diff --git a/.DS_Store b/.DS_Store
index dd306f7..bc2a648 100644
Binary files a/.DS_Store and b/.DS_Store differ
diff --git a/README.md b/README.md
index 1807172..2a81b49 100644
--- a/README.md
+++ b/README.md
@@ -6,7 +6,7 @@
  Will add member names shortly
  
  ## Function Prototypes
-Templates to create your own functions.
+Templates to create your own functions. Please commit code in either src/ (the base directopry) or src/NotWorking (please dont commit in any other subfolder).
 
  ### Wave generating function
 ```
@@ -57,10 +57,10 @@ where LOW, MED, HIGH are user-selected variables of any value.
  
  ## Useful websites
  
- - https://learningsynths.ableton.com
- - https://learningsynths.ableton.com/en/playground
- - https://blog.demofox.org/diy-synthesizer/
- - http://portaudio.com/
- - https://ccrma.stanford.edu/software/stk/
- - https://cycling74.com/products/max
- - http://msp.ucsd.edu/software.html
+ - [Learning Synths](https://learningsynths.ableton.com)
+ - [Learning Synths Playground](https://learningsynths.ableton.com/en/playground)
+ - [DIY Synthesisers](https://blog.demofox.org/diy-synthesizer/)
+ - [Port Audio (audio I/O library)](http://portaudio.com/)
+ - [Synthesis ToolKit in C++](https://ccrma.stanford.edu/software/stk/)
+ - [Max](https://cycling74.com/products/max)
+ - [Software by Miller Puckette](http://msp.ucsd.edu/software.html)
diff --git a/src/.DS_Store b/src/.DS_Store
index a262e39..385d2ba 100644
Binary files a/src/.DS_Store and b/src/.DS_Store differ
diff --git a/src/Filter/AnuragDampenTarget.m b/src/Filter/AnuragDampenTarget.m
new file mode 100644
index 0000000..3b325d3
--- /dev/null
+++ b/src/Filter/AnuragDampenTarget.m
@@ -0,0 +1,33 @@
+function output_x = AnuragDampenTarget(x, Fs,LOW, MID, HIGH) 
+%Dampen a range of frequencies a percentage around the target frequency
+%Also maintain the other frequencies so that the target is muted
+%percentage is expressed as a number between 0 and 1.
+%target is the midpoint of the frequency range to dampen
+%x and Fs are the signal and frequency
+% CONTRIBUTORS:
+% Anurag Jadhav: function creator
+
+%   Detailed explanation goes here
+    TARGET = MID;
+    AreaPercentage = 0.15;
+    len = length(x); %get length of the input
+    F = Fs * ((-len/2) : ((len/2) - 1)) / len;
+    lenf = length(F);
+    Mod_Freq = fftshift(fft(x)); %Fourier Transform of the input signal
+    output =  zeros([1,lenf]); % zero array of size Mod_freq
+    %set the bounds
+    lowerBound = (1-AreaPercentage) * TARGET;
+    upperBound = (1+AreaPercentage) * TARGET;
+%% Dampen the target frequencies and maintain the others
+    for n = 1:lenf
+        if ((lowerBound < abs(F(n))) && abs(F(n)) < upperBound)
+            output(n) = 0.5;
+        else
+            output(n) = 1;
+        end
+    end
+%%Filter the original signal and transform 
+    filtered_Mod_Freq = fftshift(Mod_Freq .* output);
+    output_x = real(ifft(filtered_Mod_Freq));
+
+end
\ No newline at end of file
diff --git a/src/Filter/AnuragEnhanceTarget.m b/src/Filter/AnuragEnhanceTarget.m
new file mode 100644
index 0000000..97d990a
--- /dev/null
+++ b/src/Filter/AnuragEnhanceTarget.m
@@ -0,0 +1,34 @@
+function output_x = AnuragEnchanceTarget(x, Fs,LOW, MID, HIGH) 
+%Amplify a range of frequencies a percentage around the target frequency
+%Also damnpen the other frequencies so that the target is easier to hear
+%percentage is expressed as a number between 0 and 1.
+%target is the midpoint of the frequency range to amplify
+%x and Fs are the signal and frequency
+% CONTRIBUTORS:
+% Anurag Jadhav: function creator
+
+%   Detailed explanation goes here
+    TARGET = MID;
+    AreaPercentage = 0.15;
+    len = length(x); %get length of the input
+    F = Fs * ((-len/2) : ((len/2) - 1)) / len;
+    lenf = length(F);
+    Mod_Freq = fftshift(fft(x)); %Fourier Transform of the input signal
+    output =  zeros([1,lenf]); % zero array of size Mod_freq
+
+    %set the bounds
+    lowerBound = (1-AreaPercentage) * TARGET;
+    upperBound = (1+AreaPercentage) * TARGET;
+%% Amplify the target frequencies and dampen the others
+    for n = 1:lenf
+        if ((lowerBound < abs(F(n))) && abs(F(n)) < upperBound)
+            output(n) = 2;
+        else
+            output(n) = 0.75;
+        end
+    end
+%%Filter the original signal and transform 
+    filtered_Mod_Freq = fftshift(Mod_Freq .* output);
+    output_x = real(ifft(filtered_Mod_Freq));
+
+end
\ No newline at end of file
diff --git a/src/Filter/bandreject_filter.m b/src/Filter/bandreject_filter.m
index 8073a9e..9d089b9 100644
--- a/src/Filter/bandreject_filter.m
+++ b/src/Filter/bandreject_filter.m
@@ -1,4 +1,4 @@
-function output_y = bandreject_filter(Input, Fs, Low, High)
+function Output = bandreject_filter(Input, Fs, Low, High)
     % A filter that lets through most frequencies unaltered
     % but attentuates the frequencies in the specified range to
     % very low levels
diff --git a/src/Filter/epic_effect_schluep.m b/src/Filter/epic_effect_schluep.m
index 2ad7109..6ff54c3 100644
--- a/src/Filter/epic_effect_schluep.m
+++ b/src/Filter/epic_effect_schluep.m
@@ -15,13 +15,22 @@ function output = epic_effect_schluep(input, Fs, LOW, MED, HIGH)
 % 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.
+n = size(input, 2);
 
-Len = length(non_stereophonic);
+non_stereophonic = input;
+
+if (n == 1) || (n == 2) 
+    non_stereophonic = input(:, 1);     % Removes the sterophonic property of the input sound
+    % by just taking the first column of data.
+    non_stereophonic = transpose(non_stereophonic); 
+end
+
+modified_input = non_stereophonic(1, :);
+
+Len = length(modified_input);
 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
+inputFreq = fftshift(fft(modified_input));   % 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));
@@ -35,7 +44,8 @@ for i = 1:length(lowAmplifyFilter)
     end
 end
 
-lowPassedInput = inputFreq .* transpose(lowAmplifyFilter); %Apply the "lowAmplifyFilter".
+lowPassedInput = inputFreq .* lowAmplifyFilter; %Apply the "lowAmplifyFilter".
+lowPassedInput = transpose(lowPassedInput);
 
 % Adding the chorus effect.
 realOutput = real(ifft(fftshift(lowPassedInput)));
@@ -53,3 +63,7 @@ for i = 1:100
 end
 
 output = output ./ 100;     % Divide by 100 to decrease the amplitude of the sound to a normal level.
+
+output = transpose(output);
+
+end
\ No newline at end of file
diff --git a/src/Filter/muffled_effect_schluep.m b/src/Filter/muffled_effect_schluep.m
index c193a38..7e611ac 100644
--- a/src/Filter/muffled_effect_schluep.m
+++ b/src/Filter/muffled_effect_schluep.m
@@ -13,13 +13,22 @@ function output = muffled_effect_schluep(input, Fs, LOW, MED, HIGH)
 % 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.
+n = size(input, 2);
 
-Len = length(non_stereophonic);
+non_stereophonic = input;
+
+if (n == 1) || (n == 2) 
+    non_stereophonic = input(:, 1);     % Removes the sterophonic property of the input sound
+    % by just taking the first column of data.
