coucou

SAR.m

@@ -6,9 +6,9 @@
 [x,fe]=audioread("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_celtic-harp-a3.wav");
 
 soundsc(x,fe);
-L1 = length(x1)
-X1 = fftshift(fft(x1))
-f1 = (-L1/2 : L1/2 - 1)*(fe1/L1)
+L1 = length(x1);
+X1 = fftshift(fft(x1));
+f1 = (-L1/2 : L1/2 - 1)*(fe1/L1);
 
 figure;
 plot(f1, log(abs(X1)),'r');
@@ -25,28 +25,28 @@ ylabel("log|X(f)|")
 soundsc(x1,fe1);
 soundsc(x2,fe2);
 
-L1 = length(x1)
-L2 = length(x2)
+L1 = length(x1);
+L2 = length(x2);
 
-X1 = fftshift(fft(x1))
-X2 = fftshift(fft(x2))
+X1 = fftshift(fft(x1));
+X2 = fftshift(fft(x2));
 
-f1 = (-L1/2 : L1/2 - 1)*(fe1/L1)
-f2 = (-L2/2 : L2/2 - 1)*(fe2/L2)
+f1 = (-L1/2 : L1/2 - 1)*(fe1/L1);
+f2 = (-L2/2 : L2/2 - 1)*(fe2/L2);
 
 figure;
 plot(f1, log(abs(X1)),'r');
 hold on;
-plot(f2, log(abs(X2)),'b')
-
-title("spectre")
-xlabel("fréquence en Hz")
-ylabel("log|X(f)|")
+plot(f2, log(abs(X2)),'b');
+title("spectre");
+xlabel("fréquence en Hz");
+ylabel("10*log|X(f)|");
 
 
 % Question 3 :
 
 freq_r=[220;442;663;885;1108;1331;1556;1782;2009];
+amplitude = [];
 
 % Durée et fréquence d'échantillonnage
 Fe = 44100;           % fréquence d'échantillonnage (standard audio)
@@ -58,7 +58,7 @@ s = zeros(size(t));
 
 % Ajouter les sinusoïdes
 for k = 1:length(freq_r)
-    s = s + sin(2*pi*freq_r(k)*t);
+    s = s + amplitude * sin(2*pi*freq_r(k)*t);
 end
 
 % Normaliser pour éviter la saturation (valeurs entre -1 et 1)
@@ -71,21 +71,49 @@ soundsc(s, Fe);
 
 % Question 4 :
 
-function s = signal(t, freq_r)
-    s = zeros(size(t));
-    for k = 1:length(freq_r)
-        s = s + sin(2*pi*freq_r(k)*t);
-    end
-end
+% Define ADSR parameters
+A = 0.1; % Attack time (seconds)
+D = 0.005; % Decay time (seconds)
+S = 0.8; % Sustain level (0 to 1)
+R = 0.75; % Release time (seconds)
+
+fs = 44100; % Sampling frequency (Hz)
+
+
+% Total duration of the envelope
+totalTime = A + D + R;
+
+% totalTime = totalTime * 6; % Extend the envelope for 6 seconds
+t = linspace(0, totalTime, totalTime * fs);
 
+
+% Generate ADSR envelope
+attack = linspace(0, 1, A * fs);
+decay = linspace(1, S, D * fs);
+release = linspace(S, 0, R * fs);
+ 
+
+% Combine all segments
+adsrEnvelope = [attack, decay, release];
+
+
+% Plot the envelope
 figure;
-plot(t, signal(t, freq_r))
+plot(t, adsrEnvelope);
+xlabel('Time (s)');
+ylabel('Amplitude');
+title('ADSR Envelope');
+
+
+[xA, feA] = audioread("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_piano2.wav");
+
+ 
+
+% Apply the ADSR envelope to the audio signal
+adsrSignal = xA(1:length(adsrEnvelope)) .* adsrEnvelope';
+sound(adsrSignal, feA);
 
-t = [0.1 0.3 0.5 0.7 1]; % Temps en secondes
-env = [signal(0.1) signal(0.3) signal(0.5) signal(0.7) signal(1)]; % Valeurs d'amplitude correspondantes
+audiowrite("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_piano2.wav", adsrSignal, fe);
 
-t_total = length(s)/Fe; % Durée totale du signal en secondes
-t_interpolated = linspace(0, t_total, length(s)); % Temps interpolé
-env_interpolated = interp1(t, env, t_interpolated); % Enveloppe interpolée
 
