Merge branch 'main' into DarellsAnnex

implemented.csv

@@ -0,0 +1,9 @@
+amplifyFreqRange.m, Filter, Connor Hsu
+DarellAmplitudeEnvelope.m, AmpEnvelope, Darrell Chua
+DarellAnnePitchEnvelope.m, PitchEnvelope, Darrel Chua / Anne Lin
+DarellbandpassFilter.m, Filter, Darrel Chua
+generate_sawtooth, Generator, Ben Zhang
+generate_sine, Generator, Arthur Lu / Benjamin Liou
+generate_square, Generator, Arthur Lu / Benjamin Liou
+generate_triangle, Generator, Arthur Lu / Benjamin Liou
+generate_white, Generator, Benjamin Liou

src/Daniel_Doan_convolution.m

@@ -0,0 +1,17 @@
+function x = Daniel_Doan_convolution(f,h)
+    %input: two 1d arrays representing two sound signals in the time domain
+    %output: the convolution of the two waves, which is the inverse FT of
+    %FT(f)*FT(h)
+    %author: Daniel Doan
+
+    %padding to ensure the entire convolution is calculated
+    pad = length(f) + length(h) - 1;
+    %take FT of f
+    F = fft(f, pad);
+    %take FT of h
+    H = fft(h, pad);
+    %multiply the two FTs
+    X = F .* H;
+    %take inverse FT of the product
+    x = ifft(X);
+end

src/DarellAnnePitchEnvelope.m

@@ -1,5 +1,21 @@
 %Written by Darell and Anne
-%This envelope uses linear calculations
+%If there is a frequency of 200Hz:
+%1. it needs to ramp up a frequency from 0Hz to the 200Hz over the attack time
+%2. It needs to ramp down to a set sustained frequency over the decay time e.g. 160Hz < 200Hz
+%3. It maintains this 160Hz until the release time
+%4. Release time: It decays from 160Hz further all the way back to 0Hz. 
+%This envelope uses logarithmic calculations
+
+% CONTRIBUTORS:
+% Person1: Darell
+% Person2: Anne
+
+% 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
+
 
 function output = DarellAnnePitchEnvelope(input, Fs, attack,decay,sustain,release) %percentages for attack, decay, sustain, release
     len = length(input);

src/Meghaj_Echo.m

@@ -0,0 +1,37 @@
+% Meghaj_Echo: input a wave (in time domain) and a frequency to induce an 
+% echo/lag effect to. The outputted wave amplifies frequencies above the 
+% cutoff and creates an echo in the frequencies below the cutoff creating 
+% a beat lag effect. Inspired by "muffled_effect_schluep" and lecture notes
+% Works best on songs that have a clear snare line with a frequency of 
+% HIGH = 1000. Use on song files like "Strong-Bassline.mp3"
+
+% CONTRIBUTORS:
+% Meghaj Vadlaputi: Function Author
+
+
+
+function y = Meghaj_Echo(x, HIGH)
+    len = length(x);
+    X = fft(x);
+    X = fftshift(X);        %Fourier transform the input wave
+    Y = zeros(1, len);
+    
+    for ind = 1:len
+        %Multiplying the Fourier transform in frequency domain by e^jw(0.05)
+        %to induce a time shift of 0.05 seconds creating the "lag" effect on
+        %frequencies below HIGH (HIGH = 1000 works best)
+        %Multiplying the remaining signal by 1.25 amplifies other
+        %frequencies to balance
+        if abs(X(ind)) < HIGH
+            Y(ind) = X(ind) + 0.5*(X(ind)*exp(1i*ind*0.05));
+        else
+            Y(ind) = 1.25*X(ind);
+        end
+    end
+    
+    
+    Y = fftshift(Y);
+    y = ifft(Y);
+    y = real(y);
+        
+end