+    non_stereophonic = transpose(non_stereophonic); 
+end
+
+modified_input = non_stereophonic(1, :);
+
+Len = length(modified_input);
 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
+inputFreq = fftshift(fft(modified_input));   % 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));
@@ -33,7 +42,8 @@ for i = 1:length(lowPassFilter)
     end
 end
 
-lowPassedInput = inputFreq .* transpose(lowPassFilter); %Apply the low-pass filter.
+lowPassedInput = inputFreq .* lowPassFilter; %Apply the low-pass filter.
+lowPassedInput = transpose(lowPassedInput);
 
 % Adding a slight reverb effect.
 realOutput = real(ifft(fftshift(lowPassedInput)));
@@ -50,6 +60,10 @@ output = (realOutput + delayedOutput) ./ 2.0;    % Adds the "realOutput" and "de
 % vectors to create a reverb effect. Divides by 2 to avoid clipping
 % effects.
 
+output = transpose(output);
+
+end
+
 
 
 
diff --git a/src/Filter/seperate_prevalent_schluep.m b/src/Filter/seperate_prevalent_schluep.m
index 3fbc7a6..a1312c8 100644
--- a/src/Filter/seperate_prevalent_schluep.m
+++ b/src/Filter/seperate_prevalent_schluep.m
@@ -14,13 +14,22 @@ function output = seperate_prevalent_schluep(input, Fs, LOW, MED, HIGH)
 % 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.
+n = size(input, 2);
 
-Len = length(non_stereophonic);
+non_stereophonic = input;
+
+if (n == 1) || (n == 2) 
+    non_stereophonic = input(:, 1);     % Removes the sterophonic property of the input sound
+    % by just taking the first column of data.
+    non_stereophonic = transpose(non_stereophonic); 
+end
+
+modified_input = non_stereophonic(1, :);
+
+Len = length(modified_input);
 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
+inputFreq = fftshift(fft(modified_input));   % 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));
@@ -53,10 +62,12 @@ for i = 1:length(bandPassFilter)
     end
 end
 
-bandPassedInput = inputFreq .* transpose(bandPassFilter); %Apply the Band-Pass Filter.
+bandPassedInput = inputFreq .* bandPassFilter; %Apply the Band-Pass Filter.
 
 output = real(ifft(fftshift(bandPassedInput)));
 
+end
+
 
 
     
diff --git a/App/.DS_Store b/src/Generators/.DS_Store
similarity index 100%
rename from App/.DS_Store
rename to src/Generators/.DS_Store
diff --git a/src/Generators/generate_cosine.m b/src/Generators/generate_cosine.m
new file mode 100644
index 0000000..8a013ed
--- /dev/null
+++ b/src/Generators/generate_cosine.m
@@ -0,0 +1,25 @@
+function x = generate_cosine(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_WAVENAME: returns a matrix of sampled WAVENAME wave
+
+% CONTRIBUTORS:
+% Mekhi Ellington: Original Creator
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty is a number between 0 and 1
+
+    % initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+    
+    % initialize a one dimensional zero matrix to be populated
+    x = zeros(1, n);
+    
+    % populate the matrix
+    for i = 1:n
+        t = i * dt;
+        x(i) = amplitude * cos(2*pi*frequency*t-phase);
+    end
+end
\ No newline at end of file
diff --git a/src/Generators/generate_heartbeat.m b/src/Generators/generate_heartbeat.m
new file mode 100644
index 0000000..40230d6
--- /dev/null
+++ b/src/Generators/generate_heartbeat.m
@@ -0,0 +1,46 @@
+function x = generate_heartbeat(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_SINE: returns a matrix of sampled heart monitor wave
+
+% CONTRIBUTORS:
+% Alex Nguyen: Original Author
+% Conner Hsu: Reviewed Code and Assisted in polishing
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty cycle should be a number between 0 and 0.5.
+    % duty of 0 or less would return 0 for all sample points
+    % duty of 0.5 or less would return a heartbeat signal since there would be
+        % equal amount of space within 1 period to have the right and left sides be equal
+    % duty of greater than 0.5 but less than 1 would return a slope starting at that duty i.e.
+        % the left side but leaving little room to the right side
+    % duty of between 1 and 2 would give a sawtooth wave in the +amplitude.
+    % duty greater than 2 would return 0 for all sample points
+   
+
+    % initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+    
+    
+    % initialize a one dimensional zero matrix to be populated
+    x = zeros(1, n);
+    
+    % populate the matrix
+    for i = 1:n
+        t = i * dt; % time at the i'th sample
+        st = mod(frequency * t - phase, 1);
+
+        if (duty-(duty / 2) < st && st < duty)
+            slope = amplitude / duty;
+            intercept = -0.5 * amplitude;
+            x(i) = slope * st + intercept;
+        elseif (duty < st && st < duty+(duty / 2)) 
+            slope = amplitude / duty;
+            intercept = -amplitude;
+            x(i) = slope * st + (1.5 * intercept);
+        else
+            x(i) = 0;
+    end
+end
\ No newline at end of file
diff --git a/src/Generators/generate_pulse.m b/src/Generators/generate_pulse.m
new file mode 100644
index 0000000..2f9c316
--- /dev/null
+++ b/src/Generators/generate_pulse.m
@@ -0,0 +1,33 @@
+function x = generate_pulse(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_puse: returns a matrix of sampled pulse wave
+
+% CONTRIBUTORS:
+% Brian Tran: Created the wave
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty is a number between 0 and 1
+
+    % initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+    % initialize a one dimensional zero matrix to be populated
+    x = zeros(1, n);
+    f0=1e+6;  % 1MHz
+    Fs=3e+6;
+    Tf=0.001;  % 1 millisecond
+    t=0:1/Fs:Tf-1/Fs;
+    td=0.1;   % duty cycle 
+    A0=10; % 10 Volts
+    F=0;
+    N=1000; % Number of points 
+
+    % populate the matrix
+    for n = 1:N
+        F=F+(1/n)*cos(n*2*pi*f0*t).*sin(n*pi*td);
+    end
+    F=F*(2*A0/pi);
+    F=F+A0*td;
+end
\ No newline at end of file
diff --git a/src/Generators/generate_trapezoid.m b/src/Generators/generate_trapezoid.m
new file mode 100644
index 0000000..6c612b5
--- /dev/null
+++ b/src/Generators/generate_trapezoid.m
@@ -0,0 +1,58 @@
+function x = generate_trapezoid(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_TRAPEZOID: returns a matrix of sampled square wave
+
+% CONTRIBUTORS:
+% Daniel Doan: Author
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty cycle should be a number between 0 and 1.
+    % duty of 0.5 would have 2 trapezoids in first half of each cycle
+    % example of wave with duty of 0.5, where the peaks are amplitude/2: 
+    % 
+    %   ____ 
+    %  /    \
+    % /      \        ________________
+    %         \      /
+    %          \____/
+    % 
+
+
+
+% initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+    
+    % initialize a one dimensional zero matrix to be populated
+    x = zeros(1, n);
+    
+    % populate the matrix
+    for i = 1:n
+        t = i * dt;
+        
+        % periodic ramp from 0 to 1
+        % progression through a cycle
+        st = mod(frequency * t - phase, 1);
+        slope = (amplitude/2) / (duty/8);
+        if(st < duty)
+            if(st <= duty/8)
+                x(i) = slope * st;
+            else
+                if(st <= 5*duty/8)
+                    x(i) = amplitude/2 - slope * (st-(3*duty/8));
+                end
+                if(st <= 3*duty/8)
+                    x(i) = amplitude/2;
+                end
+                if(st >= 5*duty/8)
+                    x(i) = -amplitude/2;
+                end
+            end
+            if(st >= 7* duty/8)
+                x(i) = -amplitude/2 + slope*(st-(7*duty/8));
+            end
+        end
+    end
+end
\ No newline at end of file
diff --git a/src/Helper/MIDIDarell.m b/src/Helper/MIDIDarell.m
index 19491c4..2253b73 100644
--- a/src/Helper/MIDIDarell.m
+++ b/src/Helper/MIDIDarell.m
@@ -1,6 +1,6 @@
 function output = MIDIDarell(dur,Fs)
 %Darell Chua Yun Da
-%   Detailed explanation goes here
+%   Use Qwerty Keys for each sound frequency.
