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DSP Quick Revision

Digital Signal Processing (DSP) involves manipulating digital signals through various techniques used in audio, image processing, and communications. Key concepts include types of signals, system properties, convolution, Z-transform, DFT/FFT, sampling theorem, and filter types (FIR and IIR). Understanding these fundamentals is essential for analyzing and implementing DSP systems.

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6 views2 pages

DSP Quick Revision

Digital Signal Processing (DSP) involves manipulating digital signals through various techniques used in audio, image processing, and communications. Key concepts include types of signals, system properties, convolution, Z-transform, DFT/FFT, sampling theorem, and filter types (FIR and IIR). Understanding these fundamentals is essential for analyzing and implementing DSP systems.

Uploaded by

Salman
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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Digital Signal Processing - Quick Revision

1. What is DSP?

DSP stands for Digital Signal Processing. It involves manipulating digital signals using computers or processors. Used in
audio processing, image processing, communications, etc.

2. Signals & Systems Basics

Signal: A function representing data, like x[n] = [1, 2, 3].


System: A process that takes a signal as input and gives output.

3. Types of Signals

- Discrete vs Continuous
- Periodic vs Aperiodic
- Even vs Odd

4. System Properties

- Linearity: Output is proportional to input


- Time-Invariant: Delay in input causes same delay in output
- Causal: Output depends only on present and past inputs
- Stable: Bounded input gives bounded output

5. Convolution

Used to find output of LTI systems: y[n] = x[n] * h[n].


Steps: Flip h[n], shift, multiply, sum.

6. Z-Transform

X(z) = sum over n of x[n] * z^(-n)


Used for analyzing discrete-time systems.
Check ROC for stability: poles must lie inside unit circle.

7. DFT & FFT

DFT: X[k] = sum over n of x[n] * e^(-j*2*pi*k*n/N)


FFT: Fast algorithm to compute DFT

8. Sampling Theorem

To avoid aliasing: Sampling rate >= 2 times the highest frequency component.

9. Filters

- FIR: Finite impulse response, always stable


Digital Signal Processing - Quick Revision

- IIR: Infinite impulse response, may be unstable

10. Realization

Implement filters using block diagrams (Direct Form I/II).

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