EC3492 DIGITAL SIGNAL PROCESSING L T P C

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COURSE OBJECTIVES:
● To learn discrete fourier transform, properties of DFT and its application to linear filtering
● To understand the characteristics of digital filters, design digital IIR and FIR filters and apply

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these filters to filter undesirable signals in various frequency bands
● To understand the effects of finite precision representation on digital filters
● To understand the fundamental concepts of multi rate signal processing and its applications
● To introduce the concepts of adaptive filters and its application to communication engineering

UNIT I DISCRETE FOURIER TRANSFORM 9

Sampling Theorem, concept of frequency in discrete-time signals, summary of analysis & synthesis
equations for FT & DTFT, frequency domain sampling, Discrete Fourier transform (DFT) - deriving
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DFT from DTFT, properties of DFT - periodicity, symmetry, circular convolution. Linear filtering using
DFT. Filtering long data sequences - overlap save and overlap add method. Fast computation of
DFT - Radix-2 Decimation-in-time (DIT) Fast Fourier transform (FFT), Decimation-in-frequency (DIF)
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Fast Fourier transform (FFT). Linear filtering using FFT.

UNIT II INFINITE IMPULSE RESPONSE FILTERS 9

Characteristics of practical frequency selective filters. characteristics of commonly used analog filters
- Butterworth filters, Chebyshev filters. Design of IIR filters from analog filters (LPF, HPF, BPF, BRF)
- Approximation of derivatives, Impulse invariance method, Bilinear transformation. Frequency
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transformation in the analog domain. Structure of IIR filter - direct form I, direct form II, Cascade,
parallel realizations.

UNIT III FINITE IMPULSE RESPONSE FILTERS 9

Design of FIR filters - symmetric and Anti-symmetric FIR filters - design of linear phase FIR filters
using Fourier series method - FIR filter design using windows (Rectangular, Hamming and Hanning
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window), Frequency sampling method. FIR filter structures - linear phase structure, direct form
realizations
 UNIT IV FINITE WORD LENGTH EFFECTS 9
Fixed point and floating point number representation - ADC - quantization - truncation and rounding
- quantization noise - input / output quantization - coefficient quantization error - product quantization
error - overflow error - limit cycle oscillations due to product quantization and summation - scaling to
prevent overflow.

UNIT V DSP APPLICATIONS 9

Multirate signal processing: Decimation, Interpolation, Sampling rate conversion by a rational factor
– Adaptive Filters: Introduction, Applications of adaptive filtering to equalization-DSP Architecture-

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Fixed and Floating point architecture principles
45 PERIODS
PRACTICAL EXERCISES:

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30 PERIODS
MATLAB / EQUIVALENT SOFTWARE PACKAGE/ DSP PROCESSOR BASED
IMPLEMENTATION
1. Generation of elementary Discrete-Time sequences
2. Linear and Circular convolutions
3. Auto correlation and Cross Correlation
4. Frequency Analysis using DFT
5. Design of FIR filters (LPF/HPF/BPF/BSF) and demonstrates the filtering operation
6. Design of Butterworth and Chebyshev IIR filters (LPF/HPF/BPF/BSF) and demonstrate the
filtering operations

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7. Study of architecture of Digital Signal Processor
8. Perform MAC operation using various addressing modes
9. Generation of various signals and random noise
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10. Design and demonstration of FIR Filter for Low pass, High pass, Band pass and Band
stop filtering
11. Design and demonstration of Butter worth and Chebyshev IIR Filters for Low pass,
High pass, Band pass and Band stop filtering
12. Implement an Up-sampling and Down-sampling operation in DSP Processor

COURSE OUTCOMES:
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At the end of the course students will be able to:

CO1:Apply DFT for the analysis of digital signals and systems
CO2:Design IIR and FIR filters
CO3: Characterize the effects of finite precision representation on digital filters
CO4:Design multirate filters
CO5:Apply adaptive filters appropriately in communication systems
TOTAL:75 PERIODS
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TEXT BOOKS:
1. 1.John G. Proakis and Dimitris G.Manolakis, Digital Signal Processing – Principles,
Algorithms and Applications, Fourth Edition, Pearson Education / Prentice Hall, 2007.
2. 2.A. V. Oppenheim, R.W. Schafer and J.R. Buck, ―Discrete-Time Signal Processingǁ, 8th
Indian Reprint, Pearson, 2004.

REFERENCES
1. Emmanuel C. Ifeachor& Barrie. W. Jervis, “Digital Signal Processing”, Second Edition,
Pearson Education / Prentice Hall, 2002.
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 2. 2.Sanjit K. Mitra, “Digital Signal Processing – A Computer Based Approach”, Tata Mc Graw
Hill, 2007.
3. 3.Andreas Antoniou, “Digital Signal Processing”, Tata Mc Graw Hill, 2006.

CO PO1 PO2 PO3 PO4 PO5 PO6 PO7 PO8 PO9 PO10 PO11 PO12 PSO1 PSO2 PSO3

1 3 3 3 3 2 2 - - - - 2 2 3 3 2

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2 3 3 3 3 2 2 - - - - 2 2 2 2 2

3 3 3 3 3 2 2 - - - - 2 2 1 2 3

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4 3 3 3 2 3 2 - - - - 1 2 2 1 2

5 3 2 2 2 3 2 - - - - 1 2 2 2 1

CO 3 3 3 3 2 2 - - - - 2 2 2 2 2

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