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IT-404 · Analog & Digital Communication/Quick Revision Short Notes

Analog & Digital Communication (IT-404) - Unit 3 Short Notes

UNIT 3: ANALOG & DIGITAL COMMUNICATION – EXAM-DRIVEN SHORT NOTES


I. SIGNALS AND SYSTEMS FUNDAMENTALS

A. Signal Classification

Category Definition Example
Continuous-time Defined for all real values of t. Analog audio signal
Discrete-time Defined only at discrete instants (integer n). Sampled signal
Deterministic Completely predictable for all t. x(t) = sin(2πft)
Random (Non-deterministic) Unpredictable; described by statistical properties. Thermal noise
Periodic x(t+T) = x(t) for some T>0 (fundamental period). cos(2πt)
Aperiodic (Non-periodic) No T exists satisfying periodicity. Rectangular pulse rect(t/τ)
Energy Signal Finite total energy: `E = ∫ x(t)
Power Signal Finite average power: `P = lim(T→∞) (1/2T) ∫ x(t)
Even x(t) = x(-t) (symmetric about y-axis). cos(t)
Odd x(t) = -x(-t) (anti-symmetric). sin(t)

[!TIP] Exam Focus: Rectangular pulse x(t) = A for |t| ≤ τ/2, 0 otherwise is an energy signal (finite duration). A periodic square wave is a power signal.

B. System Classification

Property Linear Non-linear
Definition Superposition holds: T{a*x1 + b*x2} = a*y1 + b*y2 Violates superposition.
Example y(t) = 2x(t), LTI system with h(t) y(t) = x²(t), y(t) = sin(x(t))
Property Time-Invariant (TI) Time-Variant (TV)
----------------------- ------------------------------------------------- -----------------------------------------
Definition T{x(t-t₀)} = y(t-t₀) Output shift depends on input shift.
Example y(t) = x(2t) (TV), y(t) = ∫x(τ)dτ (TI) y(t) = t*x(t)
Property Causal Non-causal
----------------------- ------------------------------------------------- -----------------------------------------
Definition Output depends only on present/past inputs. Output depends on future inputs.
Example y(t) = x(t-1), y(t) = ∫_{-∞}^t x(τ)dτ y(t) = x(t+1), y(t) = ∫_{t}^{∞} x(τ)dτ
Property Stable (BIBO) Unstable
----------------------- ------------------------------------------------- -----------------------------------------
Definition Bounded input → bounded output. Small bounded input can cause unbounded output.
Test `∫ h(t)
Example h(t) = e^{-t}u(t) (stable) h(t) = e^{t}u(t) (unstable)

C. Standard Signals & Operations

  • Unit Impulse (Dirac Delta) δ(t):

    • δ(t) = 0 for t ≠ 0, ∫_{-∞}^{∞} δ(t) dt = 1.

    • Sifting Property: ∫_{-∞}^{∞} x(t)δ(t-t₀) dt = x(t₀).

  • Unit Step u(t): u(t) = 1 for t≥0, 0 for t<0. u(t) = ∫_{-∞}^{t} δ(τ) dτ.

  • Rectangular Pulse (Gate): rect(t/τ) = 1 for |t| ≤ τ/2, 0 otherwise.

  • Triangular Pulse: Λ(t/τ) = 1 - |t|/τ for |t| ≤ τ, 0 otherwise.

  • Signum (sgn) function: sgn(t) = 1 for t>0, 0 for t=0, -1 for t<0.

  • Signal Operations:

    • Time-shifting: x(t-t₀) → delay by t₀.

    • Time-scaling: x(at) → a>1 (compression), 0<a<1 (expansion).

    • Amplitude-scaling: A*x(t).

[!TIP] Common Pitfall: x(2t+3) is shift then scale: First shift left by 3 (x(t+3)), then compress by 2 (x(2(t+3)) = x(2t+6)). Alternatively, factor: x(2(t+1.5)) → scale by 2, then shift left by 1.5.


