Application to Acoustic Echo Cancellation
        Herbert Buchner, Jacob Benesty, Tomas Gänsler, Walter Kellermann, pp.19-22, [PDF]
    *[P1-02] Double-Talk Robust Frequency Domain Echo Cancellation Algorithm with Scalable Nonlinear Reference and Error Functions
        Suehiro Shimauchi, Yoichi Haneda, Akitoshi Kataoka, pp.23-26, [PDF]
    *[P1-03] On the Application of the Unscented Kalman Filter to Speech Processing
        Sharon Gannot, Marc Moonen, pp.27-30, [PDF]
      [P1-04] On Data-Reuse Adaptive Algorithms
        Jacob Benesty, Tomas Gänsler, pp.31-34, [PDF]
      [P1-05] An Echo Canceller based on the Structure of Dual-Auxiliary Filters
        Xiongbing Ou, Zhe Chen, Fuliang Yin, pp.35-38, [PDF]
      [P1-06] Acoustic Echo Canceling in the Double-Talk Condition
        Mohammad Reza Asharif, Rui Chen, Hayato Nakamura, Katsumi Yamashita, pp.39-42, [PDF]
      [P1-07] Robust and Elegant, Purely Statistical Adaptation of Acoustic Echo Canceler and Postfilter
        Gerald Enzner, Peter Vary, pp.43-46, [PDF]
      [P1-08] Post-Filtering for Stereo Acoustic Echo Cancellation
        Markus Kallinger, Jörg Bitzer, Karl-Dirk Kammeyer, pp.47-50, [PDF]
      [P1-09] Multichannel Teleconferencing System with Multi Spatial Region Acoustic Echo Cancellation
        Kong-Aik Lee, Woon-Seng Gan, Jun Yang, Farook Sattar, pp.51-54, [PDF]
      [P1-10] Adaptive Parallel Subgradient Projection Techniques with Input Sliding Technique for Stereophonic Acoustic Echo Cancellation
        Masahiro Yukawa, Isao Yamada, pp.55-58, [PDF]
      [P1-11] The Use of Partial Update Schemes to Reduce Inter-Channel Coherence in Adaptive Stereophonic Acoustic Echo Cancellation
        Andy Wai Hoong Khong, Patrick Naylor, pp.59-62, [PDF]
      [P1-12] Stereophonic Acoustic Echo Canceller with Pre-Processing - Second-Order Pre-Processing Filter and its Convergence -
        Akihiro Hirano, Kenji Nakayama, Kazue Takebe, pp.63-66, [PDF]
      [P1-13] Room Impulse Response Variation Due to Thermal Fluctuation and its Impact on Acoustic Echo Cancellation
        Gary Elko, Eric Diethorn, Tomas Gänsler, pp.67-70, [PDF]
      [P1-14] A Method for Detecting Echo Path Variation
        Jiaquan Huo, Sven Nordholm, Zhuquan Zang, pp.71-74, [PDF]
      [P1-15] Proportionate NLMS Algorithm for Second-Order Volterra Filters and its Application to Nonlinear Echo Cancellation
        Fabian Kuech, Walter Kellermann, pp.75-78, [PDF]
      [P1-16] On Performance of Linear Adaptive Filtering Algorithms in Acoustic Echo Control in Presence of Distorting Loudspeakers
        Riitta Niemistö, Tuomo Mäkelä, pp.79-82, [PDF]