 
     freq = [330 350 370 392 415 440 466 494 523 554];
     charls = ["q" "w" "e" "r" "t" "y" "u" "i" "o" "p"];
diff --git a/src/NotWorking/fade_in.m b/src/Helper/fade_in.m
similarity index 70%
rename from src/NotWorking/fade_in.m
rename to src/Helper/fade_in.m
index 22bef76..bebb07f 100644
--- a/src/NotWorking/fade_in.m
+++ b/src/Helper/fade_in.m
@@ -12,16 +12,30 @@ function output = fade_in(input, time)
     
     len = length(input);
     
+    % need to use whole number for time
+    time = round(time);
+    
     % if time parameter longer than signal, treat time as
     % the duration of original signal
     if time > len
-        time = len
+        time = len;
+    end
+    
+    % in order to create array, time >=1
+    % if not, it's arbitrarily set to 1
+    % in which case the fade_in effect
+    % is virtually nonexistent
+    if time < 1
+        time = 1;
     end
 
     % set multiplier as 1D array
     % fade in effect: from no volume to full volume of signal
     multiplier = (1 : time) / time;
-
+    
+    while length(multiplier) < len
+        multiplier = [multiplier 1];
+    end
     % the resulting fade-in output  
     output = input .* multiplier;
 end
diff --git a/src/NotWorking/fade_out.m b/src/Helper/fade_out.m
similarity index 70%
rename from src/NotWorking/fade_out.m
rename to src/Helper/fade_out.m
index f50812b..954a1bb 100644
--- a/src/NotWorking/fade_out.m
+++ b/src/Helper/fade_out.m
@@ -11,18 +11,33 @@ function output = fade_out(input, time)
     
     len = length(input);
     
+    % need to use time as a whole number
+    time = round(time);
+    
     % if time parameter longer than signal, treat time as
     % the duration of original signal
     if time > len
         time = len
     end
 
+    % in order to create array, time >=1
+    % if not, it's arbitrarily set to 1
+    % in which case the fade_in effect
+    % is virtually nonexistent
+    if time < 1
+        time = 1;
+    end
+    
     % set multiplier as 1D array
 
     multiplier = (1 : time) / time;
 
     % fade out effect: from full volume of signal to no volume
-    multiplier = flip(multiplier)
+    multiplier = flip(multiplier);
+    
+    while length(multiplier) < len
+        multiplier = [multiplier 0];
+    end 
 
     % the resulting fade-in output  
     output = input .* multiplier;
diff --git a/src/Instruments/generate_arcoStrings.m b/src/Instruments/generate_arcoStrings.m
new file mode 100644
index 0000000..51a58fb
--- /dev/null
+++ b/src/Instruments/generate_arcoStrings.m
@@ -0,0 +1,41 @@
+function x = generate_arcoStrings(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_ARCOSTRINGS: returns a matrix of sampled waveform similar to
+% strings played arco.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency .* (1 + 0.0002 .* sin(10.*pi.*t)./t); %frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (amplitude .* sin(2.*pi.*f.*t)) ... %fundamental (first partial)
+        + (0.237992723.*amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (0.0523347308.*amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (0.124586119.*amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.0854326396.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.0578741297.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.0291070237.*amplitude .* sin(14.*pi.*f.*t)) ... %seventh partial
+        + (0.0270171595.*amplitude .* sin(16.*pi.*f.*t)) ... %eighth partial
+        + (0.0290818446.*amplitude .* sin(18.*pi.*f.*t)) ... %ninth partial
+        + (0.0356787652.*amplitude .* sin(20.*pi.*f.*t)) ... %tenth partial
+        + (0.0112928202.*amplitude .* sin(22.*pi.*f.*t)) ... %eleventh partial
+        + (0.00547645126.*amplitude .* sin(24.*pi.*f.*t)) ... %twelth partial
+        + (0.00800694943.*amplitude .* sin(26.*pi.*f.*t)) ... %thirteenth partial
+        + (0.00463295187.*amplitude .* sin(28.*pi.*f.*t)); %fourteenth partial
+    x = x .* (1 + 0.1 .* sin(10.*pi.*t)); %volume vibrato
+end
+
diff --git a/src/Instruments/generate_bassoon.m b/src/Instruments/generate_bassoon.m
new file mode 100644
index 0000000..21470ea
--- /dev/null
+++ b/src/Instruments/generate_bassoon.m
@@ -0,0 +1,45 @@
+function x = generate_bassoon(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_BASSOON: returns a matrix of sampled waveform similar to
+% a bassoon.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency .* (1 + 0.0001 .* sin(4.*pi.*t)./t); %frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (0.0934091627.*amplitude .* sin(2.*pi.*f.*t)) ... %fundamental (first partial)
+        + (0.0257187994.*amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (0.120530806.*amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.216745656.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.0561137441.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.0300031596.*amplitude .* sin(14.*pi.*f.*t)) ... %seventh partial
+        + (0.0863949447.*amplitude .* sin(16.*pi.*f.*t)) ... %eighth partial
+        + (0.0574533965.*amplitude .* sin(18.*pi.*f.*t)) ... %ninth partial
+        + (0.0337819905.*amplitude .* sin(20.*pi.*f.*t)) ... %tenth partial
+        + (0.0576176935.*amplitude .* sin(22.*pi.*f.*t)) ... %eleventh partial
+        + (0.0107298578.*amplitude .* sin(24.*pi.*f.*t)) ... %twelth partial
+        + (0.00582622433.*amplitude .* sin(26.*pi.*f.*t)) ... %thirteenth partial
+        + (0.00540916272.*amplitude .* sin(28.*pi.*f.*t)) ... %fourteenth partial
+        + (0.00615481833.*amplitude .* sin(30.*pi.*f.*t)) ... %fifteenth partial
+        + (0.00480252765.*amplitude .* sin(32.*pi.*f.*t)) ... %sixteenth partial
+        + (0.0100473934.*amplitude .* sin(34.*pi.*f.*t)) ... %seventeenth partial
+        + (0.00528278041.*amplitude .* sin(36.*pi.*f.*t)); %eighteenth partial
+    x = x .* (1 + 0.1 .* sin(4.*pi.*t)); %volume vibrato
+end
+
diff --git a/src/Instruments/generate_clarinet.m b/src/Instruments/generate_clarinet.m
new file mode 100644
index 0000000..bb86808
--- /dev/null
+++ b/src/Instruments/generate_clarinet.m
@@ -0,0 +1,38 @@
+function x = generate_clarinet(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_CLARINET: returns a matrix of sampled waveform similar to
+% a clarinet.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency; %clarinets play without vibrato
+    
+    % generate proper fundamental and overtones. Note that the clarinet has
+    % heavily suppressed odd-numbered partials
+    x = (amplitude .* sin(2.*pi.*f.*t)) ... %fundamental (first partial)
+        + (0.00290397158.*amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (0.368654023.*amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (0.0733323309.*amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.299729339.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.0483901284.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.0941300303.*amplitude .* sin(14.*pi.*f.*t)) ... %seventh partial
+        + (0.033165423.*amplitude .* sin(16.*pi.*f.*t)) ... %eighth partial
+        + (0.00803526117.*amplitude .* sin(18.*pi.*f.*t)) ... %ninth partial
+        + (0.0131383569.*amplitude .* sin(20.*pi.*f.*t)); %tenth partial
+    x = x .* (1 + 0.05 .* sin(10.*pi.*t)); %slight volume vibrato added for character
+end
+
diff --git a/src/Instruments/generate_flute.m b/src/Instruments/generate_flute.m
new file mode 100644
index 0000000..0183455
--- /dev/null
+++ b/src/Instruments/generate_flute.m
@@ -0,0 +1,34 @@
+function x = generate_flute(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_FLUTE: returns a matrix of sampled waveform similar to
+% a flute.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale and frequency
+    t = [1:n].*dt - phase;
+    f = frequency .* (1 + 0.0002 .* sin(7.*pi.*t)./t); %frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (amplitude .* sin(4.*pi.*f.*t)) ... %fundamental (first partial)
+        + (0.396821591.*amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (0.157568887.*amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (0.151279648.*amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.106818975.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.0138755668.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.0100279037.*amplitude .* sin(14.*pi.*f.*t)); %seventh partial
+    x = x .* (1 + 0.25 .* sin(7.*pi.*t)); %volume vibrato
+end
+
