Speech Enhancement Using Nonlinear Microphone Array Based on Noise Adaptive Complementary Beamforming
スポンサーリンク
概要
- 論文の詳細を見る
This paper describes an improved complementary beamforming microphone array based on the new noise adaptation algorithm. Complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this system, during a pause in the target speech, two directivity patterns of the beamformers are adapted to the noise directions of arrival so that the expectation values of each noise power spectrum are minimized in the array output. Using this technique, we can realize the directional nulls for each noise even when the number of sound sources exceeds that of microphones. To evaluate the effectiveness, speech enhancement experiments and speech recognition experiments are performed based on computer simulations with a two-element array and three sound sources under various noise conditions. In comparison with the conventional adaptive beamformer and the conventional spectral subtraction method cascaded with the adaptive beamformer, it is shown that(1)the proposed array improves the signal-to-noise ratio(SNR)of degraded speech by more than 6 dB when the interfering noise is two speakers with the input SNR of below 0 dB, (2)the proposed array improves the SNR by about 2 dB when the interfering noise is bubble noise, and(3)an improvement in the recognition rate of more than 18% is obtained when the interfering noise is two speakers or two overlapped signals of some speakers under the condition that the input SNR is 10 dB.
- 社団法人電子情報通信学会の論文
- 2000-05-25
著者
-
TAKEDA Kazuya
Department of Nuclear Engineering, School of Engineering, Tokai University
-
Takeda K
Nagoya Univ. Nagoya Jpn
-
Takeda Kazuya
Department Of Information Electronics Graduate School Of Engineering Nagoya University
-
SARUWATARI Hiroshi
Department of Dermatology, Kagoshima University Graduate School of Medical and Dental Sciences
-
Kajita S
Center For Information Media Studies Nagoya University
-
Takeda K
Center For Integrated Acoustic Information Research Graduate School Of Engineering Nagoya University
-
SARUWATARI Hiroshi
Graduate School of Information Science, Nara Institute of Science and Technology
-
Saruwatari Hiroshi
Graduate School Of Information Science Nara Institute Of Science And Technology
-
Saruwatari Hiroshi
Department Of Dermatology Kagoshima University Graduate School Of Medical And Dental Sciences
-
Saruwatari H
Graduate School Of Information Science Nara Institute Of Science And Technology
-
ITAKURA Fumitada
Center for Information Media Studies, Nagoya University
-
Itakura F
Graduate School Of Information Engineering Meijo University
-
Itakura Fumitada
Center For Information Media Studies Nagoya University
-
KAJITA Shoji
Center for Information Media Studies, Nagoya University
関連論文
- 磁化シｰトプラズマを用いたガス･ダイバｰタの基礎実験
- CENSREC-1-C : An evaluation framework for voice activity detection under noisy environments
- Genetic analysis of the ferrochelatase gene in eight Japanese patients from seven families with erythropoietic protoporphyria
- Building an Effective Speech Corpus by Utilizing Statistical Multidimensional Scaling Method
- Cost Reduction of Acoustic Modeling for Real-Environment Applications Using Unsupervised and Selective Training
- IMPROVEMENT OF CHOLEDOCHOSCOPY : CHROMOENDOCHOLEDOCHOSCOPY, AUTOFLUORESCENCE IMAGING, OR NARROW-BAND IMAGING
- Effect of Central Limit Theorem non-compliance on blind separation of speech by negentropy maximization
- Robots that can hear, understand and talk
- Probability Distribution of Time-Series of Speech Spectral Components(Audio/Speech Coding)(Applications and Implementations of Digital Signal Processing)
- AURORA-2J: An Evaluation Framework for Japanese Noisy Speech Recognition(Speech Corpora and Related Topics, Corpus-Based Speech Technologies)
- CENSREC-3: An Evaluation Framework for Japanese Speech Recognition in Real Car-Driving Environments(Speech and Hearing)
- Non-Audible Murmur (NAM) Recognition Exploiting Adaptation Techniques
- MC-32 Development of microdrive assembly process
- Multiple Regression of Log Spectra for In-Car Speech Recognition Using Multiple Distributed Microphones(Feature Extraction and Acoustic Medelings, Corpus-Based Speech Technologies)
- Objective sound quality evaluation for combination method of beamforming and spectral subtraction (応用音響)
