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Digital Signal Processing
Digital Signal Processing
S Poornachandra, B Sasikala
ISBN 9789394828285
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Designed for a first course in Digital Signal Processing, this book covers mahor topics like Discrete Fourier Transform (DFT), Fast Fourier Transform (FFT), Design of Digital Filters, Effect of Finite Word Length and Multirate Signal Processing. Written in a clear style, the book provides a lot of solved problems, illustrations and flow graphs that will facilitate easy learning of the subject.

  • Halftitle Page
  • About the Authors
  • Title Page
  • Copyright Page
  • Contents
  • Preface
  • CHAPTER 1 SIGNALS AND SYSTEMS
    • 1.1 What is DSP
    • 1.2 Introduction to Modeling
      • 1.2.1 Signals
      • 1.2.2 One-dimensional Signal
      • 1.2.3 Two-dimensional Signal
      • 1.2.4 Multi-dimensional Signal
      • 1.2.5 Sampling
      • 1.2.6 Quantization
      • 1.2.7 Coding
    • 1.3 Sampling Rate
      • 1.3.1 Sampling Theorem
      • 1.3.2 Over Sampling
      • 1.3.3 Under Sampling
      • 1.3.4 Nyquist Frequency
      • 1.3.5 Nyquist Rate Relative to Sampling
      • 1.3.6 The Aliasing Problem
      • 1.3.7 Sample and Hold Circuit
    • 1.4 Classification of Signals
      • 1.4.1 Continuous-time Signal and Discrete-time Signal
      • 1.4.2 Periodic and Aperiodic Discrete-time Signal
      • 1.4.3 Odd and Even Discrete-time Signal
      • 1.4.4 Energy Signal and Power Signal
      • 1.4.5 Deterministic Signal and Random Signal
    • 1.5 Basic Operations on Signals
      • 1.5.1 Amplitude Scaling of Signals
      • 1.5.2 Addition of Signals
      • 1.5.3 Multiplication of Signals
      • 1.5.4 Differentiation on Signals
      • 1.5.5 Integration on Signals
      • 1.5.6 Time Scaling of Signals
      • 1.5.7. Reflection of Signals
      • 1.5.8 Time Shifting of Signals
      • 1.5.9 Time Shifting and Time Scaling
    • 1.6 Types of-Signals
      • 1.6.1 Exponential Signal
      • 1.6.2 Sinusoidal Signal (Continuous-time)
      • 1.6.3 Step Function
      • 1.6.4 Impulse Function
      • 1.6.5 Ramp Function
    • 1.7 System
    • 1.8 Properties of Systems
      • 1.8.1 Continuous-time System and Discrete-time System
      • 1.8.2 Stable System and Unstable System
      • 1.8.3 Memory and Memory less System
      • 1.8.4 Invertible and Noninvertible System
      • 1.8.5 Time-invariant and Time-variant System
      • 1.8.6 Linear and Nonlinear System
      • 1.8.7 C ausal and Noncausal System
    • 1.9 Interconnection of Systems
    • 1.10 Linear Time-Invariant (LTI) System
      • 1.10.1 Discrete-Time Linear Time-Invariant System
      • 1.10.2 Representation of Discrete-time Signals in Terms of Impulses
      • 1.10.3 Convolution Sum
    • 1.11 Properties of System
      • 1.11.1 Distributive Property
      • 1.11.2 Associative Property
      • 1.11.3 Commutative Property
    • 1.12 Properties of Discrete-time LTI System
      • 1.12.1 LTI System With and Without Memory
      • 1.12.2 Invertibility of LTI System
      • 1.12.3 Stability for LTI System
      • 1.12.4 Causal System
    • 1.13 Linear Convolution
    • 1.14 Linear Convolution using Cross-Table Method
    • 1.15 Linear Convolution using Matrix Method
    • 1.16 Step Response
    • 1.17 Deconvolution
    • 1.18 Basic Systems
    • 1.19 Linear Constant Coefficient Difference Equation
      • 1.19.1 Solution to Linear Constant Coefficient Difference Equation
    • 1.20 Introduction to Correlation
      • 1.20.1 Cross-correlation
      • 1.20.2 Autocorrelation
      • 1.20.3 Properties of Cross-correlation and Autocorrelation
    • Chapter Summary
    • Review Questions
  • CHAPTER 2 FOURIER TRANSFORM
    • 2.1 Continuous-time Fourier Series
    • 2.2 Fourier Series Representation of Continuous-time Periodic Signal
    • 2.3 Fourier Series Representation (Exponential) of a Continuous-time Signal
    • 2.4 Trigonometric Representation of Continuous-time Fourier Series
    • 2.5 Properties of Continuous-time Fourier Series
      • 2.5.1 Continuous-time Fourier Series
      • 2.5.2 Linearity Property
      • 2.5.3 Time Shifting Property
      • 2.5.4 Time Reversal Property
      • 2.5.5 Time Scaling Property
      • 2.5.6 Multiplication Property
      • 2.5.7 Conjugation and Conjugate Symmetry Property
      • 2.5.8 Parseval's Relation for Continuous-time Periodic Signal
      • 2.5.9 Differentiation Property
      • 2.5.10 Integration Property
    • 2.6 Dirichlet Condition
    • 2.7 Gibb's Phenomenon
    • 2.8 Discrete-Time Fourier Series
      • 2.8.1 To Calculate Discrete-time Fourier Series Coefficient
    • 2.9 Properties of Discrete-time Fourier Series
      • 2.9.1 Linearity Property
      • 2.9.2 Time Shifting Property
      • 2.9.3 Time Reversal Property
      • 2.9.4 Time Scaling Property
      • 2.9.5 Multiplication Property
      • 2.9.6 Conjugation and Conjugate Symmetry Property
      • 2.9.7 Parseval's Relation for Discrete-time Periodic Signal
      • 2.9.8 Difference Property2.9.9 Running Sum (Integration) Property
      • 2.9.9 Running Sum (Integration) Property
    • 2.10 Continuous-Time Fourier Transform
    • 2.11 Representation of Aperiodic Signals
    • 2.12 Spectrum Analysis
    • 2.13 Properties of Continuous-time Fourier Transform
      • 2.13.1 Linearity Property
      • 2.13.2 Time Shifting Property
      • 2.13.3 Frequency Shifting Property
      • 2.13.4 Time Reversal Property
      • 2.13.5 Time Scaling Property
      • 2.13.6 Multiplication Property
      • 2.13.7 Conjugation and Conjugate Symmetry Property
