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Phase-Based Dual-Microphone Speech Enhancement Using A Prior Speech Model

机译:使用先验语音模型的基于相位的双麦克风语音增强

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This paper proposes a phase-based dual-microphone speech enhancement technique that utilizes a prior speech model. Recently, it has been shown that phase-based dual-microphone filters can result in significant noise reduction in low signal-to-noise ratio [(SNR) less than 10 dB] conditions and negligible distortion at high SNRs (greater than 10 dB), as long as a correct filter parameter is chosen at each SNR. While prior work utilizes a constant parameter for all SNRs, we present an SNR-adaptive filter parameter estimation algorithm that maximizes the likelihood of the enhanced speech features based on a prior speech model. Experimental results using the CARVUI database show significant speech recognition accuracy rate improvement over alternative techniques in low SNR situations (e.g., an improvement of 11% in word error rate (WER) over postfiltering and 23% over delay-and-sum beamforming at 0 dB) and negligible distortion at high SNRs. The proposed adaptive approach also significantly outperforms the original phase-based filter with a constant parameter. Furthermore, it improves the filter's robustness when there are errors in time delay estimation
机译:本文提出了一种利用现有语音模型的基于相位的双麦克风语音增强技术。最近,研究表明,基于相位的双麦克风滤波器可以在低信噪比[(SNR)小于10 dB]条件下显着降低噪声,而在高SNR(大于10 dB)时失真可以忽略不计。 ,只要在每个SNR选择正确的滤波器参数即可。尽管先前的工作对所有SNR使用了恒定的参数,但我们提出了一种SNR自适应滤波器参数估计算法,该算法可根据先前的语音模型最大化增强语音特征的可能性。使用CARVUI数据库的实验结果表明,与低信噪比情况下的替代技术相比,语音识别准确率有了显着提高(例如,在0 dB的情况下,误码率(WER)较后滤波提高了11%,而延迟和总和波束形成比0dB时提高了23% )和高SNR时失真可忽略不计。所提出的自适应方法还明显优于具有恒定参数的原始基于相位的滤波器。此外,当延时估计存在误差时,它可以提高滤波器的鲁棒性

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