• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.34-44
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• 2010

In this study, the algorithms for multichannel format conversion and robust representation of spatial sound information are proposed. In the spatial analysis, the directional information of sound source is estimated and sound sources are separated from stereo signal. In the spatial resynthesis, the multichannel matrixing with spatial repanning and post-scaling method are applied to represent a spatial sound. The conventional method about channel format conversion has the problem that the energy of sound source and the spatial information are not preserved in the desired channel format. Because the proposed method is designed in consideration of the target multichannel format and its resynthesized signal, the robust representation of spatial sound can be achieved in the multichannel format conversion.

• The Journal of the Acoustical Society of Korea
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• v.42 no.4
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• pp.357-363
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• 2023

In underwater signal processing, separating individual signals from mixed signals has long been a challenge due to low signal quality. The common method using Short-time Fourier transform for spectrogram analysis has faced criticism for its complex parameter optimization and loss of phase data. We propose a Triple-path Recurrent Neural Network, based on the Dual-path Recurrent Neural Network's success in long time series signal processing, to handle three-dimensional tensors from multi-channel sensor input signals. By dividing input signals into short chunks and creating a 3D tensor, the method accounts for relationships within and between chunks and channels, enabling local and global feature learning. The proposed technique demonstrates improved Root Mean Square Error and Scale Invariant Signal to Noise Ratio compared to the existing method.

• The Journal of the Acoustical Society of Korea
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• v.25 no.1
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• pp.14-20
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• 2006

In this paper. a novel technique to set up the initial weights in BSS of delayed mixtures is proposed. After analyzing Eigendecomposition for the correlation matrix of mixing data. the initial weights are set from the Eigenvectors ith delay information. The Proposed setting of initial weighting method for conventional FDICA technique improved the separation Performance. The computer simulation shows that the Proposed method achieves the improved SIR and faster convergence speed of learning curve.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.351-359
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• 2003

In this paper, we proposed a method for acoustic echo cancellation based on independent component analysis. When the large acoustic noise is picked up by the microphone, the performance of echo cancellation decreased. We used two microphones that received echo signal which is linearly mixed with the noise, then separated the echo signals from the received signals with independent component analysis algorithm. The separated echo signal is used for the reference signal of adaptive algorithm which leads to better performance of the echo cancellation. Computer simulation results show the validity of the proposed method.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.175-178
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• 1994

본 논문에서는 선형 예측 방법과 다중 대역 여기 방법의 장점을 조합하여 낮은 전송률에서 고품질의 합성음을 제공하는 LP-SMBE 부호기를 제안한다. LP-SMBE 부호기에서는 선형 예측 방법과 단순화된 여기 신호 추정방법을 이용하여 성도 특성 정보와 여기 신호를 분리 추정한다. 제안한 단순화된 여기 신호 추정 방법은 정규화된 스펙트럼 영역에서 원음 스펙트럼과 합성 스펙트럼을 비교하여 여기 신호를 추정한다. 이 방법은 기존 MBE 방법의 여기 신호 추정 방법보다 연산량이 적고, 여기 신호르 F보다 정확히 추정할 수 있다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.105-106
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• 2004

음성인식, 합성 및 분석과 같은 음성신호처리 분야에 있어서 피치검출이나 포만트검출은 매우 중요하다. 주파수 영역의 스펙트럼 신호는 잡음이 부가되는 경우에도 고조파정보와 포만트 포락선 정보를 유지하기 때문에 음성신호처리분야에서 매우 유용하다고 할 수 있다. 고조파 정보나 포만트 포락선 정보는 피치검출과 포만트 주파수 검출에 직접 이용된다 하지만 두 성분을 분리하는 방법에 따라 피치검출이나 포만트 주파수 검출에 영향을 미칠 수 있으므로 기존의 방법보다 두 성분을 더 잘 분리할 수 있는 방법이 필요한 것이다. 본 논문에서는 스펙트럼 신호를 최대한 평탄화시킴으로써 포만트의 영향을 제거하고 고조파 성분을 분리해 내어 이를 피치검출에 사용한다. LSP를 이용하여 적응적 밴드에서 평탄화를 시도하고 이를 피치 검출에 이용하였다.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.317-331
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• 2012

In this paper, we develop a multichannel blind source separation algorithm based on a beamspace transform and the multichannel non-negative matrix factorization (NMF) method. The NMF algorithm is a famous algorithm which is used to solve the source separation problems. In this paper, we consider a beamspace-time-frequency domain data model for multichannel NMF method, and enhance the conventional method using a beamspace transform. Our decomposition algorithm is applied to audio source separation, using a dataset from the international Signal Separation Evaluation Campaign 2010 (SiSEC 2010) for evaluation.

• The Journal of the Korea institute of electronic communication sciences
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• v.17 no.3
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• pp.409-416
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• 2022

This paper relates to the blind noise separation method of time-delayed convolutive mixed signals. Since the mixed model of acoustic signals in a closed space is multi-channel, a convolutive blind signal separation method is applied and time-delayed data samples of the two microphone input signals is used. For signal separation, the mixing coefficient is calculated using an inverse model rather than directly calculating the separation coefficient, and the coefficient update is performed by repeated calculations based on secondary statistical properties to estimate the speech signal. Many simulations were performed to verify the performance of the proposed blind signal separation. As a result of the simulation, noise separation using this method operates safely regardless of convolutive mixing, and PESQ is improved by 0.3 points compared to the general adaptive FIR filter structure.

• The Journal of the Acoustical Society of Korea
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• v.29 no.2
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• pp.141-147
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• 2010

In this paper, we propose a new speech separation algorithm to extract and enhance the target speech signals from mixed speech signals by utilizing both magnitude and phase information. Since the previous statistical modeling algorithms assume that the log power spectrum values of the mixed speech signals are independent in the temporal and frequency domain, discontinuities occur in the resultant separated speech signals. To reduce the discontinuities, we apply a smoothing filter in the time-frequency domain. To further improve speech separation performance, we propose a statistical model based on both magnitude and phase information of speech signals. Experimental results show that the proposed algorithm improve signal-to-interference ratio (SIR) by 1.5 dB compared with the previous magnitude-only algorithms.

• The Journal of the Acoustical Society of Korea
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• v.38 no.2
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• pp.177-183
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• 2019

In this paper, we proposed an application method of ICA (Independent Component Analysis) to passive line array sonar to separate interferences from target signals in low frequency band and compared performance of three conventional ICA algorithms. Since the low frequency signals are received through larger bearing angles than other frequency bands, neighboring beam signals can be used to perform ICA as measurement signals of the ICA. We use three ICA algorithms such as Fast ICA, NNMF (Non-negative Matrix Factorization) and JADE (Joint Approximation Diagonalization of Eigen-matrices). Through experiments on real data obtained from passive line array sonar, it is verified that the interference can be separable from target signals by the suggested method and the JADE algorithm shows the best separation performance among the three algorithms.