diff --git a/src/Instruments/generate_horn.m b/src/Instruments/generate_horn.m
new file mode 100644
index 0000000..81797a5
--- /dev/null
+++ b/src/Instruments/generate_horn.m
@@ -0,0 +1,41 @@
+function x = generate_horn(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_HORN: returns a matrix of sampled waveform similar to
+% a horn.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency .* (1 + 0.0002 .* sin(4.*pi.*t)./t); %frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (0.21707879.*amplitude .* sin(2.*pi.*f.*t)) ... %fundamental (first partial)
+        + (amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (0.234667056.*amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (0.088606568.*amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.0864485981.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.0275181724.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.0472968588.*amplitude .* sin(14.*pi.*f.*t)) ... %seventh partial
+        + (0.0246462876.*amplitude .* sin(16.*pi.*f.*t)) ... %eighth partial
+        + (0.00816134476.*amplitude .* sin(18.*pi.*f.*t)) ... %ninth partial
+        + (0.00691199377.*amplitude .* sin(20.*pi.*f.*t)) ... %tenth partial
+        + (0.00494872793.*amplitude .* sin(22.*pi.*f.*t)) ... %eleventh partial
+        + (0.0048026999.*amplitude .* sin(24.*pi.*f.*t)) ... %twelth partial
+        + (0.00335864486.*amplitude .* sin(26.*pi.*f.*t)) ... %thirteenth partial
+        + (0.00262850467.*amplitude .* sin(28.*pi.*f.*t)); %fourteenth partial
+    x = x .* (1 + 0.05 .* sin(2.*pi.*t)); %volume vibrato
+end
+
diff --git a/src/Instruments/generate_oboe.m b/src/Instruments/generate_oboe.m
new file mode 100644
index 0000000..87ff0af
--- /dev/null
+++ b/src/Instruments/generate_oboe.m
@@ -0,0 +1,37 @@
+function x = generate_oboe(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_OBOE: returns a matrix of sampled waveform similar to
+% an oboe.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency .* (1 + 0.0001 .* sin(10.*pi.*t)./t); %frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (0.274577037.*amplitude .* sin(2.*pi.*f.*t)) ... %fundamental (first partial)
+        + (0.612336031.*amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (0.105870314.*amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.00825325944.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.0767380354.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.0381430831.*amplitude .* sin(14.*pi.*f.*t)) ... %seventh partial
+        + (0.010765121.*amplitude .* sin(16.*pi.*f.*t)) ... %eighth partial
+        + (0.0024454102.*amplitude .* sin(18.*pi.*f.*t)) ... %ninth partial
+        + (0.00963544782.*amplitude .* sin(20.*pi.*f.*t)); %tenth partial
+    x = x .* (1 + 0.1 .* sin(10.*pi.*t)); %volume vibrato
+end
+
diff --git a/src/Instruments/generate_organ.m b/src/Instruments/generate_organ.m
new file mode 100644
index 0000000..125e288
--- /dev/null
+++ b/src/Instruments/generate_organ.m
@@ -0,0 +1,44 @@
+function x = generate_organ(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_ORGAN: returns a matrix of sampled waveform similar to
+% an organ.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency; %no frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (0.605975181.*amplitude .* sin(2.*pi.*f.*t)) ... %fundamental (first partial)
+        + (0.96669476.*amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (0.509982853.*amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.138074341.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.397744776.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.00348745968.*amplitude .* sin(14.*pi.*f.*t)) ... %seventh partial
+        + (0.350983754.*amplitude .* sin(16.*pi.*f.*t)) ... %eighth partial
+        + (0.0326368101.*amplitude .* sin(18.*pi.*f.*t)) ... %ninth partial
+        + (0.0121770467.*amplitude .* sin(20.*pi.*f.*t)) ... %tenth partial
+        + (0.000406870296.*amplitude .* sin(22.*pi.*f.*t)) ... %eleventh partial
+        + (0.0440001162.*amplitude .* sin(24.*pi.*f.*t)) ... %twelth partial
+        + (0.00299340289.*amplitude .* sin(26.*pi.*f.*t)) ... %thirteenth partial
+        + (0.00767241129.*amplitude .* sin(28.*pi.*f.*t)) ... %fourteenth partial
+        + (0.00337121102.*amplitude .* sin(30.*pi.*f.*t)) ... %fifteenth partial
+        + (0.0389432997.*amplitude .* sin(32.*pi.*f.*t)) ... %sixteenth partial
+        + (0.00496963004.*amplitude .* sin(64.*pi.*f.*t)); %thirty-second partial
+         %no volume vibrato
+end
+
diff --git a/src/Instruments/generate_saxophone.m b/src/Instruments/generate_saxophone.m
new file mode 100644
index 0000000..8e7bbde
--- /dev/null
+++ b/src/Instruments/generate_saxophone.m
@@ -0,0 +1,46 @@
+function x = generate_saxophone(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_SAXOPHONE: returns a matrix of sampled waveform similar to
+% a saxophone.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency .* (1 + 0.0001 .* sin(9.*pi.*t)./t); %frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (0.64011635.*amplitude .* sin(2.*pi.*f.*t)) ... %fundamental (first partial)
+        + (0.558462369.*amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (0.156636999.*amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.102886106.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.289185376.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.21277411.*amplitude .* sin(14.*pi.*f.*t)) ... %seventh partial
+        + (0.108250745.*amplitude .* sin(16.*pi.*f.*t)) ... %eighth partial
+        + (0.077212478.*amplitude .* sin(18.*pi.*f.*t)) ... %ninth partial
+        + (0.285980527.*amplitude .* sin(20.*pi.*f.*t)) ... %tenth partial
+        + (0.167644958.*amplitude .* sin(22.*pi.*f.*t)) ... %eleventh partial
+        + (0.115479073.*amplitude .* sin(24.*pi.*f.*t)) ... %twelth partial
+        + (0.0345740512.*amplitude .* sin(26.*pi.*f.*t)) ... %thirteenth partial
+        + (0.0192987651.*amplitude .* sin(28.*pi.*f.*t)) ... %fourteenth partial
+        + (0.0140386324.*amplitude .* sin(30.*pi.*f.*t)) ... %fifteenth partial
+        + (0.0102938359.*amplitude .* sin(32.*pi.*f.*t)) ... %sixteenth partial
+        + (0.0166338634.*amplitude .* sin(34.*pi.*f.*t)) ... %seventeenth partial
+        + (0.0128019786.*amplitude .* sin(36.*pi.*f.*t)) ... %eighteenth partial
+        + (0.00442408514.*amplitude .* sin(38.*pi.*f.*t)); %nineteenth partial
+    x = x .* (1 + 0.1 .* sin(9.*pi.*t)); %volume vibrato
+end
+
diff --git a/src/Instruments/generate_shepardsCarillon.m b/src/Instruments/generate_shepardsCarillon.m
new file mode 100644
index 0000000..01ad5fc
--- /dev/null
+++ b/src/Instruments/generate_shepardsCarillon.m
@@ -0,0 +1,27 @@
+function x = generate_shepardsCarillon(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_SHEPARDSCARILLON: returns a matrix of sampled waveform similar to
+% tubular bells, doubled at octaves.
+% Calls generate_tubularBells
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+    
+    % generate proper tone
+    x = generate_tubularBells(amplitude/5, frequency/16, phase, fs, duration, duty) ... %4 octaves below
+        + generate_tubularBells(amplitude/4, frequency/8, phase, fs, duration, duty) ... %3 octaves below
+        + generate_tubularBells(amplitude/3, frequency/4, phase, fs, duration, duty) ... %2 octaves below
+        + generate_tubularBells(amplitude/2, frequency/2, phase, fs, duration, duty) ... %1 octave below
+        + generate_tubularBells(amplitude, frequency, phase, fs, duration, duty) ... %unison
+        + generate_tubularBells(amplitude/2, frequency*2, phase, fs, duration, duty) ... %1 octave above
+        + generate_tubularBells(amplitude/3, frequency*4, phase, fs, duration, duty) ... %2 octaves above
+        + generate_tubularBells(amplitude/4, frequency*8, phase, fs, duration, duty) ... %3 octaves above
+        + generate_tubularBells(amplitude/5, frequency*16, phase, fs, duration, duty); %4 octaves above
+end
+
diff --git a/src/Instruments/generate_shepardsOrgan.m b/src/Instruments/generate_shepardsOrgan.m
new file mode 100644
index 0000000..9cbb33f
--- /dev/null
+++ b/src/Instruments/generate_shepardsOrgan.m
@@ -0,0 +1,33 @@
+function x = generate_shepardsOrgan(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_SHEPARDSORGAN: returns a matrix of sampled waveform similar to
+% an organ, doubled at octaves.
