《數字信號處理（第2版）（英文版）》系統地闡述了數字信號處理所涉及的信號與系統分析和系統設計的基本理論、基本分析與設計方法、基本算法和處理技術。《數字信號處理（第2版）（英文版）》共10章，主要內容包括：離散時間信號與系統的基本概念，離散時間信號與系統的變換域分析，包括z變換和離散時間傅里葉變換、連續時間信號的抽樣與重建，離散傅里葉變換及其快速算法（fft），數字濾波器實現的基本結構，iir和fir數字濾波器的設計原理與基本設計方法，數字信號處理中的有限字長效應，多抽樣率數字信號處理。《數字信號處理（第2版）（英文版）》配有多媒體電子課件、英文版教學大綱、習題指導與實驗手冊。
《數字信號處理（第2版）（英文版）》可以作為電子與通信相關專業的本科數字信號處理課程中英文雙語教學的教材，或中文授課的英文版教學參考書，也可供從事數字信號處理的工程技術人員學習參考。《數字信號處理（第2版）（英文版）》尤其適合初步開展數字信號處理課程中英文雙語授課的師生選用。
21世紀是國際化的知識經濟時代，理工科教育已發生深刻的變化，各專業涉及的新理論與新技術發展日新月異，科技的創新在很大程度上已依賴于信息的及時獲取、準確理解和有效利用。當今，數字信號處理理論和算法的研究、應用與實現技術的發展，以及其在現代信息與通信技術中的重要性和巨大潛力，已超越了初期人們所做的估計與預測。與此同時，社會對高素質信息技術人才的需求對高等學校專業基礎課程的教學質量提出了越來越高的要求，然而“數字信號處理”及其相關技術的基礎課程所能獲得的學時數反而在減少。有知名專家與學者將當今的教學改革難題歸結為：人類知識的無限積累與個人學習能力和時間的有限形成間日益尖銳的矛盾。
高等教育的國際化是當今教育與教學改革的必然趨勢，國際視野和國際交流能力已成為我國高等學校人才培養的一項基本要求。為適應教學改革的新要求，我總結了“數字信號處理”教學與科研工作20余年所積累的經驗與創新成果，并力圖繼承國內老一代專家與學者編著的優秀教材的知識體系結構嚴謹、系統性強的特色與傳統，參考了20余本國內外一流或知名高等學校的優秀教材，通過消化、吸收和創新，編寫了本書。2007年8月本書第1版由電子工業出版社出版。此后，本書在作者本人執教的重慶大學通信工程學院本科數字信號處理課程雙語教學班連續使用了4屆。本書第1版于2010年已脫銷，由于作者工作繁忙，直至今日才進行修訂工作。
我在教材內容的選擇、知識體系的組織和編排方面，做了慎重考慮——本書內容要適應我國高等學校的教學和課程設置的實際情況。面對數字信號處理知識內容迅速擴展和學時數有限的實際情況，我在編寫過程中始終貫徹的基本思想是：使讀者系統地掌握離散時間信號與系統分析與設計的基本理論；在兩種常用的數字信號處理技術方面（基于DFT的連續時間信號的頻譜分析、IIR和FIR濾波器那樣的數字信號處理系統的設計），力求使讀者對分析與設計的原理和方法有較透徹的理解與掌握；在數字信號處理系統中的有限字長效應和多抽樣率數字信號處理方面打下一定的基礎；通過進一步自學或學習更加深入的后續課程，即可較容易地擴充數字信號處理的理論知識與實際技能。
基于我使用第1版作為教材的實際經驗與體會，第2版保留了第1版中的主要內容，以適應目前本科教學的基本需要；壓縮了篇幅，以適應學時數減少的實際情況；修正了第1版中的文字與公式符號錯誤，潤色了語句文字。具體修訂情況如下：
① 基于提高課堂教學效率和提高學生分析與解決問題能力的考慮，對第1版第2、3章中一些相對較簡單的例題進行了精簡。這些例題的題目被插入到相應章的習題中，可以作為學生課后作業。
② 考慮到學時數有限的實際情況，而且第1版未介紹頻率抽樣濾波器設計的內容，刪除原6.3.4節。
③ 基于方便教師檢驗課堂教學效果的考慮，刪除第1版附錄F課后習題參考答案。為方便學生自學，課后習題參考答案可登錄華信教育資源網注冊下載。
蔡坤寶博士，重慶大學通信工程學院教授，信號與信息處理碩士學位點負貴人。長期從事信號與信息處理的教學與科研工作。近十余年來，積極探索和實施中英文雙語教學，現任重慶大學大類系列課程“信號與系統”建設項目負責人，重慶市精品課程“信號與線性系統”負責人、國家級雙語教學示范課程“信號與系統”負責人，并承擔重慶市精品課程“數字信號處理”的建設工作。
1 introduction
1.1 what is a signal?
