From 4cfaf9454487e7cb4e6c0f500c6b9a5926e5d3ef Mon Sep 17 00:00:00 2001
From: "mattis.roellinger" <mattis.roellinger@ecam.fr>
Date: Thu, 28 Mar 2024 14:09:20 +0100
Subject: [PATCH] Adding comments to the speech_analysis

---
 speech_analysis.m | 21 ++++++++++++++++++---
 1 file changed, 18 insertions(+), 3 deletions(-)

diff --git a/speech_analysis.m b/speech_analysis.m
index 8e6249b..6d3f300 100644
--- a/speech_analysis.m
+++ b/speech_analysis.m
@@ -1,42 +1,56 @@
+%loading the signal library in octave
 pkg load signal
 clear all
 close all
 
+%reading of the orignial signal in data
 [y,fs]=audioread('./data/modulator22.wav');
 
+%calculation of the time from the sampling frequency
 t=0:1/fs:length(y)/fs-1/fs;
 
+%showing the original signal
 ##figure;
 ##set(gcf, 'name','Signal of origin');
 ##plot(t,y);
 ##xlabel('time(s)');
 ##ylabel('amplitude(n.u.');
 
+%save a speed down version of the original signal
 audiowrite('./data/modulator22_speedown.wav', y , fs/2);
+
+%save the speed down version's signal and sampling frequency  and calculate it's new time duration
 [y_speedown,fs_speedown]=audioread('./data/modulator22_speedown.wav');
 t_speedown=0:1/fs_speedown:length(y_speedown)/fs_speedown-1/fs_speedown;
 
+%showing the speed down version's signal
 ##figure;
 ##set(gcf, 'name','Speed down of the signal of origin');
 ##plot(t_speedown,y_speedown);
 ##xlabel('time(s)');
 ##ylabel('amplitude(n.u.');
 
+%Apply the DFT(flase) and FFT(true) to the speed down signal
 %frequencySpectrum(y_speedown, fs_speedown, false);
 %frequencySpectrum(y_speedown, fs_speedown, true);
 
+%Illustrate the spectogram of the speed down version at 2 windows values (1 for a precise frequency read and 1 for a precise time domain)
 %spectrogram(y_speedown, fs_speedown, 5, 30);
 %spectrogram(y_speedown, fs_speedown, 5, 5);
 
+%Lowering the sampling frequency using the downsample() function
 ##y_downsampled = downsample(y, 2);
 ##fs_downsample = fs/2;
 ##audiowrite('./data/modulator22_downsample.wav', y_downsampled , fs_downsample);
 ##t_downsample=0:1/fs_downsample:length(y_downsampled)/fs_downsample-1/fs_downsample;
+
+%Lowering again the sampling frequency but with the decimate() function (applying a LP filter at the start )
 ##y_decimated = decimate(y, 2);
 ##fs_decimate = fs/2;
 ##audiowrite('./data/modulator22_decimate.wav', y_decimated , fs_decimate);
 ##t_decimate=0:1/fs_decimate:length(y_decimated)/fs_decimate-1/fs_decimate;
-##
+
+%Comparing them both on the same graph to see the effect of the LP filter on the signal itself
 ##figure;
 ##set(gcf, 'name','Decimate vs Downsample');
 ##title('Decimate vs Downsample');
@@ -46,11 +60,12 @@ t_speedown=0:1/fs_speedown:length(y_speedown)/fs_speedown-1/fs_speedown;
 ##xlabel('time(s)');
 ##ylabel('amplitude(n.u)');
 ##legend('downsample()','decimate()',"location","southeastoutside");
-##
+
+%Illustrating them with their frequencySpectrum to see the effect of the LP filter
 ##frequencySpectrum(y_decimated, fs_decimate, true);
 ##frequencySpectrum(y_downsampled, fs_downsample, true);
 
-%%%FILTER IMPLEMENTATION %%%%
+%%%IIR and FIR LP FILTER IMPLEMENTATION %%%%
 %creating a low pass filter FIR 
 N=30;
 fc=1000;