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Traitement parametrique des signaux audio dans le contexte des protheses auditives.

机译:助听器中音频信号的参数处理。

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摘要

The digital hearing aids for the hearing-impaired constitute a particularly important application field of signal processing. The last generations of hearing aids are equipped with a wireless local area network which allows a sophisticated binaural processing of the exchanged information not only to improve its intelligibility and its quality perception but also to emphasize the associated source localization acoustic cues. This technological progress has made possible the use of conventional parametric processing techniques which are widely used for speech compression in telephony.;Within the context of the first approach, we study an adaptive filtering method for noise reduction, proposed previously in the literature, and we identify some inherent problems associated with its use. To fix these problems, we suggest modifying an existing noise power spectrum estimator and combining it with this filtering method by means of a soft-decision scheme. We show that this combination is advantageous in that it provides a good tradeoff between the amount of noise reduction and the speech distortion.;Regarding the second approach, we propose two versions of an iterative method for parametric processing based on an autoregressive linear model, a predefined decision criterion and the Minimum Statistics (MS) algorithm. We show that this method, which operates in the correlation domain, allows obtaining compensated and stable spectral parameters. We present the limits associated with the use of the MS algorithm in the correlation domain. Then, a method that allows improving the precision of the noise variance estimator in the case of an audio signal contaminated by an additive white noise was developed. In comparison with the method based on the MS algorithm, we show empirically that the new proposed method gives rise to more robust spectral parameters which suitably model the formantic structure of the signal.;The sensitivity problem of the conventional parametric processing techniques to background noise is well-known. Two approaches of parametric processing in the presence of noise are often used. The first approach consists of performing the reduction of noise, generally in the frequency domain, as a preprocessing for the parametric modeling of the audio signal. The second approach allows applying a noise reduction processing directly in the correlation domain. This processing allows estimating the spectral parameters which model suitably the formantic structure of the signal. We propose innovative and relevant algorithmic solutions to determine which of both approaches allows obtaining the best fidelity of the spectral envelope of an audio signal to its formantic structure.
机译:用于听力障碍的数字助听器构成信号处理的一个特别重要的应用领域。最后几代助听器配备了无线局域网,该无线局域网允许对交换的信息进行复杂的双耳处理,不仅可以提高其清晰度和质量感知能力,而且可以强调相关的声源定位声音提示。这一技术进步使人们可以使用广泛用于电话语音压缩的常规参数处理技术。在第一种方法的背景下,我们研究了一种自适应降噪方法,该方法先前已在文献中提出,并且确定与其使用相关的一些固有问题。为了解决这些问题,我们建议修改现有的噪声功率谱估计器,并通过软判决方案将其与该滤波方法结合。我们展示了这种组合的优势,因为它在降噪量和语音失真之间提供了良好的折衷。关于第二种方法,我们提出了基于自回归线性模型的两种迭代方法用于参数处理。预定义的决策标准和最小统计(MS)算法。我们表明,该方法在相关域中运行,可以获取补偿和稳定的光谱参数。我们提出了与在相关域中使用MS算法相关的限制。然后,开发了一种在音频信号被加性白噪声污染的情况下能够提高噪声方差估计器的精度的方法。与基于MS算法的方法相比,我们从经验上证明了该新方法产生了更健壮的频谱参数,可以对信号的强结构进行适当建模。;传统参数处理技术对背景噪声的敏感性问题是知名。经常使用两种在存在噪声的情况下进行参数处理的方法。第一种方法包括执行降噪,通常在频域中,作为音频信号参数化建模的预处理。第二种方法允许直接在相关域中应用降噪处理。该处理允许估计频谱参数,该频谱参数适当地对信号的弹性结构建模。我们提出了创新且相关的算法解决方案,以确定这两种方法中的哪一种都可以使音频信号的频谱包络获得最佳的保真度。

著录项

  • 作者

    Trabelsi, Abdelaziz.;

  • 作者单位

    Ecole Polytechnique, Montreal (Canada).;

  • 授予单位 Ecole Polytechnique, Montreal (Canada).;
  • 学科 Health Sciences Audiology.;Computer Science.;Engineering Electronics and Electrical.
  • 学位 Ph.D.
  • 年度 2009
  • 页码 111 p.
  • 总页数 111
  • 原文格式 PDF
  • 正文语种 eng
  • 中图分类
  • 关键词

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