Merge branch 'develop'

FT_times.m

@@ -0,0 +1,21 @@
+clear all
+close all
+clc
+
+[y, fs] = audioread("sound/modulator22.wav");
+times = [];
+item_num = 20;
+for j = 1:item_num
+  localtime = [];
+  for i = 0:1
+    t0 = clock ();
+    frequencySpectrum_noplot(y, fs, 0); %true does FFT, false DFT
+    localtime(i+1) = etime (clock (), t0);
+  endfor
+  times = [times; localtime];
+endfor
+
+printf('Average DFT time: %d \n', mean(times(1:item_num, 1)))
+printf('Standard deviation of DFT time: %d \n', std(times(1:item_num, 1)))
+printf('Average FFT time: %d \n', mean(times(1:item_num, 2)))
+printf('Standard deviation of FFT time: %d \n', std(times(1:item_num, 2)))

downsample_analysis.m

@@ -0,0 +1,114 @@
+clear all
+close all 
+clc
+
+% Modify the desired frequency for the whole code here
+desired_freq = 4000;
+% Read and Set up coeficient
+[y, fs] = audioread("sound/modulator22.wav");
+decrease_coef = round(fs/desired_freq);
+
+% Modify Signal with downsample
+% Only keeps one sample out of decrease_coef
+down_y = downsample(y,decrease_coef);
+down_fs = fs/decrease_coef;
+audiowrite("sound/down_output.wav", down_y, down_fs);
+% Plot Down Modified 
+plot(0:1/down_fs:(length(down_y)-1)/down_fs,down_y);
+xlabel("Time (s)");
+ylabel("Amplitude");
+title("Downsample Sound Amplitude Over Time");
+
+% Modify Signal with decimate
+% Only keeps one sample out of decrease_coef
+deci_y = decimate(y,decrease_coef);
+deci_fs = fs/decrease_coef;
+audiowrite("sound/deci_output.wav", deci_y, deci_fs);
+% Plot Down Modified 
+figure;
+plot(0:1/deci_fs:(length(deci_y)-1)/deci_fs,deci_y);
+xlabel("Time (s)");
+ylabel("Amplitude");
+title("Decimate Sound Amplitude Over Time");
+
+
+% Filter Signal with FIR filter
+order = 30;
+cutoff_freq = 1000; % in Hz
+fir_b = fir1(order, cutoff_freq/(fs/2));
+% Show Filter
+figure
+freqz(fir_b)
+% Apply filter
+fir_y = filter(fir_b, 1, y);
+audiowrite("sound/fir_output.wav", fir_y, fs);
+% Plot Down Modified 
+figure;
+plot(0:1/fs:(length(fir_y)-1)/fs,fir_y);
+xlabel("Time (s)");
+ylabel("Amplitude");
+title("FIR Filter Sound Amplitude Over Time");
+
+% Filter Signal with Butter filter
+order = 8; 
+cutoff_freq = 1000; % in Hz
+[b, a] = butter(order, cutoff_freq/(fs/2), 'low');
+% Show Filter
+figure
+freqz(b)
+% Apply filter
+butt_y = filter(b, a, y);
+audiowrite("sound/butt_output.wav", butt_y, fs);
+% Plot Down Modified 
+figure;
+plot(0:1/fs:(length(butt_y)-1)/fs,butt_y);
+xlabel("Time (s)");
+ylabel("Amplitude");
+title("Butter Filter Sound Amplitude Over Time");
+
+% Test FIR Filter Stability
+% Find the transfer function
+H = tf(fir_b, 1);
+% Get the poles of the transfer function
+poles = pole(H);
+% Check if all poles are inside the unit circle
+if all(abs(poles) < 1)
+    disp('The FIR filter is stable');
+else
+    disp('The FIR filter is unstable');
+end
+% Test Butter Filter Stability
+% Find the transfer function
+H = tf(b, a);
+% Get the poles of the transfer function
+poles = pole(H);
+% Check if all poles are inside the unit circle
+if all(abs(poles) < 1)
+    disp('The Butter filter is stable');
+else
+    disp('The Butter filter is unstable');
+end
+
+% Modify Signal with downsample
+% Only keeps one sample out of decrease_coef
+down_fir_y = downsample(fir_y,decrease_coef);
+down_fir_fs = fs/decrease_coef;
+audiowrite("sound/down_fir_output.wav", down_fir_y, down_fir_fs);
+% Plot Down Modified 
+figure
+plot(0:1/down_fir_fs:(length(down_fir_y)-1)/down_fir_fs,down_fir_y);
+xlabel("Time (s)");
+ylabel("Amplitude");
+title("Downsample FIR Sound Amplitude Over Time");
+
+% Modify Signal with downsample
+% Only keeps one sample out of decrease_coef
+down_butt_y = downsample(butt_y,decrease_coef);
+down_butt_fs = fs/decrease_coef;
+audiowrite("sound/down_butt_output.wav", down_butt_y, down_butt_fs);
+% Plot Down Modified 
+figure
+plot(0:1/down_butt_fs:(length(down_butt_y)-1)/down_butt_fs,down_butt_y);
+xlabel("Time (s)");
+ylabel("Amplitude");
+title("Downsample Butter Sound Amplitude Over Time");

