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import os |
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|
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import numpy as np |
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import torch |
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try: |
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|
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import intel_extension_for_pytorch as ipex |
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if torch.xpu.is_available(): |
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from lib.infer.modules.ipex import ipex_init |
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ipex_init() |
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except Exception: |
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pass |
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import torch.nn as nn |
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import torch.nn.functional as F |
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from librosa.util import normalize, pad_center, tiny |
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from scipy.signal import get_window |
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|
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import logging |
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|
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logger = logging.getLogger(__name__) |
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|
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def window_sumsquare( |
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window, |
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n_frames, |
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hop_length=200, |
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win_length=800, |
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n_fft=800, |
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dtype=np.float32, |
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norm=None, |
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): |
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""" |
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# from librosa 0.6 |
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Compute the sum-square envelope of a window function at a given hop length. |
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This is used to estimate modulation effects induced by windowing |
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observations in short-time fourier transforms. |
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Parameters |
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---------- |
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window : string, tuple, number, callable, or list-like |
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Window specification, as in `get_window` |
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n_frames : int > 0 |
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The number of analysis frames |
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hop_length : int > 0 |
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The number of samples to advance between frames |
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win_length : [optional] |
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The length of the window function. By default, this matches `n_fft`. |
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n_fft : int > 0 |
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The length of each analysis frame. |
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dtype : np.dtype |
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The data type of the output |
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Returns |
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------- |
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wss : np.ndarray, shape=`(n_fft + hop_length * (n_frames - 1))` |
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The sum-squared envelope of the window function |
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""" |
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if win_length is None: |
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win_length = n_fft |
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|
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n = n_fft + hop_length * (n_frames - 1) |
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x = np.zeros(n, dtype=dtype) |
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|
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win_sq = get_window(window, win_length, fftbins=True) |
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win_sq = normalize(win_sq, norm=norm) ** 2 |
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win_sq = pad_center(win_sq, n_fft) |
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|
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for i in range(n_frames): |
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sample = i * hop_length |
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x[sample : min(n, sample + n_fft)] += win_sq[: max(0, min(n_fft, n - sample))] |
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return x |
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|
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class STFT(torch.nn.Module): |
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def __init__( |
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self, filter_length=1024, hop_length=512, win_length=None, window="hann" |
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): |
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""" |
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This module implements an STFT using 1D convolution and 1D transpose convolutions. |
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This is a bit tricky so there are some cases that probably won't work as working |
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out the same sizes before and after in all overlap add setups is tough. Right now, |
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this code should work with hop lengths that are half the filter length (50% overlap |
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between frames). |
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|
