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A new two-microphone Gauss-Seidel pseudo affine projection algorithm for speech quality enhancement

机译:一种新的两麦克风高斯-赛德尔伪仿射投影算法,用于增强语音质量

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摘要

This study addresses the problem of speech quality enhancement by adaptive and nonadaptive filtering algorithms. The well-known two-microphone forward blind source separation (TM-FBSS) structure has been largely studied in the literature. Several two-microphone algorithms combined with TM-FBSS have been recently proposed. In this study, we propose 2 contributions: In the first, a new two-microphone Gauss-Seidel pseudo affine projection (TM-GSPAP) algorithm is combined with TM-FBSS. In the second, we propose to use the new TM-GSPAP algorithm in speech enhancement. Furthermore, we show the efficiency of the proposed TM-GSPAP algorithm in speech enhancement when highly noisy observations are available. To validate the good performances of our algorithm, we have evaluated the adaptive filtering properties in computational complexity and convergence speed performance by system mismatch criteria. A fair comparison with adaptive and non adaptive noise reduction algorithms are also presented. The adaptive algorithms are the well-known two-microphone normalized least mean square algorithm, and the recently published two-microphone pseudo affine projection algorithm. The non adaptive algorithms are the one-microphone spectral subtraction and the two-microphone Wiener filter algorithm. We evaluate the quality of the output speech signal in each algorithm by several objective and subjective criteria as the segmental signal-to-noise ratio, cepstral distance, perceptual evaluation of speech quality, and the mean opinion score. Finally, we validate the superior performances of the proposed algorithm with physically measured signals.
机译:这项研究解决了通过自适应和非自适应滤波算法提高语音质量的问题。在文献中已经对众所周知的两麦克风前向盲源分离(TM-FBSS)结构进行了大量研究。最近已经提出了几种结合TM-FBSS的两麦克风算法。在这项研究中,我们提出了两个建议:首先,将新的两麦克风高斯-赛德尔伪仿射投影(TM-GSPAP)算法与TM-FBSS相结合。第二,我们建议在语音增强中使用新的TM-GSPAP算法。此外,当高噪声观测可用时,我们展示了提出的TM-GSPAP算法在语音增强中的效率。为了验证我们算法的良好性能,我们通过系统失配准则评估了自适应滤波属性在计算复杂度和收敛速度方面的性能。还提出了与自适应和非自适应降噪算法的合理比较。自适应算法是众所周知的两麦克风归一化最小均方算法,以及最近发布的两麦克风伪仿射投影算法。非自适应算法是一麦克风频谱减法和两麦克风维纳滤波器算法。我们通过几种客观和主观标准来评估每种算法中输出语音信号的质量,这些标准包括分段信噪比,倒谱距离,语音质量的感知评估以及平均意见得分。最后,我们通过物理测量的信号验证了所提算法的优越性能。

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