Muutke küpsiste eelistusi

E-raamat: Understanding Digital Signal Processing

Teised raamatud teemal:
  • Formaat - PDF+DRM
  • Hind: 86,44 €*
  • * hind on lõplik, st. muud allahindlused enam ei rakendu
  • Lisa ostukorvi
  • Lisa soovinimekirja
  • See e-raamat on mõeldud ainult isiklikuks kasutamiseks. E-raamatuid ei saa tagastada.
Understanding Digital Signal ProcessingSuurem pilt
Teised raamatud teemal:

DRM piirangud

  • Kopeerimine (copy/paste):

    ei ole lubatud

  • Printimine:

    ei ole lubatud

  • Kasutamine:

    Digitaalõiguste kaitse (DRM)
    Kirjastus on väljastanud selle e-raamatu krüpteeritud kujul, mis tähendab, et selle lugemiseks peate installeerima spetsiaalse tarkvara. Samuti peate looma endale  Adobe ID Rohkem infot siin. E-raamatut saab lugeda 1 kasutaja ning alla laadida kuni 6'de seadmesse (kõik autoriseeritud sama Adobe ID-ga).

    Vajalik tarkvara
    Mobiilsetes seadmetes (telefon või tahvelarvuti) lugemiseks peate installeerima selle tasuta rakenduse: PocketBook Reader (iOS / Android)

    PC või Mac seadmes lugemiseks peate installima Adobe Digital Editionsi (Seeon tasuta rakendus spetsiaalselt e-raamatute lugemiseks. Seda ei tohi segamini ajada Adober Reader'iga, mis tõenäoliselt on juba teie arvutisse installeeritud )

    Seda e-raamatut ei saa lugeda Amazon Kindle's. 

This book explains digital signal processing topics in detail, with a particular focus on ease of understanding. Accordingly, it includes a wealth of examples to aid in comprehension, and stresses simplicity.





The book is divided into four chapters, which respectively address the topics sampling of continuous time signals; multirate signal processing; the discrete Fourier transform; and filter design concepts. It provides original practical techniques to draw the spectrum of aliased signals, together with well-designed numerical examples to illustrate the operation of the fast transforms, filter algorithms, and circuit designs.





