信号处理导论（影印版）
目 录内容简介
Preface xiii
1 Sampling and Reconstruction 1
1.1 Introduction， 1
1.2 Review of Analog Signals， 1
1.3 Sampling Theorem， 4
1.3.1 Sampling Theoreml 6
1.3.2 Antialiasing Prefilters， 7
1.3.3 Hardware Limits， 8
1.4 Sampling of Sinusoids， 9
1.4.1 Analog Reconstruction and Aliasing， 10
1.4.2 Rotational Motion， 27
1.4.3 DSP Frequency Units， 30
1.5 Spectra of Sampled Signals， 30
1.5.1 Discrete-Time Fourier Transform， 31
1.5.2 Spectrum Replication， 33
1.5.3 Practical Antiallasing Prefilters， 38
1.6 Analog Reconstructors， 43
1.6.1 Ideal Reconstructors， 44
1.6.2 Staircase Reconstructors， 46
1.6.3 Anti-Image Postfilters， 47
1.7 Basic Components of DSP Systems， 54
1.8 Problems， 57
2 Quantization 63
2.1 Quantization Process， 63
2.2 Oversampling and Noise Shaping， 67
2.3 D/A Converters， 73
2.4 A/D Converters， 77
2.5 Analog and Digital Dither， 86
2.6 Problems， 93
3 Discrete-Time Systems 98
3.1 Input/Output Rules， 99
3.2 Linear/ty and Time Invariance， 103
3.3 Impulse Response， 106
3.4 FIR and IIR Filters， 108
3.5 Causality and Stability， 115
3.6 Problems， 120
4 FIR Filtering and Convolution 124
4.1 Block Processing Methods， 125
4.1.1 Convolution， 125
4.1.2 Direct Form， 126
4.1.3 Convolution Table， 129
4.1.4 LTIForm， 130
4.1.5 Matrix Form， 132
4.1.6 Flip-and-Slide Form， 134
4.1.7 Transient and Steady-State Behavior， 135
4.1.8 Convolution of Infinite Sequences， 137
4.1.9 Programming Considerations， 142
4.1.10 Overlap-Add Block Convolution Method， 146
4.2 Sample Processing Methods， 149
4.2.1 Pure Delays， 150
4.2.2 FIR Filtering in Direct Form， 155
4.2.3 Programming Considerations， 163
4.2.4 Hardware Realizations and Circular Buffers， 165
4.3 Problems， 181
5 Z-Transforms 186
5.1 Basic Properties， 186
5.2 Region of Convergence， 189
5.3 Causality and Stability， 196
5.4 Frequency Spectrum， 200
5.5 Inverse z-Transforms， 205
5.6 Problems， 214
6 Transfer Functions 217
6.1 Equivalent Descriptions of Digital Filters， 217
6.2 Transfer Functions， 217
6.3 Sinusoidal Response， 232
6.3.1 Steady-State Response， 232
6.3.2 Transient Response， 235
6.4 Pole/Zero Designs， 246
6.4.1 First-Order Filters， 246
6.4.2 Parametric Resonators and Equalizers， 248
6.4.3 Notch and Comb Filters， 253
6.5 Deconvolution， Inverse Filters， and Stability， 258
6.6 Problems， 263
7 Digital Filter Realizations 269
7.1 Direct Form， 269
7.2 Canonical Form， 275
7.3 Cascade Form， 281
7.4 Cascade to Canonical， 288
7.5 Hardware Realizations and Circular Buffers， 297
7.6 Quantization Effects in Digital Filters， 310
7.7 Problems， 311
8 Signal Processing Applications 321
8.1 Digital Waveform Generators， 321
8.1.1 Sinusoidal Generators， 321
8.1.2 Periodic Waveform Generators， 326
8.1.3 Wavetable Generators， 335
8.2 Digital Audio Effects， 355
8.2.1 Delays， Echoes， and Comb Filters， 355
8.2.2 Flanging， Chorusing， and Phasing， 360
8.2.3 Digital Reverberation， 367
8.2.4 MultRap Delays， 379
