语音与音频信号处理技术及应用详解

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"Audio and Speech Signal Processing: Technologies and Applications" 是一本关于音频与语音信号处理的入门经典著作，由Hector Perez-Meana撰写，他在墨西哥国家理工学院任职。该书由IDEA GROUP PUBLISHING出版，旨在深入浅出地介绍音频和语音信号处理的基础知识和技术应用。音频和语音信号处理是通信、多媒体、人工智能、语音识别和音频编码等多个领域的核心部分。本书涵盖了从信号采集到分析、处理和恢复的整个流程，旨在帮助读者理解这些技术的基本原理，并能应用到实际系统中。在内容上，可能包括以下几个关键知识点： 1. **信号模型**：介绍音频和语音信号的基本特征，如波形、频谱和时域表示，以及它们如何被数学模型化。 2. **信号获取**：探讨麦克风和其他音频输入设备的工作原理，以及如何将声波转化为电信号。 3. **预处理技术**：如噪声抑制、增益控制和滤波，这些都是在进一步分析之前改善信号质量的重要步骤。 4. **特征提取**：讲解如何从原始信号中提取有意义的特征，如MFCC（梅尔频率倒谱系数）用于语音识别和情感分析。 5. **信号处理算法**：涵盖傅立叶变换、拉普拉斯变换和小波变换等基础工具，以及它们在音频压缩和信号分析中的应用。 6. **语音识别**：介绍基于模板匹配、隐马尔科夫模型（HMM）等方法的语音识别系统。 7. **语音合成**：讲述如何通过文本到语音（TTS）技术将文本转换为自然的语音输出。 8. **音频编码**：讨论MP3、AAC等音频压缩标准，以及它们在存储和传输音频时的作用。 9. **应用实例**：可能包括电话系统、语音助手、听力辅助设备、音频编辑软件和多媒体播放器等领域的应用案例。 10. **最新发展**：可能会涉及现代音频和语音处理的前沿技术，如深度学习在语音识别和情感分析中的应用。这本书不仅适合初学者，也对有经验的工程师和研究人员有所启发，因为它提供了深入的理论解释和实践指导，有助于读者构建扎实的音频和语音信号处理基础。通过阅读本书，读者可以全面了解这一领域的基本概念和技术，并具备开发和优化相关应用的能力。

References

Aguilar, G., Nakano-Miyatake, M., & Perez-Meana, H. (2005). Alaryngeal speech enhance-

ment using pattern recognition techniques. IEICE Trans. Inf. & Syst., E88-D(7),

1618-1622.

Amano, F., Perez-Meana, H., De Luca, A., & Duchen, G. (1995). A multirate acoustic echo

canceler structure. IEEE Trans. on Communications, 43(7), 2173-2176.

Bender, W., Gruhl, D., Marimoto, N., & Lu. (1996).Techniques for data hiding. IBM Systems

Journal, 35, 313-336.

Bosi, M., & Goldberg, R. (2002). Introduction to digital audio coding and standards. Boston:

Kluwer Academic Publishers.

Childers, D. (2000). Speech processing and synthesis toolboxes. New York: John Wiley &

Sons.

Cox, I., Miller, M., & Bloom, J. (2001). Digital watermark: Principle and practice. New

York: Morgan Kaufmann.

Hattori, Y., Ishihara, K., Komatani, K., Ogata, T., & Okuno, H. (2004). Repeat recognition

for environmental sounds. In Proceedings of IEEE International Workshop on Robot

and Human Interaction (pp. 83-88).

Haykin, S. (1991). Adaptive lter theory. Englewood Cliffs, NJ: Prentice Hall.

Kondoz, A. M. (1994). Digital speech. Chinchester, England: Wiley & Sons.

Kuo, S., & Morgan, D. (1996). Active noise control system: Algorithms and DSP implemen-

tations. New York: John Wiley & Sons.

Lee, C., Soong, F., & Paliwal, K. (1996). Automatic speech and speaker recognition. Boston:

Kluwer Academic Publishers.

Madisetti, V., & Williams, D. (1998). The digital signal processing handbook. Boca Raton,

FL: CRC Press.

Messershmitt, D. (1984). Echo cancellation in speech and data transmission. IEEE Journal

of Selected Areas in Communications, 2(3), 283-297.

Perez-Meana, H., & Nakano-Miyatake, M. (2005). Speech and audio signal applications. In

Encyclopedia of information science and technology (pp. 2592-2596). Idea Group.

Proakis, J. (1985). Digital communications. New York: McGraw Hill.

Rabiner, L., & Biing-Hwang, J. (1993). Fundamentals of speech recognition. Englewood

Cliff, NJ: Prentice Hall.

