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Introduction to Digital Signal Processing: Discrete-time signals & sequences, Linear shift-invariant systems, Stability, and Causality, Linear constant coefficient difference equations, the Frequency domain representation of discrete-time signals and systems.
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List of Reference Books for Digital Signal Processing
- E.C. Ifeachor and B.W. Jervis, ” Digital signal processing – A practical approach”, Second edition, Pearson, 2002.
- S.K. Mitra, Digital Signal Processing, A Computer Based approach, Tata Mc GrawHill, 1998.
- P.P.Vaidyanathan, Multirate Systems & Filter Banks, Prentice-Hall, Englewood Cliffs, NJ, 1993.
- Johny R. Johnson, Introduction to Digital Signal Processing, PHI, 2006.
- John G Proakis and Manolakis, “Digital Signal Processing Principles, Algorithms and Applications”, Pearson, Fourth Edition, 2007.
- S.Salivahanan, A. Vallavaraj, C. Gnanapriya, Digital Signal Processing, TMH/McGraw HillInternational, 2007
Digital Signal Processing Syllabus has been provided below – B.Tech Elective Subject
OBJECTIVES:
• To study DFT and its computation
• To study the design techniques for digital filters
• Study the finite word length effects in signal processing
• To study the non-parametric methods of power spectrum estimations
• To study the fundamentals of digital signal processors.
UNIT -I: Discrete Fourier Transform
DFT and its properties, Relation between DTFT and DFT, FFT computations using Decimation in time and Decimation in frequency algorithms, Overlap-add and save methods
UNIT -II
Infinite Impulse Response Digital Filters Review of design of analog Butterworth and Chebyshev filters, Frequency transformation in the analog domain – Design of IIR digital filters using impulse invariance technique – Design of digital filters using bilinear transform – pre warping – Realization using direct, cascade and parallel forms.
UNIT- III
Finite Impulse Response Digital Filters Symmetric and Ant symmetric FIR filters – Linear phase FIR filters – Design using Hamming, Henning and Blackman Windows – Frequency sampling method – Realization of FIR filters – Transversal, Linear phase and Polyphasestructures.
UNIT -IV: Finite Word Length Effects
Fixed point and floating-point number representations – Comparison – Truncation and Rounding errors – Quantization noise – derivation for quantization noise power – coefficient quantization error – Product quantization error –
UNIT -V
Overflow error – Roundoff noise power – limit cycle oscillations due to product round off and overflow errors – signal scaling
UNIT -VI: Multirate Signal Processing
Introduction to Multirate signal processing-Decimation Interpolation-Polyphase implementation of FIR filters for interpolator and decimator -Multistage implementation of sampling rate conversion- Design of narrowband filters – Applications of Multirate signal processing.
Why you should study Digital Signal Processing –
- For getting the ability to apply the knowledge of Maths, Science as well as engineering
- To achieve the ability which helps to design, experiment and interpret data.
- To be able to design a system, component or a process which is focused on meeting desired needs, not on its own, but with its realistic constraints which would be political, economic, environmental, social, ethical, manufacturability, sustainability, health and safety.
- To develop the ability to work as a part of a multi – disciplinary team.
Important Questions in Digital Signal Processing
- Compute the DFT for the sequence {1, 2, 0, 0, 0, 2, 1, 1}. Using radix -2 DIF FFT and radix -2 DIT- FFT algorithm.
- Derive the equation to implement a butterfly structure In the DIT FFT algorithm.
- State and prove the time convolution property of Z-Transforms.
- Obtain the impulse response of the digital filter to correspond to an analog filter with impulse response ha(t) = 0.5 e-2t and with a sampling rate of 1.0kHz using impulse invariant method.
- Compare bilinear transformation and impulse invariant mapping.
- Explain the decimation and interpolation process with an example. Also, find the spectrum.
- The sequence x(n)=[0,2,4,6,8] is interpolated using interpolation sequence bk =[1/2,1,1/2] and the interpolation factor is 2.find the interpolated sequence y(m).
- Describe the multiplier/adder unit of the TMS320c54xx processor with a neat block diagram.
- What are interrupts? What are the classes of interrupts available in the TMS320C5xx processor?