      • 2.13.8 Parseval's Relation for Continuous-time Aperiodic Signal
      • 2.13.9 Differentiation in Time Property
      • 2.13.10 Differentiation in Frequency Property
      • 2.13.11 Integration Property
      • 2.13.12 Convolution Property
    • 2.14 Discrete-Time Fourier Transform
    • 2.15 Representation of Discrete-time Aperiodic Signals
    • 2.16 Properties of Discrete-time Fourier Transform
      • 2.16.1 Linearity Property
      • 2.16.2 Time Shifting Property
      • 2.16.3 Frequency Shifting Property
      • 2.16.4 Time Reversal Property
      • 2.16.5 Time Scaling Property
      • 2.16.6 Multiplication Property
      • 2.16.7 Conjugation and Conjugate Symmetry Property
      • 2.16.8 Parseval's Relation for Discrete-time Periodic Signal
      • 2.16.9 Difference in Time Property
      • 2.16.10 Difference in Frequency Property
      • 2.16.11 Accumulation Property
      • 2.16.12 Convolution Property
      • 2.16.13 Convergence of the Fourier Transform
      • 2.16.14
    • 2.17 Band Limited Signals
      • 2.17.1 Mathematical Analysis
      • 2.17.2 The Ideal Bandpass Signal
    • 2.18 Parameter Estimation of a Band Limited Signal
      • 2.18.1 Signal Energy
    • 2.19 Orthogonal Band Limited Signal
    • 2.20 Discrete Fourier Transform
    • 2.21 Frequency Analysis of Discrete-time Signal
    • 2.22 Properties of DFT
      • 2.22.1 Periodicity
      • 2.22.2 Linearity
      • 2.22.3 Circular Shift of a Sequence (Time-domain)
      • 2.22.4 Time Reversal of the Sequence
      • 2.22.5 Circular Shift (Frequency-domain)
      • 2.22.6 Complex-conjugate Properties
      • 2.22.7 Circular Correlation
      • 2.22.8 Parseval's Theorem
      • 2.22.9 Symmetry Properties of the DFT
      • 2.22.10 Multiplication of Two DFT
      • 2.22.11 Multiplication of Two Sequences
      • 2.22.12 Convolution of Two Sequences
    • 2.23 Circular Convolution
      • 2.23.1 Circle Method
      • 2.23.2 Matrix Method
      • 2.23.3 DFT-IDFT Method
    • 2.24 Section convolution
      • 2.24.1 Overlap-save Method
      • 2.24.2 Overlap-add Method
    • 2.25 Fast Fourier Transform (FFT)
      • 2.25.1 Radix-2 FFT Algorithm
      • 2.25.2 Decimation-in-time FFT Algorithm
      • 2.25.3 Decimation-in-frequency FFT Algorithm
    • 2.26 Inverse Fast Fourier Transform (IFFT)
    • 2.27 FFT Algonthm in Linear Filtering and Correlation
    • 2.28 Inplace Computation Using Butterfly Structure
    • 2.29 Computational Complexity
    • 2.30 Comparison Between DIT and DIF Algorithm
    • 2.31 Discrete Cosine Transform
    • Chapter Summary
    • Review Questions
  • CHAPTER 3 Z-TRANSFORM
    • 3.1 Introduction
    • 3.2 Z-Transform
    • 3.3 Region of Convergence (ROC)
      • 3.3 1 Relationship between Z-Transform and DTF
    • 3.4 Z-Transform of Finite Sequence
      • 3.4.1 Right-hand Finite Sequence
      • 3.4.2 Left-hand Finite Sequence
      • 3.4.3 Two-sided Finite Sequence
    • 3.5 Characteristic Features of Signals
      • 3.5.1 Properties of Region of Convergence (ROC)
    • 3.6 Properties of Z-Transform
      • 3.6.1 Linearity Property
      • 3.6.2 Time Shifting Property
      • 3.6.3 Time Reversal Property
      • 3.6.4 Time Scaling Property
      • 3.6.5 Multiplication Property
      • 3.6.6 Conjugation and Conjugate Symmetry Property
      • 3.6.7 Parseval's Relation
      • 3.6.8 Difference in Time Property
      • 3.6.9 Differentiation in Frequency Property
      • 3.6.10 Convolution Property
    • 3.7 Initial Value and Final Value Theorem
      • 3.7.1 Initial Value Theorem
      • 3.7.2 Final Value Theorem
    • 3.8 Inverse Z-Transform
      • 3.8.1 Power Series Method (Long-division)
      • 3.8.2 Partial Fraction Method
      • 3.8.3 Residual Method
      • 3.8.4 Convolution of Two Signals
    • 3.9 LTI System Characterized by Linear Constant Coefficient Difference Equation
    • 3.10 Relationship between Z-Transform and Fourier Transform
    • 3.11 Relationship between Z-plane and S-plane
    • Chapter Summary
    • Review Questions
  • CHAPTER 4 INFINITE IMPULSE RESPONSE FILTER
    • 4.1 Introduction
      • 4.1.1 Distinguish between Analog and Digital Filter
    • 4.2 Analog Filters
    • 4.3 Analog Domain to Digital Domain Transformation
      • 4.3.1 Impulse-Invariant Transformation Technique
      • 4.3.2 Bilinear Transformation Technique
      • 4.3.3 Approximation of Derivatives
    • 4.4 Analog Frequency Transformation
      • 4.4.1 Normalized Low Pass Filter to Desired Low Pass Filter Transformation
      • 4.4.2 Normalized Low Pass Filter to Desired High Pass Filter Transformation
      • 4.4.3 Normalized Low Pass Filter to Desired Band Pass Filter Transformation
      • 4.4.4 Normalized Low Pass Filter to Desired
    • 4.5 Digital Frequency Transformation
      • 4.5.1 Low Pass to Desired Low Pass Filter Transformation
      • 4.5.2 Low Pass to Desired High Pass Filter Transformation
      • 4.5.3 Low Pass to Desired Band Pass Filter Transformation
      • 4.5.4 Low Pass to Desired Band Elimination Filter Transformation
    • 4.6 Butterworth Filter
      • 4.6.1 Determination of Order of the Filter
    • 4.7 Chebyshev Filter
      • 4.7.1 Magnitude Response of Type-I Chebyshev Filter
      • 4.7.2 Order of the Filter
      • 4.7.3 Transfei Function of Chebyshev Filter
    • 4.8 Structure Realization—Introduction
    • 4.9 Implementation of Discrete-time System
      • 4.9.1 Structure Realization of IIR System Recursive Structure
      • 4.9.2 System Realization using Direct Form-I
      • 4.9.3 System Realization using Direct Form-II
      • 4.9.4 System Realization using Cascade Form
      • 4.9.5 System Realization using Parallel Form