    Poster 2: Sound Enhancement and Noise Reduction

    *[P2-01] Noise Reduction by Maximum a Posteriori Spectral Amplitude Estimation with Supergaussian Speech Modeling
        Thomas Lotter, Peter Vary, pp.83-86, [PDF]
      [P2-02] Speech Enhancement in the DFT Domain using Laplacian Speech Priors
        Rainer Martin, Colin Breithaupt, pp.87-90, [PDF]
      [P2-03] Implementation and Effects of Single Channel Dereverberation based on the Harmonic Structure of Speech
        Tomohiro Nakatani, Masato Miyoshi, Keisuke Kinoshita, pp.91-94, [PDF]
      [P2-04] Speech Dereverberation via Subband Implementation of Subspace Methods
        Sharon Gannot, Marc Moonen, pp.95-98, [PDF]
      [P2-05] On the Use of Linear Prediction for Dereverberation of Speech
        Nikolay Gaubitch, Patrick Naylor, Darren Ward, pp.99-102, [PDF]
      [P2-06] Limitations of FIR Multi-Microphone Speech Dereverberation in the Low-Delay Case
        Markus Hofbauer, Hans-Andrea Loeliger, pp.103-106, [PDF]
      [P2-07] Improved Artificial Low-Pass Extension of Telephone Speech
        Ulrich Kornagel, pp.107-110, [PDF]
      [P2-08] Kalman Filter-based Single Microphone Noise Canceller
        Marcel Gabrea, pp.111-114, [PDF]
      [P2-09] Regularized Optimization with Spectral Smoothing for Speech Spectral Estimation
        Karsten Vandborg Sørensen, Søren Vang Andersen, pp.115-118, [PDF]
      [P2-10] Speech Enhancement of Noisy Speech using Log-Spectral Amplitude Estimator and Harmonic Tunneling
        Hyoung-Gook Kim, Markus Schwab, Nicolas Moreau, Thomas Sikora, pp.119-122, [PDF]
      [P2-11] Speech Enhancement based on Linear Prediction Error Signals and Spectral Subtraction
        Agustín Álvarez-Marquina, Víctor Nieto-Lluis, Pedro Gómez-Vilda, Rafael Martínez-Olalla, pp.123-126, [PDF]
      [P2-12] Spectral Subtraction based on Speech/noise-Dominant Classification
        Yukihiro Nomura, Jianming Lu, Hiroo Sekiya, Takashi Yahagi, pp.127-130, [PDF]
      [P2-13] Methodology for the Design of a Robust Voice Activity Detector for Speech Enhancement
        Virginie Gilg, Christophe Beaugeant, Martin Schönle, Bernt Andrassy, pp.131-134, [PDF]
      [P2-14] Noise Suppression based on Teager Energy Operator for Improving the Robustness of ASR Front-End
        Junhui Zhao, Jingming Kuang, Xiang Xie, Huang Shilei, pp.135-138, [PDF]
      [P2-15] Effects of Harmonic Components Generated by Polynomial Preprocessor in Acoustic Echo Control
        Tuomo Mäkelä, Riitta Niemistö, pp.139-142, [PDF]

    Poster 3: Active Noise Control, Hearing Aids, and Hardware

    *[P3-01] Crosstalk-Resistant Three-Channel Noise Canceller
        Akihiro Hirano, Kenji Nakayama, Shin'ya Arai, pp.143-146, [PDF]
    *[P3-02] Spatially Pre-Processed Speech Distortion Weighted Multi-Channel Wiener Filtering for Noise Reduction in Hearing Aids
        Ann Spriet, Marc Moonen, Jan Wouters, pp.147-150, [PDF]
      [P3-03] Analysis of a Feedback-Type Active Noise Control System with Online Secondary Path Modeling and its Application to Hearing Aids
        Hideaki Sakai, Shotaro Inoue, Yoichi Hinamoto, pp.151-154, [PDF]
      [P3-04] A New Structure for Feedforward Active Noise Control Systems with Online Secondary-Path Modeling
        Muhammad Tahir Akhtar, Masahide Abe, Masayuki Kawamata, pp.155-158, [PDF]
      [P3-05] Multi-Channel Active Noise Control for All Uncertain Primary and Secondary Paths
        Yuhsuke Ohta, Akira Sano, pp.159-162, [PDF]
      [P3-06] Functional Link Artificial Neural Network for Active Control of Nonlinear Noise Processes
        Ganapati Panda, Debi Prasad Das, pp.163-166, [PDF]
      [P3-07] A Theoretical Analysis for Feedforward ANC System with State Equation Model
        Iwao Nagashiro, Toichi Machida, pp.167-170, [PDF]
      [P3-08] Comparison of Adaptive Noise Reduction Algorithms in Dual Microphone Hearing Aids
        Jean Baptiste Maj, Liesbeth Royackers, Marc Moonen, Jan Wouters, pp.171-174, [PDF]
      [P3-09] Active Noise Control using the Perturbation Method -Verification in Actual Multi-Channel Systems-
        Tasuku Ainoya, Takashi Mori, Yoshinobu Kajikawa, Yasuo Nomura, pp.175-178, [PDF]
      [P3-10] An Algorithm for Cancellation of Sidetone Oscillations
        Martin Schönle, Virginie Gilg, pp.179-182, [PDF]
      [P3-11] Real-Time TF-GSC in Nonstationary Noise Environments
        Israel Cohen, Sharon Gannot, Baruch Berdugo, pp.183-186, [PDF]
      [P3-12] Algorithm of a Single Chip Acoustic Echo Canceller using Cascaded Cross Spectral Estimation
        Marco Liem, Hyoung-Gook Kim, Otto Manck, pp.187-190, [PDF]
      [P3-13] Implementing and Evaluating an Audio Teleconferencing Terminal with Noise and Echo Reduction
        Sumitaka Sakauchi, Akira Nakagawa, Yoichi Haneda, Akitoshi Kataoka, pp.191-194, [PDF]
      [P3-14] A PC based Platform for Multichannel Real-Time Audio Processing
        Hauke Krüeger, Thomas Lotter, Gerald Enzner, Peter Vary, pp.195-198, [PDF]
      [P3-15] Perception Oriented, Delay-Controlled Echo Cancellation in IP based Telephone Networks
        Wolfgang Brandstätter, Frank Kettler, pp.199-202, [PDF]
      [P3-16] Improvement of Noise Source Identification by Phase Redundant Acoustical Holography
        Haruo Uchiyama, pp.203-206, [PDF]

