Skip to main content

Equalizer to reduce Multi-path Effects using MATLAB

 

Steps

1. Convert Bit Stream to Bipolar Format. Converts the bit stream from binary (0, 1) to bipolar format (-1, 1).

2. Define Channel Impulse Response

3. Pass Signal Through the Channel. Convolves the bipolar signal with the channel impulse response to simulate the channel effect.

4. Adds Gaussian noise to the received signal based on the specified SNR.

5. Initialize Adaptive Filter Parameters. 

  • w: Initializes the adaptive filter coefficients.
  • x_buf: Initializes the buffer for the input to the adaptive filter.
  • equalized_signal: Initializes the array to store the equalized signal.
  • P: Initializes the inverse correlation matrix.
  • 6. Adaptive Equalization Using RLS Algorithm

    Loops through each sample to perform adaptive equalization:

    • Update Input Buffer: Adds the current sample to the input buffer.
    • Calculate Gain Vector: Computes the gain vector k for the adaptive filter.
    • Calculate Error Signal: Computes the error between the original signal and the filter output.
    • Update Filter Coefficients: Updates the adaptive filter coefficients based on the error signal.
    • Update Inverse Correlation Matrix: Updates the inverse correlation matrix for the RLS algorithm.
    • Store Equalized Output: Stores the equalized signal in the output array.

    7. Plot Original and Equalized Signals 

     

    MATLAB Script

    clc;
    clear;
    close all;

    % Parameters
    bit_stream = [1, 1, 0, 0, 1, 0, 1, 1, 1, 0]; % Original bit stream
    N = length(bit_stream); % Number of samples
    filter_order = 10; % Order of the adaptive filter
    lambda = 0.99; % Forgetting factor for RLS algorithm
    delta = 1; % Initial value for the inverse correlation matrix
    SNR = 15; % SNR value in dB

    % Convert bit stream to bipolar format (-1, 1)
    original_signal = bit_stream * 2 - 1;

    % Channel impulse response
    h = [0.75, 0.05, 0.02];

    % Pass the signal through the channel
    received_signal = filter(h, 1, original_signal);

    % Add some noise
    received_signal_noisy = awgn(received_signal, SNR, 'measured');

    % Initialize the adaptive filter coefficients
    w = zeros(filter_order, 1);

    % Initialize buffer for the input to the adaptive filter
    x_buf = zeros(filter_order, 1);

    % Initialize output
    equalized_signal = zeros(N, 1);

    % Initialize the inverse correlation matrix
    P = delta * eye(filter_order);

    % Adaptive equalization using RLS
    for n = 1:N
        % Update the input buffer
        x_buf = [received_signal_noisy(n); x_buf(1:end-1)];

        % Calculate the gain vector
        k = (P * x_buf) / (lambda + x_buf' * P * x_buf);

        % Calculate the error signal
        e = original_signal(n) - w' * x_buf;

        % Update the filter coefficients
        w = w + k * e;

        % Update the inverse correlation matrix
        P = (P - k * x_buf' * P) / lambda;

        % Store the equalized output
        equalized_signal(n) = w' * x_buf;
    end

    % Plot original and equalized signals
    figure;
    subplot(2, 1, 1);
    stem(original_signal, 'filled');
    title('Original Signal');
    xlabel('Sample Index');
    ylabel('Amplitude');
    grid on;

    subplot(2, 1, 2);
    stem(equalized_signal, 'filled');
    title('Equalized Signal');
    xlabel('Sample Index');
    ylabel('Amplitude');
    grid on;
     

    Output


     

    Copy the MATLAB Code from here

     

    Further Reading



    Contact Us

    Name

    Email *

    Message *

    Popular Posts

    OFDM Symbols and Subcarriers Explained

    This article explains how OFDM (Orthogonal Frequency Division Multiplexing) symbols and subcarriers work. It covers modulation, mapping symbols to subcarriers, subcarrier frequency spacing, IFFT synthesis, cyclic prefix, and transmission. Step 1: Modulation First, modulate the input bitstream. For example, with 16-QAM , each group of 4 bits maps to one QAM symbol. Suppose we generate a sequence of QAM symbols: s0, s1, s2, s3, s4, s5, …, s63 Step 2: Mapping Symbols to Subcarriers Assume N sub = 8 subcarriers. Each OFDM symbol in the frequency domain contains 8 QAM symbols (one per subcarrier): Mapping (example) OFDM symbol 1 → s0, s1, s2, s3, s4, s5, s6, s7 OFDM symbol 2 → s8, s9, s10, s11, s12, s13, s14, s15 … OFDM sym...

    UGC NET Electronic Science Previous Year Question Papers with Solutions

    Home / Engineering & Other Exams / UGC NET 2026 PYQ ⬇️ Download Papers and Solutions 📋 Exam Pattern 💡 Preparation Tips ❓ FAQs 📊 Exam Highlights: Electronic Science (88) Feature Details Junior Research Fellowship (JRF) ₹37,000 + HRA per month Eligibility M.Sc/M.Tech in Electronics (55%) Validity of Certificate JRF (3 Years) | Lectureship (Lifetime) 📥 Download UGC NET Electronics PDFs Complete collection of previous year question papers, answer keys and explanations for Subject Code 88. Start Downloading 📂 View All Question Papers June 2025 - Question Paper Download PDF June 2025 - Solved Paper + Explanation ...

