diff --git a/App/app.mlapp b/App/app.mlapp
index 2aea4c0..ac57d04 100644
Binary files a/App/app.mlapp and b/App/app.mlapp differ
diff --git a/src/NotWorking/AnuragDampenTarget.m b/src/NotWorking/AnuragDampenTarget.m
index e15f1f3..3b325d3 100644
--- a/src/NotWorking/AnuragDampenTarget.m
+++ b/src/NotWorking/AnuragDampenTarget.m
@@ -26,7 +26,6 @@ function output_x = AnuragDampenTarget(x, Fs,LOW, MID, HIGH)
             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));
diff --git a/src/NotWorking/AnuragEnhanceTarget.m b/src/NotWorking/AnuragEnhanceTarget.m
index ad2b187..97d990a 100644
--- a/src/NotWorking/AnuragEnhanceTarget.m
+++ b/src/NotWorking/AnuragEnhanceTarget.m
@@ -15,6 +15,7 @@ function output_x = AnuragEnchanceTarget(x, Fs,LOW, MID, HIGH)
     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;
diff --git a/src/NotWorking/distortion_filter.m b/src/NotWorking/distortion_filter.m
index e60f7cd..869aab7 100644
--- a/src/NotWorking/distortion_filter.m
+++ b/src/NotWorking/distortion_filter.m
@@ -1,26 +1,30 @@
-% An audio is distorted based on the value of HIGH. Nothing of a certain
-% threshold should be played above this constant, or they are simply clipped to this 
-% value. Inspired from Meghaj_Echo.m and epic_effect_schluep.m.
-% Author: Jason Liang
-
-function y = distortion_filter(x, HIGH)
-  len = length(X);
-  X = fft(x);
-  X = fftshift(X);
-  Y = zeros(1, len);
-  
-  for ind = 1:len
-    if (X(ind) > HIGH) 
-       Y(ind) = HIGH;
-    elseif (X(ind) < -HIGH)
-      Y(ind) = -HIGH;
-    else 
-      Y(ind) = X(ind);
-    end
-  end
-  
-  Y = fftshift(Y);
-  y = ifft(Y);
-  y = real(y);
-
-end
+% An audio is distorted based on the value of HIGH. Nothing of a certain
+% threshold should be played above this constant, or they are simply clipped to this 
+% value. Inspired from Meghaj_Echo.m and epic_effect_schluep.m.
+% Author: Jason Liang
+% Mekhi Ellington: Added some comments and editted formatting
+
+function y = distortion_filter(x, HIGH)
+  len = length(X);    %Storing length of X.
+  X = fft(x);         %X is the Fourier Transform of x.
+  X = fftshift(X);    %Shifts X.
+  Y = zeros(1, len);  %Matrix of length containing zeros.
+  
+  for ind = 1:len
+    if (X(ind) > HIGH) 
+       Y(ind) = HIGH;
+
+    elseif (X(ind) < -HIGH)
+      Y(ind) = -HIGH;
+
+    else 
+      Y(ind) = X(ind);
+
+    end
+  end
+  
+  Y = fftshift(Y); %Shifts Y.
+  y = ifft(Y); %Inverse Fourier Transform of Y.
+  y = real(y); %Stores only the real part of the complex y.
+
+end
\ No newline at end of file
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 6e5c61c..67bd924 100644
--- a/src/Select/FilterSelect.m
+++ b/src/Select/FilterSelect.m
@@ -7,22 +7,26 @@
 %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 == "Option 7")
-        output = AnuragEnhanceTarget(input, Fs, LOW,MID, HIGH);
-    elseif(number == "Option 8")
-        output = AnuragDampenTarget(input, Fs, LOW,MID, 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..b33d90e 100644
--- a/src/Select/LFOSelect.m
+++ b/src/Select/LFOSelect.m
@@ -9,9 +9,9 @@
 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 == "Sawtooth")
         output = lfo_sawtooth(amplitude, frequency, phase, fs, duration, input);
-    elseif(number == "Option 2")
+    elseif(number == "Sine")
         output = lfo_sine(amplitude, frequency, phase, fs, duration, input);
     else
         output = input;
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/distortion_filter.m b/src/distortion_filter.m
deleted file mode 100644
index 869aab7..0000000
--- a/src/distortion_filter.m
+++ /dev/null
@@ -1,30 +0,0 @@
-% An audio is distorted based on the value of HIGH. Nothing of a certain
-% threshold should be played above this constant, or they are simply clipped to this 
-% value. Inspired from Meghaj_Echo.m and epic_effect_schluep.m.
-% Author: Jason Liang
-% Mekhi Ellington: Added some comments and editted formatting
-
-function y = distortion_filter(x, HIGH)
-  len = length(X);    %Storing length of X.
-  X = fft(x);         %X is the Fourier Transform of x.
-  X = fftshift(X);    %Shifts X.
-  Y = zeros(1, len);  %Matrix of length containing zeros.
-  
-  for ind = 1:len
-    if (X(ind) > HIGH) 
-       Y(ind) = HIGH;
-
-    elseif (X(ind) < -HIGH)
-      Y(ind) = -HIGH;
-
-    else 
-      Y(ind) = X(ind);
-
-    end
-  end
-  
-  Y = fftshift(Y); %Shifts Y.
-  y = ifft(Y); %Inverse Fourier Transform of Y.
-  y = real(y); %Stores only the real part of the complex y.
-
-end
\ No newline at end of file
diff --git a/src/generate_cosine.m b/src/generate_cosine.m
deleted file mode 100644
index 8a013ed..0000000
--- a/src/generate_cosine.m
+++ /dev/null
@@ -1,25 +0,0 @@
-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/generate_trapezoid.m b/src/generate_trapezoid.m
deleted file mode 100644
index 69ce591..0000000
--- a/src/generate_trapezoid.m
+++ /dev/null
@@ -1,55 +0,0 @@
-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 || st > 7*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
-        end
-    end
-end
\ No newline at end of file