+% Calls generate_organ
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+    
+    % generate proper tone
+    x = generate_organ(amplitude/8, frequency/128, phase, fs, duration, duty) ... %7 octaves below
+        + generate_organ(amplitude/7, frequency/64, phase, fs, duration, duty) ... %6 octaves below
+        + generate_organ(amplitude/6, frequency/32, phase, fs, duration, duty) ... %5 octaves below
+        + generate_organ(amplitude/5, frequency/16, phase, fs, duration, duty) ... %4 octaves below
+        + generate_organ(amplitude/4, frequency/8, phase, fs, duration, duty) ... %3 octaves below
+        + generate_organ(amplitude/3, frequency/4, phase, fs, duration, duty) ... %2 octaves below
+        + generate_organ(amplitude/2, frequency/2, phase, fs, duration, duty) ... %1 octave below
+        + generate_organ(amplitude, frequency, phase, fs, duration, duty) ... %unison
+        + generate_organ(amplitude/2, frequency*2, phase, fs, duration, duty) ... %1 octave above
+        + generate_organ(amplitude/3, frequency*4, phase, fs, duration, duty) ... %2 octaves above
+        + generate_organ(amplitude/4, frequency*8, phase, fs, duration, duty) ... %3 octaves above
+        + generate_organ(amplitude/5, frequency*16, phase, fs, duration, duty) ... %4 octaves above
+        + generate_organ(amplitude/6, frequency*32, phase, fs, duration, duty) ... %5 octaves above
+        + generate_organ(amplitude/7, frequency*64, phase, fs, duration, duty) ... %6 octaves above
+        + generate_organ(amplitude/8, frequency*128, phase, fs, duration, duty); %7 octaves above
+end
+
diff --git a/src/Instruments/generate_shepardsStrings.m b/src/Instruments/generate_shepardsStrings.m
new file mode 100644
index 0000000..4af1f07
--- /dev/null
+++ b/src/Instruments/generate_shepardsStrings.m
@@ -0,0 +1,33 @@
+function x = generate_shepardsStrings(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_SHEPARDSSTRINGS: returns a matrix of sampled waveform similar to
+% arco strings, doubled at octaves.
+% Calls generate_arcoStrings
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+    
+    % generate proper tone
+    x = generate_arcoStrings(amplitude/8, frequency/128, phase, fs, duration, duty) ... %7 octaves below
+        + generate_arcoStrings(amplitude/7, frequency/64, phase, fs, duration, duty) ... %6 octaves below
+        + generate_arcoStrings(amplitude/6, frequency/32, phase, fs, duration, duty) ... %5 octaves below
+        + generate_arcoStrings(amplitude/5, frequency/16, phase, fs, duration, duty) ... %4 octaves below
+        + generate_arcoStrings(amplitude/4, frequency/8, phase, fs, duration, duty) ... %3 octaves below
+        + generate_arcoStrings(amplitude/3, frequency/4, phase, fs, duration, duty) ... %2 octaves below
+        + generate_arcoStrings(amplitude/2, frequency/2, phase, fs, duration, duty) ... %1 octave below
+        + generate_arcoStrings(amplitude, frequency, phase, fs, duration, duty) ... %unison
+        + generate_arcoStrings(amplitude/2, frequency*2, phase, fs, duration, duty) ... %1 octave above
+        + generate_arcoStrings(amplitude/3, frequency*4, phase, fs, duration, duty) ... %2 octaves above
+        + generate_arcoStrings(amplitude/4, frequency*8, phase, fs, duration, duty) ... %3 octaves above
+        + generate_arcoStrings(amplitude/5, frequency*16, phase, fs, duration, duty) ... %4 octaves above
+        + generate_arcoStrings(amplitude/6, frequency*32, phase, fs, duration, duty) ... %5 octaves above
+        + generate_arcoStrings(amplitude/7, frequency*64, phase, fs, duration, duty) ... %6 octaves above
+        + generate_arcoStrings(amplitude/8, frequency*128, phase, fs, duration, duty); %7 octaves above
+end
+
diff --git a/src/Instruments/generate_trumpet.m b/src/Instruments/generate_trumpet.m
new file mode 100644
index 0000000..58d9309
--- /dev/null
+++ b/src/Instruments/generate_trumpet.m
@@ -0,0 +1,49 @@
+function x = generate_trumpet(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_TRUMPET: returns a matrix of sampled waveform similar to
+% a trumpet.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency .* (1 + 0.00005 .* sin(10.*pi.*t)./t); %frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (0.802698298.*amplitude .* sin(2.*pi.*f.*t)) ... %fundamental (first partial)
+        + (amplitude .* sin(4.*pi.*f.*t)) ... %second partial
+        + (0.700319932.*amplitude .* sin(6.*pi.*f.*t)) ... %third partial
+        + (0.748098515.*amplitude .* sin(8.*pi.*f.*t)) ... %fourth partial
+        + (0.909272003.*amplitude .* sin(10.*pi.*f.*t)) ... %fifth partial
+        + (0.471477725.*amplitude .* sin(12.*pi.*f.*t)) ... %sixth partial
+        + (0.200863214.*amplitude .* sin(14.*pi.*f.*t)) ... %seventh partial
+        + (0.264578051.*amplitude .* sin(16.*pi.*f.*t)) ... %eighth partial
+        + (0.282234698.*amplitude .* sin(18.*pi.*f.*t)) ... %ninth partial
+        + (0.128787879.*amplitude .* sin(20.*pi.*f.*t)) ... %tenth partial
+        + (0.100688156.*amplitude .* sin(22.*pi.*f.*t)) ... %eleventh partial
+        + (0.0874984909.*amplitude .* sin(24.*pi.*f.*t)) ... %twelth partial
+        + (0.0488047809.*amplitude .* sin(26.*pi.*f.*t)) ... %thirteenth partial
+        + (0.0261680551.*amplitude .* sin(28.*pi.*f.*t)) ... %fourteenth partial
+        + (0.0186224798.*amplitude .* sin(30.*pi.*f.*t)) ... %fifteenth partial
+        + (0.0255342267.*amplitude .* sin(32.*pi.*f.*t)) ... %sixteenth partial
+        + (0.0111674514.*amplitude .* sin(34.*pi.*f.*t)) ... %seventeenth partial
+        + (0.010020524.*amplitude .* sin(36.*pi.*f.*t)) ... %eighteenth partial
+        + (0.00694192925.*amplitude .* sin(38.*pi.*f.*t)) ... %nineteenth partial
+        + (0.00307859471.*amplitude .* sin(40.*pi.*f.*t)) ... %twentieth partial
+        + (0.00338041772.*amplitude .* sin(42.*pi.*f.*t)) ... %twentyfirst partial
+        + (0.0026862248.*amplitude .* sin(44.*pi.*f.*t)); %twentysecond partial
+    x = x .* (1 + 0.02 .* sin(10.*pi.*t)); %volume vibrato
+end
+
diff --git a/src/Instruments/generate_tubularBells.m b/src/Instruments/generate_tubularBells.m
new file mode 100644
index 0000000..4f22eee
--- /dev/null
+++ b/src/Instruments/generate_tubularBells.m
@@ -0,0 +1,38 @@
+function x = generate_tubularBells(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_TUBULARBELLS: returns a matrix of sampled waveform similar to
+% a tubular bell.