    Design of CMOS XOR/XNOR Gates

    Design of CMOS XOR/XNOR Gates The semiconductor industry has experienced rapid integration of multimedia applications into mobile electronics, leading to very high integration density in CMOS VLSI. As operating frequencies increase, power consumption, speed, silicon area, and reliability become critical considerations. The XOR-XNOR circuits are fundamental building blocks in arithmetic circuits (Full Adders, Multipliers), compressors, comparators, parity checkers, code converters, error-detecting/correcting codes, and phase detectors. Their performance directly impacts the complex circuits they are used in. Design goals include full output voltage swing, low power consumption, reduced transistor count, minimal delay, and simultaneous non-skewed outputs. Static Logic (Static CMOS) Stat...

    DFTs-OFDM vs OFDM: Why DFT-Spread OFDM Reduces PAPR Effectively (with MATLAB Code)

    Understanding PAPR in DFT-spread OFDM vs. Standard OFDM In modern wireless communications like 4G LTE and 5G NR, managing the Peak-to-Average Power Ratio (PAPR) is critical for hardware efficiency. While OFDM is the gold standard for high-speed data, its high PAPR poses significant challenges for mobile devices. This is where DFTs-OFDM (also known as SC-FDMA) comes in. DFT-spread OFDM (DFTs-OFDM) has lower Peak-to-Average Power Ratio (PAPR) because it "spreads" the data in the frequency domain before applying IFFT, making the time-domain signal behave more like a single-carrier signal rather than a multi-carrier one like OFDM. Deeper Explanation: Aspect OFDM DFTs-OFDM Signal Type Multi-carrier Single-carrier-like Process IFFT of QAM directly QAM → DFT → IFFT PAPR Level High (due to many...

    Online Simulator for ASK, FSK, and PSK Signal Generation

    Interactive Digital Signal Processing (DSP) Tutorial and Simulator for ASK, FSK, and BPSK modulation techniques. Try our new Digital Signal Processing Simulator!   •   Interactive ASK, FSK, and BPSK tools updated for 2025. Start Now Digital Modulation Visualizer: ASK, FSK, & BPSK Simulator Learn and visualize binary modulation techniques (ASK, FSK, BPSK) in real-time with adjustable carrier and sampling parameters. Perfect for DSP students and engineers. 📡 ASK Simulator 📶 FSK Simulator 🎚️ BPSK Simulator 📚 More Topics ASK Modulator FSK Modulator BPSK Modulator More Topics 1. ASK (Amplitude Shift Keying) Simulat...

    Calculation of SNR from FFT bins in MATLAB

    📘 Overview 💻 FFT Bin Method 💻 Kaiser Window 📚 Further Reading SNR Estimation Overview In digital signal processing, estimating the Signal-to-Noise Ratio (SNR) accurately is crucial. Below, we demonstrate how to calculate SNR from periodogram and FFT bins using the Kaiser Window . The beta (β) parameter is the key—it allows you to control the trade-off between main-lobe width and side-lobe levels for precise spectral analysis. 1 Define Sampling rate and Time vector 2 Compute FFT and Periodogram PSD 3 Identify Signal Bin and Frequency resolution 4 Segment Signal Power from Noise floor 5 Logarithmic calculation of SNR in dB Method 1: Estimation from FFT Bins This approach uses a Hamming window to estimate SNR directly from the spectral bins. MATLAB Source Code Copy Code clc...

    MATLAB Code for MUSIC

      MATLAB Code clc; clear; close all ; %% Step 1: Define Parameters M = 8; % Number of array sensors d = 0.5; % Sensor spacing (lambda/2) K = 2; % Number of signals N = 200; % Number of snapshots theta = [-20 30]; % True signal angles (degrees) SNR = 10; % Signal-to-noise ratio (dB) fprintf( 'Step 1: Parameters Initialized\n' ); %% Step 2: Generate Signal Sources t = 1:N; s1 = exp(1j*2*pi*0.05*t); s2 = exp(1j*2*pi*0.1*t); S = [s1; s2]; figure; plot(real(S(1,:))) title( 'Signal 1 (Real Part)' ) xlabel( 'Samples' ) ylabel( 'Amplitude' ) figure; plot(real(S(2,:))) title( 'Signal 2 (Real Part)' ) xlabel( 'Samples' ) ylabel( 'Amplitude' ) fprintf( 'Step 2: Source Signals Generated\n' ); %% Step 3: Construct Steering Matrix A = zeros(M,K); for k = 1:K A(:,k) = exp(-1j*2*pi*d*(0:M-1)'*sin(theta(k)*pi/180)); end fprintf( 'Step 3: Steering Matr...