II. FOURIER ANALYSIS

A. Continuous-Time Fourier Transform (CTFT)

Definition: X(ω) = ∫_{-∞}^{∞} x(t) e^{-jωt} dt (Frequency domain representation). Inverse: x(t) = (1/(2π)) ∫_{-∞}^{∞} X(ω) e^{jωt} dω.

Key Fourier Transform Pairs:

  1. δ(t) ↔ 1

  2. u(t) ↔ πδ(ω) + 1/(jω) (Using u(t) = (1/2)(1+sgn(t)))

  3. rect(t/τ) ↔ τ sinc(ωτ/2π) where sinc(x) = sin(πx)/(πx).

  4. e^{-at}u(t) (a>0) ↔ 1/(a+jω)

B. Fourier Transform Properties

Property Time Domain Frequency Domain Equation
Linearity a*x1(t) + b*x2(t) a*X1(ω) + b*X2(ω) T{a*x1+b*x2} = a*T{x1}+b*T{x2}
Time-shifting x(t-t₀) X(ω) e^{-jωt₀}
Frequency-shifting x(t) e^{jω₀t} X(ω-ω₀)
Time-scaling x(at) `(1/ a
Duality (Symmetry) X(t) 2π x(-ω) If x(t) ↔ X(ω), then X(t) ↔ 2π x(-ω)
Convolution x1(t) * x2(t) X1(ω) X2(ω) * denotes convolution.
Parseval's Theorem `∫ x(t) ² dt`

[!TIP] Exam Proof: Time-scaling property: Y(ω) = ∫ x(at) e^{-jωt} dt. Let τ = at, dt = dτ/a. Then Y(ω) = (1/|a|) ∫ x(τ) e^{-j(ω/a)τ} dτ = (1/|a|) X(ω/a).

C. Application to LTI Systems

  • Frequency Response: H(ω) = ∫_{-∞}^{∞} h(t) e^{-jωt} dt (Fourier Transform of impulse response h(t)).

  • System Output: Y(ω) = H(ω) X(ω).

  • Example: For h(t) = e^{-2t}u(t), H(ω) = ∫_{0}^{∞} e^{-2t} e^{-jωt} dt = 1/(2+jω).

    • Magnitude: |H(ω)| = 1/√(4+ω²), Phase: ∠H(ω) = -tan⁻¹(ω/2).

    • Plot: Low-pass filter (magnitude decreases as |ω| increases).


III. ANALOG MODULATION TECHNIQUES

A. Amplitude Modulation (AM)

  1. Standard AM (DSB-FC):

    • Expression: s_AM(t) = A_c[1 + m_a cos(ω_m t)] cos(ω_c t), where m_a is modulation index (0 ≤ m_a ≤ 1 for linear modulation).

    • Spectrum: Carrier at ω_c, Upper Sideband (USB) at ω_c+ω_m, Lower Sideband (LSB) at ω_c-ω_m.

    • Power Relations:

      • Carrier Power: P_c = A_c²/2

      • Total Power: P_T = P_c (1 + m_a²/2)

      • Sideband Power: P_sb = P_c (m_a²/2)

      • Transmission Efficiency: η = P_sb / P_T = (m_a²/2) / (1 + m_a²/2). \boxed{\eta = \frac{m_a^2}{2 + m_a^2}}

    • Over-modulation (m_a > 1): Envelope distortion, carrier phase reversal.

  2. Generation:

    • Square-law Modulator: Uses nonlinear device (diode) with i = a*v + b*v². Produces v²(t) term containing cos(ω_c t) and cos(2ω_c t). Filter out 2ω_c term to get DSB-SC. Limitation: Low power, harmonic distortion.

    • Switching Modulator (Ring Modulator): Uses diodes as switches driven by carrier. Output is product m(t)*c(t) (DSB-SC). Efficient, no carrier leakage.

  3. Suppressed Carrier Variants:

    • DSB-SC: s(t) = m(t) cos(ω_c t). Bandwidth = 2f_m. Requires coherent detection (phase sync).

