downsample_analysis.m

@@ -1,114 +0,0 @@
-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");

spectral_analysis.m

@@ -3,37 +3,32 @@ close all
 clc
 
 [y, fs] = audioread("sound/modulator22.wav");
-ranges = [17000, 21000; 30500, 36000; 41500, 46000]; 
+ranges = [17000, 20000; 29000, 37000; 41000, 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;
+word = one;
 
 n = length(word);   
 f = (0:n-1)*(fs/n);
 f1 = 0;%Hz
-f2 = 2500;%Hz
+f2 = 4000;%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 
+power = power(idx);
 [val, ind] = max(power);
 
-%lowpass for the formant, moving average
+%lowpass for the formant
+
+Fc = 2000; % define the cutoff frequency of the low-pass filter
+[b, a] = butter(6, Fc/(fs/2), 'low'); % design a 4th-order Butterworth low-pass filter
+Pxx_filt = filter(b, a, power); % apply the filter to the power spectrum
+length(Pxx_filt)
+length(f)
 
-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;
 
@@ -44,7 +39,7 @@ 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');
+plot(f, 10*log10(Pxx_filt), 'r');
 xlabel('Frequency (Hz)')
 ylabel('Power (dB)')