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Merge pull request #3 from jdalfons/Nancy
Browse files- src/speech2.py +201 -0
src/speech2.py
ADDED
@@ -0,0 +1,201 @@
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import os
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import torch
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import torch.nn as nn
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import torch.optim as optim
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import soundfile as sf
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import torchaudio
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import numpy as np
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from datasets import Dataset
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from transformers import (
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Wav2Vec2Model,
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Wav2Vec2Processor
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)
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from dotenv import load_dotenv
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from sklearn.metrics import accuracy_score
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# Charger .env pour Hugging Face API Key
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load_dotenv()
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HF_API_KEY = os.getenv("HF_API_KEY")
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if not HF_API_KEY:
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raise ValueError("Le token Hugging Face n'a pas été trouvé dans .env")
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# Définition des labels pour la classification des émotions
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LABELS = {"colere": 0, "neutre": 1, "joie": 2}
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NUM_LABELS = len(LABELS)
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# Charger le processeur et le modèle pour l'extraction de features
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model_name = "facebook/wav2vec2-large-xlsr-53-french"
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device = "cuda" if torch.cuda.is_available() else "cpu"
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processor = Wav2Vec2Processor.from_pretrained(model_name)
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feature_extractor = Wav2Vec2Model.from_pretrained(model_name).to(device)
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# Resampleur pour convertir en 16 kHz
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resampler = torchaudio.transforms.Resample(orig_freq=48_000, new_freq=16_000)
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# Définition du classifieur amélioré
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class EmotionClassifier(nn.Module):
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def __init__(self, feature_dim, num_labels):
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super(EmotionClassifier, self).__init__()
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self.fc1 = nn.Linear(feature_dim, 256)
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self.relu = nn.ReLU()
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self.dropout = nn.Dropout(0.3)
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self.fc2 = nn.Linear(256, num_labels)
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def forward(self, x):
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x = self.fc1(x)
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x = self.relu(x)
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x = self.dropout(x)
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return self.fc2(x)
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# Instancier le classifieur
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classifier = EmotionClassifier(feature_extractor.config.hidden_size, NUM_LABELS).to(device)
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# Charger les fichiers audio et leurs labels
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def load_audio_data(data_dir):
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data = []
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for label_name, label_id in LABELS.items():
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label_dir = os.path.join(data_dir, label_name)
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for file in os.listdir(label_dir):
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if file.endswith(".wav"):
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file_path = os.path.join(label_dir, file)
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data.append({"path": file_path, "label": label_id})
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return Dataset.from_list(data)
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# Chargement du dataset
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data_dir = "./dataset"
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ds = load_audio_data(data_dir)
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# Charger les fichiers audio avec SoundFile et rééchantillonner à 16 kHz
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def preprocess_audio(batch):
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speech, sample_rate = sf.read(batch["path"], dtype="float32")
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if sample_rate != 16000:
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speech = torch.tensor(speech).unsqueeze(0)
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speech = resampler(speech).squeeze(0).numpy()
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batch["speech"] = speech.tolist() # Convertir en liste pour éviter les erreurs de PyArrow
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batch["sampling_rate"] = 16000
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return batch
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ds = ds.map(preprocess_audio)
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# Vérifier la distribution des longueurs des fichiers audio
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lengths = [len(sample["speech"]) for sample in ds]
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max_length = int(np.percentile(lengths, 95))
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# Transformer l'audio en features utilisables par le modèle
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def prepare_features(batch):
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features = processor(
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batch["speech"],
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sampling_rate=16000,
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padding=True,
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truncation=True,
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max_length=max_length,
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return_tensors="pt"
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)
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batch["input_values"] = features.input_values.squeeze(0)
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batch["label"] = torch.tensor(batch["label"], dtype=torch.long)
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return batch
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ds = ds.map(prepare_features)
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# Diviser les données en train et test
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ds = ds.train_test_split(test_size=0.2)
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train_ds = ds["train"]
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test_ds = ds["test"]
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# Fonction d'évaluation sur les données de test
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def evaluate(classifier, feature_extractor, test_ds):
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classifier.eval()
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correct = 0
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total = 0
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with torch.no_grad():
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for batch in test_ds:
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input_values = processor(
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batch["speech"],
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sampling_rate=16000,
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return_tensors="pt",
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padding=True,
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truncation=True,
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max_length=max_length
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).input_values.to(device)
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features = feature_extractor(input_values).last_hidden_state.mean(dim=1)
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logits = classifier(features)
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predictions = logits.argmax(dim=-1)
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labels = torch.tensor(batch["label"], dtype=torch.long, device=device)
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correct += (predictions == labels).sum().item()
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total += 1
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return correct / total
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# Fonction d'entraînement
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def train_classifier(feature_extractor, classifier, train_ds, test_ds, epochs=10, batch_size=16):
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optimizer = optim.Adam(classifier.parameters(), lr=1e-4)
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scheduler = optim.lr_scheduler.StepLR(optimizer, step_size=3, gamma=0.7)
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loss_fn = nn.CrossEntropyLoss()
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best_accuracy = 0.0 # Variable pour stocker la meilleure accuracy
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for epoch in range(epochs):
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classifier.train()
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total_loss, correct = 0, 0
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batch_count = 0
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for i in range(0, len(train_ds), batch_size):
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batch = train_ds[i: i + batch_size]
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optimizer.zero_grad()
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input_values = processor(
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batch["speech"],
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sampling_rate=16000,
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return_tensors="pt",
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padding=True,
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truncation=True,
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max_length=max_length
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).input_values.to(device)
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with torch.no_grad():
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features = feature_extractor(input_values).last_hidden_state.mean(dim=1)
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features = (features - features.mean()) / features.std() # Normalisation
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logits = classifier(features)
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labels = torch.tensor(batch["label"], dtype=torch.long, device=device)
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if labels.numel() == 0:
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continue
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loss = loss_fn(logits, labels)
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loss.backward()
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optimizer.step()
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total_loss += loss.item()
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correct += (logits.argmax(dim=-1) == labels).sum().item()
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batch_count += 1
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train_acc = correct / len(train_ds)
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test_acc = evaluate(classifier, feature_extractor, test_ds)
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scheduler.step()
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# Sauvegarde uniquement si l'accuracy sur test est la meilleure obtenue
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if test_acc > best_accuracy:
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best_accuracy = test_acc
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torch.save({
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"classifier_state_dict": classifier.state_dict(),
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"feature_extractor_state_dict": feature_extractor.state_dict(),
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"processor": processor
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}, "best_emotion_model.pth")
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print(f"✅ Nouveau meilleur modèle sauvegardé ! Accuracy Test: {best_accuracy:.4f}")
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print(f"Epoch {epoch+1}/{epochs} - Loss: {total_loss/batch_count:.4f} - Train Accuracy: {train_acc:.4f} - Test Accuracy: {test_acc:.4f}")
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return classifier
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# Entraînement
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trained_classifier = train_classifier(feature_extractor, classifier, train_ds, test_ds, epochs=10, batch_size=16)
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print("✅ Entraînement terminé, le meilleur modèle a été sauvegardé !")
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