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DSP Solved Questions

This document provides a comprehensive overview of Digital Signal Processing (DSP), covering fundamental concepts such as discrete-time signals, sampling theorems, LTI systems, and signal classifications. It includes step-by-step solutions to key questions, mathematical representations of signals, and the architecture of a DSP system. The content is exam-oriented, featuring equations and diagrams to aid understanding.

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0% found this document useful (0 votes)
3 views8 pages

DSP Solved Questions

This document provides a comprehensive overview of Digital Signal Processing (DSP), covering fundamental concepts such as discrete-time signals, sampling theorems, LTI systems, and signal classifications. It includes step-by-step solutions to key questions, mathematical representations of signals, and the architecture of a DSP system. The content is exam-oriented, featuring equations and diagrams to aid understanding.

Uploaded by

engrhassan1234
Copyright
© All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

Digital Signal Processing

Step-by-Step Solutions to Ten Fundamental Questions

Topics covered Signals, sampling, periodicity, energy/power, LTI systems, causality, signal
categories, and DSP architecture

Notation Continuous-time: x(t) | Discrete-time: x[n]

Prepared as Exam-oriented solved notes with equations and diagrams

Key conventions
u denotes a step function, δ denotes an impulse/unit sample, and n is an integer sample index.

Digital Signal Processing - Solved Questions Page 1


1. Discrete-Time Signal
Definition. A discrete-time signal is a sequence of numbers defined only at discrete instants,
usually integer sample indices. It is written as x[n], where n ∈ Z.

x[n] = x(nTs), n = ..., -2, -1, 0, 1, 2, ...

Example. Let x[n] = sin(0.25πn). Values exist only at n = 0, 1, 2, ... . In contrast, the
continuous-time signal x(t) = sin(0.25πt) is defined for every real value of t.

Main difference: x(t) exists at every instant, whereas x[n] exists only at
selected sample instants.

2. Mathematical and Graphical Representations of Common


Signals
Signal Mathematical representation Interpretation

Unit step u(t) = 0 for t < 0; 1 for t ≥ 0 Switches from 0 to 1 at the origin.

General step s(t) = A u(t - t₀) Switches from 0 to amplitude A at t =


t₀.

Continuous impulse x(t) = Aδ(t - t₀), ∫x(t)dt = A Zero everywhere except t₀ in the
generalized sense; total area is A.

Discrete unit δ[n] = 1 for n = 0; 0 otherwise A single nonzero sample at n = 0.


impulse

Triangular tri(t/T) = 1 - |t|/T for |t| ≤ T; 0 otherwise Rises linearly to a peak and falls
linearly.

Sawtooth x(t) = (A/T)t, 0 ≤ t < T; x(t+T)=x(t) Linear ramp that resets every period
T.

Digital Signal Processing - Solved Questions Page 2


Note: A continuous-time impulse is represented by an arrow because its ideal amplitude is not
finite; its defining quantity is area. The discrete unit impulse is an ordinary sequence with one
sample equal to 1.

Digital Signal Processing - Solved Questions Page 3


3. Nyquist Sampling Theorem and Required Sampling Rate
Nyquist sampling theorem. A band-limited continuous-time signal whose highest frequency
component is fmax can be reconstructed from its samples without aliasing when the sampling
frequency satisfies:

fs ≥ 2 fmax

Step 1: Maximum signal frequency:


fmax = 5 kHz

Step 2: Apply the Nyquist condition:


fs,min = 2(5 kHz) = 10 kHz

Answer: The theoretical minimum sampling rate is 10 kHz. In practical


systems, a rate slightly greater than 10 kHz is normally selected to allow a
realizable anti-aliasing filter.

4. Unit Impulse and Unit Step Sequences


Unit impulse (unit sample) sequence:

δ[n] = { 1, n = 0 ; 0, n ≠ 0 }

Unit step sequence:

u[n] = { 1, n ≥ 0 ; 0, n < 0 }

Important relationship:
δ[n] = u[n] - u[n - 1] and u[n] = Σk=-∞n δ[k]

5. Periodic Discrete-Time Signal


Definition. A discrete-time signal x[n] is periodic if there exists a positive integer N such that:

x[n + N] = x[n] for every integer n

The smallest positive integer satisfying this condition is the fundamental period N₀.

Given: x[n] = cos(0.2πn)


Step 1: Express the angular frequency.
ω₀ = 0.2π = π/5 rad/sample

Step 2: Use the discrete-time periodicity condition. A sinusoid is periodic only if ω₀N = 2πk for
some integers N > 0 and k.

