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Microphone array signal processing for advancements in robust speech systems.

机译:麦克风阵列信号处理可增强健壮语音系统的性能。

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摘要

Speech system performance degrades significantly in distant-talking environments, where the speech signals can be severely distorted by additive noise and reverberation. Microphone array processing techniques have presented a potential alternative to close-talking microphones by providing speech enhancement through spatial filtering and directional discrimination. Different from conventional array optimization criteria, such as Minimal Variance Distortionless Reponse, Maximal Signal-to-Noise Ratio or Minimal Mean Squared Error, this thesis presents a series of task-oriented and environment-oriented microphone array solutions for real world speech system applications. Primarily, four important tasks (e.g., blind beamforming, automatic speech recognition (ASR), speech quality enhancement and integrated voice activity detection (VAD) with speech enhancement) are considered in two typical acoustic environments (e.g., in-vehicle and conference room). Our objective is to optimize the microphone array front-end at a system level, directly advancing the performance of a given task for the whole speech system. Specifically, several new algorithms and systems are proposed in this thesis: variance of spectra flux based blind beamforming to identify target speech source in in-vehicle and conference room environments, order statistic filter based squared spectra enhancement for ASR in in-vehicle environment, integrated VAD and speech quality enhancement system in in-vehicle environment, fast relative transfer function identification for speech quality enhancement and ASR in conference room, position dependent spectra conversion for speech quality enhancement and ASR in in-vehicle and conference room, discriminative training based VAD for in-vehicle environment, and an efficient real-time microphone array based speech acquisition platform. Primary theoretical analysis and promising real/simulation evaluations on the proposed algorithms are also presented in this thesis.
机译:在远距离通话环境中,语音信号的性能会大大降低,在这种情况下,语音信号会因加性噪声和混响而严重失真。麦克风阵列处理技术通过通过空间滤波和方向辨别提供语音增强功能,为近距离通话麦克风提供了一种潜在的替代方法。与常规阵列优化标准(如最小方差无失真响应,最大信噪比或最小均方误差)不同,本文针对现实世界的语音系统应用提出了一系列面向任务和面向环境的麦克风阵列解决方案。首先,在两种典型的声学环境(例如车内和会议室)中考虑了四个重要任务(例如,盲束成形,自动语音识别(ASR),语音质量增强和具有语音增强功能的集成语音活动检测(VAD)) 。我们的目标是在系统级别优化麦克风阵列前端,直接提高整个语音系统的特定任务性能。具体而言,本文提出了几种新的算法和系统:基于频谱通量的方差波束形成以识别车内和会议室环境中的目标语音源;基于阶跃统计滤波器的车内环境ASR平方频谱增强,集成车载环境中的VAD和语音质量增强系统,会议室中语音质量增强和ASR的快速相对传递函数识别,车载和会议室中语音质量增强和ASR的位置相关频谱转换,基于判别训练的VAD用于车载环境,以及基于高效实时麦克风阵列的语音采集平台。本文还对所提出的算法进行了初步的理论分析和有希望的真实/仿真评估。

著录项

  • 作者

    Yu, Tao.;

  • 作者单位

    The University of Texas at Dallas.;

  • 授予单位 The University of Texas at Dallas.;
  • 学科 Speech Communication.;Engineering Electronics and Electrical.
  • 学位 Ph.D.
  • 年度 2011
  • 页码 149 p.
  • 总页数 149
  • 原文格式 PDF
  • 正文语种 eng
  • 中图分类 康复医学;
  • 关键词

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