Speech enhancement algorithms for multimedia applications

语音增强在电话会议系统、语音识别器、人机界面、免提通信、移动电话和助听器等许多应用中扮演着重要的角色。事实上，噪音在我们周围和任何地方都存在。因此，当感兴趣的信号(通常是语音)被麦克风拾取时，它总是被噪声、混响和其他不想要的信号所污染。然后，目标是使用数字信号处理工具来清除噪声信号，而不会对所需的语音信号造成太大的损害。 语音增强问题是一个古老的问题，已经存在了40多年。这并不令人惊讶，因为与一些人可能认为的相反，这是一个非常困难的问题，主要有两个原因。首先，噪声信号的性质和特征会在时间和应用上发生巨大变化。因此，要找到在不同的实际环境中真正有效的通用算法是很费力的。其次，还可以为每个应用程序定义不同的性能衡量标准。两个感知标准被广泛用于衡量性能：质量和可理解性。前者是主观的(它反映了听者的个人偏好)，后者是客观的(它给出了听者能够正确识别的单词的百分比)。要同时满足这两方面的要求是非常困难的。 因此，本研究方案旨在开发具有一个麦克风，特别是多个麦克风(或麦克风阵列)的下一代语音增强算法，其明确的目标是在期望的语音信号几乎(或没有)失真的情况下，尽可能地减少噪声和其他干扰信号。换句话说，我们将在不影响清晰度的情况下，尽可能地提高质量。我们还将努力开发实用的算法，这些算法在现实世界中和大多数应用程序中都可以方便地使用和调整。为了实现这一目标，我们将依靠我们最近开发的一些强有力的想法和成果。

Speech enhancement plays a fundamental role in many applications such as teleconferencing systems, speech recognizers, human-machine interfaces, hands-free communications, mobile phones, and hearing aids. Indeed, the noise is around us and everywhere. So when a signal of interest (usually speech) is picked up by a microphone, it is always contaminated by noise, reverberation, and other undesired signals. The objective is then to clean up the noisy signal with digital signal processing tools without damaging much the desired speech signal. The problem of speech enhancement is an old one and has been around for more than 40 years. This should not come as a surprise since, contrary to what some people may believe, it is a very difficult problem for mainly two reasons. First, the nature and characteristics of the noise signals can change dramatically in time and application to application. It is therefore laborious to find versatile algorithms that really work in different practical environments. Second, the performance measure can also be defined differently for each application. Two perceptual criteria are widely used to measure the performance: quality and intelligibility. While the former is subjective (it reflects individual preferences of listeners), the latter is objective (it gives the percentage of words that could be correctly identified by listeners). It is very hard to satisfy both at the same time. So this research proposal aims at developing the next-generation speech enhancement algorithms with one and, especially, multiple microphones (or microphone arrays) with the clear objective to reduce the noise and other interference signals as much as possible with little (or no) distortion of the desired speech signal. In other words, we will try to improve the quality as much as possible without affecting the intelligibility. We will also try to develop practical algorithms that can be easy to use and easy to tune in the real world and for most applications. To achieve this goal, we will rely on some strong ideas and results we have developed recently.