语音去混响技术：理论与应用

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"《Speech Dereverberation》是由Patrick A. Naylor和Nikolay D. Gaubitch合著的一本专著，主要探讨了语音去混响的各种技术和方法，特别适合于学生、研究人员以及产品开发的专业人士。这本书详细介绍了在会议场景和临境通信中解决语音混响问题的关键技术，以提升语音质量和可理解性。" 在书中，作者首先对语音去混响这一主题进行了全面概述，阐述了混响如何影响语音质量和理解度，并提出了相应的评估指标。接着，他们深入探讨了混响模型，这是理解和处理混响问题的基础。在后续章节，读者可以了解到多种不同的去混响算法： 1. 基于统计模型的语音去混响算法 - 这种方法通常利用统计特性来估计并去除混响，通过建立声源与接收信号之间的数学模型，以减少混响的影响。 2. 基于LPC（线性预测编码）模型的语音去混响算法 - LPC是一种常见的信号建模技术，通过分析语音信号的线性预测系数来去除混响，提高语音清晰度。 3. 基于多麦克风特征值分解的语音去混响算法 - 利用多个麦克风捕捉的信号，通过特征值分解来分离直达声和混响信号，从而实现去混响。 4. 自适应盲多通道系统辨识 - 自适应算法能够根据环境变化自动调整，以识别和消除混响。 5. 多通道声学系统的子空间逆矩阵 - 这种方法涉及利用多通道信号处理技术，通过计算子空间逆矩阵来提取无混响的语音信号。 6. 移动目标语音的贝叶斯单通道盲去混响 - 在动态环境中，针对移动说话者的语音去混响，贝叶斯方法可以提供更有效的解决方案。 7. 不使用房间声学信息的语音去混响逆滤波 - 这种方法不依赖于房间的物理特性，而是通过逆滤波技术来消除混响。 8. 用于语音和音频信号去混响的TRINICON - TRINICON是特制的去混响技术，特别适用于语音和音频信号的处理，以提高整体音质。这本书的每一章都深入浅出地解释了相关理论和技术，同时提供了丰富的实证研究和实例分析，对于理解并实施语音去混响技术具有很高的参考价值。此外，它还遵循了版权法规，确保合法复制和使用。通过学习本书，读者将能掌握处理语音混响问题的关键工具和策略，进而改进语音通信系统的性能。

xviii List of Contributors

Walter Kellermann

University of Erlangen-Nuremberg, Germany

e-mail: wk@lnt.de

Andy Khong

Nanyang Technological University, Singapore

e-mail: AndyKhong@ntu.edu.sg

Keisuke Kinoshita

NTT Communication Science Laboratories, Japan

e-mail: kinoshita@cslab.kecl.ntt.co.jp

Masato Miyoshi

NTT Communication Science Laboratories, Japan

e-mail: miyo@cslab.kecl.ntt.co.jp

Tomohiro Nakatani

NTT Communication Science Laboratories, Japan

e-mail: nak@cslab.kecl.ntt.co.jp

Patrick A. Naylor

Imperial College London, UK

e-mail: p.naylor@imperial.ac.uk

Mark R.P. Thomas

Imperial College London, UK

e-mail: mark.r.thomas@imperial.ac.uk

Jimi Y.-C. Wen

Imperial College London, UK

e-mail: jimi.wen@imperial.ac.uk

Takuya Yoshioka

NTT Communication Science Laboratories, Japan

e-mail: takuya@cslab.kecl.ntt.co.jp

Chapter 1

Introduction

Patrick A. Naylor and Nikolay D. Gaubitch

Abstract Acoustic reve rberation will be intro duced in this chap ter in the context

of telecommunication. The adverse effects on speech caused by reverberation are

problematic, in particular, in hands-free terminals operating typically at arms-length

from the talker’s lips. This introductory chapter will provide a system description of

room reverberation and will formulate mathematically the dereverberation pr oblem

in its most direct form so as to introduce and underpin the more detailed presen-

tation in subsequent chapters. Elements of room acoustics will also be introduced

where needed, though detailed study of acoustics is not the aim of this text.

At the time of writing this, dereverberation is a topic of study with many impor-

tant research questions remaining as yet unanswered. Whilst reviewing the relevant

literature later in this cha pter, it is intended both to describe the state-of-the-art and

to highlight some of the signiﬁcant open issues. Whereas the former aims to con-

solidate, perhaps for the ﬁrst time, the known achievements to date o f the research

community, the latter aims to highlight potential avenues of future research.

1.1 Background

One can conﬁdently deduce that the phenomenon of reverberation has been known

to mankind since the time of prehistoric cave dwellers. Sound reﬂection effects are

believed to have inﬂuenced prehisto ric cave art [90, 91]. Reverberation is also used

in several well known cases by other species, such as bats for navigation during

ﬂight.