    Poster 4: Microphone Array and Sound Separation 1

    *[P4-01] Time Delay Estimation using Spatial Correlation Techniques
        Jingdong Chen, Jacob Benesty, Yiteng (Arden) Huang, pp.207-210, [PDF]
    *[P4-02] Blind Source Separation When Speech Signals Outnumber Sensors using a Sparseness-Mixing Matrix Estimation (SMME)
        Audrey Blin, Shoko Araki, Shoji Makino, pp.211-214, [PDF]
    *[P4-03] Approaches for Time Difference of Arrival Estimation in a Noisy and Reverberant Environment
        Tsvi Gregory Dvorkind, Sharon Gannot, pp.215-218, [PDF]
      [P4-04] Array Geometry Arrangement for Frequency Domain Blind Source Separation
        Ryo Mukai, Hiroshi Sawada, Sebastien F. G. M. de la Kethulle de Ryhove, Shoko Araki, Shoji Makino, pp.219-222, [PDF]
      [P4-05] Least-Squares Error Beamforming using Minimum Statistics and Multichannel Frequency-Domain Adaptive Filtering
        Robert Aichner, Wolfgang Herbordt, Herbert Buchner, Walter Kellermann, pp.223-226, [PDF]
      [P4-06] DOA Estimation of Speech Signal with a Small Number of Microphone Array in Real Acoustical Environment
        Yusuke Hioka, Nozomu Hamada, pp.227-230, [PDF]
      [P4-07] Geometrical Understanding of the PCA Subspace Method for Overdetermined Blind Source Separation
        Stefan Winter, Hiroshi Sawada, Shoji Makino, pp.231-234, [PDF]
      [P4-08] Estimation of Direction of Arrival using Matching Pursuit and its Application to Source Separation
        Yasuhiro Oikawa, Yoshio Yamasaki, pp.235-238, [PDF]
      [P4-09] A New Adaptive Blocking Matrix with Exact FIR Structure for Robust Generalized Sidelobe Canceller
        Zhaorong Hou, Ying Jia, pp.239-242, [PDF]
      [P4-10] Adaptive Beamformer based on Average Vowel / Consonant Spectrum with Phoneme Identification
        Masato Nakayama, Takanobu Nishiura, Hideki Kawahara, pp.243-246, [PDF] [PostScript]
      [P4-11] Robust Spatial Estimation of the Signal-to-Interference Ratio for Non-Stationary Mixtures
        Wolfgang Herbordt, Tim Trini, Walter Kellermann, pp.247-250, [PDF]
      [P4-12] High-Fidelity Blind Separation of Acoustic Signals using SIMO-Model-based ICA with Information-Geometric Learning
        Tomoya Takatani, Tsuyoki Nishikawa, Hiroshi Saruwatari, Kiyohiro Shikano, pp.251-254, [PDF]
      [P4-13] Concurrent Speech Signal Separation based on Frequency Domain Binaural Model
        Yoshifumi Chisaki, Takashi Nakanishi, Hidetoshi Nakashima, Tsuyoshi Usagawa, pp.255-258, [PDF]
      [P4-14] Differential Microphone Arrays for Spectral Subtraction
        Marc Ihle, pp.259-262, [PDF]
      [P4-15] Sound Source Localization using a Pinna-based Profile Fitting Method
        Osamu Ichikawa, Tetsuya Takiguchi, Masafumi Nishimura, pp.263-266, [PDF]