    • SSB-SC: Transmits only one sideband (USB or LSB). Bandwidth = f_m, Power savings (no carrier, one sideband).

      • Generation:

        • Filter Method: Generate DSB-SC, then use sharp bandpass filter.

        • Phasing Method (Hilbert Transform): s(t) = m(t)cos(ω_c t) ∓ ĥm(t)sin(ω_c t), where ĥm(t) is Hilbert transform (90° phase-shifted) of m(t).

    • VSB-SC: Transmits one full sideband + vestige of other. Used in TV broadcasting (to preserve low-frequency video components). Filter has a gradual transition band.

B. Angle Modulation (FM & PM)

  1. Fundamentals:

    • General Expression: s(t) = A_c cos[ω_c t + φ(t)].

    • Frequency Modulation (FM): φ(t) = k_f ∫ m(τ) dτ. Instantaneous frequency: ω_i(t) = ω_c + k_f m(t).

      • Modulation Index (β): β = Δf / f_m = (k_f A_m)/ω_m. Δf is peak frequency deviation.

      • Single-tone FM: s(t) = A_c cos(ω_c t + β sin ω_m t).

    • Phase Modulation (PM): φ(t) = k_p m(t). β = k_p A_m.

    • Relationship: FM is the integral of PM. For single-tone, β_FM = β_PM / (ω_m).

    • Deviation Ratio (for wideband FM): D = Δf / f_{max} (where f_{max} is max modulating frequency).

  2. Generation:

    • Direct Method: VCO (Voltage Controlled Oscillator) where ω_out ∝ v_in. Armstrong Method: Phase modulator + differentiator → FM.

    • Indirect Method: Direct PM + frequency multiplication (to increase Δf).

  3. Demodulation:

    • Ratio Detector: Uses a transformer and diodes. Output across R1 (no carrier recovery needed). Advantage: Less sensitive to amplitude variations.

    • Foster-Seeley Discriminator: Uses a tuned transformer (primary & secondary 90° out of phase). Output across R is proportional to dφ/dt (frequency). Requires carrier for operation.

    • Balanced Slope Detector: Two slope detectors in balance. Simple but poor linearity.

  4. Pre-emphasis & De-emphasis:

    • Need: FM has better SNR for high-frequency components? Actually, noise power increases with frequency. Pre-emphasis boosts high-frequency modulating signal before transmission to improve post-demodulation SNR.

    • Transfer Functions:

      • Pre-emphasis (High-pass): H_{pre}(f) = 1 + j(f/f_x) (simple RC differentiator).

      • De-emphasis (Low-pass): H_{de}(f) = 1 / (1 + j(f/f_x)) (RC integrator).

    • f_x is the emphasis frequency (typically 2.12 kHz or 75 µs time constant).

[!TIP] Key Difference: Ratio Detector does not require a separate carrier recovery circuit (uses the transmitted carrier), while Foster-Seeley does.


IV. RADIO RECEIVERS

A. Superheterodyne Receiver

Block Diagram: RF Amp → Mixer (with LO) → IF Amp → Detector → Audio Amp.

  • RF Amplifier: Selects desired station, provides initial gain.

  • Mixer & Local Oscillator: Heterodynes RF to fixed Intermediate Frequency (IF). f_IF = |f_RF - f_LO|.

  • IF Amplifier: Provides most gain and selectivity (using fixed-tuned filters).

  • Detector: Demodulates IF signal (AM envelope detector, FM discriminator).

  • Audio Amplifier: Boosts audio signal for speaker.

Image Frequency: f_image = f_LO + f_IF (for high-side injection) or f_LO - f_IF (low-side). An undesired signal at f_image also produces f_IF output. Rejection achieved by RF front-end filtering.

B. TRF (Tuned Radio Frequency) Receiver

Block Diagram: RF Amp (tuned) → Detector → Audio Amp. All stages tuned to f_c.

  • Operation: Each RF stage is tuned to the carrier frequency.

