diff --git a/DarellsAnnex/app.mlapp b/DarellsAnnex/app.mlapp
index 3fb8dca..763205b 100644
Binary files a/DarellsAnnex/app.mlapp and b/DarellsAnnex/app.mlapp differ
diff --git a/src/DarellAnnePitchEnvelope.m b/src/DarellAnnePitchEnvelope.m
new file mode 100644
index 0000000..dc147f6
--- /dev/null
+++ b/src/DarellAnnePitchEnvelope.m
@@ -0,0 +1,57 @@
+%Written by Darell and Anne
+
+function output = DarellAnnePitchEnvelope(input, Fs, attack,decay,sustain,release) %percentages for attack, decay, sustain, release
+    len = length(input);
+    T = (len-1)/Fs;
+    attacktime = attack * T * Fs;
+    decaytime = attacktime + decay * T * Fs;
+    sustaintime = (T - (release * T)) * Fs;
+    
+    output =  zeros([1,len]); 
+    
+    tcounter = 1;
+    %attack phase
+    curr = attacktime;
+    while tcounter <= curr
+        ncount = round(curr*log(tcounter)/log(curr)+1);
+        output(tcounter) = input(ncount);
+        tcounter = tcounter+1;
+    end
+    
+    %decay phase
+    prevcur = curr;
+    tcounter = prevcur;
+    curr = decaytime;
+    while tcounter <= curr
+        ncount = round(sustain*curr*(1-log(tcounter)/log(prevcur)) + prevcur);
+        tcounter = round(tcounter);
+        output(tcounter)
+        output(tcounter) = input(ncount);
+        tcounter = tcounter+1;
+    end
+    
+    %sustain phase
+    prevncount = ncount;
+    prevcur = curr;
+    tcounter = prevcur;
+    curr = sustaintime;
+    while tcounter <= curr
+        ncount = round(sustain*(tcounter - prevcur) + prevncount);
+        tcounter = round(tcounter);
+        output(tcounter) = input(ncount);
+        tcounter = tcounter+1;
+    end
+
+    %release phase
+    prevncount = ncount;
+    prevcur = curr;
+    tcounter = prevcur;
+    curr = Fs;
+    while tcounter <= Fs
+        ncount = round(curr*(1-log(tcounter)/log(prevcur)) + prevncount);
+        tcounter = round(tcounter);
+        output(tcounter) = input(ncount);
+        tcounter = tcounter+1;
+    end
+
+end
\ No newline at end of file
diff --git a/src/FilterSelect.m b/src/FilterSelect.m
new file mode 100644
index 0000000..abe6325
--- /dev/null
+++ b/src/FilterSelect.m
@@ -0,0 +1,8 @@
+function output = FilterSelect(input,Fs,LOW,MED,HIGH,number)
+    if(number == "Option 1")
+        output = DarellbandpassFilter(input,Fs,LOW,MED,HIGH);
+    else
+        output = input;
+    end
+end 
+
diff --git a/src/Main_test.m b/src/Main_test.m
index 31f53eb..1ebd646 100644
--- a/src/Main_test.m
+++ b/src/Main_test.m
@@ -13,7 +13,7 @@ x = x + generate_sine(amplitude, (frequency+5), phase, fs, duration, duty);
 
 %play over 5 counts, should hear both frequencies, 5 beats per second between the 2 frequencies
 playtime = 5;
-play_continuous(x, fs, playtime)
+%play_continuous(x, fs, playtime)
 
 LOW = 0;
 HIGH = frequency + 1;
@@ -22,7 +22,14 @@ x = DarellbandpassFilter(x,fs,LOW,MED,HIGH);
 
 %play over 5 counts, should only hear 200hz
 playtime = 5;
-play_continuous(x, fs, playtime)
+%play_continuous(x, fs, playtime)
+
+attack = 0.2;
+decay = 0.1;
+sustain = 0.8;
+release = 0.4;
+
+x = DarellAnnePitchEnvelope(x, Fs, attack,decay,sustain,release);
 