    • 4.10 Lattice Structure of IIR System
    • 4.11 Lattice Ladder Structure of IIR System
    • Chapter Summary
    • Review Questions
  • CHAPTER 5 FINITE IMPULSE RESPONSE FILTER
    • 5.1 Introduction
    • 5.2 Phase Delay and Group Delay
    • 5.3 Linear Phase Transfer Function
      • 5.3.1 Symmetric: Impulse Response with Odd Length
      • 5.3.2 Symmetric Impulse Response with Even Length
      • 5.3.3 Antisymmetric Impulse Response with Odd Length
      • 5.3.4 Antisymmetric Impulse Response with Even Length
    • 5.4 Design of FIR Filter- Fourier Method
    • 5.5 Design of FIR Filter- Windowing Techniques
    • 5.6 The Triangular Window (Bartlett Window)
    • 5.7 Raised Cosine Window
    • 5.8 Hanning Window
    • 5.9 Hamming Window
    • 5.10 Blackman Window
    • 5.11 Kaiser Window
      • 5.11.1 Kaiser Window Technique for High Pass Filter
      • 5:11.2 Kaiser Window Technique for Band Pass Filter
      • 5.11.3 Kaiser Window Technique for Band Stop Filter
    • 5.12 FIR Filter Design Using Frequency Sampling Technique
    • 5.13 Equiripple Linear-phase FIR Filter
    • 5.14 FIR Differentiators
    • 5.15 Hilbert Transform
      • 5.15.1 FIR vs. IIR
    • 5.16 Structure Realization of FIR System
      • 5.16.1 Direct Form Realization of FIR System
      • 5.16.2 Cascade Structure Realization of FIR System
      • 5.16.3 Realization Structure of Linear Phase FIR System
      • 5.16.4 Lattice Structure for FIR Filter
    • Chapter Summary
    • Review Questions
  • CHAPTER 6 FINITE WORD LENGTH EFFECT
    • 6.1 Introduction
    • 6.2 Fixed-Point Numbers
      • 6.2.1 Sign-magnitude Format
      • 6.2.2 One's-complement Format
      • 6.2.3 Two's-complement Format
      • 6.2.4 Fixed-point Addition
      • 6.2.5 Fixed-point Multiplication
    • 6.3 Floating-Point Numbers
      • 6.3.1 The Floating-point Representation of Integer Part
      • 6.3.2 The Floating-point Representation of Fractional Part
      • 6.3.3 IEEE Single Precision and Double Precision Format
      • 6.3.4 Floating-Point Addition
      • 6.3.5 Floating-point Multiplication
      • 6.3.6 Comparative Study of Fixed-point and Floating-point Representations
      • 6.3.7 Dynamic Range, Resolution and Precision
    • 6.4 Quantization Error
      • 6.4.1 Truncation
      • 6.4.2 Rounding
      • 6.4.3 Quantization Error in Floating-point Representation
      • 6.4.4 Probability Density Function of Quantization Error
    • 6.5 Variance Estimation of Quantization Error
      • 6.5.1 SNR Calculation
      • 6.5.2 Effect of Scaling on SNR
    • 6.6 Finite Word Length Effect on IIR Filter
      • 6.6.1 Finite Word Length Effect on Filter Structure
      • 6.6.2 Effect of Finite Word Length on Direct Form-I Structure
      • 6.6.3 Effect of Finite Word Length on Direct Form-1! Structure
    • 6.7 Product Quantization Error in IIR Filter
    • 6.8 Mathematical Analysis of Steady-state Output Noise
    • 6.9 Dynamic Scaling to Prevent Overflow
      • 6.9.1 Alternate Method
    • 6.10 Limit-cycle Oscillations in Recursive Systems
    • 6.11 Rounding off Error in DFT Computation
    • 6.12 Rounding off Error in FFT Computation
    • Solved University Questions
    • Chapter Summary
    • Review Questions
  • CHAPTER 7 MULTIRATE DIGITAL SIGNAL PROCESSING
    • 7.1 Introduction
    • 7.2 Decimator (Down-sampler)
      • 7.2.1 Spectral Analysis of Decimator
    • 7.3 Interpolation (Up-sampling)
      • 7.3.1 Spectral Analysis of Interpolator
    • 7.4 Sampling Rate
    • 7.5 Noble Identities
      • 7.5.1 Noble Identity for Down-sampler
      • 7.5.2 Noble identity for Up-sampler
    • 7.6 Polyphase Decomposition for Decimator
    • 7.7 Polyphase Decomposition for Interpolator
    • 7.8 Multistage Design of Decimator and Interpolator
      • 7.8.1 Multistage Implementation of Decimator
      • 7.8.2 Multistage Implementation of Interpolator
    • 7.9 Comb Filter
    • 7.10 Applications of Multirate Signal Processing
      • 7.10.1 Phase Shifter
      • 7.10.2 Subband Coding of Speech Signal
      • 7.10.3 Multirate Narrowband Digital Filtering
    • 7.11 Quadrature Mirror Filter
      • 7.11.1 Two-channel QMF Bank
      • 7.11.2 Mathematical Analysis of Two-channel QMF
      • 7.11.3 Alias-free Realization
    • Chapter Summary
    • Review Questions
  • CHAPTER 8 SPEECH PROCESSING AND COMPRESSION
    • 8.1 Introduction
    • 8.2 Production of Speech Waveform
      • 8.2.1 Important Points to Remember
    • 8.3 Speech Recognition
      • 8.3.1 Training Stage
      • 8.3.2 Recognition Stage
    • 8.4 Linear Predictive Coding
      • 8.4.1 Basic Linear Prediction Model
    • 8.5 The Cepstrum: A Method for Speech Analysis
      • 8.5.1 Subband Coding of Speech Signals Vocoder
    • 8.6 Transmultiplexers
      • 8.6.1 FDM to TDM Transmultiplexer
      • 8.6.2 TDM to FDM Transmultiplexer
    • 8.7 Speech Compression
    • 8.8 Coding Technique
    • 8.9 Speech Coding Techniques
    • Chapter Summary
    • Review Questions
  • CHAPTER 9 ADAPTIVE FILTERS
    • 9.1 Introduction
      • 9.1.1 Need for Adaptive Filter
      • 9.1.2 Characteristics of Adaptive Filter
    • 9.2 Performance Measures of Adaptive Filter
    • 9.3 Open-loop and Closed-loop Adaptation
      • 9.3.1 Open-loop Adaptation
      • 9.3.2 Closed-loop Adaptation
      • 9.3.3 Factors that Determining the Choice of Adaptation
    • 9.4 Approaches for Deriving Recursive Algorithms
      • 9.4.1 Stochastic Gradient Approach
      • 9.4.2 Least-squares Estimation
      • 9.4.3 How to Choose an Adaptive Filter
    • 9.5 The Adaptive Linear Combiner
    • 9.6 Gradient Search Methods