  • Limitations vs. Superhet:

    1. Poor Selectivity: Requires multiple high-Q RF stages, difficult to achieve constant bandwidth over tuning range.

    2. Poor Sensitivity: Gains at RF are low; high gain at RF causes instability (oscillations).

    3. Instability: High gain at high frequencies prone to feedback oscillations.

    4. No Image Rejection: No frequency conversion to a fixed IF.

C. Receiver Performance Parameters

  • Sensitivity: Minimum input signal power required at antenna to produce a usable output (specified SNR at output). Sensitivity ∝ (SNR_required * k T B) / (G * modulation efficiency).

  • Selectivity: Ability to separate adjacent channels. Measured by bandwidth of IF filters (e.g., 6 dB bandwidth).

  • Fidelity: Faithfulness of audio reproduction. Limited by bandwidth of entire receiver (IF, audio stages).


V. PULSE MODULATION TECHNIQUES

A. Basic Pulse Modulation

Type Modulated Parameter Generation Principle Key Feature
PAM Pulse amplitude Sample m(t) at t=nTs. Amplitude follows m(t).
PWM Pulse width (duration) Compare m(t) with sawtooth (triangular) carrier. Width ∝ m(t).
PPM Pulse position (time) Differentiate PWM → zero-crossings mark positions. Position ∝ m(t). Often derived from PWM.
  • PAM Types:

    • Natural Sampling: Top of pulses follow m(t).

    • Flat-top Sampling: Pulse held constant between samples (easier to handle, requires reconstruction filter).

B. Delta Modulation (DM)

  • Principle: 1-bit quantizer. Compares input with integrated output (previous quantized level + step Δ).

    • If m(t) > pred. output: Output +Δ (bit 1).

    • If m(t) < pred. output: Output -Δ (bit 0).

  • Encoder/Decoder: Simple, low bit rate (1/Ts).

  • Waveforms: Staircase approximation of m(t).

  • Limitations:

    1. Slope Overload Distortion: When dm/dt > Δ/Ts (step too small to follow steep slope).

    2. Granular (Idle) Noise: When dm/dt ≈ 0, output oscillates between ±Δ due to quantization noise.

C. Adaptive Delta Modulation (ADM)

  • Principle: Variable step size Δ.

    • Increase Δ for steep slopes (reduce slope overload).

    • Decrease Δ for gentle slopes (reduce granular noise).

  • Merits over DM: Reduces both distortions.

  • Demerits: More complex logic to adapt step size (e.g., based on recent bit pattern).

D. Time Division Multiplexing (TDM)

  • Synchronous TDM: Each signal gets a fixed time slot in a repeating frame.

  • Block Diagram:

    • Multiplexer: Parallel-to-serial switch, samples each input in round-robin.

    • Demultiplexer: Serial-to-parallel switch, distributes bits to appropriate output lines using sync.

  • Applications: PCM telephony (E1/T1 lines), digital audio (AES/EBU).


VI. SAMPLING THEORY AND QUANTIZATION

A. Nyquist-Shannon Sampling Theorem (Low-pass)

  • Statement: A bandlimited signal with max frequency f_max can be perfectly reconstructed from its samples if sampled at f_s ≥ 2 f_max.

  • Nyquist Rate: f_N = 2 f_max.

  • Proof Concept:

    1. Sampling x(t) by δ_T(t) = Σ δ(t-nT) (with T=1/f_s) in time domain is multiplication.

    2. In frequency, X_s(ω) = (1/T) Σ X(ω - kω_s) (periodic repetition of X(ω) with period ω_s = 2π f_s).

    3. To avoid aliasing (overlap of spectral replicas), need ω_s ≥ 2ω_max → f_s ≥ 2 f_max.

  • Consequences of Undersampling (f_s < 2f_max): Spectral replicas overlap → aliasing (irreversible distortion). Anti-aliasing filter (low-pass) required before sampling.

B. Sampling of Bandpass Signals

  • Bandpass Sampling Theorem: For signal bandlimited to [f_L, f_H], minimum f_s can be as low as 2B (where B = f_H - f_L) if f_L is sufficiently high.