Tapia-Sánchez, D., Bustamante, R., Pérez-Meana, H., & Nakano-Miyatake, M. (2005). Single

channel active noise canceller algorithm using discrete cosine transform. Journal of

Signal Processing, 9(2), 141-151.

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Acknowledgments

The editor would like to acknowledge the help of all involved in the collation and review

process of the book, without whose support the project could not have been satisfactorily

completed.

Deep appreciation and gratitude is due to the National Polytechnic Institute of Mexico, for

ongoing sponsorship in terms of generous allocation of online and off-line Internet, WWW,

hardware and software resources, and other editorial support services for coordination of

this yearlong project.

Most of the authors of chapters included in this also served as referees for articles written

by other authors. Thanks go to all those who provided constructive and comprehensive

reviews that contributed to improve the chapter contents. I also would like to thanks to Dr.

Tomohiko Taniguchi of Fujitsu Laboratories Ltd. of Japan, for taking some time of his very

busy schedule to write the foreword of this book.

Special thanks also go to all the staff at Idea Group Inc., whose contributions throughout the

whole process from inception of the initial idea to nal publication have been invaluable. In

particular, to Kristin Roth who continuously prodded via e-mail for keeping the project on

schedule and to Mehdi Khosrow-Pour, whose enthusiasm motivated me to initially accept

his invitation for taking on this project.

Special thanks go to my wife, Dr. Mariko Nakano-Miyatake, of the National Polytechnic

Institute of Mexico, who assisted me during the reviewing process, read a semi-nal draft

of the manuscript, and provided helpful suggestions for enhancing its content; also I would

like to thank her for her unfailing support and encouragement during the months it took to

give birth to this book.

In closing, I wish to thank all of the authors for their insights and excellent contributions to

this book. I also want to thank all of the people who assisted me in the reviewing process. Fi-

nally, I want to thank my daughter Anri for her love and support throughout this project.

Hector Perez-Meana, PhD

National Polytechnic Institute

Mexico City, Mexico

December 2006

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2 Perez-Meana & Nakano-Miyatake

of Idea Group Inc. is prohibited.

Introduction

The advances of the VLSI technology have allowed the development of high performance

digital signal processing (DSP) devices, enabling the implementation of very efcient and

sophisticated algorithms, which have been successfully used in the solution of a large amount

of practical problems in several elds of science and engineering. Thus, signal processing

techniques have been used with great success in telecommunications to solve the echo prob-

lem in telecommunications and teleconference systems (Amano, Perez-Meana, De Luca, &

Duchen, 1995), to solve the inter-symbol interference in high speed data communications

systems (Proakis, 1985), as well as to develop efcient coders that allow the storage and

transmission of speech and audio signals with a low bit rate keeping at the same time a high

sound quality (Bosi & Golberg, 2002; Kondoz, 1994). Signal processing algorithms have

also been used for speech and audio signal enhancement and restoration (Childers, 2000;

Davis, 2002) to reduce the noise produced by air conditioning equipment and motors (Kuo &

Morgan, 1996), and so forth, and to develop electronic mufers (Kuo & Morgan, 1996) and

headsets with active noise control (Davis, 2002). In the educational eld, signal processing

algorithms that allow the time scale modication of speech signals have been used to assist

the foreign language students during their learning process (Childers, 2000). These systems

have also been used to improve the hearing capability of elder people (Davis, 2002).

The digital technology allows an easy and error free reproduction of any digital material,

allowing the illegal reproduction of audio and video material. Because this fact represents

a huge economical loss for the entertainment industry, many efforts have been carried out

to solve this problem. Among the several possible solutions, the watermarking technology

appears to be a desirable alternative for copyright protection (Bassia, Pitas, & Nikoladis,

2001; Bender, Gruhl, Marimoto, & Lu, 1996). As a result, several audio and speech water-

marking algorithms have been proposed during the last decade, and this has been a subject

of active research during the last several years. Some of these applications are analyzed in

the remaining chapters of this book.

This chapter presents an overview of signal processing systems to storage, transmission,

enhancement, protection, and reproduction of speech and audio signals that have been suc-

cessfully used in telecommunications, audio, access control, and so forth.

Adaptive.Echo.Cancellation

A very successful speech signal processing application is the adaptive echo cancellation

used to reduce a common but undesirable phenomenon in most telecommunications sys-

tems, called echo. Here, when mismatch impedance is present in any telecommunications

system, a portion of the transmitted signal is reected to the transmitter as an echo, which

represents an impairment that degrades the system quality (Messershmitt, 1984). In most

telecommunications systems, such as a telephone circuit, the echo is generated when the

long distant portion consisting of two one-directional channels (four wires) is connected

with a bidirectional channel (two wires) by means of a hybrid transformer, as shown in

Figure 1. If the hybrid impedance is perfectly balanced, the two one-directional channels are

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