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Keyword Arguments: |
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filter_length {int} -- Length of filters used (default: {1024}) |
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hop_length {int} -- Hop length of STFT (restrict to 50% overlap between frames) (default: {512}) |
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win_length {[type]} -- Length of the window function applied to each frame (if not specified, it |
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equals the filter length). (default: {None}) |
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window {str} -- Type of window to use (options are bartlett, hann, hamming, blackman, blackmanharris) |
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(default: {'hann'}) |
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""" |
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super(STFT, self).__init__() |
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self.filter_length = filter_length |
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self.hop_length = hop_length |
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self.win_length = win_length if win_length else filter_length |
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self.window = window |
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self.forward_transform = None |
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self.pad_amount = int(self.filter_length / 2) |
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scale = self.filter_length / self.hop_length |
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fourier_basis = np.fft.fft(np.eye(self.filter_length)) |
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|
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cutoff = int((self.filter_length / 2 + 1)) |
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fourier_basis = np.vstack( |
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[np.real(fourier_basis[:cutoff, :]), np.imag(fourier_basis[:cutoff, :])] |
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) |
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forward_basis = torch.FloatTensor(fourier_basis[:, None, :]) |
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inverse_basis = torch.FloatTensor( |
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np.linalg.pinv(scale * fourier_basis).T[:, None, :] |
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) |
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|
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assert filter_length >= self.win_length |
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|
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fft_window = get_window(window, self.win_length, fftbins=True) |
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fft_window = pad_center(fft_window, size=filter_length) |
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fft_window = torch.from_numpy(fft_window).float() |
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forward_basis *= fft_window |
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inverse_basis *= fft_window |
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|
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self.register_buffer("forward_basis", forward_basis.float()) |
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self.register_buffer("inverse_basis", inverse_basis.float()) |
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|
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def transform(self, input_data): |
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"""Take input data (audio) to STFT domain. |
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|
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Arguments: |
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input_data {tensor} -- Tensor of floats, with shape (num_batch, num_samples) |
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|
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Returns: |
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magnitude {tensor} -- Magnitude of STFT with shape (num_batch, |
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num_frequencies, num_frames) |
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phase {tensor} -- Phase of STFT with shape (num_batch, |
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num_frequencies, num_frames) |
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""" |
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num_batches = input_data.shape[0] |
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num_samples = input_data.shape[-1] |
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|
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self.num_samples = num_samples |
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input_data = input_data.view(num_batches, 1, num_samples) |
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|
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input_data = F.pad( |
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input_data.unsqueeze(1), |
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(self.pad_amount, self.pad_amount, 0, 0, 0, 0), |
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mode="reflect", |
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).squeeze(1) |
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|
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forward_transform = F.conv1d( |
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input_data, self.forward_basis, stride=self.hop_length, padding=0 |
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) |
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|
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cutoff = int((self.filter_length / 2) + 1) |
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real_part = forward_transform[:, :cutoff, :] |
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imag_part = forward_transform[:, cutoff:, :] |
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|
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magnitude = torch.sqrt(real_part**2 + imag_part**2) |
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return magnitude |
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|
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def inverse(self, magnitude, phase): |
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"""Call the inverse STFT (iSTFT), given magnitude and phase tensors produced |
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by the ```transform``` function. |
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|
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Arguments: |
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magnitude {tensor} -- Magnitude of STFT with shape (num_batch, |
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num_frequencies, num_frames) |
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phase {tensor} -- Phase of STFT with shape (num_batch, |
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num_frequencies, num_frames) |
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|
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Returns: |
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inverse_transform {tensor} -- Reconstructed audio given magnitude and phase. Of |
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shape (num_batch, num_samples) |
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""" |
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recombine_magnitude_phase = torch.cat( |
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[magnitude * torch.cos(phase), magnitude * torch.sin(phase)], dim=1 |
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) |
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|
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inverse_transform = F.conv_transpose1d( |
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recombine_magnitude_phase, |
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self.inverse_basis, |
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stride=self.hop_length, |
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padding=0, |
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) |
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|
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if self.window is not None: |
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window_sum = window_sumsquare( |
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self.window, |
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magnitude.size(-1), |
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hop_length=self.hop_length, |
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win_length=self.win_length, |
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n_fft=self.filter_length, |
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dtype=np.float32, |
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) |
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|
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approx_nonzero_indices = torch.from_numpy( |
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np.where(window_sum > tiny(window_sum))[0] |
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) |
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window_sum = torch.from_numpy(window_sum).to(inverse_transform.device) |
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inverse_transform[:, :, approx_nonzero_indices] /= window_sum[ |
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approx_nonzero_indices |
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] |
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|
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|
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inverse_transform *= float(self.filter_length) / self.hop_length |
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|
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inverse_transform = inverse_transform[..., self.pad_amount :] |
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inverse_transform = inverse_transform[..., : self.num_samples] |
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inverse_transform = inverse_transform.squeeze(1) |
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|
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return inverse_transform |
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|
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def forward(self, input_data): |
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"""Take input data (audio) to STFT domain and then back to audio. |
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|
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Arguments: |
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input_data {tensor} -- Tensor of floats, with shape (num_batch, num_samples) |
|
|
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Returns: |
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reconstruction {tensor} -- Reconstructed audio given magnitude and phase. Of |
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shape (num_batch, num_samples) |
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""" |
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self.magnitude, self.phase = self.transform(input_data) |
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reconstruction = self.inverse(self.magnitude, self.phase) |
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return reconstruction |
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|
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|
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from time import time as ttime |
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|
|
|
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class BiGRU(nn.Module): |
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def __init__(self, input_features, hidden_features, num_layers): |
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super(BiGRU, self).__init__() |
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self.gru = nn.GRU( |
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input_features, |
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hidden_features, |
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num_layers=num_layers, |
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batch_first=True, |
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bidirectional=True, |
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) |
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|
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def forward(self, x): |
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return self.gru(x)[0] |
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|
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|
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class ConvBlockRes(nn.Module): |
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def __init__(self, in_channels, out_channels, momentum=0.01): |
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super(ConvBlockRes, self).__init__() |
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self.conv = nn.Sequential( |
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nn.Conv2d( |
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in_channels=in_channels, |
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out_channels=out_channels, |
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kernel_size=(3, 3), |
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stride=(1, 1), |
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padding=(1, 1), |
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bias=False, |
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), |
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nn.BatchNorm2d(out_channels, momentum=momentum), |
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nn.ReLU(), |
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nn.Conv2d( |
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in_channels=out_channels, |
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out_channels=out_channels, |
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kernel_size=(3, 3), |
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stride=(1, 1), |
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padding=(1, 1), |
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bias=False, |
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), |
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nn.BatchNorm2d(out_channels, momentum=momentum), |
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nn.ReLU(), |
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) |
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if in_channels != out_channels: |
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self.shortcut = nn.Conv2d(in_channels, out_channels, (1, 1)) |
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self.is_shortcut = True |
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else: |
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self.is_shortcut = False |
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|
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def forward(self, x): |
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if self.is_shortcut: |
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return self.conv(x) + self.shortcut(x) |
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else: |
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return self.conv(x) + x |
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|
|
|
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class Encoder(nn.Module): |
|
def __init__( |
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self, |
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in_channels, |
|
in_size, |
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n_encoders, |
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kernel_size, |
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n_blocks, |
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out_channels=16, |
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momentum=0.01, |
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): |
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super(Encoder, self).__init__() |
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self.n_encoders = n_encoders |
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self.bn = nn.BatchNorm2d(in_channels, momentum=momentum) |
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self.layers = nn.ModuleList() |
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self.latent_channels = [] |
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for i in range(self.n_encoders): |
|
self.layers.append( |
|
ResEncoderBlock( |
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in_channels, out_channels, kernel_size, n_blocks, momentum=momentum |
|
) |
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) |
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self.latent_channels.append([out_channels, in_size]) |
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in_channels = out_channels |
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out_channels *= 2 |
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in_size //= 2 |
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self.out_size = in_size |
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self.out_channel = out_channels |
|
|
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def forward(self, x): |
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concat_tensors = [] |
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x = self.bn(x) |
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for i in range(self.n_encoders): |
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_, x = self.layers[i](x) |
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concat_tensors.append(_) |
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return x, concat_tensors |
|
|
|
|
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class ResEncoderBlock(nn.Module): |
|
def __init__( |
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self, in_channels, out_channels, kernel_size, n_blocks=1, momentum=0.01 |
|
): |
|
super(ResEncoderBlock, self).__init__() |
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self.n_blocks = n_blocks |
|
self.conv = nn.ModuleList() |
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self.conv.append(ConvBlockRes(in_channels, out_channels, momentum)) |
|
for i in range(n_blocks - 1): |
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self.conv.append(ConvBlockRes(out_channels, out_channels, momentum)) |
|
self.kernel_size = kernel_size |
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if self.kernel_size is not None: |
|
self.pool = nn.AvgPool2d(kernel_size=kernel_size) |
|
|
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def forward(self, x): |
|
for i in range(self.n_blocks): |
|
x = self.conv[i](x) |
|
if self.kernel_size is not None: |
|
return x, self.pool(x) |
|
else: |
|
return x |
|
|
|
|
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class Intermediate(nn.Module): |
|
def __init__(self, in_channels, out_channels, n_inters, n_blocks, momentum=0.01): |
|