  • Condition: f_s ≥ 2B and f_s chosen so that spectral replicas do not overlap. f_s must satisfy: f_s = 2f_H / n for some integer n such that f_L ≥ (n-1)f_s / 2.

C. Quantization

  • Need: Convert continuous amplitude samples to discrete levels for digital transmission (PCM).

  • Uniform Quantization: L levels, step size Δ = (x_max - x_min)/L.

    • Quantization Error (Noise) e_q: e_q = x_q - x, where x_q is quantized value.

    • Assumed uniformly distributed over [-Δ/2, Δ/2] for large L and high probability of overload.

    • Variance (Power): σ_q² = Δ²/12.

  • Signal-to-Quantization-Noise Ratio (SQNR):

    • For a full-scale sinusoidal signal x(t) = A sin(ωt), signal power P_x = A²/2.

    • Δ = 2A/L (for L levels over [-A, A]).

    • σ_q² = (2A/L)² / 12 = A²/(3L²).

    • SQNR = P_x / σ_q² = (A²/2) / (A²/(3L²)) = (3/2) L².

    • In dB: SQNR_dB = 10 log10(3L²/2) ≈ 10 log10(L²) + 1.76 = 20 log10(L) + 1.76.

    • Since L = 2^n (n bits), \boxed{SQNR_{dB} \approx 6n + 1.76}. Approx. 6 dB per bit.


VII. DIGITAL MODULATION TECHNIQUES

A. Binary Modulation

  1. Amplitude Shift Keying (ASK):

    • Coherent ASK (On-Off Keying): s(t) = A_c cos(ω_c t) for bit 1, 0 for bit 0.

    • Non-coherent ASK: Envelope detection (simple, but poorer performance).

    • Bandwidth: Approximately 2/T_b (same as baseband).

  2. Frequency Shift Keying (FSK):

    • Generation: Two oscillators at f1 and f2, switched by data. Or VCO controlled by data.

    • Spectrum: Two discrete tones. Bandwidth (approx.): B ≈ |f1 - f2| + 2f_m = 2Δf + 2f_m (using Carson's rule for continuous phase).

    • Continuous Phase FSK (CPFSK): Phase is continuous at bit transitions (e.g., MSK).

  3. Binary Phase Shift Keying (BPSK):

    • Expression: s(t) = ±A_c cos(ω_c t) (0° for 1, 180° for 0).

    • Generation: Multiply binary data ±1 by carrier.

    • Coherent Detection: Correlator or matched filter with reference carrier. Optimal.

    • Bandwidth: Same as ASK (≈ 2/T_b).

B. Differential and Quadrature Modulation

  1. Differential BPSK (DBPSK):

    • Principle: s(t) = A_c cos(ω_c t + θ_n), where θ_n = θ_{n-1} + π(1 - b_n) (phase change depends on current bit b_n).

    • Receiver (Differential Detection): Delay current symbol by T_b, multiply with previous symbol. Output polarity gives b_n.

    • Advantage over BPSK: No need for absolute carrier phase synchronization. Disadvantage: ~3 dB worse BER in AWGN.

  2. Quadrature Phase Shift Keying (QPSK):

    • Generation: Two BPSK streams on quadrature carriers (cos(ω_c t) and sin(ω_c t)).

      • I(t) = A_c * d_I(t) (even bits), Q(t) = A_c * d_Q(t) (odd bits).

      • s(t) = I(t)cos(ω_c t) - Q(t)sin(ω_c t).

    • Constellation: 4 points (±1, ±1) on a circle (constant envelope). 2 bits/symbol.

    • Bandwidth: Half of BPSK for same bit rate (since symbol rate R_s = R_b / 2). B ≈ R_s = R_b / 2.

    • Offset QPSK (OQPSK): Stagger I and Q channels by T_b/2. Prevents 180° phase jumps (which cause spectral regrowth). Better for nonlinear amplifiers.