+
+% CONTRIBUTORS:
+% Brandon Roberts: Original author
+% MuseScore: Reference source for overtone information
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% duty does not apply for instruments
+% amplitude decays exponentially at varying rates for different paritals
+   
+
+    % setup needed variables
+    n = fs * duration; % total data points
+    dt = 1 / fs; % space between samples
+    
+    % initialize template time scale
+    t = [1:n].*dt - phase;
+    f = frequency; %no frequency vibrato
+    
+    % generate proper fundamental and overtones
+    x = (0.105668016.*amplitude .* sin(pi.*f.*t))./(2.^(t./10)) ... %hum tone
+        + (0.143724696.*amplitude .* sin(2.*pi.*f.*t))./(2.^(t./5)) ... %strike tone
+        + (0.12.*amplitude .* sin(12./5.*pi.*f.*t))./(2.^(t./3)) ... %minor third overtone
+        + (0.10.*amplitude .* sin(3.*pi.*f.*t))./(2.^(t./2.5)) ... %perfect fifth overtone
+        + (0.8.*amplitude .* sin(4.*pi.*f.*t))./(2.^(t./2)) ... %perfect octave overtone
+        + (0.6.*amplitude .* sin(5.*pi.*f.*t))./(2.^(t./1.5)) ... %major third overtone
+        + (0.4.*amplitude .* sin(8.*pi.*f.*t))./(2.^(t)) ... %second perfect fifth overtone
+        + (0.4.*amplitude .* sin(2.2.*pi.*f.*t))./(2.^(4.*t)) ... %transient 1
+        + (0.4.*amplitude .* sin(1.8.*pi.*f.*t))./(2.^(8.*t)) ... %transient 2
+        + (0.4.*amplitude .* sin(2.7.*pi.*f.*t))./(2.^(10.*t));%transient 3
+    x = x .* (1 + 0.02 .* sin(10.*pi.*t)); %volume vibrato
+end
+
diff --git a/src/LFO/lfo_freq_saw.m b/src/LFO/lfo_freq_saw.m
new file mode 100644
index 0000000..361a741
--- /dev/null
+++ b/src/LFO/lfo_freq_saw.m
@@ -0,0 +1,38 @@
+function output = lfo_freq_saw(amplitude, frequency, phase, fs, duration, input)
+%LFO_FREQ_SAW Modulates the frequency of an input with a saw LFO
+
+% CONTRIBUTORS:
+% Neelay Joglekar: Original author
+
+% SOURCES:
+% Code inspired by Benjamin Liou's lfo functions
+% Code also inspired by Darren and Anne's pitch envelopes
+% Source for frequency modulation equation:
+%   https://dsp.stackexchange.com/questions/2349/help-with-algorithm-for-modulating-oscillator-pitch-using-lfo
+
+% DOCUMENTATION:
+% Frequency is modulated by saw wave with given amplitude (in octaves, not Hz)
+
+    % initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+
+    % create output array
+    output = zeros([1, n]);
+    
+    x = 0.0; % the theoretical input index if the input were not discrete
+    % populate output
+    for i = 1:n
+        t = i * dt; % time at the i'th sample
+
+        % Increment x based off of saw wave output
+        omega_ratio = 2 ^ (amplitude * (2 * mod((t - phase / (2*pi)) * frequency, 1) - 1));
+        x = mod(x + omega_ratio, n);
+
+        % Linearly interpolate the actual indicies adjacent to x
+        % to get an output value
+        x_0 = floor(x) + 1;
+        x_1 = mod(x_0 + 1, n) + 1;
+        output(i) = (input(x_1) - input(x_0)) * (x - x_0) + input(x_0);
+    end
+end
diff --git a/src/LFO/lfo_freq_sine.m b/src/LFO/lfo_freq_sine.m
new file mode 100644
index 0000000..bef74a5
--- /dev/null
+++ b/src/LFO/lfo_freq_sine.m
@@ -0,0 +1,39 @@
+function output = lfo_freq_sine(amplitude, frequency, phase, fs, duration, input)
+%LFO_FREQ_SINE Modulates the frequency of an input with a sine LFO
+
+% CONTRIBUTORS:
+% Neelay Joglekar: Original author
+
+% SOURCES:
+% Code inspired by Benjamin Liou's lfo functions
+% Code also inspired by Darren and Anne's pitch envelopes
+% Source for frequency modulation equation:
+%   https://dsp.stackexchange.com/questions/2349/help-with-algorithm-for-modulating-oscillator-pitch-using-lfo
+
+% DOCUMENTATION:
+% Frequency is modulated by sine wave with given amplitude (in octaves, not Hz)
+
+    % initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+
+    % create output array
+    output = zeros([1, n]);
+    
+    x = 0.0; % the theoretical input index if the input were not discrete
+    % populate output
+    for i = 1:n
+        t = i * dt; % time at the i'th sample
+
+        % Increment x based off of sine wave output
+        omega_ratio = 2 ^ (amplitude * sin(2 * pi * frequency * t - phase));
+        x = mod(x + omega_ratio, n);
+
+        % Linearly interpolate the actual indicies adjacent to x
+        % to get an output value
+        x_0 = floor(x) + 1;
+        x_1 = mod(x_0 + 1, n) + 1;
+        output(i) = (input(x_1) - input(x_0)) * (x - x_0) + input(x_0);
+    end
+end
+
diff --git a/src/LFO/lfo_freq_square.m b/src/LFO/lfo_freq_square.m
new file mode 100644
index 0000000..f413e6c
--- /dev/null
+++ b/src/LFO/lfo_freq_square.m
@@ -0,0 +1,38 @@
+function output = lfo_freq_square(amplitude, frequency, phase, fs, duration, input)
+%LFO_FREQ_SQUARE Modulates the frequency of an input with a square LFO
+
+% CONTRIBUTORS:
+% Neelay Joglekar: Original author
+
+% SOURCES:
+% Code inspired by Benjamin Liou's lfo functions
+% Code also inspired by Darren and Anne's pitch envelopes
+% Source for frequency modulation equation:
+%   https://dsp.stackexchange.com/questions/2349/help-with-algorithm-for-modulating-oscillator-pitch-using-lfo
+
+% DOCUMENTATION:
+% Frequency is modulated by square wave with given amplitude (in octaves, not Hz)
+
+    % initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+
+    % create output array
+    output = zeros([1, n]);
+    
+    x = 0.0; % the theoretical input index if the input were not discrete
+    % populate output
+    for i = 1:n
+        t = i * dt; % time at the i'th sample
+
+        % Increment x based off of square wave output
+        omega_ratio = 2 ^ (amplitude * (2 * round(mod((t - phase / (2*pi)) * frequency, 1)) - 1));
+        x = mod(x + omega_ratio, n);
+
+        % Linearly interpolate the actual indicies adjacent to x
+        % to get an output value
+        x_0 = floor(x) + 1;
+        x_1 = mod(x_0 + 1, n) + 1;
+        output(i) = (input(x_1) - input(x_0)) * (x - x_0) + input(x_0);
+    end
+end
\ No newline at end of file
diff --git a/src/NotWorking/distortion_filter.m b/src/NotWorking/distortion_filter.m
new file mode 100644
index 0000000..7a83763
--- /dev/null
+++ b/src/NotWorking/distortion_filter.m
@@ -0,0 +1,23 @@
+function y = distortion_filter(input, fs, LOW, MED, HIGH)
+    %Author: Jason Liang
+    %input: 1D array representing the sound signal in the time domain
+    %fs: sampling frequency
+    %low: lower bound of frequencies
+    %med: median of all frequencies
+    %high: upper bound of frequencies
+    %y: output signal 
+    %This function distorts a signal based on the constant LOW; all
+    %frequencies in between the specified range are clipped by the constant
+    %LOW.
