Discrete-Time Speech Signal Processing: Principles and Practice (IE-Hardcover)
Thomas F. Quatieri
- 出版商: Prentice Hall
- 出版日期: 2001-10-31
- 定價: $1,190
- 售價: 9.8 折 $1,166
- 語言: 英文
- 頁數: 779
- 裝訂: Paperback
- ISBN: 9861541322
- ISBN-13: 9789861541327
-
相關分類:
Matlab、數位影像處理 Digital-image
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商品描述
Essential principles, practical examples, current applications, and leading-edge research.
In this book, Thomas F. Quatieri presents the field's most intensive, up-to-date tutorial and reference on discrete-time speech signal processing. Building on his MIT graduate course, he introduces key principles, essential applications, and state-of-the-art research, and he identifies limitations that point the way to new research opportunities.
Quatieri provides an excellent balance of theory and application, beginning with a complete framework for understanding discrete-time speech signal processing. Along the way, he presents important advances never before covered in a speech signal processing text book, including sinusoidal speech processing, advanced time-frequency analysis, and nonlinear aeroacoustic speech production modeling. Coverage includes:
- Speech production and speech perception: a dual view
- Crucial distinctions between stochastic and deterministic problems
- Pole-zero speech models
- Homomorphic signal processing
- Short-time Fourier transform analysis/synthesis
- Filter-bank and wavelet analysis/synthesis
- Nonlinear measurement and modeling techniques
The book's in-depth applications coverage includes speech coding, enhancement, and modification; speaker recognition; noise reduction; signal restoration; dynamic range compression, and more. Principles of Discrete-Time Speech Processing also contains an exceptionally complete series of examples and Matlab exercises, all carefully integrated into the book's coverage of theory and applications.
Table of Contents
1. Introduction.
2. A Discrete-Time Signal Processing Framework.
3. Production and Classification of Speech Sounds.
4. Acoustics of Speech Production.
5. Analysis and Synthesis of Pole-Zero Speech Models.
Appendix 5.A: Properties of Stochastic Processes.
Appendix 5.B: Derivation of the Lattice Filter in Linear Prediction Analysis.
6. Homomorphic Signal Processing.
7. Short-Time Fourier Transform Analysis and Synthesis.
Appendix 7.A: FBS Method with Multiplicative Modification.
8. Filter-Bank Analysis/Synthesis.
9. Sinusoidal Analysis/Synthesis.
Appendix 9.A: Derivation of the Sinewave Model.
Appendix 9.B: Derivation of Optimal Cubic Phase Parameters.
10. Frequency-Domain Pitch Estimation.
11. Nonlinear Measurement and Modeling Techniques.
12. Speech Coding.
13. Speech Enhancement.
Appendix 13.A: Stochastic-Theoretic parameter Estimation.
14. Speaker Recognition.
Appendix 14.A: Expectation-Maximization (EM) Estimation.
Glossary.
Speech Signal Processing.
Units.
Databases.
Index.
About the Author.
商品描述(中文翻譯)
本書由Thomas F. Quatieri撰寫,是關於離散時間語音信號處理的最全面、最新的教程和參考資料。他在麻省理工學院的研究生課程的基礎上,介紹了關鍵原則、基本應用和最新研究,並指出了指向新研究機會的限制。
Quatieri在理論和應用之間提供了很好的平衡,從完整的框架開始理解離散時間語音信號處理。在此過程中,他提出了以往語音信號處理教材中從未涵蓋的重要進展,包括正弦語音處理、先進的時頻分析和非線性空氣聲學語音生成建模。內容包括:
- 語音生成和語音知覺:雙重觀點
- 隨機和確定性問題的關鍵區別
- 極點-零點語音模型
- 同調信號處理
- 短時傅立葉變換分析/合成
- 濾波器組和小波分析/合成
- 非線性測量和建模技術
本書深入涵蓋了語音編碼、增強、修改、語者識別、降噪、信號恢復、動態範圍壓縮等應用。《離散時間語音處理原理》還包含了一系列非常完整的例子和Matlab練習,所有這些都被精心融入到理論和應用的內容中。
目錄
1. 簡介
2. 離散時間信號處理框架。
3. 語音聲音的生成和分類。
4. 語音生成的聲學。
5. 極點-零點語音模型的分析和合成。
附錄 5.A:隨機過程的性質。
附錄 5.B:線性預測分析中格子濾波器的推導。
6. 同調信號處理。