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Create lid.py
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lid.py
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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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# microphone
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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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# file upload
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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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# set device
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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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]
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