Time domain blind source separation of non-stationary convolved signals by utilizing geometric beamforming

R. Aichner, S. Araki, S. Makino, T. Nishikawa, H. Saruwatari
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引用次数: 55

Abstract

We propose a time-domain blind source separation (BSS) algorithm that utilizes geometric information such as sensor positions and assumed locations of sources. The algorithm tackles the problem of convolved mixtures by explicitly exploiting the non-stationarity of the acoustic sources. The learning rule is based on second-order statistics and is derived by natural gradient minimization. The proposed initialization of the algorithm is based on the null beamforming principle. This method leads to improved separation performance, and the algorithm is able to estimate long unmixing FIR filters in the time domain due to the geometric initialization. We also propose a post-filtering method for dewhitening which is based on the scaling technique in frequency-domain BSS. The validity of the proposed method is shown by computer simulations. Our experimental results confirm that the algorithm is capable of separating real-world speech mixtures and can be applied to short learning data sets down to a few seconds. Our results also confirm that the proposed dewhitening post-filtering method maintains the spectral content of the original speech in the separated output.
基于几何波束形成的非平稳卷积信号时域盲源分离
我们提出了一种时域盲源分离(BSS)算法,该算法利用几何信息,如传感器位置和源的假设位置。该算法通过明确地利用声源的非平稳性来解决卷积混合问题。学习规则是基于二阶统计量,并由自然梯度最小化导出。该算法的初始化基于零波束形成原理。该方法提高了分离性能,并且由于几何初始化,该算法能够在时域估计长解混FIR滤波器。我们还提出了一种基于频域BSS标度技术的后滤波去白化方法。计算机仿真结果表明了该方法的有效性。我们的实验结果证实,该算法能够分离真实世界的语音混合,并且可以应用于短至几秒的学习数据集。我们的结果也证实了所提出的去白后滤波方法在分离后的输出中保持了原始语音的频谱内容。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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