moved non functional Filters to NotWorking

src/NotWorking/epic_effect_schluep.m

@@ -0,0 +1,55 @@
+function output = epic_effect_schluep(input, Fs, LOW, MED, HIGH)
+% epic_effect_schluep: Outputs a more "epic" version of the input sound.
+% This is done by amplifying low frequencies and by implementing a chorus
+% effect. A chorus effect is created when multiple copies of sound delayed
+% by a small, random amount are added to the original signal. Works best on
+% songs with a stronger bassline.
+% Try this function out with "Strong_Bassline.mp3".
+
+% CONTRIBUTORS:
+% Nicolas Schluep: Function Author
+
+% DOCUMENTATION:
+% input: The input sound in the time-domain.
+% Fs: The sampling rate of the input signal. A typical value is 44100 Hz.
+% HIGH: The maximum frequency the filter will amplify. A typical value for 
+% this variable is 1000 Hz.
+
+non_stereophonic = input(:, 1);     % Removes the sterophonic property of the input sound
+% by just taking the first column of data.
+
+Len = length(non_stereophonic);
+F = Fs * ((-Len/2) : ((Len/2) - 1)) / Len;  % Creating the array of frequencies
+% which the FFT Shifted version of the signal can be plotted against.
+inputFreq = fftshift(fft(non_stereophonic));   % Creates the Fourier Transform of the
+% input signal. fftshift() makes it such that the zero frequency is at the 
+% center of the array.
+lowAmplifyFilter = zeros(1, length(inputFreq));
+
+% Creating a filter which amplifies lower frequencies.
+for i = 1:length(lowAmplifyFilter)
+    if abs(F(i)) < HIGH
+        lowAmplifyFilter(i) = 1.25;
+    else
+        lowAmplifyFilter(i) = 1.00;
+    end
+end
+
+lowPassedInput = inputFreq .* transpose(lowAmplifyFilter); %Apply the "lowAmplifyFilter".
+
+% Adding the chorus effect.
+realOutput = real(ifft(fftshift(lowPassedInput)));
+output = realOutput;
+
+% Adding 100 randomly delayed signals to the original signal which creates the chorus effect.
+for i = 1:100
+    currentDelay = 0.003 * rand();  % The current delay of the sound in seconds.
+    currentIndex = round(currentDelay * Fs);    % Find the first index where the sound should start playing.
+    delayedOutput = [zeros(currentIndex, 1); realOutput];      % Adds "currentIndex" zeros to the front of the 
+    % "realOutput" vector to create a slightly delayed version of the signal.
+    delayedOutput = delayedOutput(1:length(realOutput));    % Truncates the "delayedOutput"
+    % vector so that it can be added to the "realOutput" vector.
+    output = output + delayedOutput; 
+end
+
+output = output ./ 100;     % Divide by 100 to decrease the amplitude of the sound to a normal level.