(π/5)N = 2πk ⇒ N = 10k

Step 3: Select the smallest positive integer by taking k = 1.

N₀ = 10

Verification: x[n+10] = cos(0.2πn + 2π) = cos(0.2πn) = x[n].

Digital Signal Processing - Solved Questions Page 4


Digital Signal Processing - Solved Questions Page 5
6. Energy Signal and Power Signal
For a discrete-time signal x[n], the total energy and average power are defined as:

Energy: E = Σn=-∞∞ |x[n]|²

Average power: P = limN→∞ [1/(2N+1)] Σn=-NN |x[n]|²

Type Condition Typical behavior Example

Energy signal 0 < E < ∞ and P = 0 Usually decays with time or x[n] = (1/2)ⁿu[n]. Here E =
has finite duration. Σ(1/4)ⁿ = 4/3.

Power signal E = ∞ and 0 < P < ∞ Usually periodic or constant x[n] = 1. Here P = 1 and E =
and continues indefinitely. ∞.

Important: A nonzero signal cannot normally be both an energy signal and a power signal. Some
signals, such as x[n] = n, are neither because both energy and average power are infinite.

7. Linear Time-Invariant (LTI) System


A system T{·} is linear when it satisfies superposition:

T{a x₁[n] + b x₂[n]} = aT{x₁[n]} + bT{x₂[n]}

It is time-invariant when shifting the input causes exactly the same shift in the output. If y[n]
= T{x[n]}, then:

T{x[n - n₀]} = y[n - n₀]

A system satisfying both conditions is called an LTI system.

Why the LTI property is important:


• An LTI system is completely characterized by its impulse response h[n].

• The output is found by convolution: y[n] = x[n] * h[n].

• Complex inputs can be decomposed into simpler components and analyzed using
superposition.

• In the frequency domain, convolution becomes multiplication: Y(e^{jω}) =


X(e^{jω})H(e^{jω}).

• Many filters, communication channels, and physical systems are accurately approximated
as LTI systems.

Example: y[n] = x[n] + 0.5x[n-1] is an LTI system.

Digital Signal Processing - Solved Questions Page 6


8. Causal and Non-Causal Systems
Property Causal system Non-causal system

Dependence Output at n depends only on present Output at n depends on at least one


and past input samples. future input sample.

Real-time Can be implemented in real time. Cannot be implemented in real time


implementation unless future data are already stored.

Example y[n] = x[n] + x[n-1] y[n] = x[n+1]

Reason Uses samples at n and n-1 only. Requires the future sample at n+1.

9. Analog, Digital, Discrete, and Continuous Signals


The terms describe two different issues: time axis and amplitude values.
“Continuous/discrete” primarily describes the time variable, while “analog/digital” usually
describes how the information is represented.

Signal type Time variable Amplitude Typical notation /


example

Continuous-time Defined at every real time May be continuous or, in x(t), e.g., temperature versus
t. unusual models, discrete. time.

Discrete-time Defined only at selected May still take any real x[n], e.g., exact samples of a
instants n. amplitude. sensor voltage.

Analog Usually continuous in time Continuous range of Microphone voltage before


and amplitude. values. digitization.

Digital Discrete in time and Finite or countable 16-bit PCM audio samples.
quantized in amplitude; amplitude levels.
encoded by symbols/bits.

Key distinction: Every ordinary digital signal is discrete-time, but a


discrete-time signal is not necessarily digital until its amplitude has also been
quantized and encoded.

Digital Signal Processing - Solved Questions Page 7


10. Basic Elements of a Digital Signal Processing System
A complete DSP system converts an analog input into digital samples, processes those
samples, and—when an analog output is required—converts the result back to continuous
time.

Block Function

Analog input / transducer Produces a continuous-time electrical signal from sound, temperature,
pressure, light, etc.

Anti-aliasing low-pass filter Removes frequency components above the chosen Nyquist limit before
sampling.

Sample-and-hold Takes samples and holds each value long enough for accurate conversion.

ADC Performs sampling, quantization, and binary encoding.

Digital signal processor Executes algorithms such as filtering, compression, detection, equalization,
or control.

DAC Converts numerical samples into a staircase-like analog waveform.

Reconstruction low-pass filter Smooths the DAC output and removes high-frequency spectral images.

Analog output / actuator Delivers the processed physical output, such as sound from a loudspeaker.

Compact conversion chain: Analog signal → anti-aliasing filter → sampling/ADC


→ digital processing → DAC → reconstruction filter → analog signal.

End of solved notes

Digital Signal Processing - Solved Questions Page 8

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