Fixed Functions Nicolas Schluep

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.
 
+n = size(input, 2);
+
+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
 
-Len = length(non_stereophonic);
+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

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.
 
+n = size(input, 2);
+
+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
 
-Len = length(non_stereophonic);
+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
+
 
 
 

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.
 
+n = size(input, 2);
+
+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
 
-Len = length(non_stereophonic);
+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
+
 
 
     

src/Select/FilterSelect.m

@@ -11,17 +11,14 @@ function output = FilterSelect(input,Fs,LOW,MED,HIGH,number)
         output = DarellbandpassFilter(input,Fs,LOW,MED,HIGH);
     elseif(number == "AmplifyRange")
         output = amplifyFreqRange(input, Fs, LOW, MED, HIGH);
-    %{
     elseif(number == "EpicEffect")
         output = epic_effect_schluep(input, Fs, LOW, MED, HIGH);
     elseif(number == "MuffledEffect")
         output = muffled_effect_schluep(input, Fs, LOW, MED, HIGH);
     elseif(number == "SeparatePrevalent")
         output = seperate_prevalent_schluep(input, Fs, LOW, MED, HIGH);
-    %}
     elseif(number == "IdealBandReject")
         output = bandreject_filter(input, Fs, LOW, HIGH);
-    
     elseif(number == "EnchanceTarget")
         output = AnuragEnhanceTarget(input, Fs, LOW, MED, HIGH);
     elseif(number == "DampenTarget")