      • 9.6.1 Concept of Gradient Search Methods
      • 9.6.2 Gradient Search Algorithm
      • 9.6.3 Newton's Descent Method for Gradient Search
      • 9.6.4 Steepest Descent Method for Gradient Search
    • 9.7 The LMS Algorithm
      • 9.7.1 Convergence of LMS Algorithm
    • 9.8 The RLS Algorithm
      • 9.8.1 Recursive Computation of ф(n) and z(n)
      • 9.8.2 Matrix Inversion Lemma
      • 9.8.3 Time Update for the lap-Weight Vector
      • 9.8.4 Distinguish between RLS and LMS Algorithm
    • 9.9 Application of Adaptive Filter
      • 9.9.1 Adaptive Modeling and System Identification
      • 9.9.2 Inverse Modeling and Adaptive Equalization
      • 9.9.3 Adaptive Prediction
      • 9.9.4 Adaptive Interference Cancellation
      • 9.9.5 Adaptive Sidelobe Cancellation
    • Chapter Summary
    • Review Questions
  • CHAPTER 10 MUSICAL SOUND PROCESSING
    • 10.1 Introduction
    • 10.2 Time Domain Operations
      • 10.2.1 Single Echo Generating Filter
      • 10.2.2 Multiple Echo Generating Filter
      • 10.2.3 Infinite Echo Generating Filter
      • 10.2.4 Reverberation Structure
      • 10.2.5 IIR Echoes Filter with Reverberation
      • 10.2.6 Teeth Filter
      • 10.2.7 Flanging
      • 10.2.8 Chorus Effect
      • 10.2.9 Phasing Effect
    • 10.3 Frequency Domain Operations
      • 10.3.1 First Order Filter
      • 10.3.2 Shelving Filter
      • 10.3.3 Second Order Filter
      • 10.3.4 Equalizer
    • Chapter Summary
    • Review questions
  • CHAPTER 11 IMAGE ENHANCEMENT
    • 11.1 Introduction to Image Processing
    • 11.2 Basic Building Blocks of Image Processing System
      • 11.2.1 Image Acquisition and Formation
      • 11.2.2 Image Enhancement
      • 11.2.3 Image Restoration
      • 11.2.4 Image Compression
      • 11.2.5 Image Morphological Processing
      • 11.2.6 Image Segmentation
      • 11.2.7 Image Representation and Description
      • 11.2.8 Image Recognition
    • 11.3 Image Enhancement Techniques
      • 11.3.1 Basic operations of Image Enhancement Technique
      • 11.3.2 Broad Areas for Analysis of Image Enhancement Techniques
    • 11.4 Spatial-Domain Approach
      • 11.4.1 Gray Level Transformation
      • 11.4.2 Basic Gray Level Transformation Functions
      • 11.4.3 Piecewise Linear Transformation Functions
      • 11.4.4 Histogram Processing
      • 11.4.5 Local Enhancement Techniques
      • 11.4.6 Image Enhancement using Arithmetic/ Logical Operation
      • 11.4.7 Spatial "Filtering
    • 11.5 Frequency-domain Analysis
      • 11.5.1 Lowpass Frequency-domain Filter
      • 11.5.2 Highpass Filter
      • 11.5.3 Hbmomorphic Filter
    • Chapter Summary
    • Review Questions
  • CHAPTER 12 DIGITAL SIGNAL PROCESSOR
    • 12.1 Introduction
    • 12.2 Programmable Digital Signal Processing
    • 12.3 Multiplier Accumulator
      • 12.3.1 Overflow and Underflow in MAC Unit
    • 12.4 Computer Architecture
      • 12.4.1 Von-Neumann Architecture
      • 12.4.2 Harvard Architecture
      • 12.4.3 Modified-Harvard Architecture
    • 12.5 On-chip (Cache) Memory
    • 12.6 Pipelining
    • 12.7 Pipeline Structure Processor
      • 12.7.1 Single Instruction Multiple Data Architecture
      • 12.7.2 Very Long Instruction Word Architecture
      • 12.7.3 Superscalar Processing Architecture
    • 12.8 Computer Configuration
      • 12.8.1 Restricted Instruction Set Computer
      • 12.8.2 Complex Instruction Set Computer
    • 12.9 Addressing Modes
      • 12.9.1 Immediate Addressing
      • 12.9.2 Absolute Addressing
      • 12.9.3 Accumulator Addressing
      • 12.9.4 Direct Addressing
      • 12.9.5 Indirect Addressing
      • 12.9.6 Circular Addressing
      • 12.9.7 Bit Reversal Addressing
      • 12.9.8 Memory-Mapped Register Addressing
      • 12.9.9 Stack Addressing
      • 12.9.10 Replication
    • 12.10 Introduction to Processor
    • 12.11 First Generation TMS320C1X Processor
      • 12.11.1 Architecture
      • 12.11.2 The Arithmetic and Logic Unit
    • 12.12 Second-Generation TMS320C2X Processor
    • 12.13 TMS320C3X Digital Signal Processor
      • 12.13.2 Memory Architecture
    • 12.14 TMS320C4X Digital Signal Processor
      • 12.14.1 Architecture
      • 12.14.2 Memory Architecture
      • 12.14.3 Internal Bus Operation
      • 12.14.4 Communication Port
      • 12.14.5 DMA Coprocessor
    • 12.15 TMS320C5X Digital Signal Processor
      • 12.15.1 Central Processing Unit (CPU)
      • 12.15.2 On-Chip Memory
      • 12.15.3 On-Chip Peripherals
      • 12.15.4 Addressing Modes
      • 12.15.5 TMS320C54XX Instructions
    • 12.16 TMS320C6X Digital Signal Processor
      • 12.16.1 TMS320C64X Processor
      • 12.16.2 TMS320C67X Processor
    • 12.17 Code Composer Studio
      • 12.7.1 What is DSK
      • 12.7.2 Introduction to Code Composer Studio
    • Chapter Summary
    • Review Questions
B.E., M.Tech., Ph.D. Jerusalem College of Engineering Chennai pcmed8@yahoo. comB.E., M.S. Department of Electronics and Communication Engineering Cresent Engineering College Chennai
B.E., M.Tech., Ph.D. Jerusalem College of Engineering Chennai pcmed8@yahoo. comB.E., M.S. Department of Electronics and Communication Engineering Cresent Engineering College Chennai
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Description

Designed for a first course in Digital Signal Processing, this book covers mahor topics like Discrete Fourier Transform (DFT), Fast Fourier Transform (FFT), Design of Digital Filters, Effect of Finite Word Length and Multirate Signal Processing. Written in a clear style, the book provides a lot of solved problems, illustrations and flow graphs that will facilitate easy learning of the subject.

