Merge branch 'estevan' into main

buttering.m

@@ -0,0 +1,18 @@
+clear all 
+close all
+pkg load signal
+[y,fs] = audioread('modulator22.wav');
+sound(y, fs);
+n = length(y); 
+t=0:1/fs:(n-1)/fs; % time range
+spectrogram(y,fs,5,30);
+
+[y, fs] = audioread('modulator22.wav');
+n = 8;
+Wn = 0.0453;
+[b, a] = butter(n, Wn);
+
+y = filter(b, a, y);
+spectrogram(y,fs,5,30);
+sound(y, fs);
+audiowrite('butter.wav',y,fs);

chanvocoder.m

@@ -0,0 +1,37 @@
+function y = chanvocoder(carrier, modul, chan, numband, overlap)
+% y = chanvocoder(carrier, modul, chan, numband, overlap)
+% The Channel Vocoder modulates the carrier signal with the modulation signal
+% chan = number of channels         (e.g., 512)
+% numband = number of bands (<chan) (e.g., 32)
+% overlap = window overlap          (e.g., 1/4)
+
+if numband>chan, error('# bands must be < # channels'), end
+[rc, cc]   = size(carrier); if cc>rc, carrier = carrier'; end
+[rm, cm]   = size(modul);   if cm>rm, modul = modul'; end
+st         = min(rc,cc);                         % stereo or mono?
+if st~= min(rm,cm), error('carrier and modulator must have same number of tracks'); end
+len        = min(length(carrier),length(modul)); % find shortest length
+carrier    = carrier(1:len,1:st);                % shorten carrier if needed
+modul      = modul(1:len,1:st);                  % shorten modulator if needed
+L          = 2*chan;                             % window length/FFT length
+w          = hanning(L); if st==2, w=[w w]; end  % window/ stereo window
+bands      = 1:round(chan/numband):chan;         % indices for frequency bands     
+bands(end) = chan;
+y          = zeros(len,st);                      % output vector
+ii         = 0;
+while ii*L*overlap+L <= len
+    ind    = round([1+ii*L*overlap:ii*L*overlap+L]);
+    FFTmod = fft( modul(ind,:) .* w );    % window & take FFT of modulator
+    FFTcar = fft( carrier(ind,:) .* w );  % window & take FFT of carrier  
+    syn    = zeros(chan,st);              % place for synthesized output
+    for jj = 1:numband-1                  % for each frequency band
+        b        = [bands(jj):bands(jj+1)-1]; % current band
+        syn(b,:) = FFTcar(b,:)*diag(mean(abs(FFTmod(b,:))));
+    end                                   % take product of spectra
+    midval   = FFTmod(1+L/2,:).*FFTcar(1+L/2,:); % midpoint is special
+    synfull  = [syn; midval; flipud( conj( syn(2:end,:) ) );]; % + and - frequencies
+    timsig   = real( ifft(synfull) );     % invert back to time
+    y(ind,:) = y(ind,:) + timsig;         % add back into time waveform   
+    ii       = ii+1;
+end
+y = 0.8*y/max(max(abs(y)));               % normalize output

decimato.m

@@ -0,0 +1,14 @@
+clear all 
+close all
+pkg load signal
+[x, fs] = audioread('modulator22.wav');
+sound(x, fs);
+y = decimate(x, 5);
+new_fs = fs/5;
+display(new_fs);
+audiowrite('decimated.wav', y, new_fs);
+spectrogram(x,fs,5,30);
+title('original');
+spectrogram(y,fs,5,30);
+title('decimato');
+sound(y, new_fs);

downsampl.m

@@ -0,0 +1,15 @@
+clear all 
+close all
+pkg load signal
+[x, fs] = audioread('butter.wav');
+sound(x, fs);
+new_fs = fs/5;
+display(new_fs);
+r = fs/new_fs;
+y = downsample(x, r);
+audiowrite('down.wav', y, new_fs);
+spectrogram(x,fs,5,30);
+title('original');
+spectrogram(y,fs,5,30);
+title('dowsampled');
+sound(y, new_fs);

fir.m

@@ -0,0 +1,19 @@
+clear all 
+close all
+pkg load signal
+[y,fs] = audioread('modulator22.wav');
+sound(y, fs);
+n = length(y); 
+t=0:1/fs:(n-1)/fs; % time range
+spectrogram(y,fs,5,30);
+
+[y, fs] = audioread('modulator22.wav');
+n = 30;
+fc = 1000/(fs/2);
+w = hamming(n+1);
+b = fir1(n, fc, w);
+filtered_signal = filter(b, 1, y);
+spectrogram(filtered_signal,fs,5,30);
+sound(filtered_signal, fs);
+audiowrite('FIR.wav',filtered_signal,fs);
+

last.m

@@ -0,0 +1,21 @@
+pkg load signal
+
+[x, fs] = audioread('carrier22.wav');
+recording_duration = 5; % in seconds % in Hz
+
+
+printf('Start recording...\n');
+recording = record(5, 22400);
+printf('Recording finished.\n');
+audiowrite('recording.wav', recording, fs);
+plot(recording);
+recording1= 'recording.wav';
+carrierfile = 'carrier22.wav';
+outfile = 'vocodedsound.wav';
+
+[modul, sr1] = audioread(recording1);
+[carrier, sr2] = audioread(carrierfile);
+if sr2 ~= fs, disp('your sampling rates dont match'); end
+
+y = chanvocoder(carrier, modul, 512, 16, 0.2);
+audiowrite('outfile.wav', y, sr1);

