| import gradio as gr |
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| import torch |
| import torch.nn as nn |
| import torch.nn.functional as F |
| import torchaudio |
| from transformers import AutoConfig, Wav2Vec2FeatureExtractor |
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|
| import librosa |
| import IPython.display as ipd |
| import numpy as np |
| import pandas as pd |
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| device = torch.device("cuda" if torch.cuda.is_available() else "cpu") |
| model_name_or_path = "m3hrdadfi/wav2vec2-base-100k-voxpopuli-gtzan-music" |
| config = AutoConfig.from_pretrained(model_name_or_path) |
| feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(model_name_or_path) |
| sampling_rate = feature_extractor.sampling_rate |
| model = Wav2Vec2ForSpeechClassification.from_pretrained(model_name_or_path).to(device) |
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| def speech_file_to_array_fn(path, sampling_rate): |
| speech_array, _sampling_rate = torchaudio.load(path) |
| resampler = torchaudio.transforms.Resample(_sampling_rate) |
| speech = resampler(speech_array).squeeze().numpy() |
| return speech |
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|
| def predict(path, sampling_rate): |
| speech = speech_file_to_array_fn(path, sampling_rate) |
| inputs = feature_extractor(speech, sampling_rate=sampling_rate, return_tensors="pt", padding=True) |
| inputs = {key: inputs[key].to(device) for key in inputs} |
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| with torch.no_grad(): |
| logits = model(**inputs).logits |
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| scores = F.softmax(logits, dim=1).detach().cpu().numpy()[0] |
| outputs = [{"Label": config.id2label[i], "Score": f"{round(score * 100, 3):.1f}%"} for i, score in enumerate(scores)] |
| return outputs |
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| path = "La Campanella.mp3" |
| outputs = predict(path, sampling_rate) |
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| iface = gr.Interface(fn=predict, inputs=path, outputs=predict(path, sampling_rate)) |
| iface.launch() |
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