    Poster 5: Microphone Array and Sound Separation 2

    *[P5-01] Design of Broadband Beamformers Robust Against Microphone Position Errors
        Simon Doclo, Marc Moonen, pp.267-270, [PDF]
    *[P5-02] Blind Separation of More Speech than Sensors with Less Distortion by Combining Sparseness and ICA
        Shoko Araki, Shoji Makino, Audrey Blin, Ryo Mukai, Hiroshi Sawada, pp.271-274, [PDF]
    *[P5-03] Blind Source Separation for Convolutive Mixtures Exploiting Nongaussianity, Nonwhiteness, and Nonstationarity
        Herbert Buchner, Robert Aichner, Walter Kellermann, pp.275-278, [PDF]
      [P5-04] Blind Separation for Convolutive Mixture of Many Voices
        Kiyotoshi Matsuoka, Yoshihisa Ohba, Yasunobu Toyota, Satoshi Nakashima, pp.279-282, [PDF]
      [P5-05] Identification and Tracking of Active Speaker's Position in Noisy Environments
        Tomasz Maciej Rutkowski, Masahiro Yokoo, Danilo P. Mandic, pp.283-286, [PDF]
      [P5-06] Problems in Blind Separation of Convolutive Speech Mixtures by Negentropy Maximization
        Raj Kishore Prasad, Hiroshi Saruwatari, Kiyohiro Shikano, pp.287-290, [PDF]
      [P5-07] Introducing New Mechanism in the Learning Process of FDICA-based Speech Separation
        Masahiro Furukawa, Yusuke Hioka, Takuro Ema, Nozomu Hamada, pp.291-294, [PDF]
      [P5-08] Speaker Localization Exploiting Spatial-Temporal Information
        Tsvi Gregory Dvorkind, Sharon Gannot, pp.295-298, [PDF]
      [P5-09] Evaluation of Blind Separation and Deconvolution for Convolutive Speech Mixture using SIMO-Model-based ICA
        Hiroaki Yamajo, Hiroshi Saruwatari, Tomoya Takatani, Tsuyoki Nishikawa, Kiyohiro Shikano, pp.299-302, [PDF]
      [P5-10] Blind Source Separation for Convolutive Mixtures based on Complexity Minimization
        Sebastien F. G. M. de la Kethulle de Ryhove, Ryo Mukai, Hiroshi Sawada, Shoji Makino, pp.303-306, [PDF]
      [P5-11] Speech Enhancement Employing Adaptive Beamformer with Recursively Updated Soft Constraints
        Hai Q. Dam, Sven Nordholm, Nedelko Grbic, Hai Huyen Dam, pp.307-310, [PDF]
      [P5-12] Spectral Smoothing for Frequency-Domain Blind Source Separation
        Hiroshi Sawada, Ryo Mukai, Sebastien F. G. M. de la Kethulle de Ryhove, Shoko Araki, Shoji Makino, pp.311-314, [PDF]
      [P5-13] Considering the Second Peak in the GCC Function for Multi-Source TDOA Estimation with a Microphone Array
        Dirk Bechler, Kristian Kroschel, pp.315-318, [PDF]
      [P5-14] Detection of Speech Events in Real Environments Through Fusion of Audio and Video Information using Bayesian Networks
        Takashi Yoshimura, Futoshi Asano, Youichi Motomura, Hideki Asoh, Naoyuki Ichimura, Kiyoshi Yamamoto, Satoshi Nakamura, pp.319-322, [PDF]