| import torch | |
| from langchain_core.tools import tool | |
| from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline | |
| from .data_helpers import get_file_path | |
| def transcribe_audio_file(file_name: str) -> str: | |
| """ | |
| Transcribes an audio file to text. | |
| Args: | |
| file_name: The name of the audio file. This is simply the file name, | |
| not the full path. | |
| Returns: | |
| The transcribed text. | |
| """ | |
| # Specific setting for local run with GPU busy for the LLM (ollama) | |
| cuda_available = False | |
| device = "cuda:0" if cuda_available else "cpu" | |
| torch_dtype = torch.float16 if cuda_available else torch.float32 | |
| model_id = "openai/whisper-large-v3-turbo" | |
| model = AutoModelForSpeechSeq2Seq.from_pretrained( | |
| model_id, | |
| torch_dtype=torch_dtype, | |
| low_cpu_mem_usage=True, | |
| use_safetensors=True | |
| ) | |
| model.to(device) | |
| processor = AutoProcessor.from_pretrained(model_id) | |
| pipe = pipeline( | |
| "automatic-speech-recognition", | |
| model=model, | |
| tokenizer=processor.tokenizer, | |
| feature_extractor=processor.feature_extractor, | |
| torch_dtype=torch_dtype, | |
| device=device, | |
| ) | |
| generate_kwargs = { | |
| "return_timestamps": True, | |
| } | |
| file_path = get_file_path(file_name) | |
| result = pipe(file_path, generate_kwargs=generate_kwargs) | |
| return result["text"] | |