ajouter des trucs

tmp2.py

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+import librosa
+import os
+import numpy as np
+import math
+from scipy.io import wavfile
+import wave
+from scipy.fftpack import fft,dct
+
+from matplotlib.patches import ConnectionPatch
+import matplotlib.pyplot as plt
+import numpy as np
+import scipy.spatial.distance as dist
+import pyaudio
+import wave
+def dp(distmat):
+    N,M = distmat.shape
+    # Initialisons the cost matrix
+    costmat =np.zeros((N+1,M+1))
+    for i in range (1,N+1):
+        costmat[i,0]=np.inf
+    for i in range (1,M+1):
+        costmat[0,i]=np.inf
+
+    for i in range (N):
+        for j in range (M):
+            #on calcule le cout minimal pour chaque chemin.pour atteindre the costmat[i][j] il y a trois chemins possibles on choisit celui de cout minimal
+            penalty = [
+              costmat[i,j],     # cas T==0
+              costmat[i,j+1] ,  # cas T==1
+              costmat[i+1,j]]   # cas T==2
+            ipenalty = np.argmin(penalty)
+            costmat[i+1,j+1] = distmat[i,j] + penalty[ipenalty]
+
+    #enlever les valeurs de l infini
+    costmat = costmat[1: , 1:]
+    return (costmat, costmat[-1, -1]/(N+M))
+
+def divsignalaudiobis(signal):
+    long_signal = 20 # 20 ms
+    recouvrement = 10 # 10 ms
+    long_echantillon = long_signal*sr//1000
+    recouvrement_echantillon = recouvrement*sr//1000
+    nb_echantillon = int(np.ceil((len(signal) - long_echantillon)/recouvrement_echantillon) + 1)
+    long_a_completer = recouvrement_echantillon*(nb_echantillon -1) + long_echantillon - len(signal)
+
+    if (long_a_completer != 0):
+        echantillon_data = np.pad(signal,(0,long_a_completer),mode='constant') # on complète le dernier échantillon par des 0
+
+    else :
+        nb_echantillon -= 1
+
+    echantillon_data = np.append(echantillon_data[0], echantillon_data[1:])
+    data = np.zeros((nb_echantillon, long_echantillon))
+    for i in range(nb_echantillon):
+        echantillon_i = echantillon_data[i*recouvrement_echantillon : i*recouvrement_echantillon + long_echantillon]
+        data[i,:] = echantillon_i
+    return data
+
+
+def myfft(signal, fe):
+    n= len(signal)
+    Te = 1/fe
+    S = [0 + 0j]*(450)
+    for l in range(50, 500):
+        f = l
+        for i in range (n):
+            t= Te * i
+            S[l-50] += signal[i]*np.exp(-2*math.pi*f*t*1j)
+        S[l-50] = abs(S[l-50])/n
+    return S
+
+def puissance_spec(signal):
+    amplitude_fft = np.absolute(signal)
+    return (amplitude_fft**2)/44100 # long fft = 512
+
+def BankFiltre(rate, puis_spec):
+    freq_min = 20
+    freq_max = rate//2
+    freq_mel_min = 1000*np.log2(1 +freq_min/1000)
+    freq_mel_max = 1000*np.log2(1+ freq_max/1000)
+    nb_filtre = 40 # on prend en général 40 filtres
+    mel_points = np.linspace(freq_mel_min, freq_mel_max, 42)
+    hz_points = 1000*(2**(mel_points/1000)-1)# on convertit en hz
+
+    bankf = np.zeros((nb_filtre, int(np.floor(22050 +1))))
+
+    for m in range(1, nb_filtre +1): # pour chaque filtre, on fait :
+        f_m_min = int(math.floor(hz_points[m-1]))   # point de gauche
+        f_m = int(math.floor(hz_points[m]))             # sommet
+        f_m_max = int(math.floor(hz_points[m+1]))  # point de droite
+
+        for k in range(f_m_min, f_m):
+            bankf[m - 1, k] =  (k - hz_points[m - 1]) / (hz_points[m] - hz_points[m - 1])
+        for k in range(f_m, f_m_max):
+            bankf[m - 1, k] =  ((hz_points[m + 1]) - k) / (hz_points[m + 1] - hz_points[m])
+
+    filter_bank = np.dot(puis_spec,np.transpose(bankf)) # Produit vectoriel/matriciel #ipdb
+    filter_bank = np.where(filter_bank == 0, np.finfo(float).eps, filter_bank) # attention à 0 dans le log.