src/bandreject_filter.m

@@ -0,0 +1,62 @@
+function output_y = 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
+    % (basically exliminates them)
+    % By Yalu Ouyang
+    
+
+    % Input: the input signal in the time domain
+    % Fs: the sampling frequency
+    % Low: the lower limit of the specified range
+    % High: the upper limit of the specified range
+    % Returns Output: the filtered signal in the time domain
+
+    len = length(Input);
+    
+    F = Fs * (-len/2 : (len/2 - 1)) / len ;
+    
+    % modified signal in the frequency domain
+    % using Fourier Transform
+    mod_freq = fftshift(fft(Input));
+    
+    len_f = length(mod_freq);
+    
+    % use this array to record the frequencies
+    % that should pass through
+    % 0 indicates reject
+    % 1 indicates pass
+    multiplier =  zeros([1,len_f]); 
+    
+    for index = 1 : len_f
+
+        % within range of band reject
+        % so elminate these frequencies
+        if ((Low < abs(F(index))) && (abs(F(index)) < High))
+            multiplier(index) = 0;
+        
+        % outside of specified range
+        % so shoudln't be altered
+        else
+            multiplier(index) = 1;
+        end
+    end
+    
+    % filtered signal in the frequency domain
+    filtered_mod_freq = fftshift(mod_freq .* multiplier);
+    
+    % convert signal back to the time domain
+    Output = real(ifft(filtered_mod_freq));
+    
+end
+
+% This function is useful for eliminating
+% unwanted signals that have frequencies close to the 
+% median frequency of the original signal
+% (consider overall frequencies as one part, 
+% this elminates the middle portion)
+
+% Fourier transform is applied in this function
+% to make it easier to eliminate specified 
+% frequencies of the signal
+% (easier to do so in the frequency domain)

src/epic_effect_schluep.m

@@ -0,0 +1,55 @@
+function output = epic_effect_schluep(input, Fs, LOW, MED, HIGH)
+% epic_effect_schluep: Outputs a more "epic" version of the input sound.
+% This is done by amplifying low frequencies and by implementing a chorus
+% effect. A chorus effect is created when multiple copies of sound delayed
+% by a small, random amount are added to the original signal. Works best on
+% songs with a stronger bassline.
+% Try this function out with "Strong_Bassline.mp3".
+
+% CONTRIBUTORS:
+% Nicolas Schluep: Function Author
+
+% DOCUMENTATION:
+% input: The input sound in the time-domain.
+% Fs: The sampling rate of the input signal. A typical value is 44100 Hz.
+% HIGH: The maximum frequency the filter will amplify. A typical value for 
+% this variable is 1000 Hz.
+
+non_stereophonic = input(:, 1);     % Removes the sterophonic property of the input sound
+% by just taking the first column of data.
+
+Len = length(non_stereophonic);
+F = Fs * ((-Len/2) : ((Len/2) - 1)) / Len;  % Creating the array of frequencies
+% which the FFT Shifted version of the signal can be plotted against.
+inputFreq = fftshift(fft(non_stereophonic));   % Creates the Fourier Transform of the
+% input signal. fftshift() makes it such that the zero frequency is at the 
+% center of the array.
+lowAmplifyFilter = zeros(1, length(inputFreq));
+
+% Creating a filter which amplifies lower frequencies.
+for i = 1:length(lowAmplifyFilter)
+    if abs(F(i)) < HIGH
+        lowAmplifyFilter(i) = 1.25;
+    else
+        lowAmplifyFilter(i) = 1.00;
+    end
+end
+
+lowPassedInput = inputFreq .* transpose(lowAmplifyFilter); %Apply the "lowAmplifyFilter".
+
+% Adding the chorus effect.
+realOutput = real(ifft(fftshift(lowPassedInput)));
+output = realOutput;
+
+% Adding 100 randomly delayed signals to the original signal which creates the chorus effect.
+for i = 1:100
+    currentDelay = 0.003 * rand();  % The current delay of the sound in seconds.
+    currentIndex = round(currentDelay * Fs);    % Find the first index where the sound should start playing.
+    delayedOutput = [zeros(currentIndex, 1); realOutput];      % Adds "currentIndex" zeros to the front of the 
+    % "realOutput" vector to create a slightly delayed version of the signal.
+    delayedOutput = delayedOutput(1:length(realOutput));    % Truncates the "delayedOutput"
+    % vector so that it can be added to the "realOutput" vector.
+    output = output + delayedOutput; 
+end
+
+output = output ./ 100;     % Divide by 100 to decrease the amplitude of the sound to a normal level.