ouaiouai.m

@@ -0,0 +1,40 @@
+pkg load signal
+[y,fs] = audioread('modulator22.wav');
+audiowrite('outmodulator22.wav',y,fs/2);
+n = length(y); 
+t=0:1/fs:(n-1)/fs; % time range
+i=1;
+ldft=[];
+while (i<6)
+  [power,duration]=frequencySpectrum(y, fs, false);
+  disp(i);
+  disp (duration);
+  ldft(i)=duration;
+  i=i+1;
+end
+disp (ldft);
+lfft=[];
+i=1;
+while (i<6)
+  [power,duration]=frequencySpectrum(y, fs, true);
+  disp(i);
+  disp (duration);
+  lfft(i)=duration;
+  i=i+1;
+end
+disp (lfft);
+i=1;
+ratio=[];
+while (i<6)
+  ratio(i)=(lfft(i))/(ldft(i));
+  i=i+1;
+end
+disp('ratio');
+disp(ratio);
+avgratio=mean(ratio);
+disp('avgratio');
+disp(avgratio);
+
+stdratio= std(ratio);
+disp('stdratio');
+disp(stdratio);

spectrogram.m

@@ -0,0 +1,38 @@
+function spectrogram(signal, samplingFreq, step_size, window_size)
+%%%%%%%%%%%%%%%%%%%%%%%
+%function spectrogram(signal, samplingFreq, step_size, window_size)
+% ex.:  spectrogram(signal, samplingFreq, step_size, window_size)
+%
+% Task: Plot the spectrogram of a given signal
+%
+% Inputs:
+%	-signal: temporal signal to analyse 
+%	-samplingFreq: sampling frequency of the temporal signal
+% 	-step_size: how often the power spectrum will be computed in ms
+%	-window_size: size of the analysing window in ms
+%
+% Ouput: None
+%
+% author: Guillaume Gibert (guillaume.gibert@ecam.fr)
+% date: 14/03/2023
+%%%%%%%%%%%%%%%%%%%%%%%
+
+figure;
+	subplot(2,1,1);
+t=0:1/samplingFreq:length(signal)/samplingFreq-1/samplingFreq;
+plot(t, signal');
+xlim([0 length(signal)/samplingFreq-1/samplingFreq]);
+ylabel('amplitude (norm. unit)');
+	subplot(2,1,2);
+step = fix(step_size*samplingFreq/1000);     % one spectral slice every step_size ms
+window = fix(window_size*samplingFreq/1000);  % window_size ms data window
+fftn = 2^nextpow2(window); % next highest power of 2
+[S, f, t] = specgram(signal, fftn, samplingFreq, window, window-step);
+S = abs(S(2:fftn*4000/samplingFreq,:)); % magnitude in range 0<f<=4000 Hz.
+S = S/max(S(:));           % normalize magnitude so that max is 0 dB.
+S = max(S, 10^(-40/10));   % clip below -40 dB.
+S = min(S, 10^(-3/10));    % clip above -3 dB.
+imagesc (t, f, log(S));    % display in log scale
+set (gca, "ydir", "normal"); % put the 'y' direction in the correct direction
+xlabel('time (s)');
+ylabel('frequency (Hz)');

speech_analysis.m

@@ -3,6 +3,7 @@ pkg load signal
 audiowrite('outmodulator22.wav',y,fs/2);
 n = length(y); 
 t=0:1/fs:(n-1)/fs; % time range
+<<<<<<< HEAD
 figure;
 plot(t,y);
 xlabel('time (s)');
@@ -12,3 +13,40 @@ ylabel('amplitude');
 [power,duration]=frequencySpectrum(y, fs, false);
 figure;
 plot(duration,power);
+=======
+i=1;
+ldft=[];
+while (i<6)
+  [power,duration]=frequencySpectrum(y, fs, false);
+  disp(i);
+  disp (duration);
+  ldft(i)=duration;
+  i=i+1;
+end
+disp (ldft);
+lfft=[];
+i=1;
+while (i<6)
+  [power,duration]=frequencySpectrum(y, fs, true);
+  disp(i);
+  disp (duration);
+  lfft(i)=duration;
+  i=i+1;
+end
+disp (lfft);
+i=1;
+ratio=[];
+while (i<6)
+  ratio(i)=(lfft(i))/(ldft(i));
+  i=i+1;
+end
+disp('ratio');
+disp(ratio);
+avgratio=mean(ratio);
+disp('avgratio');
+disp(avgratio);
+
+stdratio= std(ratio);
+disp('stdratio');
+disp(stdratio);
+>>>>>>> estevan

vocoder.m

@@ -0,0 +1,9 @@
+pkg load signal
+modfile = 'modulator22.wav';
+carfile = 'carrier22.wav';
+outfile = 'vocodedsound.wav'
+[modul, sr1] = audioread(modfile);
+[carrier, sr2] = audioread(carfile);
+if sr1~=sr2, disp('your sampling rates dont match'); end
+y = chanvocoder(carrier, modul, 512, 16, .2);
+audiowrite('outfile.wav',y,sr1)