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An End-to-End Deep Learning Approach to Simultaneous Speech Dereverberation and Acoustic Modeling for Robust Speech Recognition

机译:端到端深度学习方法可同时进行语音去混响和声学建模,以实现可靠的语音识别

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We propose an integrated end-to-end automatic speech recognition (ASR) paradigm by joint learning of the front-end speech signal processing and back-end acoustic modeling. We believe that “only good signal processing can lead to top ASR performance” in challenging acoustic environments. This notion leads to a unified deep neural network (DNN) framework for distant speech processing that can achieve both high-quality enhanced speech and high-accuracy ASR simultaneously. Our goal is accomplished by two techniques, namely: (i) a reverberation-time-aware DNN based speech dereverberation architecture that can handle a wide range of reverberation times to enhance speech quality of reverberant and noisy speech, followed by (ii) DNN-based multicondition training that takes both clean-condition and multicondition speech into consideration, leveraging upon an exploitation of the data acquired and processed with multichannel microphone arrays, to improve ASR performance. The final end-to-end system is established by a joint optimization of the speech enhancement and recognition DNNs. The recent REverberant Voice Enhancement and Recognition Benchmark (REVERB) Challenge task is used as a test bed for evaluating our proposed framework. We first report on superior objective measures in enhanced speech to those listed in the 2014 REVERB Challenge Workshop on the simulated data test set. Moreover, we obtain the best single-system word error rate (WER) of 13.28% on the 1-channel REVERB simulated data with the proposed DNN-based pre-processing algorithm and clean-condition training. Leveraging upon joint training with more discriminative ASR features and improved neural network based language models, a low single-system WER of 4.46% is attained. Next, a new multi-channel-condition joint learning and testing scheme delivers a state-of-the-art WER of 3.76% on the 8-channel simulated data with a single ASR system. Finally, we also report on a preliminary yet promising experimentation with the REVERB real test data.
机译:通过联合学习前端语音信号处理和后端声学建模,我们提出了一种集成的端到端自动语音识别(ASR)范例。我们相信,在充满挑战的声学环境中,“只有良好的信号处理才能带来出色的ASR性能”。这个想法导致了用于远程语音处理的统一的深度神经网络(DNN)框架,该框架可以同时实现高质量的增强语音和高精度ASR。我们的目标是通过两种技术实现的,即:(i)基于混响时间感知DNN的语音混响架构,可以处理多种混响时间以提高混响和嘈杂语音的语音质量,其次是(ii)DNN-基于多条件的训练,它同时考虑了干净条件和多条件语音,并利用了利用多通道麦克风阵列获取和处理的数据来提高ASR性能。最终的端到端系统是通过语音增强和识别DNN的联合优化而建立的。最近的混响语音增强和识别基准(REVERB)挑战任务被用作评估我们提出的框架的测试平台。我们首先在2014年REVERB挑战研讨会上通过模拟数据测试集报告了增强语音中的卓越客观指标。此外,通过提出的基于DNN的预处理算法和干净条件训练,我们在1通道REVERB模拟数据上获得了13.28%的最佳单系统字错误率(WER)。利用具有更多判别性ASR功能和改进的基于神经网络的语言模型的联合训练,可以实现4.46%的低单系统WER。接下来,新的多通道条件联合学习和测试方案通过单个ASR系统在8通道模拟数据上提供了3.76%的最新WER。最后,我们还报告了有关REVERB真实测试数据的初步但有希望的实验。

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