+    
+    len = length(input);
+    f = fs*(-len/2:len/2-1)/len;
+    
+    outputW = fftshift(fft(input));
+    for i = 1:length(outputW)
+       if ((LOW < abs(F(n))) && HIGH > abs(F(n))) 
+           outputW(i) = outputW(i) / LOW;
+       end
+    end
+    y = real(ifft(fftshift(outputW)));
+end
diff --git a/src/NotWorking/flanger.m b/src/NotWorking/flanger.m
new file mode 100644
index 0000000..ce4fa51
--- /dev/null
+++ b/src/NotWorking/flanger.m
@@ -0,0 +1,44 @@
+function output = flanger(input, time_delay, osc_freq, amp, Fs)
+
+
+% DOCUMENTATION:
+% FLANGER combines a single oscillating delayed signal with the original signal
+% Interference between original and delayed signals create "flanging" effect
+% Rocommedned delay time: 0-15 ms
+% Recommended oscilating fequency: 0.1 - 5 Hz
+
+% input: 1D array containing audio signal
+% time_delay: maxinum value of the oscillating signal
+% osc_freq: frequence of oscillation
+% amp: amplitude of output signal
+% fs: sampling frequency
+
+% CONTRIBUTORS:
+% Lance Zhu: Function Author
+
+% SOURCES:
+% Code inspired by information provided in:
+% https://users.cs.cf.ac.uk/Dave.Marshall/CM0268/PDF/10_CM0268_Audio_FX.pdf
+
+
+% create index array
+len=length(input);
+index=1:len;
+
+% create reference wave to create oscillation
+% taking only absoluted value for the reference wave
+osc_wave = abs(sin(2*pi*index*(osc_freq/Fs)));
+% convert delay in time to delay in samples
+sample_delay=round(time_delay*Fs);
+% create empty output array
+output = zeros(len,1);
+
+% Generating delay for each sample from reference oscillation wave
+% start loop at sample_delay to make sure delayed sample is from t>=0
+for i=sample_delay:len
+% i_delay: delay for current sample
+i_delay=round(osc_wave(i)*sample_delay);
+% combine delayed sample with original sample
+output(i) = (amp*input(i))+amp*(input(i-i_delay));
+
+end
\ No newline at end of file
diff --git a/src/NotWorking/generate_keyboard.m b/src/NotWorking/generate_keyboard.m
new file mode 100644
index 0000000..9547577
--- /dev/null
+++ b/src/NotWorking/generate_keyboard.m
@@ -0,0 +1,49 @@
+function x = generate_keyboard(amplitude, frequency, phase, fs, duration, duty)
+% GENERATE_KEYBOARD: returns a matrix of a sum of sine waves
+
+% CONTRIBUTORS:
+% Ryan Goh: author
+
+% DOCUMENTATION:
+% fs is the sampling frequency: how many sample points per second
+% duration is time in seconds
+% amplitude, phase, and duty are not used in this function
+
+% This function's purpose is to generate a wave that can simulate the sound of a digital keyboard.
+% Sound is best suited for frequencies near the middle octave of the
+% piano(261.6 - 523.2). Frequencies too high or too low do not sound
+% similar to the intended sound.
+
+% Referenced Arthur Lu and Benjamin Liou's generate sine function. Also
+% adapted wave model from https://dsp.stackexchange.com/questions/46598/mathematical-equation-for-the-sound-wave-that-a-piano-makes
+
+
+    % initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+    
+    % initialize a one dimensional zero matrix to be populated
+    x = zeros(1, n);
+    
+    % populate the matrix
+    for i = 1:n
+        t = i * dt; % time at the i'th sample
+        
+        %wave model of keyboard sound. 
+        %Original version from https://dsp.stackexchange.com/questions/46598/mathematical-equation-for-the-sound-wave-that-a-piano-makes
+        y = sin(2 * pi * frequency * t) * exp(-0.002 * 2 * pi * frequency * t) ...
+        + (sin(3*2*pi*frequency*t)*exp(-0.002 * 2 * pi * frequency * t)) / 2 ...
+        + (sin(4*2*pi*frequency*t)*exp(-0.002 * 2 * pi * frequency * t)) / 8 ...
+        + (sin(5 * 2 * pi * frequency * t) * exp(-0.002 * 2 * pi * frequency * t)) / 16 ...
+        + (sin(6 * 2 * pi * frequency * t) * exp(-0.002 * 2 * pi * frequency * t)) / 25 ...
+        + (sin(7 * 2 * pi * frequency * t) * exp(-0.002 * 2 * pi * frequency * t)) / 50 ...
+        + (sin(8 * 2 * pi * frequency * t) * exp(-0.002 * 2 * pi * frequency * t)) / 100 ...
+        + (sin(9 * 2 * pi * frequency * t) * exp(-0.002 * 2 * pi * frequency * t)) / 200;
+        
+        %Further adjustments to improve sound
+        y = y + y * y * y;
+        y = y * (1 + 16*t*exp(-1000*t));
+        
+        x(i) = 0.15 * y;%multiplied by fixed amplitude value of 0.15
+    end
+end
\ No newline at end of file
diff --git a/src/NotWorking/generate_lullaby.m b/src/NotWorking/generate_lullaby.m
new file mode 100644
index 0000000..83df45d
--- /dev/null
+++ b/src/NotWorking/generate_lullaby.m
@@ -0,0 +1,66 @@
+function x = generate_lullaby(amplitude, frequency, phase, fs, duration, duty)
+    % Generates sine waves that play 'twinkle twinkle little star' 
+    % by using the frequency parameter as the root note for the melody.
+    % Currently the duty and duration parameters are not used within the
+    % function as there is no need for the former and it is currently
+    % difficult to constrain the melody duration based on user input.
+    % A function such as this one could be useful as a demo for what a
+    % melody sounds like on a synthesizer, and could even find use
+    % in toys designed to help small children sleep.
+
+    % CONTRIBUTORS:
+    % generate_lullaby created by Gabriel Diaz
+
+    % DOCUMENTATION:
+    % phase shift is in number of periods
+    % fs is the sampling frequency: how many sample points per second
+    % duration is time in seconds
+    % duty is a number between 0 and 1
+
+    %NOTE: whatever's passed as the duration parameter doesn't get used at
+    %the moment, since it's currently difficult to sync every note in the
+    %melody to play in shorter duartions such that the entire song lasts about
+    %as long as 'duration'
+    duration = 0;
+    duty = 0; %duty also isn't used in this function
+
+    %current durations that the song will work with
+    note_duration = 0.4;
+    song_duration = note_duration * 48;
+
+    %frequency multiples for all notes in melody, assuming that frequency is
+    %the root note of this chord:
+    maj_second_freq = frequency * 1.125;
+    maj_third_freq = frequency * 1.25;
+    fourth_freq = frequency * 1.33;
+    fifth_freq = frequency * 1.5;
+    maj_sixth_freq = frequency * 1.7;
+
+    % initialize local variables from input arguments
+    n = fs * note_duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+    t = [1:n].*dt - phase; %initialize time scale template
+
+    %sine waves for every note in 'frequency's' respective major scale: 
+    first = amplitude * sin(2 * pi * frequency * t - phase);
+    second = amplitude * sin(2 * pi * maj_second_freq * t - phase);
+    third = amplitude * sin(2 * pi * maj_third_freq * t - phase);
+    fourth = amplitude * sin(2 * pi * fourth_freq * t - phase);
+    fifth = amplitude * sin(2 * pi * fifth_freq * t - phase);
+    sixth = amplitude * sin(2 * pi * maj_sixth_freq * t - phase);
+
+    %each one of the three melodies making up the song:
+    melody1 = [first,first,fifth,fifth,sixth,sixth,fifth,fifth]; %size 8 notes
+    melody2 = [fourth,fourth,third,third,second,second,first,first]; %size 8 notes
+    melody3 = [fifth,fifth,fourth,fourth,third,third,second,second]; %size 8 notes
+
+    % populate the matrix(total size is 44 notes wide, which means the song's
+    % duration must fit within the specified duration parameter
+    x = [melody1,melody2,melody3,melody3,melody1,melody2];
+
+    %For testing
+    %t = 0:dt:song_duration;
+    %t(n) = [];
+    %plot(t, x);
+    %sound(x, fs);
+end
\ No newline at end of file
diff --git a/src/NotWorking/lfo_square.m b/src/NotWorking/lfo_square.m
new file mode 100644
index 0000000..8480ebc
--- /dev/null
+++ b/src/NotWorking/lfo_square.m
@@ -0,0 +1,37 @@
+function x = lfo_square(amplitude, frequency, phase, fs, duration, input)
+% LFO_SQUARE: modulates an input matrix to a square
+
+% CONTRIBUTORS:
+% Aleksandra Desens: Author
+
+% DOCUMENTATION:
+% phase shift is in number of periods
+% fs is the sampling frequency - below 20 Hz
+% duration - same as input
+