-s_envelope = s .* env_interpolated; % Signal avec enveloppe ADSR
\ No newline at end of file
+% Question 5 : 
\ No newline at end of file

question_1.m

+% Question 1 :
+
+[x,fe]=audioread("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_celtic-harp-a3.wav");
+
+soundsc(x,fe);
+L1 = length(x1);
+X1 = fftshift(fft(x1));
+f1 = (-L1/2 : L1/2 - 1)*(fe1/L1);
+
+figure;
+plot(f1, log(abs(X1)),'r');
+title("spectre")
+xlabel("fréquence en Hz")
+ylabel("log|X(f)|")
\ No newline at end of file

question_2.m

+% Question 2 :
+
+[x1,fe1]=audioread("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_piano1.wav");
+[x2,fe2]=audioread("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_piano2.wav");
+
+soundsc(x1,fe1);
+soundsc(x2,fe2);
+
+L1 = length(x1);
+L2 = length(x2);
+
+X1 = fftshift(fft(x1));
+X2 = fftshift(fft(x2));
+
+f1 = (-L1/2 : L1/2 - 1)*(fe1/L1);
+f2 = (-L2/2 : L2/2 - 1)*(fe2/L2);
+
+figure;
+plot(f1, log(abs(X1)),'r');
+hold on;
+plot(f2, log(abs(X2)),'b');
+title("spectre");
+xlabel("fréquence en Hz");
+ylabel("10*log|X(f)|");
\ No newline at end of file

question_3.asv

+% Question 3 :
+
+freq_r=[220;442;663;885;1108;1331;1556;1782;2009];
+amplitude = [8.89;8.62;6.89;7.25;6.22;6.51;6.35;4.74;5.85];
+Amplitude = [7259; 5541; 982; 1408; ]
+
+% Durée et fréquence d'échantillonnage
+Fe = 44100;           % fréquence d'échantillonnage (standard audio)
+duree = 1;            % durée du signal en secondes
+t = 0:1/Fe:duree;     % vecteur temps
+
+% Initialiser le signal composite
+s = zeros(size(t));
+
+% Ajouter les sinusoïdes
+for k = 1:length(freq_r)
+    s = s + amplitude(k) * sin(2*pi*freq_r(k)*t);
+end
+
+% Normaliser pour éviter la saturation (valeurs entre -1 et 1)
+s = s / max(abs(s));
+
+% Jouer le son
+soundsc(s, Fe);
+
+figure;
+plot(t,s)
\ No newline at end of file

question_3.m

+% Question 3 :
+
+freq_r=[220;442;663;885;1108;1331;1556;1782;2009];
+Amplitude = [7259; 5541; 982; 1408; 502; 672; 573; 114; 347];
+
+% Durée et fréquence d'échantillonnage
+Fe = 44100;           % fréquence d'échantillonnage (standard audio)
+duree = 1;            % durée du signal en secondes
+t = 0:1/Fe:duree;     % vecteur temps
+
+% Initialiser le signal composite
+s = zeros(size(t));
+
+% Ajouter les sinusoïdes
+for k = 1:length(freq_r)
+    s = s + Amplitude(k) * sin(2*pi*freq_r(k)*t);
+end
+
+% Normaliser pour éviter la saturation (valeurs entre -1 et 1)
+s = s / max(abs(s));
+
+% Jouer le son
+soundsc(s, Fe);
+
+figure;
+plot(t,s)
\ No newline at end of file

question_4.m

+% Question 4
+
+% Define ADSR parameters
+
+A = 0.1; % Attack time (seconds)
+D = 0.005; % Decay time (seconds)
+S = 0.8; % Sustain level (0 to 1)
+R = 0.75; % Release time (seconds)
+fs = 44100; % Sampling frequency (Hz)
+
+% Total duration of the envelope
+totalTime = A + D + R;
+
+% totalTime = totalTime * 6; % Extend the envelope for 6 seconds
+t = linspace(0, totalTime, totalTime * fs);
+ 
+% Generate ADSR envelope
+attack = linspace(0, 1, A * fs);
+decay = linspace(1, S, D * fs);
+release = linspace(S, 0, R * fs);
+
+ % Combine all segments
+adsrEnvelope = [attack, decay, release];
+ 
+% Plot the envelope
+plot(t, adsrEnvelope);
+xlabel('Time (s)');
+ylabel('Amplitude');
+title('ADSR Envelope');
+[x, fe] = audioread("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_piano2.wav");
+
+% Apply the ADSR envelope to the audio signal
+adsrSignal = x(1:length(adsrEnvelope)) .* adsrEnvelope';
+sound(adsrSignal, fe);
+audiowrite("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_piano2.wav", adsrSignal, fe);
\ No newline at end of file

untitled3.m

+
+[xA, feA] = audioread("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_piano2.wav");
+
+ 
+
+% Apply the ADSR envelope to the audio signal
+adsrSignal = abs(ifftshift(ifft(xA)));
+sound(adsrSignal, feA);
+
+% Plot the envelope
+
+figure;
+plot(feA, adsrSignal);
+xlabel('Time (s)');
+ylabel('Amplitude');
+title('ADSR Envelope');
+
+audiowrite("C:\Users\camil\Documents\IMT_A\semestre_6\electrical engineering\tp-audio-ee-etudiant-c24leray\src\wav\single_tone_piano2.wav", adsrSignal, fe);