asr.py

import librosa
from transformers import Wav2Vec2ForCTC, AutoProcessor
import torch
import logging

# Set up logging
logging.basicConfig(level=logging.DEBUG)

ASR_SAMPLING_RATE = 16_000

MODEL_ID = "facebook/wav2vec2-large-960h-lv60-self"

try:
    processor = AutoProcessor.from_pretrained(MODEL_ID)
    model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
    logging.info("ASR model and processor loaded successfully.")
except Exception as e:
    logging.error(f"Error loading ASR model or processor: {e}")

def transcribe(audio):
    try:
        if audio is None:
            return "ERROR: You have to either use the microphone or upload an audio file"
        
        audio_samples = librosa.load(audio, sr=ASR_SAMPLING_RATE, mono=True)[0]
        inputs = processor(audio_samples, sampling_rate=ASR_SAMPLING_RATE, return_tensors="pt")
        
        # Set language ID for Faroese
        language_id = 'fao'  # ISO 639-3 code for Faroese
        processor.tokenizer.set_lang(language_id)

        device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
        model.to(device)
        inputs = inputs.to(device)

        with torch.no_grad():
            outputs = model(**inputs).logits

        ids = torch.argmax(outputs, dim=-1)[0]
        transcription = processor.decode(ids)
        
        logging.info("Transcription completed successfully.")
        return transcription
    except Exception as e:
        logging.error(f"Error during transcription: {e}")
        return "ERROR"