Table of contents
  • Halftitle Page
  • About the Authors
  • Title Page
  • Copyright Page
  • Contents
  • Preface
  • CHAPTER 1 SIGNALS AND SYSTEMS
    • 1.1 What is DSP
    • 1.2 Introduction to Modeling
      • 1.2.1 Signals
      • 1.2.2 One-dimensional Signal
      • 1.2.3 Two-dimensional Signal
      • 1.2.4 Multi-dimensional Signal
      • 1.2.5 Sampling
      • 1.2.6 Quantization
      • 1.2.7 Coding
    • 1.3 Sampling Rate
      • 1.3.1 Sampling Theorem
      • 1.3.2 Over Sampling
      • 1.3.3 Under Sampling
      • 1.3.4 Nyquist Frequency
      • 1.3.5 Nyquist Rate Relative to Sampling
      • 1.3.6 The Aliasing Problem
      • 1.3.7 Sample and Hold Circuit
    • 1.4 Classification of Signals
      • 1.4.1 Continuous-time Signal and Discrete-time Signal
      • 1.4.2 Periodic and Aperiodic Discrete-time Signal
      • 1.4.3 Odd and Even Discrete-time Signal
      • 1.4.4 Energy Signal and Power Signal
      • 1.4.5 Deterministic Signal and Random Signal
    • 1.5 Basic Operations on Signals
      • 1.5.1 Amplitude Scaling of Signals
      • 1.5.2 Addition of Signals
      • 1.5.3 Multiplication of Signals
      • 1.5.4 Differentiation on Signals
      • 1.5.5 Integration on Signals
      • 1.5.6 Time Scaling of Signals
      • 1.5.7. Reflection of Signals
      • 1.5.8 Time Shifting of Signals
      • 1.5.9 Time Shifting and Time Scaling
    • 1.6 Types of-Signals
      • 1.6.1 Exponential Signal
      • 1.6.2 Sinusoidal Signal (Continuous-time)
      • 1.6.3 Step Function
      • 1.6.4 Impulse Function
      • 1.6.5 Ramp Function
    • 1.7 System
    • 1.8 Properties of Systems
      • 1.8.1 Continuous-time System and Discrete-time System
      • 1.8.2 Stable System and Unstable System
      • 1.8.3 Memory and Memory less System
      • 1.8.4 Invertible and Noninvertible System
      • 1.8.5 Time-invariant and Time-variant System
      • 1.8.6 Linear and Nonlinear System
      • 1.8.7 C ausal and Noncausal System
    • 1.9 Interconnection of Systems
    • 1.10 Linear Time-Invariant (LTI) System
      • 1.10.1 Discrete-Time Linear Time-Invariant System
      • 1.10.2 Representation of Discrete-time Signals in Terms of Impulses
      • 1.10.3 Convolution Sum
    • 1.11 Properties of System
      • 1.11.1 Distributive Property
      • 1.11.2 Associative Property
      • 1.11.3 Commutative Property
    • 1.12 Properties of Discrete-time LTI System
      • 1.12.1 LTI System With and Without Memory
      • 1.12.2 Invertibility of LTI System
      • 1.12.3 Stability for LTI System
      • 1.12.4 Causal System
    • 1.13 Linear Convolution
    • 1.14 Linear Convolution using Cross-Table Method
    • 1.15 Linear Convolution using Matrix Method
    • 1.16 Step Response
    • 1.17 Deconvolution
    • 1.18 Basic Systems
    • 1.19 Linear Constant Coefficient Difference Equation
      • 1.19.1 Solution to Linear Constant Coefficient Difference Equation
    • 1.20 Introduction to Correlation
      • 1.20.1 Cross-correlation
      • 1.20.2 Autocorrelation
      • 1.20.3 Properties of Cross-correlation and Autocorrelation
    • Chapter Summary
    • Review Questions
  • CHAPTER 2 FOURIER TRANSFORM
    • 2.1 Continuous-time Fourier Series
    • 2.2 Fourier Series Representation of Continuous-time Periodic Signal
    • 2.3 Fourier Series Representation (Exponential) of a Continuous-time Signal
    • 2.4 Trigonometric Representation of Continuous-time Fourier Series
    • 2.5 Properties of Continuous-time Fourier Series
      • 2.5.1 Continuous-time Fourier Series
      • 2.5.2 Linearity Property
      • 2.5.3 Time Shifting Property
      • 2.5.4 Time Reversal Property
      • 2.5.5 Time Scaling Property
      • 2.5.6 Multiplication Property
      • 2.5.7 Conjugation and Conjugate Symmetry Property
      • 2.5.8 Parseval's Relation for Continuous-time Periodic Signal
      • 2.5.9 Differentiation Property
      • 2.5.10 Integration Property
    • 2.6 Dirichlet Condition
    • 2.7 Gibb's Phenomenon
    • 2.8 Discrete-Time Fourier Series
      • 2.8.1 To Calculate Discrete-time Fourier Series Coefficient
    • 2.9 Properties of Discrete-time Fourier Series
      • 2.9.1 Linearity Property
      • 2.9.2 Time Shifting Property
      • 2.9.3 Time Reversal Property
      • 2.9.4 Time Scaling Property
      • 2.9.5 Multiplication Property
      • 2.9.6 Conjugation and Conjugate Symmetry Property
      • 2.9.7 Parseval's Relation for Discrete-time Periodic Signal
      • 2.9.8 Difference Property2.9.9 Running Sum (Integration) Property
      • 2.9.9 Running Sum (Integration) Property
    • 2.10 Continuous-Time Fourier Transform
    • 2.11 Representation of Aperiodic Signals
    • 2.12 Spectrum Analysis
    • 2.13 Properties of Continuous-time Fourier Transform
      • 2.13.1 Linearity Property
      • 2.13.2 Time Shifting Property
      • 2.13.3 Frequency Shifting Property
      • 2.13.4 Time Reversal Property
      • 2.13.5 Time Scaling Property
      • 2.13.6 Multiplication Property
      • 2.13.7 Conjugation and Conjugate Symmetry Property
      • 2.13.8 Parseval's Relation for Continuous-time Aperiodic Signal
      • 2.13.9 Differentiation in Time Property
      • 2.13.10 Differentiation in Frequency Property
      • 2.13.11 Integration Property
      • 2.13.12 Convolution Property
    • 2.14 Discrete-Time Fourier Transform
    • 2.15 Representation of Discrete-time Aperiodic Signals