frequencySpectrum_noplot.m

@@ -0,0 +1,44 @@
+function [power, duration] = frequencySpectrum_noplot(signal, fs, pad)
+%%%%%%%%%%%%%%%%%%
+%function power = frequencySpectrum(signal, fs, pad)
+%
+% Task: Display the power spectrum (lin and log scale) of a given signal
+%
+% Input:
+%	- signal: the input signal to process
+%	- fs: the sampling rate
+%	-pad: boolean if true, signal is padded with 0 to the next power of 2 -> FFT instead of DFT
+%
+% Output: 
+%	- power: the power spectrum
+%	
+%
+% Guillaume Gibert, guillaume.gibert@ecam.fr
+% 25/04/2022
+%%%%%%%%%%%%%%%%%%
+
+n = length(signal);        % number of samples
+
+if (pad)
+	n = 2^nextpow2(n);
+end
+
+tic
+y = fft(signal, n);% compute DFT of input signal
+duration = toc;
+
+power = abs(y).^2/n;    % power of the DFT
+
+[val, ind] = max(power); % find the mx value of DFT and its index
+
+t=0:1/fs:(n-1)/fs; % time range
+%pad signal with zeros
+if (pad)
+	signal = [ signal; zeros( n-length(signal), 1)];
+end
+
+f = (0:n-1)*(fs/n);     % frequency range
+
+
+
+

sound/desktop.ini

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+[.ShellClassInfo]
+IconResource=C:\Program Files\Google\Drive File Stream\72.0.3.0\GoogleDriveFS.exe,23

spectral_analysis.m

@@ -0,0 +1,51 @@
+clear all
+close all
+clc
+
+[y, fs] = audioread("sound/modulator22.wav");
+ranges = [17000, 21000; 30500, 36000; 41500, 46000]; 
+one = y(ranges(1,1):ranges(1,2));
+two = y(ranges(2,1):ranges(2,2));
+three = y(ranges(3,1):ranges(3,2));
+
+word = three;
+
+n = length(word);   
+f = (0:n-1)*(fs/n);
+f1 = 0;%Hz
+f2 = 2500;%Hz
+idx = find(f >= f1 & f <= f2); %define the index of the freq range
+f = f(idx);
+y = fft(word, n);% compute DFT of input signal
+power = abs(y).^2/n;
+power = power(idx);%limit the signal to the frequency ROI 
+[val, ind] = max(power);
+
+%lowpass for the formant, moving average
+
+for j = 1:length(idx)
+  tot = 0;
+  for k = j-18:j
+    if k <=0
+      tot = tot + power(1);
+    else
+      tot = tot + power(k);
+    endif
+  endfor
+  power_avg(j) = 1/6*(tot);
+endfor
+
+figure;
+
+subplot(1,2,1) % time plot 
+plot(0:1/fs:(length(word)-1)/fs,word);
+xlabel('Time (s)');
+ylabel('Amplitude (a.u.)');
+
+subplot(1,2,2) % freq range plot 
+plot(f,10*log10(power/power(ind))); hold on;
+plot(f, 10*log10(power_avg/power(ind)), 'r');
+xlabel('Frequency (Hz)')
+ylabel('Power (dB)')
+
+

spectrogram_analysis.m

@@ -0,0 +1,8 @@
+clear all
+close all
+clc
+
+[y, fs] = audioread("sound/modulator22.wav");
+step_size = 5; %ms
+window_size = 30;%ms, ideal value 25
+spectrogram(y, fs, step_size, window_size)

speech_analysis.m

@@ -0,0 +1,11 @@
+clear all
+close all 
+clc
+
+[y, fs] = audioread("sound/modulator22.wav");
+plot(0:1/fs:(length(y)-1)/fs,y);
+xlabel("Time (s)");
+ylabel("Amplitude");
+title("Sound Amplitude Over Time");
+
+audiowrite("sound/output.wav", y, fs/2);