• Title/Summary/Keyword: Blind Speech Source Separation

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An Acoustic Echo Canceller for Stereo Using Blind Signal Separation (암묵신호분리를 이용한 스테레오 음향반향제거기)

  • Lee, Haeng Woo
    • Journal of Korea Society of Digital Industry and Information Management
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    • v.8 no.3
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    • pp.125-131
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    • 2012
  • This paper is on a stereo acoustic echo canceller with the blind signal separation. The convergence speed of the stereo acoustic echo canceller is deteriorated due to mixing two residual signals in the update signal of each echo canceller. To solve this problem, we are to use the blind signal separation(BSS) method separating the mixed signals. The blind signal separation method can extracts the source signals by means of the iterative computations with two input signals. We had verified performances of the proposed acoustic echo canceller for stereo through simulations. The results of simulations show that the acoustic echo canceller for stereo using this algorithm operates stably without divergence in the normal state. And, when the speech signals were inputted, this echo canceller achieved about 3dB higher ERLE in the case of using the BSS algorithm than the case of not using the BSS algorithm. But this echo canceller didn't get good performances in the case of inputting the white noises as stereo signals.

Online blind source separation and dereverberation of speech based on a joint diagonalizability constraint (공동 행렬대각화 조건 기반 온라인 음원 신호 분리 및 잔향제거)

  • Yu, Ho-Gun;Kim, Do-Hui;Song, Min-Hwan;Park, Hyung-Min
    • The Journal of the Acoustical Society of Korea
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    • v.40 no.5
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    • pp.503-514
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    • 2021
  • Reverberation in speech signals tends to significantly degrade the performance of the Blind Source Separation (BSS) system. Especially in online systems, the performance degradation becomes severe. Methods based on joint diagonalizability constraints have been recently developed to tackle the problem. To improve the quality of separated speech, in this paper, we add the proposed de-reverberation method to the online BSS algorithm based on the constraints in reverberant environments. Through experiments on the WSJCAM0 corpus, the proposed method was compared with the existing online BSS algorithm. The performance evaluation by the Signal-to-Distortion Ratio and the Perceptual Evaluation of Speech Quality demonstrated that SDR improved from 1.23 dB to 3.76 dB and PESQ improved from 1.15 to 2.12 on average.

Convolutive source separation in noisy environments (잡음 환경하에서의 음성 분리)

  • Jang Inseon;Choi Seungjin
    • Proceedings of the KSPS conference
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    • 2003.10a
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    • pp.97-100
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    • 2003
  • This paper addresses a method of convolutive source separation that based on SEONS (Second Order Nonstationary Source Separation) [1] that was originally developed for blind separation of instantaneous mixtures using nonstationarity. In order to tackle this problem, we transform the convolutive BSS problem into multiple short-term instantaneous problems in the frequency domain and separated the instantaneous mixtures in every frequency bin. Moreover, we also employ a H infinity filtering technique in order to reduce the sensor noise effect. Numerical experiments are provided to demonstrate the effectiveness of the proposed approach and compare its performances with existing methods.

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Speech Enhancement for Voice commander in Car environment (차량환경에서 음성명령어기 사용을 위한 음성개선방법)

  • 백승권;한민수;남승현;이봉호;함영권
    • Journal of Broadcast Engineering
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    • v.9 no.1
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    • pp.9-16
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    • 2004
  • In this paper, we present a speech enhancement method as a pre-processor for voice commander under car environment. For the friendly and safe use of voice commander in a running car, non-stationary audio signals such as music and non-candidate speech should be reduced. Ow technique is a two microphone-based one. It consists of two parts Blind Source Separation (BSS) and Kalman filtering. Firstly, BSS is operated as a spatial filter to deal with non-stationary signals and then car noise is reduced by kalman filtering as a temporal filter. Algorithm Performance is tested for speech recognition. And the results show that our two microphone-based technique can be a good candidate to a voice commander.

Blind Audio Source Separation Based On High Exploration Particle Swarm Optimization

  • KHALFA, Ali;AMARDJIA, Nourredine;KENANE, Elhadi;CHIKOUCHE, Djamel;ATTIA, Abdelouahab
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.13 no.5
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    • pp.2574-2587
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    • 2019
  • Blind Source Separation (BSS) is a technique used to separate supposed independent sources of signals from a given set of observations. In this paper, the High Exploration Particle Swarm Optimization (HEPSO) algorithm, which is an enhancement of the Particle Swarm Optimization (PSO) algorithm, has been used to separate a set of source signals. Compared to PSO algorithm, HEPSO algorithm depends on two additional operators. The first operator is based on the multi-crossover mechanism of the genetic algorithm while the second one relies on the bee colony mechanism. Both operators have been employed to update the velocity and the position of the particles respectively. Thus, they are used to find the optimal separating matrix. The proposed method enhances the overall efficiency of the standard PSO in terms of good exploration and performance. Based on many tests realized on speech and music signals supplied by the BSS demo, experimental results confirm the robustness and the accuracy of the introduced BSS technique.

