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Digital signal processing for multichannel audio equalization and signal cancellation.

机译:用于多声道音频均衡和信号消除的数字信号处理。

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摘要

This thesis presents new algorithms in important areas of acoustics: (i) multiple listener equalization, and (ii) selective audio signal cancellation.; Equalization of acoustical responses at multiple locations is important for delivery of high quality audio to listeners in an enclosed environment (e.g., theaters, automobiles, etc.). An equalization filter designed to compensate for the room effects at a single location performs poorly at other locations in a room since room responses vary significantly with changing source-receiver positions. A good equalization filter should minimize the effects of the room at multiple listener locations in the room.; The thesis proposes a multiple listener equalization filter using the fuzzy c-means clustering technique. Since the length of the measured room response can be quite large (depending on the room reverberation time), the thesis further shows that, by applying psycho-acoustic techniques, the filter orders can be reduced significantly, for practical Digital Signal Processing (DSP) implementations, without significantly affecting the equalization performance at the multiple listener positions. Finally, it is shown that the proposed filter outperforms existing standard filters in terms of objective measures of equalization performance and subjective listening tests.; Selectively canceling signals at specific locations within an acoustical environment with multiple listeners is of significant importance for home theater, automobile and other applications. The traditional sound absorption/noise cancellation approaches are impractical for such applications because they require either, bulky or uncomfortable ear-defenders to be inserted into the ear canal for source signal absorption, or multiple secondary sources to anti-phase the primary audio source for relatively low frequency acoustic monotone signals, or high volume passive sound absorption materials for relatively high frequencies in the audio band.; In this thesis, an alternative method is presented for signal cancellation, by pre-processing the acoustical signal with a digital filter known as the eigenfilter. The eigenfilter uses information based on the positional variations in the room impulse responses so as to achieve audio signal cancellation. Also examined are the theoretical properties of an eigenfilter, and the performance tradeoffs (e.g., spectral distortion and variation of room impulse responses).; Conclusions and future directions are provided appropriately at the end of relevant chapters.
机译:本文提出了在声学的重要领域中的新算法:(i)多听众均衡,和(ii)选择性音频信号消除。对于在封闭环境(例如剧院,汽车等)中向听众传递高质量音频,在多个位置均衡声音响应非常重要。设计用于补偿单个位置的房间效果的均衡滤波器在房间的其他位置表现不佳,因为房间响应随源接收器位置的变化而显着变化。一个好的均衡滤波器应使房间中多个收听者位置的房间影响最小化。本文提出了一种基于模糊c均值聚类技术的多听者均衡滤波器。由于所测量的房间响应的长度可能很大(取决于房间的混响时间),因此本文进一步表明,对于实用的数字信号处理(DSP),通过应用心理声学技术,可以显着降低滤波器阶数实现,而不会显着影响多个侦听器位置的均衡性能。最后,表明在均衡性能和主观听觉测试的客观指标方面,所提出的滤波器优于现有的标准滤波器。在具有多个收听者的声学环境中的特定位置处选择性地消除信号对于家庭影院,汽车和其他应用非常重要。对于这些应用,传统的消音/消噪方法是不切实际的,因为它们需要将笨重或不舒服的护耳器插入耳道中以吸收源信号,或者需要多个次要源来使主音频源反相以相对低频声学单调信号,或用于音频频带中相对较高频率的高音量无源吸声材料;在本文中,提出了一种通过使用称为特征滤波器的数字滤波器对声学信号进行预处理来消除信号的另一种方法。本征滤波器使用基于房间脉冲响应中的位置变化的信息,从而实现音频信号消除。还研究了本征滤波器的理论特性和性能折衷(例如,频谱失真和房间脉冲响应的变化)。在相关章节的结尾适当地提供了结论和未来的方向。

著录项

  • 作者

    Bharitkar, Sunil.;

  • 作者单位

    University of Southern California.;

  • 授予单位 University of Southern California.;
  • 学科 Engineering Electronics and Electrical.
  • 学位 Ph.D.
  • 年度 2004
  • 页码 116 p.
  • 总页数 116
  • 原文格式 PDF
  • 正文语种 eng
  • 中图分类 无线电电子学、电信技术;
  • 关键词

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