56 lines
2.4 KiB
Matlab
56 lines
2.4 KiB
Matlab
function output = epic_effect_schluep(input, Fs, LOW, MED, HIGH)
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% epic_effect_schluep: Outputs a more "epic" version of the input sound.
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% This is done by amplifying low frequencies and by implementing a chorus
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% effect. A chorus effect is created when multiple copies of sound delayed
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% by a small, random amount are added to the original signal. Works best on
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% songs with a stronger bassline.
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% Try this function out with "Strong_Bassline.mp3".
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% CONTRIBUTORS:
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% Nicolas Schluep: Function Author
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% DOCUMENTATION:
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% input: The input sound in the time-domain.
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% Fs: The sampling rate of the input signal. A typical value is 44100 Hz.
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% HIGH: The maximum frequency the filter will amplify. A typical value for
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% this variable is 1000 Hz.
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non_stereophonic = input(:, 1); % Removes the sterophonic property of the input sound
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% by just taking the first column of data.
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Len = length(non_stereophonic);
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F = Fs * ((-Len/2) : ((Len/2) - 1)) / Len; % Creating the array of frequencies
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% which the FFT Shifted version of the signal can be plotted against.
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inputFreq = fftshift(fft(non_stereophonic)); % Creates the Fourier Transform of the
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% input signal. fftshift() makes it such that the zero frequency is at the
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% center of the array.
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lowAmplifyFilter = zeros(1, length(inputFreq));
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% Creating a filter which amplifies lower frequencies.
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for i = 1:length(lowAmplifyFilter)
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if abs(F(i)) < HIGH
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lowAmplifyFilter(i) = 1.25;
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else
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lowAmplifyFilter(i) = 1.00;
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end
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end
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lowPassedInput = inputFreq .* transpose(lowAmplifyFilter); %Apply the "lowAmplifyFilter".
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% Adding the chorus effect.
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realOutput = real(ifft(fftshift(lowPassedInput)));
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output = realOutput;
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% Adding 100 randomly delayed signals to the original signal which creates the chorus effect.
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for i = 1:100
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currentDelay = 0.003 * rand(); % The current delay of the sound in seconds.
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currentIndex = round(currentDelay * Fs); % Find the first index where the sound should start playing.
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delayedOutput = [zeros(currentIndex, 1); realOutput]; % Adds "currentIndex" zeros to the front of the
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% "realOutput" vector to create a slightly delayed version of the signal.
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delayedOutput = delayedOutput(1:length(realOutput)); % Truncates the "delayedOutput"
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% vector so that it can be added to the "realOutput" vector.
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output = output + delayedOutput;
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end
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output = output ./ 100; % Divide by 100 to decrease the amplitude of the sound to a normal level.
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