Overlapped Subband-Based Independent Vector Analysis

  • Jang, Gil-Jin;Lee, Te-Won
    • The Journal of the Acoustical Society of Korea
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    • v.27 no.1E
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    • pp.30-34
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    • 2008
  • An improvement to the existing blind signal separation (BSS) method has been made in this paper. The proposed method models the inherent signal dependency observed in acoustic object to separate the real-world convolutive sound mixtures. The frequency domain approach requires solving the well known permutation problem, and the problem had been successfully solved by a vector representation of the sources whose multidimensional joint densities have a certain amount of dependency expressed by non-spherical distributions. Especially for speech signals, we observe strong dependencies across neighboring frequency bins and the decrease of those dependencies as the bins become far apart. The non-spherical joint density model proposed in this paper reflects this property of real-world speech signals. Experimental results show the improved performances over the spherical joint density representations.

Implementation of Environmental Noise Remover for Speech Signals (배경 잡음을 제거하는 음성 신호 잡음 제거기의 구현)

  • Kim, Seon-Il;Yang, Seong-Ryong
    • 전자공학회논문지 IE
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    • v.49 no.2
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    • pp.24-29
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    • 2012
  • The sounds of exhaust emissions of automobiles are independent sound sources which are nothing to do with voices. We have no information for the sources of voices and exhaust sounds. Accordingly, Independent Component Analysis which is one of the Blind Source Separaton methods was used to segregate two source signals from each mixed signals. Maximum Likelyhood Estimation was applied to the signals came through the stereo microphone to segregate the two source signals toward the maximization of independence. Since there is no clue to find whether it is speech signal or not, the coefficients of the slope was calculated by the autocovariances of the signals in frequcency domain. Noise remover for speech signals was implemented by coupling the two algorithms.

Frequency Domain Blind Source Seperation Using Cross-Correlation of Input Signals (입력신호 상호상관을 이용한 주파수 영역 블라인드 음원 분리)

  • Sung Chang Sook;Park Jang Sik;Son Kyung Sik;Park Keun-Soo
    • Journal of Korea Multimedia Society
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    • v.8 no.3
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    • pp.328-335
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    • 2005
  • This paper proposes a frequency domain independent component analysis (ICA) algorithm to separate the mixed speech signals using a multiple microphone array By estimating the delay timings using a input cross-correlation, even in the delayed mixture case, we propose a good initial value setting method which leads to optimal convergence. To reduce the calculation, separation process is performed at frequency domain. The results of simulations confirms the better performances of the proposed algorithm.

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Microphone Array Based Speech Enhancement Using Independent Vector Analysis (마이크로폰 배열에서 독립벡터분석 기법을 이용한 잡음음성의 음질 개선)

  • Wang, Xingyang;Quan, Xingri;Bae, Keunsung
    • Phonetics and Speech Sciences
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    • v.4 no.4
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    • pp.87-92
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    • 2012
  • Speech enhancement aims to improve speech quality by removing background noise from noisy speech. Independent vector analysis is a type of frequency-domain independent component analysis method that is known to be free from the frequency bin permutation problem in the process of blind source separation from multi-channel inputs. This paper proposed a new method of microphone array based speech enhancement that combines independent vector analysis and beamforming techniques. Independent vector analysis is used to separate speech and noise components from multi-channel noisy speech, and delay-sum beamforming is used to determine the enhanced speech among the separated signals. To verify the effectiveness of the proposed method, experiments for computer simulated multi-channel noisy speech with various signal-to-noise ratios were carried out, and both PESQ and output signal-to-noise ratio were obtained as objective speech quality measures. Experimental results have shown that the proposed method is superior to the conventional microphone array based noise removal approach like GSC beamforming in the speech enhancement.

Comparison of Independent Component Analysis and Blind Source Separation Algorithms for Noisy Data (잡음환경에서 독립성분 분석과 암묵신호분리 알고리즘의 성능비교)

  • O, Sang-Hun;Cichocki, Andrzej;Choe, Seung-Jin;Lee, Su-Yeong
    • Journal of the Institute of Electronics Engineers of Korea CI
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    • v.39 no.2
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    • pp.10-20
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    • 2002
  • Various blind source separation (BSS) and independent component analysis (ICA) algorithms have been developed. However, comparison study for BSS/ICA algorithms has not been extensively carried out yet. The main objective of this paper is to compare various promising BSS/ICA algorithms in terms of several factors such as robustness to sensor noise, computational complexity, the conditioning of the mixing matrix, the number of sensors, and the number of training patterns. We propose several benchmarks which are useful for the evaluation of the algorithm. This comparison study will be useful for real-world applications, especially EEG/MEG analysis and separation of miked speech signals.