diff --git a/src/Strong_Bassline.mp3 b/src/Strong_Bassline.mp3
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index 0000000..4d5f73e
Binary files /dev/null and b/src/Strong_Bassline.mp3 differ
diff --git a/src/epic_effect_schluep.m b/src/epic_effect_schluep.m
new file mode 100644
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+++ b/src/epic_effect_schluep.m
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+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.
diff --git a/src/muffled_effect_schluep.m b/src/muffled_effect_schluep.m
new file mode 100644
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+++ b/src/muffled_effect_schluep.m
@@ -0,0 +1,56 @@
+function output = muffled_effect_schluep(input, Fs, LOW, MED, HIGH)
+% muffled_effect_schluep: Outputs a muffled version of the the original input
+% sound in the time domain. This makes it sound as if the audio is being 
+% played in another room.
+% 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 that the low-pass filter will let pass. 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.
+lowPassFilter = zeros(1, length(inputFreq));
+
+% Creating Low Pass Filter
+for i = 1:length(lowPassFilter)
+    if abs(F(i)) < HIGH
+        lowPassFilter(i) = 1;
+    else
+        lowPassFilter(i) = 0;
+    end
+end
+
+lowPassedInput = inputFreq .* transpose(lowPassFilter); %Apply the low-pass filter.
+
+% Adding a slight reverb effect.
+realOutput = real(ifft(fftshift(lowPassedInput)));
+
+delay = 0.001;     % The delay of the sound in seconds.
+index = round(delay*Fs);    % Find the first index where sound should start playing
+% by multiplying the delay by the sampling frequency.
+delayedOutput = [zeros(index, 1); realOutput];      % Adds "index" 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 = (realOutput + delayedOutput) ./ 2.0;    % Adds the "realOutput" and "delayedOutput"
+% vectors to create a reverb effect. Divides by 2 to avoid clipping
+% effects.
+
+
+
+
+
diff --git a/src/seperate_prevalent_schluep.m b/src/seperate_prevalent_schluep.m
new file mode 100644
index 0000000..3fbc7a6
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+++ b/src/seperate_prevalent_schluep.m
@@ -0,0 +1,62 @@
+function output = seperate_prevalent_schluep(input, Fs, LOW, MED, HIGH)
+% seperate_prevalent_schluep: Attempts to seperate the most prevalent frequencies
+% from the input sound by finding the most prevalent frequencies and applying
+% a band-pass filter to a small region around those frequencies.
+% 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 distance around the maximum frequency value that will
+% not be attenuated. A good range of values is usually 250-500 Hz, but it
+% depends on the input sound.
+
+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.
+bandPassFilter = zeros(1, length(inputFreq));
+
+maxAmplitude = 0;
+mostPrevalentFrequency = 0;
+
+% Finding the most prevalent frequency.
+for i = round(length(inputFreq)/2):length(inputFreq)
+    if inputFreq(i) > maxAmplitude
+        maxAmplitude = inputFreq(i);
+        mostPrevalentFrequency = F(i);
+    end
+end
+
+% Determining maximum and minimum frequency values for the Band Pass filter.
+maxFrequency = mostPrevalentFrequency + HIGH;
+minFrequency = mostPrevalentFrequency - HIGH;
+
+if minFrequency < 0.0
+    minFrequency = 0.0;
+end
+
+% Creating the Band-Pass filter.
+for i = 1:length(bandPassFilter)
+    if (abs(F(i)) < maxFrequency) && (abs(F(i)) > minFrequency)
+        bandPassFilter(i) = 1;
+    else
+        bandPassFilter(i) = 0;
+    end
+end
+
+bandPassedInput = inputFreq .* transpose(bandPassFilter); %Apply the Band-Pass Filter.
+
+output = real(ifft(fftshift(bandPassedInput)));
+
+
+
+