Readers of this book should already have some basic understanding of signals and transforms. They will learn fundamental concepts for signals and systems, as the focus is more on digital signal processing concepts rather than continuous time signal processing topics.
1 Sampling of Continuous Time Signals
1(70)
1.1 Sampling Operation for Continuous Time Signals
2(4)
1.1.1 Sampling Frequency
4(1)
1.1.2 Mathematical Characterization of the Sampling Operation
5(1)
1.2 Sampling Operation
6(5)
1.2.1 The Fourier Transform of the Product Signal
7(4)
1.3 How to Draw Fourier Transforms of Product Signal and Digital Signal
11(19)
1.3.1 Drawing the Fourier Transform of Digital Signal
25(5)
1.4 Aliasing (Spectral Overlapping)
30(15)
1.4.1 The Meaning of the Aliasing (Overlapping)
33(6)
1.4.2 Drawing the Frequency Response of Digital Signal in Case of Aliasing (Practical Method)
39(6)
1.5 Reconstruction of an Analog Signal from Its Samples
45(10)
1.5.1 Approximation of the Reconstruction Filter
51(4)
1.6 Discrete Time Processing of Continuous Time Signals
55(6)
1.7 Continuous Time Processing of Digital Signals
61(7)
1.8 Problems
68(3)
2 Multirate Signal Processing
71(74)
2.1 Sampling Rate Reduction by an Integer Factor (Downsampling, Compression)
72(25)
2.1.1 Fourier Transform of the Downsampled Signal
75(3)
2.1.2 How to Draw the Frequency Response of Downsampled Signal
78(2)
2.1.3 Aliasing in Downsampling
80(3)
2.1.4 Interpretation of the Downsampling in Terms of the Sampling Period
83(9)
2.1.5 Drawing the Fourier Transform of Downsampled Signal in Case of Aliasing (Practical Method)
92(5)
2.2 Upsampling: Increasing the Sampling Rate by an Integer Factor
97(31)
2.2.1 Upsampling (Expansion)
97(1)
2.2.2 Mathematical Formulization of Upsampling
98(1)
2.2.3 Frequency Domain Analysis of Upsampling
99(4)
2.2.4 Interpolation
103(4)
2.2.5 Mathematical Analysis of Interpolation
107(4)
2.2.6 Approximation of the Ideal Interpolation Filter
111(15)
2.2.7 Anti-aliasing Filter
126(2)
2.3 Practical Implementations of C/D and D/C Converters
128(11)
2.3.1 C/D Conversion
129(1)
2.3.2 Sample and Hold
130(4)
2.3.3 Quantization and Coding
134(2)
2.3.4 D/C Converter
136(3)
2.4 Problems
139(6)
3 Discrete Fourier Transform
145(88)
3.1 Manipulation of Digital Signals
146(12)
3.1.1 Manipulation of Periodic Digital Signals
149(1)
3.1.2 Shifting of Periodic Digital Signals
149(7)
3.1.3 Some Well Known Digital Signals
156(2)
3.2 Review of Signal Types
158(7)
3.3 Convolution of Periodic Digital Signals
165(5)
3.3.1 Alternative Method to Compute the Periodic Convolution
166(4)
3.4 Sampling of Fourier Transform
170(2)
3.5 Discrete Fourier Transform
172(26)
3.5.1 Aliasing in Time Domain
182(2)
3.5.2 Matrix Representation of DFT and Inverse DFT
184(1)
3.5.3 Properties of the Discrete Fourier Transform
185(3)
3.5.4 Circular Convolution
188(10)
3.6 Practical Calculation of the Linear Convolution
198(9)
3.6.1 Evaluation of Convolution Using Overlap-Add Method
199(5)
3.6.2 Overlap-Save Method
204(3)
3.7 Computation of the Discrete Fourier Transform
207(18)
3.7.1 Fast Fourier Transform (FFT) Algorithms
207(1)
3.7.2 Decimation in Time FFT Algorithm
207(10)
3.7.3 Decimation in Frequency FFT Algorithm
217(8)
3.8 Total Computation Amount of the FFT Algorithm
225(5)
3.9 Problems
230(3)
4 Analog and Digital Filter Design
233(66)
4.1 Review of Systems
233(7)
4.1.1 Z-Transform
236(3)
4.1.2 Laplace Transform
239(1)
4.2 Transformation Between Continuous and Discrete Time Systems
240(13)
4.2.1 Conversion of Transfer Functions of LTI Systems
245(1)
4.2.2 Forward Difference Transformation Method
246(2)
4.2.3 Bilinear Transformation
248(5)
4.3 Analogue Filter Design
253(20)
4.3.1 Ideal Filters
254(4)
4.3.2 Practical Analog Filter Design
258(2)
4.3.3 Practical Filter Design Methods
260(12)
4.3.4 Analog Frequency Transformations
272(1)
4.4 Implementation of Analog Filters
273(10)
4.4.1 Low Pass Filter Circuits
273(6)
4.4.2 Analog High-Pass Filter Circuit Design
279(3)
4.4.3 Analog Bandpass Active Filter Circuits
282(1)
4.4.4 Analog Bandstop Active Filter Circuits
282(1)
4.5 Infinite Impulse Response (IIR) Digital Filter Design (Low Pass)
283(7)
4.5.1 Generalized Linear Phase Systems
289(1)
4.6 Finite Impulse Response (FIR) Digital Filter Design
290(7)
4.6.1 FIR Filter Design Techniques
291(6)
4.7 Problems
297(2)
Bibliography 299(2)
Index 301
Orhan Gazi is an associate professor in electronic and communication engineering department, Cankaya University.

He got his BS, MS, and PhD degrees all in electrical and electronics engineering from Middle East Technical University, Ankara-Turkey, in 1996, 2001, and 2007 respectively.





His research area involves signal processing, information theory, and forward error correction. Recently he is studying on polar channel codes and preparing publications in this area.
Lisainfo e-raamatute kohta Lisainfo e-raamatute kohta
Püsilink: https://www.kriso.ee/db/97898110496202e.html
Märksõnad: