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import torch |
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import librosa |
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import torchaudio |
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import numpy as np |
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def compute_mel_spectrogram(audio, stft_processor): |
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return stft_processor.compute_mel_spectrogram(torch.autograd.Variable(torch.clip(torch.FloatTensor(audio).unsqueeze(0), -1, 1), requires_grad=False)).squeeze(0).numpy().astype(np.float32) |
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def pad_spectrogram(spectrogram, target_length=1024): |
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pad_amount = target_length - spectrogram.shape[0] |
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spectrogram = torch.nn.functional.pad(spectrogram, (0, 0, 0, pad_amount)) if pad_amount > 0 else spectrogram[:target_length, :] |
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if spectrogram.size(-1) % 2 != 0: spectrogram = spectrogram[..., :-1] |
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return spectrogram |
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def pad_waveform(waveform, segment_length): |
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waveform_length = waveform.shape[-1] |
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assert waveform_length > 100 |
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if segment_length is None or waveform_length == segment_length: return waveform |
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elif waveform_length > segment_length: return waveform[:, :segment_length] |
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padded_waveform = np.zeros((1, segment_length)) |
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padded_waveform[:, :waveform_length] = waveform |
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return padded_waveform |
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def normalize(waveform): |
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waveform -= np.mean(waveform) |
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return (waveform / (np.max(np.abs(waveform)) + 1e-8)) * 0.5 |
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def process_audio(y, sr, segment_length): |
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normalized_waveform = normalize(torchaudio.functional.resample(torch.from_numpy(y), orig_freq=sr, new_freq=16000).numpy())[None, ...] |
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return 0.5 * (pad_waveform(normalized_waveform, segment_length) / np.max(np.abs(normalized_waveform))) |
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def load_audio(audio_path, stft_processor, device=None): |
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y, sr = librosa.load(audio_path, sr=None) |
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duration = len(y) / sr |
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return pad_spectrogram(torch.FloatTensor(compute_mel_spectrogram(torch.FloatTensor(process_audio(y, sr, int(duration * 102.4) * 160)[0, ...]), stft_processor).T), int(duration * 102.4)).unsqueeze(0).unsqueeze(0).to(device), duration |