src/fade_in.m

@@ -0,0 +1,29 @@
+function output = fade_in(input, time)
+    % Creates a fade-in sound effect that lasts a given
+    % time parameter of the input sound signal
+    % By Yalu Ouyang
+    
+
+    % input: a 1D array that represents the sound signal in the time domain
+    % time: how long the fade in effect should last
+    % Shouldn't be longer than the input signal (in which case the function
+    % treats it as the duration of the signal)
+    % Returns modified signal in the time domain (output).
+    
+    len = length(input);
+    
+    % if time parameter longer than signal, treat time as
+    % the duration of original signal
+    if time > len
+        time = len
+    end
+
+    % set multiplier as 1D array
+    % fade in effect: from no volume to full volume of signal
+    multiplier = (1 : time) / time;
+
+    % the resulting fade-in output  
+    output = input .* multiplier;
+end
+
+% This is useful for making videos, specifically the intro part 

src/fade_out.m

@@ -0,0 +1,32 @@
+function output = fade_out(input, time)
+    % Creates a fade-out sound effect that lasts a given
+    % time parameter of the input sound signal
+    % By Yalu Ouyang
+    
+    % input: a 1D array that represents the sound signal in the time domain
+    % time: how long the fade out effect should last
+    % Shouldn't be longer than the input signal 
+    % (in which case the function treats it as the duration of the signal)
+    % Returns modified signal in the time domain (output).
+    
+    len = length(input);
+    
+    % if time parameter longer than signal, treat time as
+    % the duration of original signal
+    if time > len
+        time = len
+    end
+
+    % set multiplier as 1D array
+
+    multiplier = (1 : time) / time;
+
+    % fade out effect: from full volume of signal to no volume
+    multiplier = flip(multiplier)
+
+    % the resulting fade-in output  
+    output = input .* multiplier;
+end
+
+
+% This is useful for creating videos, specifically the outro part

src/generate_halfCircles.m

@@ -0,0 +1,44 @@
+function x = generate_halfCircles(amplitude, frequency, phase, fs, duration, duty)
+    % Generates half circles.
+
+    % By Conner Hsu
+
+    % DOCUMENTATION:
+    % amplitude scales how tall the half circle is.
+    % 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 or less would return 0 for all sample points
+        % duty of 0.25 would return a half circle for first quarter of each cycle
+        % then return 0 for the remaining 3/4ths
+    % duty of 1 would return all +amplitude
+
+    % 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
+        
+        % periodic ramp from 0 to 1
+        % progression through a cycle
+        st = mod(frequency * t - phase, 1);
+        
+        if(st < duty)
+            x(i) = sqrt((duty/2)^2-(st-duty/2)^2)/2*amplitude;
+        else
+            x(i) = 0;
+        end
+    end
+    %Testing code.
+    %t = 0:dt:duration;
+    %t(n) = [];
+    %plot(t, x);
+    sound(x, fs);
+end

src/lfo_sawtooth.m

@@ -0,0 +1,37 @@
+function x = lfo_sawtooth(amplitude, frequency, phase, fs, duration, input)
+% LFO_SAWTOOTH: modulates an input matrix to a sawtooth parameter
+
+% CONTRIBUTORS:
+% Ben Zhang: Function Author
+
+% DOCUMENTATION:
+% frequency is below 20Hz for people to hear the sound
+% fs and duration should be 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 lfo, which will be used to modulate the input
+    lfo = zeros(1, n);
+    period = 1 / frequency; % period of the wave
+    slope = 2 * amplitude / period; % the incline slope from start to amplitude
+     
+    % populate lfo matrix
+    for i = 1:n
+        t = i * dt; % time at the i'th sample
+        st = mod(frequency * t - phase, 1); % Progression through cycle
+        %part before the straght vertical line
+        if(st < period /2) 
+            lfo(i) = amplitude * slope;
+        %part after the straght vertical line
+        else
+            lfo(i) = amplitude * (slope - 1);
+    
+    end
+    
+    % modulate input
+    x = lfo .* input;
+    
+end