1.2 what is a system?
1.3 what is signal processing?
1.4 classification of signals
1.4.1 deterministic and random signals
1.4.2 continuous-time and discrete-time signals
1.4.3 periodic signals and nonperiodic signals
1.4.4 energy signals and power signals
1.5 overview of digital signal processing
2 discrete-time signals and systems
2.1 discrete-time signals: sequences
2.1.1 operation on sequences
2.2 basic sequences
2.2.1 some basic sequences
2.2.2 periodicity of sequences
2.2.3 representation of arbitrary sequences
2.3 discrete-time systems
2.3.1 classification of discrete-time systems
2.4 time-domain representations of lti systems
2.4.1 the linear convolution sum
2.4.2 interconnections of lti systems
2.4.3 stability condition of lti systems
2.4.4 causality condition of lti systems
2.4.5 causal and anticausal sequences
2.5 linear constant-coefficient difference equations
2.5.1 recursive solution of difference equations
2.5.2 classical solution of difference equations
2.5.3 zero-input response and zero-state response
2.5.4 the impulse response of causal lti systems
2.5.5 recursive solution of impulse responses
2.5.6 classification of lti discrete-time systems
problems
3 transform-domain analysis of discrete-time signals and systems
3.1 the z-transform
3.1.1 definition of the z-transform
3.1.2 a general shape of the region of convergence
3.1.3 uniqueness of the z-transform
3.2 relation between the rocs and sequence types
3.3 the z-transform of basic sequences
3.4 the inverse z-transform
3.4.1 contour integral method
3.4.2 partial fraction expansion method
3.4.3 long division method
3.4.4 power series expansion method
3.5 properties of the z-transform
3.6 the discrete-time fourier transform
3.6.1 definition of the discrete-time fourier transform
3.6.2 convergence criteria
3.6.3 properties of the discrete-time fourier transform
3.6.4 symmetry properties of the discrete-time fourier transform
3.7 transform-domain analysis of lti discrete-time systems
3.7.1 the frequency response of systems
3.7.2 the transfer function of lti systems
3.7.3 geometric evaluation of the frequency response
3.8 sampling of continuous-time signals
3.8.1 periodic sampling
3.8.2 reconstruction of bandlimited signals
3.9 relations of the z-transform to the laplace transform
problems
4 the discrete fourier transform
4.1 the discrete fourier series
4.2 properties of the discrete fourier series
4.2.1 evaluation of the periodic convolution sum
4.3 the discrete fourier transform
4.4 properties of the discrete fourier transform
4.4.1 circular convolution theorems
4.5 linear convolutions evaluated by the circular convolution
4.6 linear time-invariant systems implemented by the dft
4.7 sampling and reconstruction in the z-domain
4.8 fourier analysis of continuous-time signals using the dft
4.8.1 fourier analysis of nonperiodic continuous-time signals
4.8.2 practical considerations
4.8.3 spectral analysis of sinusoidal signals
problems
5 fast fourier transform algorithms
5.1 direct computation and efficiency improvement of the dft
5.2 decimation-in-time fft algorithm with radix-2
5.2.1 butterfly-branch transmittance of the decimation-in-time fft
5.2.2 in-place computations
5.3 decimation-in-frequency fft algorithm with radix-2
5.4 computational method of the inverse fft
problems
6 digital filter structures
6.1 description of the digital filter structures
6.2 basic structures for iir digital filters
6.2.1 direct form i
6.2.2 direct form ii
6.2.3 cascade form
6.2.4 parallel form
6.3 basic structures for fir digital filters
6.3.1 direct forms
6.3.2 cascade forms
6.3.3 linear-phase forms
problems
7 design techniques of digital iir filters
7.1 preliminary considerations
7.1.1 frequency response of digital filters
7.2 discrete-time systems characterized by phase properties
7.3 allpass systems
7.3.1 nonminimum-phase systems represented by a cascade connection
7.3.2 group delay of the minimum-phase systems
7.3.3 energy delay of the minimum-phase systems
7.4 analog-to-digital filter transformations
7.4.1 impulse invariance transformation
7.4.2 step invariance transformation
7.4.3 bilinear transformation
7.5 design of analog prototype filters
7.5.1 analog butterworth lowpass filters
7.5.2 analog chebyshev lowpass filters
7.6 design of lowpass iir digital filters
7.6.1 design of lowpass digital filters using the impulse invariance
7.6.2 design of lowpass digital filters using the bilinear transformation
7.7 design of iir digital filters using analog frequency transformations
7.7.1 design of bandpass iir digital filters
7.7.2 design of bandstop iir digital filters
7.7.3 design of highpass iir digital filters
7.8 design of iir digital filters using digital frequency transformations
7.8.1 lowpass-to-lowpass transformation
7.8.2 lowpass-to-highpass transformation
7.8.3 lowpass-to-bandpass transformation
7.8.4 lowpass-to-bandstop transformation
problems
8 design of fir digital filters
8.1 properties of linear phase fir filters
8.1.1 the impulse response of linear-phase fir filters
8.1.2 the frequency response of linear-phase fir filters
8.1.3 characteristics of amplitude functions
8.1.4 constraints on zero locations
8.2 design of linear-phase fir filters using windows
8.2.1 basic techniques
8.2.2 window functions
8.2.3 design of linear-phase fir lowpass filters using windows
8.2.4 design of linear-phase fir bandpass filters using windows
8.2.5 design of linear-phase fir highpass filters using windows
8.2.6 design of linear-phase fir bandstop filters using windows
problems
9 finite-wordlength effects in digital signal processing
9.1 binary number representation with its quantization errors
9.1.1 fixed-point binary representation of numbers
9.1.2 floating-point representation
9.1.3 errors from truncation and rounding
9.1.4 statistical model of the quantization errors
9.2 analysis of the quantization errors in a/d conversion
9.2.1 statistical model of the quantization errors
9.2.2 transmission of the quantization noise through lti systems
9.3 coefficient quantization effects in digital filters
9.3.1 coefficient quantization effects in iir digital filters
9.3.2 statistical analysis of coefficient quantization effects
9.3.3 coefficient quantization effects in fir filters
9.4 round-off effects in digital filters
9.4.1 round-off effects in fixed-point realizations of iir filters
9.4.2 dynamic range scaling in fixed-point implementations of iir filters
9.5 limit-cycle oscillations in realizations of iir digital filters
9.5.1 zero-input limit cycle oscillations
9.5.2 limit cycles due to overflow
9.6 round-off errors in fft algorithms
9.6.1 round-off errors in the direct dft computation
9.6.2 round-off errors in fixed-point fft realization
problems
10 multirate digital signal processing
10.1 sampling rate changed by an integer factor
10.1.1 downsampling with an integer factor m
10.1.2 decimation by an integer factor m
10.1.3 upsampling with an integer factor l
10.1.4 interpolation by an integer factor l
10.2 sampling rate conversion by a rational factor
10.3 efficient structures for sampling rate conversion
10.3.1 equivalent cascade structures
10.3.2 polyphase decompositions
10.3.3 polyphase realization of decimation filters
10.3.4 polyphase realization of interpolation filters
problems
appendix a tables for the z-transform
appendix b table for properties of the discrete-time fourier transform
appendix c table for properties of the discrete fourier series
appendix d table for properties of the discrete fourier transform
appendix e table for the normalized butterworth lowpass filters
references
新書資訊