UNIT 2: DIGITAL SIGNAL PROCESSING - CORE CONCEPTS & ANALYSIS
1.0 Discrete-Time Signals & Sequences
1.1 Representation of Sequences
A discrete-time signal is represented as a sequence of samples x[n], where n is an integer (time index).
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Graphical: Plot of amplitude vs. integer index
n. -
Mathematical: Functional definition (e.g.,
x[n] = α^n u[n]). -
Tabular: List of values for specific
n(e.g.,x[n] = {2, 1, -1, 0}forn=0,1,2,3).
1.2 Length of a Discrete-Time Signal
For a finite-duration sequence, the length L is the number of non-zero samples.
Example:
x[n] = {1, 2, 3}forn=0,1,2has lengthL = 3.
1.3 Classification of Discrete-Time Signals
| Class | Definition | Example |
|---|---|---|
| Even | x[n] = x[-n] |
x[n] = cos(ωn) |
| Odd | x[n] = -x[-n] |
x[n] = sin(ωn) |
| Periodic | x[n] = x[n+N] for all n, smallest N is period. |
x[n] = e^{jωn} with ω=2πk/N |
| Aperiodic | No finite N satisfies periodicity. |
x[n] = α^n u[n], ` |
| Energy | `E = Σ_{n=-∞}^{∞} | x[n] |
| Power | `P = lim_{N→∞} (1/(2N+1)) Σ_{n=-N}^{N} | x[n] |
1.4 Fundamental Sequence Operations
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Addition:
y[n] = x₁[n] + x₂[n](sample-by-sample). -
Multiplication:
y[n] = x₁[n] * x₂[n]. -
Time-Shifting:
x[n - n₀]→ delay ifn₀>0, advance ifn₀<0. -
Time-Reversal:
x[-n](fold aboutn=0). -
Scaling:
α * x[n](amplitude scaling).
1.5 Nyquist Sampling Theorem
To perfectly reconstruct a band-limited continuous-time signal
x_c(t)with maximum frequencyf_max(Hz) from its samplesx[n] = x_c(nT_s), the sampling frequencyf_s = 1/T_smust satisfy:
$$ f_s \geq 2 f_{\text{max}} $$
The minimum rate f_s = 2f_max is the Nyquist rate. f_Nyquist = 2f_max is the Nyquist frequency. Aliasing occurs if f_s < 2f_max.
2.0 Z-Transform & System Analysis
2.1 Definition
The bilateral (two-sided) Z-transform of a sequence x[n] is:
$$ X(z) = \mathcal{Z}\{x[n]\} = \sum_{n=-\infty}^{\infty} x[n] z^{-n} $$
where z is a complex variable. The unilateral (one-sided) Z-transform uses sum from n=0 to ∞.
2.2 Region of Convergence (ROC)
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Definition: Set of
zvalues for which the Z-transform sum converges (is finite). -
Properties: ROC is an annular region (ring) in the z-plane, bounded by poles. It never contains poles.
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Determination: Find
|z|values that make the infinite sum converge (often using geometric series).
2.3 Causal & Anti-Causal Systems
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Causal System:
x[n] = 0forn < 0. ROC is exterior of the outermost pole (|z| > r_max). -
Anti-Causal System:
x[n] = 0forn > 0. ROC is interior of the innermost pole (|z| < r_min). -
Two-Sided System: ROC is an annular ring between poles.
2.4 Properties of Z-Transform (Causal Signals)
Let x[n] ↔ X(z), ROC: |z| > r₀.
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Linearity:
a x₁[n] + b x₂[n] ↔ a X₁(z) + b X₂(z), ROC ≥ intersection. -
Time-Shifting:
x[n - n₀] ↔ z^{-n₀} X(z), ROC same asX(z). -
Scaling in z:
a^n x[n] ↔ X(z/a), ROC scaled:|z| > |a|r₀. -
Convolution:
x₁[n] * x₂[n] ↔ X₁(z) X₂(z), ROC is intersection of individual ROCs. -
Initial Value Theorem (IVT): For causal
x[n],x[0] = lim_{z→∞} X(z). -
Final Value Theorem (FVT): If poles of
(z-1)X(z)are inside unit circle,lim_{n→∞} x[n] = lim_{z→1} (z-1) X(z).
2.5 Inverse Z-Transform Methods
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Long Division (Power Series): Expand
X(z)asΣ x[k] z^{-k}. ROC determines causality. -
Partial Fraction Expansion: For rational
X(z), decompose into simpler terms and use known pairs. -
Contour Integration (Residue Method):
x[n] = (1/(2πj)) ∮ X(z) z^{n-1} dz.
2.6 Stability Determination
An LTI system is BIBO stable iff its impulse response h[n] is absolutely summable: Σ_{n=-∞}^{∞} |h[n]| < ∞.
Using Z-transform: For causal systems, stability ⇔ ROC includes the unit circle (
|z|=1). For general systems, ROC must include|z|=1.