src/muffled_effect_schluep.m

@@ -0,0 +1,56 @@
+function output = muffled_effect_schluep(input, Fs, LOW, MED, HIGH)
+% muffled_effect_schluep: Outputs a muffled version of the the original input
+% sound in the time domain. This makes it sound as if the audio is being 
+% played in another room.
+% Try this function out with "Strong_Bassline.mp3".
+
+% CONTRIBUTORS:
+% Nicolas Schluep: Function Author
+
+% DOCUMENTATION:
+% input: The input sound in the time-domain.
+% Fs: The sampling rate of the input signal. A typical value is 44100 Hz.
+% HIGH: The maximum frequency that the low-pass filter will let pass. A
+% typical value for this variable is 1000 Hz.
+
+non_stereophonic = input(:, 1);     % Removes the sterophonic property of the input sound
+% by just taking the first column of data.
+
+Len = length(non_stereophonic);
+F = Fs * ((-Len/2) : ((Len/2) - 1)) / Len;  % Creating the array of frequencies
+% which the FFT Shifted version of the signal can be plotted against.
+inputFreq = fftshift(fft(non_stereophonic));   % Creates the Fourier Transform of the
+% input signal. fftshift() makes it such that the zero frequency is at the 
+% center of the array.
+lowPassFilter = zeros(1, length(inputFreq));
+
+% Creating Low Pass Filter
+for i = 1:length(lowPassFilter)
+    if abs(F(i)) < HIGH
+        lowPassFilter(i) = 1;
+    else
+        lowPassFilter(i) = 0;
+    end
+end
+
+lowPassedInput = inputFreq .* transpose(lowPassFilter); %Apply the low-pass filter.
+
+% Adding a slight reverb effect.
+realOutput = real(ifft(fftshift(lowPassedInput)));
+
+delay = 0.001;     % The delay of the sound in seconds.
+index = round(delay*Fs);    % Find the first index where sound should start playing
+% by multiplying the delay by the sampling frequency.
+delayedOutput = [zeros(index, 1); realOutput];      % Adds "index" zeros to the front of the 
+% "realOutput" vector to create a slightly delayed version of the signal.
+delayedOutput = delayedOutput(1:length(realOutput));    % Truncates the "delayedOutput"
+% vector so that it can be added to the "realOutput" vector.
+
+output = (realOutput + delayedOutput) ./ 2.0;    % Adds the "realOutput" and "delayedOutput"
+% vectors to create a reverb effect. Divides by 2 to avoid clipping
+% effects.
+
+
+
+
+

src/seperate_prevalent_schluep.m

@@ -0,0 +1,62 @@
+function output = seperate_prevalent_schluep(input, Fs, LOW, MED, HIGH)
+% seperate_prevalent_schluep: Attempts to seperate the most prevalent frequencies
+% from the input sound by finding the most prevalent frequencies and applying
+% a band-pass filter to a small region around those frequencies.
+% Try this function out with "Strong_Bassline.mp3".
+
+% CONTRIBUTORS:
+% Nicolas Schluep: Function Author
+
+% DOCUMENTATION:
+% input: The input sound in the time-domain.
+% Fs: The sampling rate of the input signal. A typical value is 44100 Hz.
+% HIGH: The maximum distance around the maximum frequency value that will
+% not be attenuated. A good range of values is usually 250-500 Hz, but it
+% depends on the input sound.
+
+non_stereophonic = input(:, 1);     % Removes the sterophonic property of the input sound
+% by just taking the first column of data.
+
+Len = length(non_stereophonic);
+F = Fs * ((-Len/2) : ((Len/2) - 1)) / Len;  % Creating the array of frequencies
+% which the FFT Shifted version of the signal can be plotted against.
+inputFreq = fftshift(fft(non_stereophonic));   % Creates the Fourier Transform of the
+% input signal. fftshift() makes it such that the zero frequency is at the 
+% center of the array.
+bandPassFilter = zeros(1, length(inputFreq));
+
+maxAmplitude = 0;
+mostPrevalentFrequency = 0;
+
+% Finding the most prevalent frequency.
+for i = round(length(inputFreq)/2):length(inputFreq)
+    if inputFreq(i) > maxAmplitude
+        maxAmplitude = inputFreq(i);
+        mostPrevalentFrequency = F(i);
+    end
+end
+
+% Determining maximum and minimum frequency values for the Band Pass filter.
+maxFrequency = mostPrevalentFrequency + HIGH;
+minFrequency = mostPrevalentFrequency - HIGH;
+
+if minFrequency < 0.0
+    minFrequency = 0.0;
+end
+
+% Creating the Band-Pass filter.
+for i = 1:length(bandPassFilter)
+    if (abs(F(i)) < maxFrequency) && (abs(F(i)) > minFrequency)
+        bandPassFilter(i) = 1;
+    else
+        bandPassFilter(i) = 0;
+    end
+end
+
+bandPassedInput = inputFreq .* transpose(bandPassFilter); %Apply the Band-Pass Filter.
+
+output = real(ifft(fftshift(bandPassedInput)));
+
+
+
+    

unimplemented.csv

@@ -0,0 +1,14 @@
+add_sine.m
+amplify.m
+bandreject_filter.m
+Daniel_Doan_convolution.m
+epic_effect_schluep.m
+fade_in.m
+fade_out.m
+generate_halfCircles.m
+lfo_sawtooth.m
+lfo_sine.m
+muffled_effect_schluep.m
+Petha_Hsu_PitchOffset.m
+reverse.m
+seperate_prevalent_schluep.m