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Speech dereverberation for enhancement and recognition using dynamic features constrained deep neural networks and feature adaptation

机译:使用动态特征增强和识别的语音去混响约束深度神经网络和特征自适应

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This paper investigates deep neural networks (DNN) based on nonlinear feature mapping and statistical linear feature adaptation approaches for reducing reverberation in speech signals. In the nonlinear feature mapping approach, DNN is trained from parallel clean/distorted speech corpus to map reverberant and noisy speech coefficients (such as log magnitude spectrum) to the underlying clean speech coefficients. The constraint imposed by dynamic features (i.e., the time derivatives of the speech coefficients) are used to enhance the smoothness of predicted coefficient trajectories in two ways. One is to obtain the enhanced speech coefficients with a least square estimation from the coefficients and dynamic features predicted by DNN. The other is to incorporate the constraint of dynamic features directly into the DNN training process using a sequential cost function. In the linear feature adaptation approach, a sparse linear transform, called cross transform, is used to transform multiple frames of speech coefficients to a new feature space. The transform is estimated to maximize the likelihood of the transformed coefficients given a model of clean speech coefficients. Unlike the DNN approach, no parallel corpus is used and no assumption on distortion types is made. The two approaches are evaluated on the REVERB Challenge 2014 tasks. Both speech enhancement and automatic speech recognition (ASR) results show that the DNN-based mappings significantly reduce the reverberation in speech and improve both speech quality and ASR performance. For the speech enhancement task, the proposed dynamic feature constraint help to improve cepstral distance, frequency-weighted segmental signal-to-noise ratio (SNR), and log likelihood ratio metrics while moderately degrades the speech-to-reverberation modulation energy ratio. In addition, the cross transform feature adaptation improves the ASR performance significantly for clean-condition trained acoustic models.
机译:本文研究了基于非线性特征映射和统计线性特征自适应方法的深度神经网络(DNN),以减少语音信号的混响。在非线性特征映射方法中,从并行的干净/失真语音语料库训练DNN,以将混响和嘈杂的语音系数(例如对数幅度谱)映射到底层的干净语音系数。动态特征(即语音系数的时间导数)施加的约束以两种方式用于增强预测系数轨迹的平滑度。一种方法是根据DNN预测的系数和动态特征,以最小二乘估计获得增强的语音系数。另一种是使用顺序成本函数将动态特征的约束直接合并到DNN训练过程中。在线性特征自适应方法中,称为交叉变换的稀疏线性变换用于将语音系数的多个帧变换到新的特征空间。给定干净语音系数的模型,估计变换以最大化变换系数的可能性。与DNN方法不同,没有使用并行语料库,也没有对失真类型进行假设。在REVERB Challenge 2014任务中评估了这两种方法。语音增强和自动语音识别(ASR)结果均表明,基于DNN的映射显着减少了语音的混响,并提高了语音质量和ASR性能。对于语音增强任务,提出的动态特征约束有助于改善倒谱距离,频率加权分段信噪比(SNR)和对数似然比指标,同时适度降低语音与混响调制能量比。此外,交叉变换特征自适应可显着改善在干净条件下训练的声学模型的ASR性能。

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