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Efficient Blind Dereverberation and Echo Cancellation Based on Independent Component Analysis for Actual Acoustic Signals

机译:基于独立分量分析的实际声信号有效盲去混响和回声消除

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摘要

This letter presents a new algorithm for blind dereverberation and echo cancellation based on independent component analysis (ICA) for actual acoustic signals. We focus on frequency domain ICA (FD-ICA) because its computational cost and speed of learning convergence are sufficiently reasonable for practical applications such as hands-free speech recognition. In applying conventional FD-ICA as a preprocessing of automatic speech recognition in noisy environments, one of the most critical problems is how to cope with reverberations. To extract a clean signal from the reverberant observation, we model the separation process in the short-time Fourier transform domain and apply the multiple input/output inverse-filtering theorem (MINT) to the FD-ICA separation model. A naive implementation of this method is computationally expensive, because its time complexity is the second order of reverberation time. Therefore, the main issue in dereverberation is to reduce the high computational cost of ICA. In this letter, we reduce the computational complexity to the linear order of the reverberation time by using two techniques: (1) a separation model based on the independence of delayed observed signals with MINT and (2) spatial sphering for preprocessing. Experiments show that the computational cost grows in proportion to the linear order of the reverberation time and that our method improves the word correctness of automatic speech recognition by 10 to 20 points in a RT_(20) = 670 ms reverberant environment.
机译:这封信提出了一种基于独立分量分析(ICA)的用于实际声音信号的盲去混响和回声消除的新算法。我们专注于频域ICA(FD-ICA),因为它的计算成本和学习收敛速度对于免提语音识别等实际应用而言是足够合理的。在嘈杂环境中将常规FD-ICA用作自动语音识别的预处理过程中,最关键的问题之一是如何应对混响。为了从混响观测中提取出清晰的信号,我们在短时傅立叶变换域中对分离过程进行建模,并将多重输入/输出逆滤波定理(MINT)应用于FD-ICA分离模型。由于该方法的时间复杂度是混响时间的第二阶,因此该方法的简单实现在计算上非常昂贵。因此,去混响的主要问题是减少ICA的高计算成本。在这封信中,我们通过使用两种技术将计算复杂度降低到混响时间的线性顺序:(1)基于具有MINT的延迟观测信号的独立性的分离模型,以及(2)用于预处理的空间球化。实验表明,计算成本与混响时间的线性顺序成正比,并且我们的方法在RT_(20)= 670 ms混响环境中将自动语音识别的单词正确性提高了10到20个点。

著录项

  • 来源
    《Neural computation》 |2012年第1期|p.234-272|共39页
  • 作者单位

    Department of Intelligence Science and Technology, Graduate School of Informatics, Kyoto University, Kyoto 606-8501, Japan;

    Honda Research Institute Japan Co., Wako, Saitama 351-0188, Japan;

    Department of Intelligence Science and Technology, Graduate School of Informatics, Kyoto University, Kyoto 606-8501, Japan;

    Department of Electrical Engineering and Computer Science, Graduate School of Engineering, Nagoya University, Nagoya 464-8603, Japan;

    Department of Intelligence Science and Technology, Graduate School of Informatics, Kyoto University, Kyoto 606-8501, Japan;

    Department of Intelligence Science and Technology, Graduate School of Informatics, Kyoto University, Kyoto 606-8501, Japan;

  • 收录信息 美国《科学引文索引》(SCI);美国《化学文摘》(CA);
  • 原文格式 PDF
  • 正文语种 eng
  • 中图分类
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