import librosa import torch import numpy as np import langid # Language detection library from transformers import Wav2Vec2ForCTC, AutoProcessor ASR_SAMPLING_RATE = 16_000 MODEL_ID = "facebook/mms-1b-all" # openai/whisper-large-v3-turbo #ASR_SAMPLING_RATE = 16_000 #MODEL_ID = "openai/whisper-large-v3-turbo" # Load MMS Model processor = AutoProcessor.from_pretrained(MODEL_ID) model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID) model.eval() def detect_language(text): """Detects language using langid (fast & lightweight).""" lang, _ = langid.classify(text) return lang if lang in ["en", "sw"] else "en" # Default to English def transcribe_auto(audio_data=None): if not audio_data: return "<>" # Process Microphone Input if isinstance(audio_data, tuple): sr, audio_samples = audio_data audio_samples = (audio_samples / 32768.0).astype(np.float32) if sr != ASR_SAMPLING_RATE: audio_samples = librosa.resample(audio_samples, orig_sr=sr, target_sr=ASR_SAMPLING_RATE) # Process File Upload Input else: if not isinstance(audio_data, str): return "<>" audio_samples = librosa.load(audio_data, sr=ASR_SAMPLING_RATE, mono=True)[0] inputs = processor(audio_samples, sampling_rate=ASR_SAMPLING_RATE, return_tensors="pt") # **Step 1: Transcribe without Language Detection** with torch.no_grad(): outputs = model(**inputs).logits ids = torch.argmax(outputs, dim=-1)[0] raw_transcription = processor.decode(ids) # **Step 2: Detect Language from Transcription** detected_lang = detect_language(raw_transcription) lang_code = "eng" if detected_lang == "en" else "swh" # **Step 3: Reload Model with Correct Adapter** processor.tokenizer.set_target_lang(lang_code) model.load_adapter(lang_code) # **Step 4: Transcribe Again with Correct Adapter** with torch.no_grad(): outputs = model(**inputs).logits ids = torch.argmax(outputs, dim=-1)[0] final_transcription = processor.decode(ids) return f"{final_transcription}"