+    % initialize local variables from input arguments
+    n = fs * duration; % number of samples (length of matrix)
+    dt = 1 / fs; % sampling period: time between two sample points
+    
+    % initialize a one dimensional zero matrix to be populated
+    lfo = zeros(1, n);
+    
+    % populate the matrix
+    for i = 1:n
+        
+        t = i * dt;
+        duty = 0.5;  % duty of 0.5 generates a square wave
+
+        st = mod((frequency * t - phase), 1);
+
+        if (st < duty) 
+            lfo(i) = amplitude;
+        else
+            lfo(i) = -amplitude;
+
+    end
+
+    % modulate input
+    x = lfo .* input;
+
+end
diff --git a/src/Select/AmpEnvelopeSelect.m b/src/Select/AmpEnvelopeSelect.m
index 885204a..6f9e0aa 100644
--- a/src/Select/AmpEnvelopeSelect.m
+++ b/src/Select/AmpEnvelopeSelect.m
@@ -7,7 +7,7 @@
 %Pass-through function used by app
 
 function output = AmpEnvelopeSelect(input, Fs, attack,decay,sustain,release,number)
-    if(number == "Option 1")
+    if(number == "Linear")
         output = DarellAmplitudeEnvelope(input, Fs, attack,decay,sustain,release);
     else
         output = input;
diff --git a/src/Select/FilterSelect.m b/src/Select/FilterSelect.m
index 659a0d1..20aecaf 100644
--- a/src/Select/FilterSelect.m
+++ b/src/Select/FilterSelect.m
@@ -7,18 +7,22 @@
 %Pass-through function used by app
 
 function output = FilterSelect(input,Fs,LOW,MED,HIGH,number)
-    if(number == "Option 1")
+    if(number == "IdealBandPass")
         output = DarellbandpassFilter(input,Fs,LOW,MED,HIGH);
-    elseif(number == "Option 2")
+    elseif(number == "AmplifyRange")
         output = amplifyFreqRange(input, Fs, LOW, MED, HIGH);
-    elseif(number == "Option 3")
+    elseif(number == "EpicEffect")
         output = epic_effect_schluep(input, Fs, LOW, MED, HIGH);
-    elseif(number == "Option 4")
+    elseif(number == "MuffledEffect")
         output = muffled_effect_schluep(input, Fs, LOW, MED, HIGH);
-    elseif(number == "Option 5")
+    elseif(number == "SeparatePrevalent")
         output = seperate_prevalent_schluep(input, Fs, LOW, MED, HIGH);
-    elseif(number == "Option 6")
+    elseif(number == "IdealBandReject")
         output = bandreject_filter(input, Fs, LOW, HIGH);
+    elseif(number == "EnchanceTarget")
+        output = AnuragEnhanceTarget(input, Fs, LOW, MED, HIGH);
+    elseif(number == "DampenTarget")
+        output = AnuragDampenTarget(input, Fs, LOW, MED, HIGH);
     else
         output = input;
     end
diff --git a/src/Select/LFOSelect.m b/src/Select/LFOSelect.m
index fbe132e..05bdfb7 100644
--- a/src/Select/LFOSelect.m
+++ b/src/Select/LFOSelect.m
@@ -9,10 +9,16 @@
 function output = LFOSelect(amplitude, frequency, phase, fs, duration, input,number)
 %UNTITLED Summary of this function goes here
 %   Detailed explanation goes here
-    if(number == "Option 1")
+    if(number == "AmplitudeSawtooth")
         output = lfo_sawtooth(amplitude, frequency, phase, fs, duration, input);
-    elseif(number == "Option 2")
+    elseif(number == "AmplitudeSine")
         output = lfo_sine(amplitude, frequency, phase, fs, duration, input);
+    elseif(number == "FrequencySine")
+        output = lfo_freq_sine(amplitude, frequency, phase, fs, duration, input);
+    elseif(number == "FrequencySquare")
+        output = lfo_freq_square(amplitude, frequency, phase, fs, duration, input);
+    elseif(number == "FrequencySawtooth")
+        output = lfo_freq_saw(amplitude, frequency, phase, fs, duration, input);
     else
         output = input;
     end
diff --git a/src/Select/OffsetSelect.m b/src/Select/OffsetSelect.m
index a8b982c..7bd053c 100644
--- a/src/Select/OffsetSelect.m
+++ b/src/Select/OffsetSelect.m
@@ -7,9 +7,9 @@
 %Pass-through function used by app
 
 function output = OffsetSelect(input,value,number)
-    if(number == "Option 1")
+    if(number == "Echo")
         output =  Meghaj_Echo(input, value);
-    elseif(number == "Option 2")
+    elseif(number == "PitchOffset")
         output = Petha_Hsu_PitchOffset(input, value);
     else
         output = input;
diff --git a/src/Select/PitchEnvelopeSelect.m b/src/Select/PitchEnvelopeSelect.m
index 997d30f..9d65ae9 100644
--- a/src/Select/PitchEnvelopeSelect.m
+++ b/src/Select/PitchEnvelopeSelect.m
@@ -7,9 +7,9 @@
 %Pass-through function used by app
 
 function output = PitchEnvelopeSelect(input, Fs, attack,decay,sustain,release,number)
-    if(number == "Option 1")
+    if(number == "Logarithmic")
         output = DarellAnnePitchEnvelope(input, Fs, attack,decay,sustain,release);
-    elseif(number == "Option 2")
+    elseif(number == "Linear")
         output = DarellAnneLinearPitchEnvelope(input, Fs, attack,decay,sustain,release);
     else
         output = input;
diff --git a/src/Select/SoundGeneratorSelect.m b/src/Select/SoundGeneratorSelect.m
index 3fe715a..35ebd64 100644
--- a/src/Select/SoundGeneratorSelect.m
+++ b/src/Select/SoundGeneratorSelect.m
@@ -1,26 +1,62 @@
-%Written by Darell 
+%Written by Darell, edited by Arthur Lu
 
 % CONTRIBUTORS:
 % Person1: Darell
+% Person2: Arthur Lu
 
 % DOCUMENTATION:
 %Pass-through function used by app
 
 function output = SoundGeneratorSelect(amplitude, frequency, phase, fs, duration, duty,number)
-    if(number == "Option 1")
+    if(number == "Sine")
         output = generate_sine(amplitude, frequency, phase, fs, duration, duty);
-    elseif(number == "Option 2")
+    elseif(number == "Square")
         output = generate_square(amplitude, frequency, phase, fs, duration, duty);
-    elseif(number == "Option 3")
+    elseif(number == "Triangle")
         output = generate_triangle(amplitude, frequency, phase, fs, duration, duty);
-    elseif(number == "Option 4")
+    elseif(number == "Sawtooth")
         output = generate_sawtooth(amplitude, frequency, phase, fs, duration, duty);
-    elseif(number == "Option 5")
+    elseif(number == "WhiteNoise")
         output = generate_white(amplitude, fs, duration);
-    elseif(number == "Option 6")
+    elseif(number == "HalfCircle")
         output = generate_halfCircles(amplitude, frequency, phase, fs, duration, duty);
-    else
-        output = 0;
-    end
-end 
+    elseif(number == "Trapezoid")
+        output = generate_trapezoid(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Pulse")
+        output = generate_pulse(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Cosine")
+        output = generate_cosine(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Heartbeat")
+        output = generate_heartbeat(amplitude, frequency, phase, fs, duration, duty);
 
+    % instruments:
+    elseif(number == "ArcoStrings")
+        output = generate_arcoStrings(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Bassoon")
+        output = generate_bassoon(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Clarinet")
+        output = generate_clarinet(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Flute")
+        output = generate_flute(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Horn")
+        output = generate_horn(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Oboe")
+        output = generate_oboe(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Organ")
+        output = generate_organ(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Saxophone")
+        output = generate_saxophone(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "ShepardsCarillon")
+        output = generate_shepardsCarillon(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "ShepardsOrgan")
+        output = generate_shepardsOrgan(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "ShepardsStrings")
+        output = generate_shepardsStrings(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "Trumpet")
+        output = generate_trumpet(amplitude, frequency, phase, fs, duration, duty);
+    elseif(number == "TubularBells")
+        output = generate_tubularBells(amplitude, frequency, phase, fs, duration, duty);
+    else
+        output = zeros(1, fs * duration);
+    end
+end
\ No newline at end of file