8.2.5 Compressors， Limiters， Expanders， and Gates， 384
8.3 Noise Reduction and Signal Enhancement， 388
8.3.1 Noise Reduction Filters， 388
8.3.2 Notch and Comb Filters， 404
8.3.3 Line and Frame Combs for Digital TV， 416
8.3.4 Signal Averaging， 429
8.3.5 Savitzky-Golay Smoothing Filters， 434
8.4 Problems， 462
9 DFT/FFTAlgorithms 472
9.1 Frequency Resolution and Windowing， 472
9.2 DTFT Computation， 483
9.2.1 DTFT at a Single Frequency， 483
9.2.2 DTFT over Frequency Range， 486
9.2.3 DFT， 488
9.2.4 Zero Padding， 490
9.3 Physical versus Computational Resolution， 491
9.4 Matrix Form of DFT， 495
9.5 Modulo-N Reduction， 497
9.6 Inverse DFT， 505
9.7 Sampling of Periodic Signals and the DFT， 508
9.8 FFT， 513
9.9 Fast Convolution， 524
9.9.1 Circular Convolution， 524
9.9.2 Overlap-Add and Overlap-Save Methods， 530
9.10 Problems， 533
10 FIR Digital Filter Design 541
10.1 Window Method， 541
10.1.1 Ideal Filters， 541
10.1.2 Rectangular Window， 544
10.1.3 Hamming Window， 549
10.2 Kaiser Window， 551
10.2.1 Kaiser Window for Filter Design， 551
10.2.2 Kaiser Window for Spectral Analysis， 565
10.3 Frequency Sampling Method， 567
10.4 Other FIR Design Methods， 568
10.5 Problems， 569
11 IIR Digital Filter Design 573
11.1 Bilinear Transformation， 573
11.2 First-Order Lowpass and Highpass Filters， 576
11.3 Second-Order Peaking and Notching Filters， 583
11.4 Parametric Equalizer Filters， 592
11.5 Comb Filters， 601
11.6 Higher-Order Filters， 604
11.6.1 AnalogLowpassButterworthFilters， 605
11.6.2 Digital Lowpass Filters， 611
11.6.3 Digital Highpass Filters， 614
11.6.4 Digital Bandpass Filters， 618
11.6.5 Digital Bandstop Filters， 623
11.6.6 Chebyshev Filter Design， 626
11.7 Problems， 640
12 Interpolation， Decimation， and Oversampling 644
12.1 interpolation and Oversampling， 644
12.2 interpolation Filter Design， 650
12.2.1 Direct Form， 650
12.2.2 Polyphase Form， 652
12.2.3 Frequency Domain Characteristics， 657
12.2.4 Kaiser Window Designs， 660
12.2.5 Multistage Designs， 661
12.3 Linear and Hold interpolators， 669
12.4 Design Examples， 674
12.4.1 4-foldinterpolators， 674
12.4.2 Multistage 4-fold Interpolators， 678
12.4.3 DAC Equalization， 683
12.4.4 Postfilter Design and Equalization， 687
12.4.5 Multistage Equalization， 691
12.5 Decimation and Oversampling， 699
12.6 Sampling Rate Converters， 704
12.7 Noise Shaping Quantizers， 712
12.8 Problems， 720
13 Appendices 728
A Random Signals， 728
A.1 Autocorrelation Functions and Power Spectra， 728
A.2 Filtering of Random Signals， 732
B Random Number Generators， 734
B.1 Uniform and Gaussian Generators， 734
B.2 Low-Frequency Noise Generators， 740
B.3 1/f Noise Generators， 745
B.4 Problems， 749
C Complex Arithmetic in C， 752
D MATLAB Functions， 755
References 773
Index 790
1 Sampling and Reconstruction 1
1.1 Introduction， 1
1.2 Review of Analog Signals， 1
1.3 Sampling Theorem， 4
1.3.1 Sampling Theoreml 6
1.3.2 Antialiasing Prefilters， 7
1.3.3 Hardware Limits， 8
