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Spherical microphone array processing for acoustic parameter estimation and signal enhancement

机译:用于声学参数估计和信号增强的球形麦克风阵列处理

摘要

In many distant speech acquisition scenarios, such as hands-free telephony or teleconferencing, the desired speech signal is corrupted by noise and reverberation. This degrades both the speech quality and intelligibility, making communication difficult or even impossible. Speech enhancement techniques seek to mitigate these effects and extract the desired speech signal.ududThis objective is commonly achieved through the use of microphone arrays, which take advantage of the spatial properties of the sound field in order to reduce noise and reverberation. Spherical microphone arrays, where the microphones are arranged in a spherical configuration, usually mounted on a rigid baffle, are able to analyze the sound field in three dimensions; the captured sound field can then be efficiently described in the spherical harmonic domain (SHD).ududIn this thesis, a number of novel spherical array processing algorithms are proposed, based in the SHD. In order to comprehensively evaluate these algorithms under a variety of conditions, a method is developed for simulating the acoustic impulse responses between a sound source and microphones positioned on a rigid spherical array placed in a reverberant environment.ud udThe performance of speech enhancement algorithms can often be improved by taking advantage of additional a priori information, obtained by estimating various acoustic parameters. Methods for estimating two such parameters, the direction of arrival (DOA) of a source (static or moving) and the signal-to-diffuse energy ratio, are introduced.ududFinally, the signals received by a microphone array can be filtered and summed by a beamformer. A tradeoff beamformer is proposed, which achieves a balance between speech distortion and noise reduction. The beamformer weights depend on the noise statistics, which cannot be directly observed and must be estimated. An estimation algorithm is developed for this purpose, exploiting the DOA estimates previously obtained to differentiate between desired and interfering coherent sources.
机译:在许多遥远的语音采集方案中,例如免提电话或电话会议,所需的语音信号会被噪声和混响破坏。这降低了语音质量和清晰度,使通信变得困难甚至无法进行。语音增强技术试图减轻这些影响并提取所需的语音信号。 ud ud此目标通常是通过使用麦克风阵列来实现的,该麦克风阵列利用声场的空间特性来减少噪声和混响。球形麦克风阵列(通常将麦克风以球形配置排列,通常安装在刚性挡板上)能够分析三维声场。因此,可以在球谐域(SHD)中有效地描述捕获的声场。 ud ud本文基于SHD提出了许多新颖的球面阵列处理算法。为了在各种条件下全面评估这些算法,开发了一种方法,用于模拟声源与位于混响环境中的刚性球形阵列上的麦克风之间的声学​​脉冲响应。 ud ud语音增强算法的性能通常可以通过利用通过估计各种声学参数而获得的附加先验信息来改进“噪声”。介绍了用于估计两个这样的参数的方法,即源的到达方向(DOA)(静态或移动)和信噪比。 ud ud最后,可以对麦克风阵列接收的信号进行滤波并由波束形成器求和。提出了一种折衷的波束形成器,它可以在语音失真和降噪之间取得平衡。波束成形器的权重取决于噪声统计数据,该噪声统计数据无法直接观察,必须进行估算。为此目的,开发了一种估计算法,它利用先前获得的DOA估计来区分所需和干扰相干源。

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    Jarrett Daniel;

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  • 年度 2014
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