    • 2.16 Properties of Discrete-time Fourier Transform
      • 2.16.1 Linearity Property
      • 2.16.2 Time Shifting Property
      • 2.16.3 Frequency Shifting Property
      • 2.16.4 Time Reversal Property
      • 2.16.5 Time Scaling Property
      • 2.16.6 Multiplication Property
      • 2.16.7 Conjugation and Conjugate Symmetry Property
      • 2.16.8 Parseval's Relation for Discrete-time Periodic Signal
      • 2.16.9 Difference in Time Property
      • 2.16.10 Difference in Frequency Property
      • 2.16.11 Accumulation Property
      • 2.16.12 Convolution Property
      • 2.16.13 Convergence of the Fourier Transform
      • 2.16.14
    • 2.17 Band Limited Signals
      • 2.17.1 Mathematical Analysis
      • 2.17.2 The Ideal Bandpass Signal
    • 2.18 Parameter Estimation of a Band Limited Signal
      • 2.18.1 Signal Energy
    • 2.19 Orthogonal Band Limited Signal
    • 2.20 Discrete Fourier Transform
    • 2.21 Frequency Analysis of Discrete-time Signal
    • 2.22 Properties of DFT
      • 2.22.1 Periodicity
      • 2.22.2 Linearity
      • 2.22.3 Circular Shift of a Sequence (Time-domain)
      • 2.22.4 Time Reversal of the Sequence
      • 2.22.5 Circular Shift (Frequency-domain)
      • 2.22.6 Complex-conjugate Properties
      • 2.22.7 Circular Correlation
      • 2.22.8 Parseval's Theorem
      • 2.22.9 Symmetry Properties of the DFT
      • 2.22.10 Multiplication of Two DFT
      • 2.22.11 Multiplication of Two Sequences
      • 2.22.12 Convolution of Two Sequences
    • 2.23 Circular Convolution
      • 2.23.1 Circle Method
      • 2.23.2 Matrix Method
      • 2.23.3 DFT-IDFT Method
    • 2.24 Section convolution
      • 2.24.1 Overlap-save Method
      • 2.24.2 Overlap-add Method
    • 2.25 Fast Fourier Transform (FFT)
      • 2.25.1 Radix-2 FFT Algorithm
      • 2.25.2 Decimation-in-time FFT Algorithm
      • 2.25.3 Decimation-in-frequency FFT Algorithm
    • 2.26 Inverse Fast Fourier Transform (IFFT)
    • 2.27 FFT Algonthm in Linear Filtering and Correlation
    • 2.28 Inplace Computation Using Butterfly Structure
    • 2.29 Computational Complexity
    • 2.30 Comparison Between DIT and DIF Algorithm
    • 2.31 Discrete Cosine Transform
    • Chapter Summary
    • Review Questions
  • CHAPTER 3 Z-TRANSFORM
    • 3.1 Introduction
    • 3.2 Z-Transform
    • 3.3 Region of Convergence (ROC)
      • 3.3 1 Relationship between Z-Transform and DTF
    • 3.4 Z-Transform of Finite Sequence
      • 3.4.1 Right-hand Finite Sequence
      • 3.4.2 Left-hand Finite Sequence
      • 3.4.3 Two-sided Finite Sequence
    • 3.5 Characteristic Features of Signals
      • 3.5.1 Properties of Region of Convergence (ROC)
    • 3.6 Properties of Z-Transform
      • 3.6.1 Linearity Property
      • 3.6.2 Time Shifting Property
      • 3.6.3 Time Reversal Property
      • 3.6.4 Time Scaling Property
      • 3.6.5 Multiplication Property
      • 3.6.6 Conjugation and Conjugate Symmetry Property
      • 3.6.7 Parseval's Relation
      • 3.6.8 Difference in Time Property
      • 3.6.9 Differentiation in Frequency Property
      • 3.6.10 Convolution Property
    • 3.7 Initial Value and Final Value Theorem
      • 3.7.1 Initial Value Theorem
      • 3.7.2 Final Value Theorem
    • 3.8 Inverse Z-Transform
      • 3.8.1 Power Series Method (Long-division)
      • 3.8.2 Partial Fraction Method
      • 3.8.3 Residual Method
      • 3.8.4 Convolution of Two Signals
    • 3.9 LTI System Characterized by Linear Constant Coefficient Difference Equation
    • 3.10 Relationship between Z-Transform and Fourier Transform
    • 3.11 Relationship between Z-plane and S-plane
    • Chapter Summary
    • Review Questions
  • CHAPTER 4 INFINITE IMPULSE RESPONSE FILTER
    • 4.1 Introduction
      • 4.1.1 Distinguish between Analog and Digital Filter
    • 4.2 Analog Filters
    • 4.3 Analog Domain to Digital Domain Transformation
      • 4.3.1 Impulse-Invariant Transformation Technique
      • 4.3.2 Bilinear Transformation Technique
      • 4.3.3 Approximation of Derivatives
    • 4.4 Analog Frequency Transformation
      • 4.4.1 Normalized Low Pass Filter to Desired Low Pass Filter Transformation
      • 4.4.2 Normalized Low Pass Filter to Desired High Pass Filter Transformation
      • 4.4.3 Normalized Low Pass Filter to Desired Band Pass Filter Transformation
      • 4.4.4 Normalized Low Pass Filter to Desired
    • 4.5 Digital Frequency Transformation
      • 4.5.1 Low Pass to Desired Low Pass Filter Transformation
      • 4.5.2 Low Pass to Desired High Pass Filter Transformation
      • 4.5.3 Low Pass to Desired Band Pass Filter Transformation
      • 4.5.4 Low Pass to Desired Band Elimination Filter Transformation
    • 4.6 Butterworth Filter
      • 4.6.1 Determination of Order of the Filter
    • 4.7 Chebyshev Filter
      • 4.7.1 Magnitude Response of Type-I Chebyshev Filter
      • 4.7.2 Order of the Filter
      • 4.7.3 Transfei Function of Chebyshev Filter
    • 4.8 Structure Realization—Introduction
    • 4.9 Implementation of Discrete-time System
      • 4.9.1 Structure Realization of IIR System Recursive Structure
      • 4.9.2 System Realization using Direct Form-I
      • 4.9.3 System Realization using Direct Form-II
      • 4.9.4 System Realization using Cascade Form
      • 4.9.5 System Realization using Parallel Form
    • 4.10 Lattice Structure of IIR System
    • 4.11 Lattice Ladder Structure of IIR System