 attack = 0.2;
 decay = 0.2;
diff --git a/src/Petha_Hsu_PitchOffset.m b/src/Petha_Hsu_PitchOffset.m
index 908accf..feaa922 100644
--- a/src/Petha_Hsu_PitchOffset.m
+++ b/src/Petha_Hsu_PitchOffset.m
@@ -1,50 +1,50 @@
-% Petha_Hsu_PitchOffset: input a wave and pitch offset (in percentage)
-% This function takes a input sound signal and increases the pitch by the
-% offset percentage. The output is not 100% accurate. It is only an
-% estimation as the information we the function works with is very limited.
-% If 100% output is expected, more information like the frequency and type
-% of wave would be required. And the estimation only works for
-% f_offset > 0.5. Below 0.5, the output is problematic. Again, the output
-% is just a good estimation.
-
-% CONTRIBUTORS:
-% Pethaperumal Natarajan: I figured a way to increase the pitch of input
-% signal by using fourier transform and shifting the frequency using a for
-% loop. Then I used inverse transform to get a sound signal back in the time
-% domain.
-% Wesley Hsu: I helped solve the problem of being unable to lower frequency
-% for the loop using the floor function. This allowed rounding to maintain
-% the lower signals that were needed when the code returned the signal back
-% into the time domain.
-
-function y = Petha_Hsu_PitchOffset(x, f_offset)
-    len = length(x);
-    X = fft(x);
-    X = fftshift(X);        %Fourier transform the input wave
-    Y = zeros(1, len);
-    
-    midpoint = len/2;
-    for i = 1:len
-        %Shifting the Fourier transform in frequency domain to adjust the
-        %frequency of signal.
-        %Floor function is used as signals must be integers and not
-        %doubles.
-        if floor((i - midpoint) / f_offset + midpoint) < 1 || floor((i - midpoint) / f_offset + midpoint) > len
-            continue;
-        end
-        Y(i) = X(floor((i - midpoint) / f_offset + midpoint));
-    end
-    
-    %Plotted graphs to troubleshoot the problem.
-    %Fs = 44800;
-    %f = Fs *(-len/2 : len/2 -1) / len;
-    %tiledlayout(1,3); nexttile;
-    %plot(f, abs(X)); title("input"); nexttile;
-    %plot(f, abs(Y)); title("output"); nexttile;
-    
-    
-    Y = fftshift(Y);
-    y = ifft(Y);
-    y = real(y);
-        
+% Petha_Hsu_PitchOffset: input a wave and pitch offset (in percentage)
+% This function takes a input sound signal and increases the pitch by the
+% offset percentage. The output is not 100% accurate. It is only an
+% estimation as the information we the function works with is very limited.
+% If 100% output is expected, more information like the frequency and type
+% of wave would be required. And the estimation only works for
+% f_offset > 0.5. Below 0.5, the output is problematic. Again, the output
+% is just a good estimation.
+
+% CONTRIBUTORS:
+% Pethaperumal Natarajan: I figured a way to increase the pitch of input
+% signal by using fourier transform and shifting the frequency using a for
+% loop. Then I used inverse transform to get a sound signal back in the time
+% domain.
+% Wesley Hsu: I helped solve the problem of being unable to lower frequency
+% for the loop using the floor function. This allowed rounding to maintain
+% the lower signals that were needed when the code returned the signal back
+% into the time domain.
+
+function y = Petha_Hsu_PitchOffset(x, f_offset)
+    len = length(x);
+    X = fft(x);
+    X = fftshift(X);        %Fourier transform the input wave
+    Y = zeros(1, len);
+    
+    midpoint = len/2;
+    for i = 1:len
+        %Shifting the Fourier transform in frequency domain to adjust the
+        %frequency of signal.
+        %Floor function is used as signals must be integers and not
+        %doubles.
+        if floor((i - midpoint) / f_offset + midpoint) < 1 || floor((i - midpoint) / f_offset + midpoint) > len
+            continue;
+        end
+        Y(i) = X(floor((i - midpoint) / f_offset + midpoint));
+    end
+    
+    %Plotted graphs to troubleshoot the problem.
+    %Fs = 44800;
+    %f = Fs *(-len/2 : len/2 -1) / len;
+    %tiledlayout(1,3); nexttile;
+    %plot(f, abs(X)); title("input"); nexttile;
+    %plot(f, abs(Y)); title("output"); nexttile;
+    
+    
+    Y = fftshift(Y);
+    y = ifft(Y);
+    y = real(y);
+        
 end
\ No newline at end of file
diff --git a/src/PitchEnvelopeSelect.m b/src/PitchEnvelopeSelect.m
index 1cbe9d8..91bc91c 100644
--- a/src/PitchEnvelopeSelect.m
+++ b/src/PitchEnvelopeSelect.m
@@ -1,6 +1,6 @@
 function output = PitchEnvelopeSelect(input, Fs, attack,decay,sustain,release,number)
     if(number == "Option 1")
-        output = DarellAmplitudeEnvelope(input, Fs, attack,decay,sustain,release);
+        output = DarellAnnePitchEnvelope(input, Fs, attack,decay,sustain,release);
     else
         output = input;
     end