1.4 Sampling of Sinusoids， 9
1.4.1 Analog Reconstruction and Aliasing， 10
1.4.2 Rotational Motion， 27
1.4.3 DSP Frequency Units， 30
1.5 Spectra of Sampled Signals， 30
1.5.1 Discrete-Time Fourier Transform， 31
1.5.2 Spectrum Replication， 33
1.5.3 Practical Antiallasing Prefilters， 38
1.6 Analog Reconstructors， 43
1.6.1 Ideal Reconstructors， 44
1.6.2 Staircase Reconstructors， 46
1.6.3 Anti-Image Postfilters， 47
1.7 Basic Components of DSP Systems， 54
1.8 Problems， 57
2 Quantization 63
2.1 Quantization Process， 63
2.2 Oversampling and Noise Shaping， 67
2.3 D/A Converters， 73
2.4 A/D Converters， 77
2.5 Analog and Digital Dither， 86
2.6 Problems， 93
3 Discrete-Time Systems 98
3.1 Input/Output Rules， 99
3.2 Linear/ty and Time Invariance， 103
3.3 Impulse Response， 106
3.4 FIR and IIR Filters， 108
3.5 Causality and Stability， 115
3.6 Problems， 120
4 FIR Filtering and Convolution 124
4.1 Block Processing Methods， 125
4.1.1 Convolution， 125
4.1.2 Direct Form， 126
4.1.3 Convolution Table， 129
4.1.4 LTIForm， 130
4.1.5 Matrix Form， 132
4.1.6 Flip-and-Slide Form， 134
4.1.7 Transient and Steady-State Behavior， 135
4.1.8 Convolution of Infinite Sequences， 137
4.1.9 Programming Considerations， 142
4.1.10 Overlap-Add Block Convolution Method， 146
4.2 Sample Processing Methods， 149
4.2.1 Pure Delays， 150
4.2.2 FIR Filtering in Direct Form， 155
4.2.3 Programming Considerations， 163
4.2.4 Hardware Realizations and Circular Buffers， 165
4.3 Problems， 181
5 Z-Transforms 186
5.1 Basic Properties， 186
5.2 Region of Convergence， 189
5.3 Causality and Stability， 196
5.4 Frequency Spectrum， 200
5.5 Inverse z-Transforms， 205
5.6 Problems， 214
6 Transfer Functions 217
6.1 Equivalent Descriptions of Digital Filters， 217
6.2 Transfer Functions， 217
6.3 Sinusoidal Response， 232
6.3.1 Steady-State Response， 232
6.3.2 Transient Response， 235
6.4 Pole/Zero Designs， 246
6.4.1 First-Order Filters， 246
6.4.2 Parametric Resonators and Equalizers， 248
6.4.3 Notch and Comb Filters， 253
6.5 Deconvolution， Inverse Filters， and Stability， 258
6.6 Problems， 263
7 Digital Filter Realizations 269
7.1 Direct Form， 269
7.2 Canonical Form， 275
7.3 Cascade Form， 281
7.4 Cascade to Canonical， 288
7.5 Hardware Realizations and Circular Buffers， 297
7.6 Quantization Effects in Digital Filters， 310
7.7 Problems， 311
8 Signal Processing Applications 321
8.1 Digital Waveform Generators， 321
8.1.1 Sinusoidal Generators， 321
8.1.2 Periodic Waveform Generators， 326
8.1.3 Wavetable Generators， 335
8.2 Digital Audio Effects， 355
8.2.1 Delays， Echoes， and Comb Filters， 355
8.2.2 Flanging， Chorusing， and Phasing， 360
8.2.3 Digital Reverberation， 367
8.2.4 MultRap Delays， 379
8.2.5 Compressors， Limiters， Expanders， and Gates， 384
8.3 Noise Reduction and Signal Enhancement， 388
8.3.1 Noise Reduction Filters， 388
8.3.2 Notch and Comb Filters， 404
8.3.3 Line and Frame Combs for Digital TV， 416
8.3.4 Signal Averaging， 429