    • Chapter Summary
    • Review Questions
  • CHAPTER 5 FINITE IMPULSE RESPONSE FILTER
    • 5.1 Introduction
    • 5.2 Phase Delay and Group Delay
    • 5.3 Linear Phase Transfer Function
      • 5.3.1 Symmetric: Impulse Response with Odd Length
      • 5.3.2 Symmetric Impulse Response with Even Length
      • 5.3.3 Antisymmetric Impulse Response with Odd Length
      • 5.3.4 Antisymmetric Impulse Response with Even Length
    • 5.4 Design of FIR Filter- Fourier Method
    • 5.5 Design of FIR Filter- Windowing Techniques
    • 5.6 The Triangular Window (Bartlett Window)
    • 5.7 Raised Cosine Window
    • 5.8 Hanning Window
    • 5.9 Hamming Window
    • 5.10 Blackman Window
    • 5.11 Kaiser Window
      • 5.11.1 Kaiser Window Technique for High Pass Filter
      • 5:11.2 Kaiser Window Technique for Band Pass Filter
      • 5.11.3 Kaiser Window Technique for Band Stop Filter
    • 5.12 FIR Filter Design Using Frequency Sampling Technique
    • 5.13 Equiripple Linear-phase FIR Filter
    • 5.14 FIR Differentiators
    • 5.15 Hilbert Transform
      • 5.15.1 FIR vs. IIR
    • 5.16 Structure Realization of FIR System
      • 5.16.1 Direct Form Realization of FIR System
      • 5.16.2 Cascade Structure Realization of FIR System
      • 5.16.3 Realization Structure of Linear Phase FIR System
      • 5.16.4 Lattice Structure for FIR Filter
    • Chapter Summary
    • Review Questions
  • CHAPTER 6 FINITE WORD LENGTH EFFECT
    • 6.1 Introduction
    • 6.2 Fixed-Point Numbers
      • 6.2.1 Sign-magnitude Format
      • 6.2.2 One's-complement Format
      • 6.2.3 Two's-complement Format
      • 6.2.4 Fixed-point Addition
      • 6.2.5 Fixed-point Multiplication
    • 6.3 Floating-Point Numbers
      • 6.3.1 The Floating-point Representation of Integer Part
      • 6.3.2 The Floating-point Representation of Fractional Part
      • 6.3.3 IEEE Single Precision and Double Precision Format
      • 6.3.4 Floating-Point Addition
      • 6.3.5 Floating-point Multiplication
      • 6.3.6 Comparative Study of Fixed-point and Floating-point Representations
      • 6.3.7 Dynamic Range, Resolution and Precision
    • 6.4 Quantization Error
      • 6.4.1 Truncation
      • 6.4.2 Rounding
      • 6.4.3 Quantization Error in Floating-point Representation
      • 6.4.4 Probability Density Function of Quantization Error
    • 6.5 Variance Estimation of Quantization Error
      • 6.5.1 SNR Calculation
      • 6.5.2 Effect of Scaling on SNR
    • 6.6 Finite Word Length Effect on IIR Filter
      • 6.6.1 Finite Word Length Effect on Filter Structure
      • 6.6.2 Effect of Finite Word Length on Direct Form-I Structure
      • 6.6.3 Effect of Finite Word Length on Direct Form-1! Structure
    • 6.7 Product Quantization Error in IIR Filter
    • 6.8 Mathematical Analysis of Steady-state Output Noise
    • 6.9 Dynamic Scaling to Prevent Overflow
      • 6.9.1 Alternate Method
    • 6.10 Limit-cycle Oscillations in Recursive Systems
    • 6.11 Rounding off Error in DFT Computation
    • 6.12 Rounding off Error in FFT Computation
    • Solved University Questions
    • Chapter Summary
    • Review Questions
  • CHAPTER 7 MULTIRATE DIGITAL SIGNAL PROCESSING
    • 7.1 Introduction
    • 7.2 Decimator (Down-sampler)
      • 7.2.1 Spectral Analysis of Decimator
    • 7.3 Interpolation (Up-sampling)
      • 7.3.1 Spectral Analysis of Interpolator
    • 7.4 Sampling Rate
    • 7.5 Noble Identities
      • 7.5.1 Noble Identity for Down-sampler
      • 7.5.2 Noble identity for Up-sampler
    • 7.6 Polyphase Decomposition for Decimator
    • 7.7 Polyphase Decomposition for Interpolator
    • 7.8 Multistage Design of Decimator and Interpolator
      • 7.8.1 Multistage Implementation of Decimator
      • 7.8.2 Multistage Implementation of Interpolator
    • 7.9 Comb Filter
    • 7.10 Applications of Multirate Signal Processing
      • 7.10.1 Phase Shifter
      • 7.10.2 Subband Coding of Speech Signal
      • 7.10.3 Multirate Narrowband Digital Filtering
    • 7.11 Quadrature Mirror Filter
      • 7.11.1 Two-channel QMF Bank
      • 7.11.2 Mathematical Analysis of Two-channel QMF
      • 7.11.3 Alias-free Realization
    • Chapter Summary
    • Review Questions
  • CHAPTER 8 SPEECH PROCESSING AND COMPRESSION
    • 8.1 Introduction
    • 8.2 Production of Speech Waveform
      • 8.2.1 Important Points to Remember
    • 8.3 Speech Recognition
      • 8.3.1 Training Stage
      • 8.3.2 Recognition Stage
    • 8.4 Linear Predictive Coding
      • 8.4.1 Basic Linear Prediction Model
    • 8.5 The Cepstrum: A Method for Speech Analysis
      • 8.5.1 Subband Coding of Speech Signals Vocoder
    • 8.6 Transmultiplexers
      • 8.6.1 FDM to TDM Transmultiplexer
      • 8.6.2 TDM to FDM Transmultiplexer
    • 8.7 Speech Compression
    • 8.8 Coding Technique
    • 8.9 Speech Coding Techniques
    • Chapter Summary
    • Review Questions
  • CHAPTER 9 ADAPTIVE FILTERS
    • 9.1 Introduction
      • 9.1.1 Need for Adaptive Filter
      • 9.1.2 Characteristics of Adaptive Filter
    • 9.2 Performance Measures of Adaptive Filter
    • 9.3 Open-loop and Closed-loop Adaptation
      • 9.3.1 Open-loop Adaptation
      • 9.3.2 Closed-loop Adaptation
      • 9.3.3 Factors that Determining the Choice of Adaptation
    • 9.4 Approaches for Deriving Recursive Algorithms
      • 9.4.1 Stochastic Gradient Approach
      • 9.4.2 Least-squares Estimation
