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import streamlit as st | |
import torch | |
import tempfile | |
import os | |
from transformers import WhisperProcessor, WhisperForConditionalGeneration | |
from datasets import load_dataset | |
# Configuration de l'interface Streamlit | |
st.title("🔊 Transcription Audio avec Whisper Fine-tuné") | |
st.write("Upload un fichier audio et laisse ton modèle fine-tuné faire le travail !") | |
# 🔹 Charger le modèle fine-tuné et le processeur | |
def load_model(): | |
model_name = "SimpleFrog/whisper_finetuned" # Remplace par ton nom de repo sur Hugging Face | |
processor = WhisperProcessor.from_pretrained(model_name) | |
model = WhisperForConditionalGeneration.from_pretrained(model_name) | |
model.eval() # Mode évaluation | |
return processor, model | |
processor, model = load_model() | |
# 🔹 Upload d'un fichier audio | |
uploaded_file = st.file_uploader("Upload un fichier audio", type=["mp3", "wav", "m4a"]) | |
if uploaded_file is not None: | |
# Sauvegarder temporairement l'audio | |
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as temp_audio: | |
temp_audio.write(uploaded_file.read()) | |
temp_audio_path = temp_audio.name | |
# Charger et traiter l'audio | |
st.write("📄 **Transcription en cours...**") | |
audio_input = processor(temp_audio_path, return_tensors="pt", sampling_rate=16000) | |
input_features = audio_input.input_features | |
# Générer la transcription | |
with torch.no_grad(): | |
predicted_ids = model.generate(input_features) | |
# Décoder la sortie | |
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0] | |
# Afficher la transcription | |
st.subheader("📝 Transcription :") | |
st.text_area("", transcription, height=200) | |
# Supprimer le fichier temporaire après l'affichage | |
os.remove(temp_audio_path) | |
st.write("🔹 Modèle fine-tuné utilisé :", "SimpleFrog/whisper_finetuned") | |