8.3.5 Savitzky-Golay Smoothing Filters， 434
8.4 Problems， 462
9 DFT/FFTAlgorithms 472
9.1 Frequency Resolution and Windowing， 472
9.2 DTFT Computation， 483
9.2.1 DTFT at a Single Frequency， 483
9.2.2 DTFT over Frequency Range， 486
9.2.3 DFT， 488
9.2.4 Zero Padding， 490
9.3 Physical versus Computational Resolution， 491
9.4 Matrix Form of DFT， 495
9.5 Modulo-N Reduction， 497
9.6 Inverse DFT， 505
9.7 Sampling of Periodic Signals and the DFT， 508
9.8 FFT， 513
9.9 Fast Convolution， 524
9.9.1 Circular Convolution， 524
9.9.2 Overlap-Add and Overlap-Save Methods， 530
9.10 Problems， 533
10 FIR Digital Filter Design 541
10.1 Window Method， 541
10.1.1 Ideal Filters， 541
10.1.2 Rectangular Window， 544
10.1.3 Hamming Window， 549
10.2 Kaiser Window， 551
10.2.1 Kaiser Window for Filter Design， 551
10.2.2 Kaiser Window for Spectral Analysis， 565
10.3 Frequency Sampling Method， 567
10.4 Other FIR Design Methods， 568
10.5 Problems， 569
11 IIR Digital Filter Design 573
11.1 Bilinear Transformation， 573
11.2 First-Order Lowpass and Highpass Filters， 576
11.3 Second-Order Peaking and Notching Filters， 583
11.4 Parametric Equalizer Filters， 592
11.5 Comb Filters， 601
11.6 Higher-Order Filters， 604
11.6.1 AnalogLowpassButterworthFilters， 605
11.6.2 Digital Lowpass Filters， 611
11.6.3 Digital Highpass Filters， 614
11.6.4 Digital Bandpass Filters， 618
11.6.5 Digital Bandstop Filters， 623
11.6.6 Chebyshev Filter Design， 626
11.7 Problems， 640
12 Interpolation， Decimation， and Oversampling 644
12.1 interpolation and Oversampling， 644
12.2 interpolation Filter Design， 650
12.2.1 Direct Form， 650
12.2.2 Polyphase Form， 652
12.2.3 Frequency Domain Characteristics， 657
12.2.4 Kaiser Window Designs， 660
12.2.5 Multistage Designs， 661
12.3 Linear and Hold interpolators， 669
12.4 Design Examples， 674
12.4.1 4-foldinterpolators， 674
12.4.2 Multistage 4-fold Interpolators， 678
12.4.3 DAC Equalization， 683
12.4.4 Postfilter Design and Equalization， 687
12.4.5 Multistage Equalization， 691
12.5 Decimation and Oversampling， 699
12.6 Sampling Rate Converters， 704
12.7 Noise Shaping Quantizers， 712
12.8 Problems， 720
13 Appendices 728
A Random Signals， 728
A.1 Autocorrelation Functions and Power Spectra， 728
A.2 Filtering of Random Signals， 732
B Random Number Generators， 734
B.1 Uniform and Gaussian Generators， 734
B.2 Low-Frequency Noise Generators， 740
B.3 1/f Noise Generators， 745
B.4 Problems， 749
C Complex Arithmetic in C， 752
D MATLAB Functions， 755
References 773
Index 790
目 录内容简介
《信号处理导论(影印版)》以清晰、直观的文体全面介绍了数字信号处理（DSP）的基本原理和算法，并通过大量实例展示了信号处理理论的应用；如：数字信号发生器（包括波表发生器）、数字音响效果处理器、降噪和信号增强、随机噪声发生器等。《信号处理导论(影印版)》实用性极强，全书没有繁琐的公式推导，但提供了100个C语言函数和MATLAB函数，以及编程中的考虑，使读者能方便地进行软件实现和算法仿填 ，同时还介绍了DSP硬件实现的方法。全书有350个习题，其中75个上机实验。此外，还有几个一般的DSP文献少有介绍和内容如：环形缓冲器，参量均衡器设计、音响效果处理、Savitzky-Golay平滑滤波器和噪声整形等。《信号处理导论(影印版)》适用于不同层次的读者如：大学生、研究生、工程技术人员以及DSP爱好者。
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