There is evidence of comprehension of the notion of reﬂected speech occurring

in Plato’s Republic [76]: “And what if sound echoed off the prison wall opposite

them? When any of the passers-by spoke, don’t you think they’d be bound to assume

that the sound came from a passing shadow?”. Pioneering scientiﬁc work on sound

and acoustics in the 19th century was undertaken by, for example, Rayleigh [80] and

Imperial College London, UK

2 P.A. Naylor and N.D. Gaubitch

Sabine [81]. In the 20th century initial efforts in the understanding of reverberation

of speech were provided by Bolt [8] and the effects of single echoes by Haas [37].

The level to which humans employ reverberation during everyday life is unclear.

There is some evidence to suggest that, through the use of two ears, spatial process-

ing is used to enhan c e speech intelligibility and enab les a useful degree of source

separation to be achieved in human speech perception [12].

In music audio processing, the sense of ‘space’ that can be created by stereo or

surround sound reproduction adds greatly to realism and often makes recorded mu-

sic more attractive and enjoyable. We might then ask ourselves the question: since

reverbera tion is present in everyday life experience of sound and in some important

cases is effective in aiding speech communication, why should we be interested in

removing reverberation from speech using dereverberation processing? The answer

to this question is dependent on the application context.

There is a continuously growing demand for high quality hands-free speech input

for various telecommunication applications [71, 82]. One driving force behind this

development is the rapidly increasing use of portable devices such as mobile tele-

phones, Personal Digital Assistant (PDA) devices and laptop computers equipped

for Voice Over Internet Protocol (VoIP) [63]. Furthermore, there is a continuous

worldwide expansion of broadband internet access [5]. These factors have paved

the way for several advanced speech applications such as wideband teleconferenc-

ing with automatic camera steering, automatic speech-to-text conversion, speaker

identiﬁcation, voice-controlled device operation and car interior commu nication

systems [82]. Another important application where speech obtained from a distant

talker is of interest is that of hearing aids [82].

1.2 Effects of Reverberation

When speech signals are obtained in an enclosed space by one or more microphones

positioned at a distance from the talker, the observed signal consists of a superpo-

sition of many delayed and attenuated copies of th e speech sig nal due to multiple

reﬂections from the surrounding walls and other objects, as illustrated in Fig. 1.1.

We here deﬁne the direct-path as the acoustic propagation path from the talker to

the microphone without reﬂections. We also note that a delay of the superimposed

copies arises because all other propagation paths are longer than the direct-path and

that additional attenuation occurs at each reﬂection due to frequency dependent ab-

sorption. The perceptual effects of reverberation can be summarized as:

1. The box effect – the reverberated speech signal can be viewed as the same source

signal coming from several different sources positioned at different locations

in the r oom and thus arriving at different times and with different intensities

[3]. This adds spaciousness to the sound [56] and makes the talker sound as if

positioned “inside a box”.

2. The distant talker effect – the perceived spaciousness explained in the previous

point makes the talker sound far away from the microphone.

4 P.A. Naylor and N.D. Gaubitch

In hands-free speech acquisition, the talker’s lips are typically located at a dis-

tance of 0.3-3 m from the microphone. In such a scenario, the speech signal is

affected by the u ser’s surrounding environment, which results in the following three

distinct effects [71, 82]:

(i) Additive measurement noise due to, for example, other audible talkers or pass-

ing trafﬁc. When the noise level is comparable to or greater than the speech

level, it is difﬁcult for a listener to distinguish the desired speech signal from

the noise, and thus intelligibility and listening comfort are reduced.

(ii) Acoustic echoes due to speech from a far-end talker which is picked up by the

near-end microphones and retransmitted back to the far-end talker with delay.

This re sults in the talker hearing an echo of their own voice, which greatly

disturbs the communication.

(iii) Reverberation that arises whenever sound is produced in enclosed spaces, such

as ofﬁces and other rooms, due to reﬂections from walls and surrounding ob-

jects.

These components jointly contribute to an overall degradation in the quality of

the observed speech signals, which signiﬁcantly reduce the perceived speech qual-

ity for the listener and the performance of applications such as speech recogniz-

ers [71]. Speech enhancement and acoustic echo cancellation are two widely re-

searched ﬁelds that address problems (i) and (ii) respectively. Several signiﬁcant

contributions have been made in these areas [6, 7, 11, 13, 30, 41, 60] and many

algorithms have been implemented and are in use in commercial applications [77].

The problem of reverberation on the other hand, received much less attention in the

literature until recently. Nevertheless, ﬁnding solutions to this problem is essential

for the future development of applications with hands-free speech acquisition. This

indeed motivates the focus of the forthcoming chapters of this book.

1.4 System Description

A generic system diagram for multichan nel dereverberation is shown in Fig. 1.2.

The speech signal, s(n), from the talker propagates through acoustic channels, H

(z)

for m = 1toM. The output of each channel is observed using M microphones to

give signals x

(n). All noise in the system is assumed additive and is represented

(n).

The observed signal, x

(n), at microphone m can be described as the superposi-

tion of (i) the direct-path signal, which propagates by line-of-sight from the talker

to the microphone with corresponding attenuation and propagation delay and (ii) a

theoretically inﬁnite set of reﬂections of the talker signal arriving at the microphone

at later time instances [56] with attenuation dependent on the properties of the re-

ﬂecting surfaces. This can be expressed as

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