+    return filter_bank
+
+def mfcc(signal, rate):
+    data = divsignalaudiobis(signal)
+    data_fft = np.fft.rfft(data, 44100)
+    data_puiss = puissance_spec(data_fft)
+    data_filtre = BankFiltre(rate, data_puiss)
+    pre_mfcc = np.log(data_filtre)
+    mfcc = dct(pre_mfcc, type=2, axis=1, norm="ortho")[:, 0 : 13] # on ne garde que les 13 premiers
+    #return mfcc
+    return mfcc
+def calculate_dtw_cost(mfccs_query , mfccs_train):
+    distmat = dist.cdist(mfccs_query, mfccs_train,"cosine")
+    costmat,mincost = dp(distmat)
+    return mincost
+def recognize_speech(audio_query, audio_train_list, sr):#sr frequence d echantillonnage
+    # Calculate MFCCs for query audio
+    mfccs_query = mfcc(audio_query, sr)
+
+    # Calculate DTW cost for each audio in training data
+    dtw_costs = []
+    for audio_train in audio_train_list:
+        mfccs_train = mfcc(audio_train, sr)
+        mincost = calculate_dtw_cost(mfccs_query, mfccs_train)
+        dtw_costs.append(mincost)
+
+    # Find index of word with lowest DTW cost
+    index = np.argmin(dtw_costs)
+
+    # Return recognized word
+    return index
+
+# Example usage
+
+def record_audio(filename, duration, sr):
+    chunk = 1024
+    sample_format = pyaudio.paInt16
+    channels = 1
+    record_seconds = duration
+    filename = f"{filename}.wav"
+
+    p = pyaudio.PyAudio()
+
+    stream = p.open(format=sample_format,
+                    channels=channels,
+                    rate=sr,
+                    frames_per_buffer=chunk,
+                    input=True)
+
+    frames = []
+
+    print(f"Enregistrement en cours...")
+
+    for i in range(0, int(sr / chunk * record_seconds)):
+        data = stream.read(chunk)
+        frames.append(data)
+
+    stream.stop_stream()
+    stream.close()
+
+    p.terminate()
+
+    print("Enregistrement terminé")
+
+    wf = wave.open(filename, "wb")
+    wf.setnchannels(channels)
+    wf.setsampwidth(p.get_sample_size(sample_format))
+    wf.setframerate(sr)
+    wf.writeframes(b"".join(frames))
+    wf.close()
+
+    print(f"Fichier enregistré sous {filename}")
+
+def coupe_silence(signal):
+
+    t = 0
+
+    if signal[t] == 0 :
+
+        p = 0
+
+        while signal[t+p] == 0 :
+
+            if p == 88 :
+
+                signal = signal[:t] + signal[t+p:]
+
+                coupe_silence(signal)
+
+            else :
+
+                p = p+1
+sr = 44100  # fréquence d'échantillonnage
+duration = 6  # durée d'enregistrement en secondes
+filename = "audio_query"  # nom du fichier à enregistrer
+
+record_audio(filename, duration, sr)
+audio_query, sr = librosa.load('C:\\Users\\HP\\audio_query.wav', sr=sr)
+coupe_silence(audio_query)
+audio_train_list = [librosa.load('C:\\Users\\HP\\Documents\\cool.wav', sr=sr)[0], librosa.load('C:\\Users\\HP\\Documents\\formidable.wav', sr=sr)[0], librosa.load('C:\\Users\\HP\\Documents\\cest mauvais.wav', sr=sr)[0] , librosa.load('C:\\Users\\HP\\Documents\\un.wav', sr=sr)[0], librosa.load('C:\\Users\\HP\\Documents\\parfait.wav', sr=sr)[0]]
+recognized_word_index = recognize_speech(audio_query, audio_train_list, sr)
+print(f'Recognized word: {recognized_word_index}')