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.
- 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
- If you have a collection of audio files, you may also look for the "audioDatastore" function.
- 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
- 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:
Hope you find this helpful! Thanks.