      • 9.4.3 How to Choose an Adaptive Filter
    • 9.5 The Adaptive Linear Combiner
    • 9.6 Gradient Search Methods
      • 9.6.1 Concept of Gradient Search Methods
      • 9.6.2 Gradient Search Algorithm
      • 9.6.3 Newton's Descent Method for Gradient Search
      • 9.6.4 Steepest Descent Method for Gradient Search
    • 9.7 The LMS Algorithm
      • 9.7.1 Convergence of LMS Algorithm
    • 9.8 The RLS Algorithm
      • 9.8.1 Recursive Computation of ф(n) and z(n)
      • 9.8.2 Matrix Inversion Lemma
      • 9.8.3 Time Update for the lap-Weight Vector
      • 9.8.4 Distinguish between RLS and LMS Algorithm
    • 9.9 Application of Adaptive Filter
      • 9.9.1 Adaptive Modeling and System Identification
      • 9.9.2 Inverse Modeling and Adaptive Equalization
      • 9.9.3 Adaptive Prediction
      • 9.9.4 Adaptive Interference Cancellation
      • 9.9.5 Adaptive Sidelobe Cancellation
    • Chapter Summary
    • Review Questions
  • CHAPTER 10 MUSICAL SOUND PROCESSING
    • 10.1 Introduction
    • 10.2 Time Domain Operations
      • 10.2.1 Single Echo Generating Filter
      • 10.2.2 Multiple Echo Generating Filter
      • 10.2.3 Infinite Echo Generating Filter
      • 10.2.4 Reverberation Structure
      • 10.2.5 IIR Echoes Filter with Reverberation
      • 10.2.6 Teeth Filter
      • 10.2.7 Flanging
      • 10.2.8 Chorus Effect
      • 10.2.9 Phasing Effect
    • 10.3 Frequency Domain Operations
      • 10.3.1 First Order Filter
      • 10.3.2 Shelving Filter
      • 10.3.3 Second Order Filter
      • 10.3.4 Equalizer
    • Chapter Summary
    • Review questions
  • CHAPTER 11 IMAGE ENHANCEMENT
    • 11.1 Introduction to Image Processing
    • 11.2 Basic Building Blocks of Image Processing System
      • 11.2.1 Image Acquisition and Formation
      • 11.2.2 Image Enhancement
      • 11.2.3 Image Restoration
      • 11.2.4 Image Compression
      • 11.2.5 Image Morphological Processing
      • 11.2.6 Image Segmentation
      • 11.2.7 Image Representation and Description
      • 11.2.8 Image Recognition
    • 11.3 Image Enhancement Techniques
      • 11.3.1 Basic operations of Image Enhancement Technique
      • 11.3.2 Broad Areas for Analysis of Image Enhancement Techniques
    • 11.4 Spatial-Domain Approach
      • 11.4.1 Gray Level Transformation
      • 11.4.2 Basic Gray Level Transformation Functions
      • 11.4.3 Piecewise Linear Transformation Functions
      • 11.4.4 Histogram Processing
      • 11.4.5 Local Enhancement Techniques
      • 11.4.6 Image Enhancement using Arithmetic/ Logical Operation
      • 11.4.7 Spatial "Filtering
    • 11.5 Frequency-domain Analysis
      • 11.5.1 Lowpass Frequency-domain Filter
      • 11.5.2 Highpass Filter
      • 11.5.3 Hbmomorphic Filter
    • Chapter Summary
    • Review Questions
  • CHAPTER 12 DIGITAL SIGNAL PROCESSOR
    • 12.1 Introduction
    • 12.2 Programmable Digital Signal Processing
    • 12.3 Multiplier Accumulator
      • 12.3.1 Overflow and Underflow in MAC Unit
    • 12.4 Computer Architecture
      • 12.4.1 Von-Neumann Architecture
      • 12.4.2 Harvard Architecture
      • 12.4.3 Modified-Harvard Architecture
    • 12.5 On-chip (Cache) Memory
    • 12.6 Pipelining
    • 12.7 Pipeline Structure Processor
      • 12.7.1 Single Instruction Multiple Data Architecture
      • 12.7.2 Very Long Instruction Word Architecture
      • 12.7.3 Superscalar Processing Architecture
    • 12.8 Computer Configuration
      • 12.8.1 Restricted Instruction Set Computer
      • 12.8.2 Complex Instruction Set Computer
    • 12.9 Addressing Modes
      • 12.9.1 Immediate Addressing
      • 12.9.2 Absolute Addressing
      • 12.9.3 Accumulator Addressing
      • 12.9.4 Direct Addressing
      • 12.9.5 Indirect Addressing
      • 12.9.6 Circular Addressing
      • 12.9.7 Bit Reversal Addressing
      • 12.9.8 Memory-Mapped Register Addressing
      • 12.9.9 Stack Addressing
      • 12.9.10 Replication
    • 12.10 Introduction to Processor
    • 12.11 First Generation TMS320C1X Processor
      • 12.11.1 Architecture
      • 12.11.2 The Arithmetic and Logic Unit
    • 12.12 Second-Generation TMS320C2X Processor
    • 12.13 TMS320C3X Digital Signal Processor
      • 12.13.2 Memory Architecture
    • 12.14 TMS320C4X Digital Signal Processor
      • 12.14.1 Architecture
      • 12.14.2 Memory Architecture
      • 12.14.3 Internal Bus Operation
      • 12.14.4 Communication Port
      • 12.14.5 DMA Coprocessor
    • 12.15 TMS320C5X Digital Signal Processor
      • 12.15.1 Central Processing Unit (CPU)
      • 12.15.2 On-Chip Memory
      • 12.15.3 On-Chip Peripherals
      • 12.15.4 Addressing Modes
      • 12.15.5 TMS320C54XX Instructions
    • 12.16 TMS320C6X Digital Signal Processor
      • 12.16.1 TMS320C64X Processor
      • 12.16.2 TMS320C67X Processor
    • 12.17 Code Composer Studio
      • 12.7.1 What is DSK
      • 12.7.2 Introduction to Code Composer Studio
    • Chapter Summary
    • Review Questions
Biographical note
B.E., M.Tech., Ph.D. Jerusalem College of Engineering Chennai pcmed8@yahoo. comB.E., M.S. Department of Electronics and Communication Engineering Cresent Engineering College Chennai
B.E., M.Tech., Ph.D. Jerusalem College of Engineering Chennai pcmed8@yahoo. comB.E., M.S. Department of Electronics and Communication Engineering Cresent Engineering College Chennai
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