# %% Utilisation de Whisper pour la transcription de podcasts en français from pathlib import Path import numpy as np import torch import torchaudio import tqdm from transformers import ( AutoModelForCausalLM, AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline, ) # %% File paths audio_paths = ["METTRE LES LIENS DES FICHIERS MP3 OU WAV ICI"] audio_dir = "data" # %% load PyTorch device = "cuda:0" if torch.cuda.is_available() else "cpu" torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 # %% Load model model_name_or_path = "bofenghuang/whisper-large-v3-french" processor = AutoProcessor.from_pretrained(model_name_or_path) model = AutoModelForSpeechSeq2Seq.from_pretrained( model_name_or_path, torch_dtype=torch_dtype, low_cpu_mem_usage=True, ) model.to(device) # %% Load draft model assistant_model_name_or_path = "bofenghuang/whisper-large-v3-french-distil-dec2" assistant_model = AutoModelForCausalLM.from_pretrained( assistant_model_name_or_path, torch_dtype=torch_dtype, low_cpu_mem_usage=True, ) assistant_model.to(device) # %% Init pipeline pipe = pipeline( "automatic-speech-recognition", model=model, feature_extractor=processor.feature_extractor, tokenizer=processor.tokenizer, torch_dtype=torch_dtype, device=device, generate_kwargs={"assistant_model": assistant_model}, max_new_tokens=128, ) # %% Transcript function def transcript(audio_dir, audio_path): # Load audio model_sr = 16000 speech, sr = torchaudio.load(Path(audio_dir) / audio_path) speech_16000 = torchaudio.functional.resample(speech, orig_freq=sr, new_freq=model_sr) speech_16000 = speech_16000.squeeze() # Run pipeline result = pipe(np.array(speech_16000)) # Save text result to file transcript_path = f'whisper-large/{audio_path.replace(".mp3", "_transcript_whisper.txt")}' with open(transcript_path, "w") as f: f.write(result["text"]) return None # %% Transcription loop for audio_path in tqdm.tqdm(audio_paths): transcript(audio_dir, audio_path)