- Fast Convergence Blind Source Separation Using Frequency Subband Interpolation by Null Beamforming
- Rapid Compensation of Temperature Fluctuation Effect for Multichannel Sound Field Reproduction System
- Development, Long-Term Operation and Portability of a Real-Environment Speech-Oriented Guidance System
- On-Line Relaxation Algorithm Applicable to Acoustic Fluctuation for Inverse Filter in Multichannel Sound Reproduction System(Sound Field Reproduction, Multi-channel Acoustic Signal Processing)
- Iterative Inverse Filter Relaxation Algorithm for Adaptation to Acoustic Fluctuation in Sound Reproduction System
- Sound Reproduction System Including Adaptive Compensation of Temperature Fluctuation Effect for Broad-Band Sound Control(Special Section on Digital Signal Processing)
- Cutaneous Mycobacterium intracellulare infection in a bone marrow transplantation recipient
- Interface for Barge-in Free Spoken Dialogue System Combining Adaptive Sound Field Control and Microphone Array(Speech and Hearing)
- SNR and sub-band SNR estimation based on Gaussian mixture modeling in the log power domain with application for speech enhancements (第6回音声言語シンポジウム)
- Driver's irritation detection using speech recognition results (音声・第10回音声言語シンポジウム)
- Driver's irritation detection using speech recognition results (音声言語情報処理)
- Driver's irritation detection using speech recognition results (言語理解とコミュニケーション・第10回音声言語シンポジウム)
- A Self-Generator Method for Initial Filters of SIMO-ICA Applied to Blind Separation of Binaural Sound Mixtures(Blind Source Separation, Multi-channel Acoustic Signal Processing)
- Multistage SIMO-Model-Based Blind Source Separation Combining Frequency-Domain ICA and Time-Domain ICA(Adaptive Signal Processing and Its Applications)
- サブバンドに含まれる周波数成分の瞬時周波数に基づく推定
- Lack of Interaction Between Cefdinir and Calcium Polycarbophil : In vitro and In vivo Studies
- Predicting the Degradation of Speech Recognition Performance from Sub-band Dynamic Ranges (特集 音声言語情報処理とその応用)
- A model of perceptual distance for group delays based on ellipsoidal mapping
- The effect of group delay spectrum on timbre
- Direction of Arrival Estimation Using Nonlinear Microphone Array
- Speech Enhancement Using Nonlinear Microphone Array Based on Noise Adaptive Complementary Beamforming
- Speech Enhancement Using Nonlinear Microphone Array Based on Complementary Beamforming (Special Section on Digital Signal Processing)
- Noise Robust Speech Recognition Using Subband-Crosscorrelation Analysis
- An Acoustically Oriented Vocal-Tract Model
- Evaluation of Extremely Small Sound Source Signals Used in Speaking-Aid System with Statistical Voice Conversion
- Adaptive Nonlinear Regression Using Multiple Distributed Microphones for In-Car Speech Recognition(Speech Enhancement, Multi-channel Acoustic Signal Processing)
- Improvements of the One-to-Many Eigenvoice Conversion System
- Esophageal Speech Enhancement Based on Statistical Voice Conversion with Gaussian Mixture Models
- Adaptive Training for Voice Conversion Based on Eigenvoices
- Blind Separation and Deconvolution for Convolutive Mixture of Speech Combining SIMO-Model-Based ICA and Multichannel Inverse Filtering(Engineering Acoustics)
- Subband-Based Blind Separation for Convolutive Mixtures of Speech(Engineering Acoustics)
- Overdetermined Blind Separation for Real Convolutive Mixtures of Speech Based on Multistage ICA Using Subarray Processing(Speech/Acoustic Signal Processing)(Digital Signal Processing)
- Stable Learning Algorithm for Blind Separation of Temporally Correlated Acoustic Signals Combining Multistage ICA and Linear Prediction(Digital Signal Processing)
- Blind Source Separation of Acoustic Signals Based on Multistage ICA Combining Frequency-Domain ICA and Time-Domain ICA
- Fast-Convergence Algorithm for Blind Source Separation Based on Array Signal Processing
- An Iterative Inverse Filter Design Method for the Multichannel Sound Field Sound Field Reproduction System(Special Section on Acoustic Signal Processing)
- On the use of two-mass vocal cord model in characterizing the stress speech (音声)
- Particle Size Distribution Measurement of Free-Falling Fine Particles in a Dusty Plasma Experiment
- Semi-Blind Optimization Scheme of Joint Suppression of Background Noise and Late Reverberation
- Relaxation behavior of laser-peening residual stress under tensile loading investigated by X-ray and neutron diffraction