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