  3. Quadrature Amplitude Modulation (QAM):

    • Principle: Independent amplitude modulation of I and Q channels. s(t) = I(t)cos(ω_c t) - Q(t)sin(ω_c t).

    • Constellation: Square grid (e.g., 16-QAM: 16 points, 4 bits/symbol).

    • Generation & Detection: I-Q Modulator/Demodulator block diagram.

      • Modulator: Data split → I & Q → amplitude scale → multiply with cos/sin → add.

      • Demodulator: Coherent detection: multiply with cos/sin, low-pass filter, sample.

    • Comparison with QPSK:

      • QPSK is 4-QAM (points at corners of square, constant radius).

      • Higher-order QAM (16, 64) has varying amplitude → more sensitive to noise and nonlinear distortion.

      • QAM has higher bandwidth efficiency (bits/s/Hz) but worse BER.

C. M-ary Modulation

  • M-ary PSK: M equally spaced phases on a circle. Bandwidth efficiency = log2(M) bits/symbol. BER increases with M for same E_b/N_0.

  • General Concept: Increase M → more bits per symbol → higher bandwidth efficiency, but requires higher E_b/N_0 for same BER.


VIII. PERFORMANCE ANALYSIS & MULTIPLEXING

A. Signal-to-Noise Ratio (SNR) Analysis (Coherent Detection)

  • DSB-SC:

    • Demodulator output: y(t) = (2/A_c) [m(t) * cos(ω_c t)] * cos(ω_c t) = m(t) + (1/A_c)[m(t) cos(2ω_c t)].

    • After LPF: y(t) = m(t). Noise power at output: N_0 (baseband).

    • SNR_out = (P_m) / (N_0 B), where P_m is message power.

  • SSB (USB):

    • s(t) = m(t)cos(ω_c t) ∓ ĥm(t)sin(ω_c t).

    • Coherent demodulation with cos(ω_c t) and sin(ω_c t) recovers m(t) and ±ĥm(t).

    • SNR_out = (P_m) / (N_0 B) (same as DSB-SC, but bandwidth B = f_m instead of 2f_m → better SNR for same noise power spectral density).

  • Comparison: SSB has half the bandwidth of DSB-SC for same message, so for same total noise power N_0 B, SSB has 3 dB better SNR.

B. Multiplexing Techniques

  • Frequency Division Multiplexing (FDM):

    • Concept: Each signal modulates a different carrier frequency (f_c1, f_c2, ...). Spectra are separated in frequency domain.

    • Application: Analog systems (AM radio bands, cable TV).

    • Process: Modulate → translate spectra → sum. At receiver: bandpass filter → demodulate.

  • Time Division Multiplexing (TDM): (See Section V.D) → Digital domain.

C. Bandwidth and Spectral Efficiency

  • Bandwidth Calculations (approx.):

    • AM (DSB-FC): B = 2 f_m.

    • DSB-SC: B = 2 f_m.

    • SSB: B = f_m.

    • FM (Carson's Rule): B ≈ 2(Δf + f_m).

    • BPSK/QPSK: B ≈ R_s (for practical pulses). R_s = R_b / log2(M).

    • FSK: B ≈ |f1-f2| + 2f_m = 2Δf + 2f_m.

  • Bandwidth Efficiency (η): η = (bit rate) / (bandwidth) = R_b / B (bits/s/Hz).

    • BPSK: η ≈ 1 (if B ≈ R_b).

    • QPSK: η ≈ 2 (since R_b = 2 R_s, B ≈ R_s).

    • M-ary: η = log2(M) (theoretical max).

[!TIP] Exam Formula: For raised cosine filtering with roll-off α, B = R_s (1+α)/2 for QPSK. So η = 1/(1+α) (e.g., α=0.2 → η ≈ 0.83 bits/s/Hz for QPSK? Wait: R_b = 2 R_s, B = R_s(1+α), so η = 2R_s / (R_s(1+α)) = 2/(1+α). For α=0.2, η = 2/1.2 ≈ 1.67 bits/s/Hz).

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