3.0 Discrete Fourier Transform (DFT)
3.1 Definition of N-point DFT
For a finite-length sequence x[n], 0 ≤ n ≤ N-1:
$$ X[k] = \sum_{n=0}^{N-1} x[n] e^{-j(2π/N)kn} = \sum_{n=0}^{N-1} x[n] W_N^{kn} $$
where W_N = e^{-j(2π/N)} is the twiddle factor. k = 0, 1, ..., N-1.
3.2 DFT as Orthogonal Transform Pair
The DFT and its inverse (IDFT) form an orthogonal transform pair:
$$ x[n] = \frac{1}{N} \sum_{k=0}^{N-1} X[k] e^{j(2π/N)kn} = \frac{1}{N} \sum_{k=0}^{N-1} X[k] W_N^{-kn} $$
The basis vectors {W_N^{kn}} are orthogonal over n=0,...,N-1.
3.3 Computation of DFT for Specific Sequences
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Unit Impulse:
δ[n]→X[k] = 1for allk(DC value only). -
Unit Step (within length):
u[n]for0≤n≤N-1→X[k] = Nfork=0,0fork≠0(after considering periodicity). -
Constant:
x[n]=1→X[0]=N,X[k]=0fork≠0.
3.4 Relationship: DFT vs. DTFT
The Discrete-Time Fourier Transform (DTFT) of x[n] is X(e^{jω}) = Σ_{n=-∞}^{∞} x[n] e^{-jωn}, periodic with 2π.
- DFT samples the DTFT at
Nequally spaced frequencies over[0, 2π):
$$ X[k] = X(e^{jω}) \big|_{\omega = (2πk/N)} $$
- For a finite-length
x[n], the DTFT is continuous and periodic. The DFT provides a discrete-frequency representation.
4.0 Digital Filter Structures & Realizations
4.1 FIR Filters
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Direct Form (Transversal): Implements the difference equation
y[n] = Σ_{k=0}^{M} b_k x[n-k].-
Structure:
Mdelay elements,M+1multipliers,Madders. -
Complexity:
M+1multipliers,Madders.
graph LR X[n] --> B0[ b0 ]; B0 --> Add; Delay1[ z^-1 ] --> B1[ b1 ]; B1 --> Add; Delay2[ z^-1 ] --> BM[ bM ]; BM --> Add; Add --> Y[n]; Delay1 --> Delay2; -
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Cascade Form: Factors transfer function
H(z) = Π H_i(z). EachH_i(z)is a small FIR section (often 1st or 2nd order). Better for fixed-point implementation (less sensitivity to coefficient quantization).
4.2 IIR Filters
Transfer function: H(z) = Σ_{k=0}^{M} b_k z^{-k} / (1 - Σ_{k=1}^{N} a_k z^{-k}).
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Direct Form I: Separately implements numerator (FIR) and denominator (IIR) sections, then adds. Requires
max(M,N)delays. -
Direct Form II: Shares delay elements between numerator and denominator. More efficient.
- Complexity:
M+N+1multipliers,M+Nadders,max(M,N)delays.
Example Realization (Direct Form II): For
H(z) = (b0 + b1 z^{-1} + b2 z^{-2}) / (1 - a1 z^{-1} - a2 z^{-2}):graph LR X[n] --> Add1; Add1 --> Delay1[ z^-1 ]; Delay1 --> Delay2[ z^-1 ]; Delay2 --> Add2; Add2 --> Mult_b2[b2]; Mult_b2 --> Add3; Delay1 --> Mult_b1[b1]; Mult_b1 --> Add3; Add1 --> Mult_b0[b0]; Mult_b0 --> Add3; Add3 --> Y[n]; Add2 --> Mult_a2[-a2]; Mult_a2 --> Add1; Delay2 --> Mult_a1[-a1]; Mult_a1 --> Add1; - Complexity:
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Cascade/Parallel Forms: Similar to FIR, but sections are 1st/2nd order IIR sections (poles/zeros). Cascade is common for stability (each section stable).
4.3 Reduction of Computational Complexity
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Exploit Symmetry: For linear-phase FIR,
h[n] = h[M-n]. Reduces multipliers by ~half. -
Use Efficient Structures: Direct Form II for IIR uses fewer delays than Direct Form I.
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Polyphase Decomposition: For decimation/interpolation, reduces computation rate.
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Example: For
M=4symmetric FIR (h[0]=h[4],h[1]=h[3],h[2]), direct form needs 5 mults. Symmetric form:y[n] = h[2]*(x[n-2]+x[n-2]) + h[1]*(x[n-1]+x[n-3]) + h[0]*(x[n]+x[n-4])→ 3 multipliers.
4.4 Realization Example (Given IIR)
Given: H(z) = (0.44z^2 + 0.362z + 0.02) / (z^2 + 0.4z + 0.18z - 0.2) → Correct denominator: likely z^2 + 0.4z + 0.18? Assuming H(z) = (0.44z^2 + 0.362z + 0.02) / (z^2 + 0.4z + 0.18).
