How can i insert the audio and convert it to digtal signal using Adaptive Delta Modulation ?

How can i insert the audio and convert it to digtal signal using Adaptive Delta Modulation ?

Answer 1

[Saurav]

Hello,

It appears that you wish to use MATLAB to import an audio file and use the Adaptive Delta Modulation method to convert it to a digital signal.

Adaptive Delta Modulation (ADM) or Continuously Variable Slope Delta-Modulation (CVSD), is used to approximate the input signal with a single bit per signal sample with the addition of an adaptive step-size.

1. The "audioread" function in MATLAB can be used to import an audio file. To find out more about this function, visit this link: https://www.mathworks.com/help/releases/R2023b/matlab/ref/audioread.html
2. If you have a collection of audio files, you may also look for the "audioDatastore" function.
3. Define your Adaptive Delta Modulation parameters, such as step size and initial quantized value. Visit the following page for detailed information about the ADM techniques: https://www.mathworks.com/help/releases/R2023b/dsp/ug/comparison-of-ldm-cvsd-and-adpcm.html
4. Implement the ADM logic in a loop or using a custom function.

It can be implemented in the following way:

% Read an audio file
[originalSignal, Fs] = audioread('Free_Test_Data_2MB_WAV.wav'); 
t1 = (0:length(originalSignal)-1)/Fs; % Time vector

% Assuming a mono channel audio signal for simplicity
originalSignal = originalSignal(:,1);

% ADM parameters
initialStepSize = 0.02; % Initial step size
increaseFactor = 1.1; % Factor by which step size is increased
decreaseFactor = 0.9; % Factor by which step size is decreased
maxStepSize = 0.1; % Maximum step size
minStepSize = 0.001; % Minimum step size

% Initialize variables
stepSize = initialStepSize;
encodedSignal = zeros(size(originalSignal)); % Initialize encoded signal
decodedSignal = zeros(size(originalSignal)); % Initialize decoded signal
decodedSignal(1) = originalSignal(1); % Set the first sample

% ADM Encoding and Decoding loop
for i = 2:length(originalSignal)
    % Calculate the difference between the current sample and the previous decoded value
    difference = originalSignal(i) - decodedSignal(i-1);
    
    % Check if the signal has increased or decreased
    if difference >= 0
        encodedSignal(i) = 1;
        decodedSignal(i) = decodedSignal(i-1) + stepSize;
        % If the previous bit was the same, increase the step size
        if encodedSignal(i) == encodedSignal(i-1)
            stepSize = min(stepSize * increaseFactor, maxStepSize);
        end
    else
        encodedSignal(i) = 0;
        decodedSignal(i) = decodedSignal(i-1) - stepSize;
        % If the previous bit was different, decrease the step size
        if encodedSignal(i) ~= encodedSignal(i-1)
            stepSize = max(stepSize * decreaseFactor, minStepSize);
        end
    end
end

% Plot the original and the decoded signal
figure;
plot(t1, originalSignal, 'b', t1, decodedSignal, 'r');
legend('Original Signal', 'Decoded Signal');
xlabel('Time [s]');
ylabel('Amplitude');
title('ADM');

% Calculate and plot the difference (error) signal
errorSignal = originalSignal - decodedSignal;
figure;
plot(t1, errorSignal);
title('Error Signal');
xlabel('Time [s]');
ylabel('Error Amplitude');

A few points that need to be considered:

• The implementation shown above is merely an example and does not have every feature of a complete ADM system.
• Depending on the particular needs of your application and the properties of the signal you are dealing with, modify the parameters and the adaptation mechanism (step-size algorithm).
• Be sure to substitute your own "Audio_file.wav" for "Free_Test_Data_2MB_WAV.wav."

Refer to the following link to learn more about “audioDatastore” function:

• https://www.mathworks.com/help/releases/R2023b/audio/ref/audiodatastore.html

Hope you find this helpful! Thanks.