from transformers import Wav2Vec2ForSequenceClassification, AutoFeatureExtractor import torch import librosa import numpy as np model_id = "facebook/mms-lid-1024" processor = AutoFeatureExtractor.from_pretrained(model_id) model = Wav2Vec2ForSequenceClassification.from_pretrained(model_id) LID_SAMPLING_RATE = 16_000 LID_TOPK = 10 LID_THRESHOLD = 0.33 LID_LANGUAGES = {} with open(f"data/lid/all_langs.tsv") as f: for line in f: iso, name = line.split(" ", 1) LID_LANGUAGES[iso] = name def identify(audio_data = None): if not audio_data: return "<>" if isinstance(audio_data, tuple): # microphone sr, audio_samples = audio_data audio_samples = (audio_samples / 32768.0).astype(np.float32) if sr != LID_SAMPLING_RATE: audio_samples = librosa.resample( audio_samples, orig_sr=sr, target_sr=LID_SAMPLING_RATE ) else: # file upload isinstance(audio_data, str) audio_samples = librosa.load(audio_data, sr=LID_SAMPLING_RATE, mono=True)[0] inputs = processor( audio_samples, sampling_rate=LID_SAMPLING_RATE, return_tensors="pt" ) # set device if torch.cuda.is_available(): device = torch.device("cuda") elif ( hasattr(torch.backends, "mps") and torch.backends.mps.is_available() and torch.backends.mps.is_built() ): device = torch.device("mps") else: device = torch.device("cpu") model.to(device) inputs = inputs.to(device) with torch.no_grad(): logit = model(**inputs).logits logit_lsm = torch.log_softmax(logit.squeeze(), dim=-1) scores, indices = torch.topk(logit_lsm, 5, dim=-1) scores, indices = torch.exp(scores).to("cpu").tolist(), indices.to("cpu").tolist() iso2score = {model.config.id2label[int(i)]: s for s, i in zip(scores, indices)} if max(iso2score.values()) < LID_THRESHOLD: return "Low confidence in the language identification predictions. Output is not shown!" return {LID_LANGUAGES[iso]: score for iso, score in iso2score.items()} LID_EXAMPLES = [ ["upload/english.mp3"], ["upload/tamil.mp3"], ["upload/burmese.mp3"], ]