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Direct Form II:
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Write in negative powers:
H(z) = (0.44 + 0.362 z^{-1} + 0.02 z^{-2}) / (1 + 0.4 z^{-1} + 0.18 z^{-2}). -
Difference equation:
y[n] = -0.4 y[n-1] - 0.18 y[n-2] + 0.44 x[n] + 0.362 x[n-1] + 0.02 x[n-2]. -
Draw structure with 2 delays (shared), 5 multipliers (
-0.4, -0.18, 0.44, 0.362, 0.02), 2 adders.
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5.0 Random Signals & Optimal Filtering
5.1 Classification of Stochastic Processes
| Property | Strict-Sense Stationary (SSS) | Wide-Sense Stationary (WSS) |
|---|---|---|
| Definition | All statistics (all PDFs) invariant to time shift. | 1. Mean constant: E[x[n]] = μ_x (independent of n).<br>2. Autocorrelation depends only on lag k: R_x[m,n] = R_x[m-n] = R_x[k]. |
| Implication | SSS ⇒ WSS. WSS ⇏ SSS. | Weaker condition, sufficient for many linear systems analyses. |
| Example | Gaussian process with constant mean & autocovariance. | Any process with constant mean & autocorrelation function. |
5.2 Optimal Filtering (Wiener Filter)
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Concept: Design a filter
H(z)to estimate a desired signald[n]from an observed signalx[n], minimizing the mean square error (MSE)E\{|e[n]|^2\}, wheree[n] = d[n] - ŷ[n]. -
Wiener-Hopf Equations: For FIR Wiener filter of length
M:
$$ \mathbf{R}_x \mathbf{h} = \mathbf{p}_{xd} $$
where `R_x` is the Toeplitz autocorrelation matrix of `x[n]`, `p_xd` is the cross-correlation vector between `x[n]` and `d[n]`, `h` is the optimal coefficient vector.
- Illustrative Example (Simplest Case): Estimate
d[n] = x[n-1]fromx[n]corrupted by noisev[n](WSS, uncorrelated withx). Optimal FIR (1-tap) Wiener filter:h_opt = R_x[1] / (R_x[0] + R_v[0]).
6.0 Digital Filter Design Techniques
6.1 Window Method for FIR Design
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Principle:
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Start with an ideal frequency response
H_d(e^{jω})(e.g., ideal low-pass with cutoffω_c). -
Find its infinite-duration impulse response
h_d[n]via IDFT (often sinc function). -
Truncate
h_d[n]to finite lengthMusing a windoww[n]:h[n] = h_d[n] w[n],0 ≤ n ≤ M-1.
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Common Windows & Gibbs Phenomenon:
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Rectangular: Simple truncation. Highest sidelobe (~-13 dB), slow roll-off. Severe Gibbs (ringing).
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Hamming:
w[n] = 0.54 - 0.46 cos(2πn/(M-1)). Sidelobe ~-41 dB, better stopband attenuation, wider transition band. -
Hanning:
w[n] = 0.5 - 0.5 cos(2πn/(M-1)). Sidelobe ~-31 dB.
Gibbs Phenomenon: Undershoot/overshoot near discontinuities in
H_d(e^{jω})due to truncation. Window choice trades transition width for stopband attenuation. -
6.2 Parks-McClellan (Equiripple) Algorithm
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Principle: Minimax optimization. Designs FIR filter that minimizes the maximum error (ripple) in both passband and stopband simultaneously, subject to specified ripple magnitudes
δ_p, δ_s. -
Process: Uses Remez exchange algorithm. Alternation theorem: optimal error alternates between
±δat(L+2)frequencies (forL-th order filter). -
Comparison with Window Method:
| Aspect | Window Method | Parks-McClellan | | :--- | :--- | :--- | | Design Control | Indirect (via window &
M). Transition width fixed byM. | Direct: specifyω_p, ω_s, δ_p, δ_s. | | Optimality | Suboptimal. | Optimal (minimizes maximum ripple). | | Ripple | Passband/stopband ripples not independently controllable. | Equiripple in both bands (by design). | | Transition | Wider for low sidelobes. | Narrowest possible for givenδ_p, δ_s. | | Complexity | Simple, analytical. | Iterative, requires software (MATLABfirpm). |
7.0 Applications of DSP
Broad application areas include:
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Speech/Audio Processing: Compression (MP3), noise reduction, recognition.
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Communications: Modulation/demodulation, equalization, channel coding.
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Image/Video Processing: Compression (JPEG, MPEG), enhancement, object detection.
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Biomedical: ECG/EEG analysis, MRI, hearing aids.
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Radar/Sonar: Target detection, tracking, beamforming.
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Control Systems: Digital controllers, system identification.
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Consumer Electronics: Mobile phones, digital cameras, music players.
[!TIP]
Exam Focus: Be prepared to derive Z-transform of finite sequences, determine ROC and deduce causality/stability, compute N-point DFT for simple sequences (impulse, step), draw Direct Form II for given IIR
H(z), and differentiate SSS vs WSS with definitions. For design questions, compare Window vs Parks-McClellan in terms of optimality and control. Always state assumptions (e.g., causal signal for IVT/FVT).