super(Intermediate, self).__init__() |
|
self.n_inters = n_inters |
|
self.layers = nn.ModuleList() |
|
self.layers.append( |
|
ResEncoderBlock(in_channels, out_channels, None, n_blocks, momentum) |
|
) |
|
for i in range(self.n_inters - 1): |
|
self.layers.append( |
|
ResEncoderBlock(out_channels, out_channels, None, n_blocks, momentum) |
|
) |
|
|
|
def forward(self, x): |
|
for i in range(self.n_inters): |
|
x = self.layers[i](x) |
|
return x |
|
|
|
|
|
class ResDecoderBlock(nn.Module): |
|
def __init__(self, in_channels, out_channels, stride, n_blocks=1, momentum=0.01): |
|
super(ResDecoderBlock, self).__init__() |
|
out_padding = (0, 1) if stride == (1, 2) else (1, 1) |
|
self.n_blocks = n_blocks |
|
self.conv1 = nn.Sequential( |
|
nn.ConvTranspose2d( |
|
in_channels=in_channels, |
|
out_channels=out_channels, |
|
kernel_size=(3, 3), |
|
stride=stride, |
|
padding=(1, 1), |
|
output_padding=out_padding, |
|
bias=False, |
|
), |
|
nn.BatchNorm2d(out_channels, momentum=momentum), |
|
nn.ReLU(), |
|
) |
|
self.conv2 = nn.ModuleList() |
|
self.conv2.append(ConvBlockRes(out_channels * 2, out_channels, momentum)) |
|
for i in range(n_blocks - 1): |
|
self.conv2.append(ConvBlockRes(out_channels, out_channels, momentum)) |
|
|
|
def forward(self, x, concat_tensor): |
|
x = self.conv1(x) |
|
x = torch.cat((x, concat_tensor), dim=1) |
|
for i in range(self.n_blocks): |
|
x = self.conv2[i](x) |
|
return x |
|
|
|
|
|
class Decoder(nn.Module): |
|
def __init__(self, in_channels, n_decoders, stride, n_blocks, momentum=0.01): |
|
super(Decoder, self).__init__() |
|
self.layers = nn.ModuleList() |
|
self.n_decoders = n_decoders |
|
for i in range(self.n_decoders): |
|
out_channels = in_channels // 2 |
|
self.layers.append( |
|
ResDecoderBlock(in_channels, out_channels, stride, n_blocks, momentum) |
|
) |
|
in_channels = out_channels |
|
|
|
def forward(self, x, concat_tensors): |
|
for i in range(self.n_decoders): |
|
x = self.layers[i](x, concat_tensors[-1 - i]) |
|
return x |
|
|
|
|
|
class DeepUnet(nn.Module): |
|
def __init__( |
|
self, |
|
kernel_size, |
|
n_blocks, |
|
en_de_layers=5, |
|
inter_layers=4, |
|
in_channels=1, |
|
en_out_channels=16, |
|
): |
|
super(DeepUnet, self).__init__() |
|
self.encoder = Encoder( |
|
in_channels, 128, en_de_layers, kernel_size, n_blocks, en_out_channels |
|
) |
|
self.intermediate = Intermediate( |
|
self.encoder.out_channel // 2, |
|
self.encoder.out_channel, |
|
inter_layers, |
|
n_blocks, |
|
) |
|
self.decoder = Decoder( |
|
self.encoder.out_channel, en_de_layers, kernel_size, n_blocks |
|
) |
|
|
|
def forward(self, x): |
|
x, concat_tensors = self.encoder(x) |
|
x = self.intermediate(x) |
|
x = self.decoder(x, concat_tensors) |
|
return x |
|
|
|
|
|
class E2E(nn.Module): |
|
def __init__( |
|
self, |
|
n_blocks, |
|
n_gru, |
|
kernel_size, |
|
en_de_layers=5, |
|
inter_layers=4, |
|
in_channels=1, |
|
en_out_channels=16, |
|
): |
|
super(E2E, self).__init__() |
|
self.unet = DeepUnet( |
|
kernel_size, |
|
n_blocks, |
|
en_de_layers, |
|
inter_layers, |
|
in_channels, |
|
en_out_channels, |
|
) |
|
self.cnn = nn.Conv2d(en_out_channels, 3, (3, 3), padding=(1, 1)) |
|
if n_gru: |
|
self.fc = nn.Sequential( |
|
BiGRU(3 * 128, 256, n_gru), |
|
nn.Linear(512, 360), |
|
nn.Dropout(0.25), |
|
nn.Sigmoid(), |
|
) |
|
else: |
|
self.fc = nn.Sequential( |
|
nn.Linear(3 * nn.N_MELS, nn.N_CLASS), nn.Dropout(0.25), nn.Sigmoid() |
|
) |
|
|
|
def forward(self, mel): |
|
|
|
mel = mel.transpose(-1, -2).unsqueeze(1) |
|
x = self.cnn(self.unet(mel)).transpose(1, 2).flatten(-2) |
|
x = self.fc(x) |
|
|
|
return x |
|
|
|
|
|
from librosa.filters import mel |
|
|
|
|
|
class MelSpectrogram(torch.nn.Module): |
|
def __init__( |
|
self, |
|
is_half, |
|
n_mel_channels, |
|
sampling_rate, |
|
win_length, |
|
hop_length, |
|
n_fft=None, |
|
mel_fmin=0, |
|
mel_fmax=None, |
|
clamp=1e-5, |
|
): |
|
super().__init__() |
|
n_fft = win_length if n_fft is None else n_fft |
|
self.hann_window = {} |
|
mel_basis = mel( |
|
sr=sampling_rate, |
|
n_fft=n_fft, |
|
n_mels=n_mel_channels, |
|
fmin=mel_fmin, |
|
fmax=mel_fmax, |
|
htk=True, |
|
) |
|
mel_basis = torch.from_numpy(mel_basis).float() |
|
self.register_buffer("mel_basis", mel_basis) |
|
self.n_fft = win_length if n_fft is None else n_fft |
|
self.hop_length = hop_length |
|
self.win_length = win_length |
|
self.sampling_rate = sampling_rate |
|
self.n_mel_channels = n_mel_channels |
|
self.clamp = clamp |
|
self.is_half = is_half |
|
|
|
def forward(self, audio, keyshift=0, speed=1, center=True): |
|
factor = 2 ** (keyshift / 12) |
|
n_fft_new = int(np.round(self.n_fft * factor)) |
|
win_length_new = int(np.round(self.win_length * factor)) |
|
hop_length_new = int(np.round(self.hop_length * speed)) |
|
keyshift_key = str(keyshift) + "_" + str(audio.device) |
|
if keyshift_key not in self.hann_window: |
|
self.hann_window[keyshift_key] = torch.hann_window(win_length_new).to( |
|
|
|
audio.device |
|
) |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
if hasattr(self, "stft") == False: |
|
|
|
self.stft = STFT( |
|
filter_length=n_fft_new, |
|
hop_length=hop_length_new, |
|
win_length=win_length_new, |
|
window="hann", |
|
).to(audio.device) |
|
magnitude = self.stft.transform(audio) |
|
|
|
|
|
if keyshift != 0: |
|
size = self.n_fft // 2 + 1 |
|
resize = magnitude.size(1) |
|
if resize < size: |
|
magnitude = F.pad(magnitude, (0, 0, 0, size - resize)) |
|
magnitude = magnitude[:, :size, :] * self.win_length / win_length_new |
|
mel_output = torch.matmul(self.mel_basis, magnitude) |
|
if self.is_half == True: |
|
mel_output = mel_output.half() |
|
log_mel_spec = torch.log(torch.clamp(mel_output, min=self.clamp)) |
|
|
|
return log_mel_spec |
|
|
|
|
|
class RMVPE: |
|
def __init__(self, model_path, is_half, device=None): |
|
self.resample_kernel = {} |
|
self.resample_kernel = {} |
|
self.is_half = is_half |
|
if device is None: |
|
device = "cuda" if torch.cuda.is_available() else "cpu" |
|
self.device = device |
|
self.mel_extractor = MelSpectrogram( |
|
is_half, 128, 16000, 1024, 160, None, 30, 8000 |
|
).to(device) |
|
if "privateuseone" in str(device): |
|
import onnxruntime as ort |
|
|
|
ort_session = ort.InferenceSession( |
|
"%s/rmvpe.onnx" % os.environ["rmvpe_root"], |
|
providers=["DmlExecutionProvider"], |
|
) |
|
self.model = ort_session |
|
else: |
|
model = E2E(4, 1, (2, 2)) |
|
ckpt = torch.load(model_path, map_location="cpu") |
|
model.load_state_dict(ckpt) |
|
model.eval() |
|
if is_half == True: |
|
model = model.half() |
|
self.model = model |
|
self.model = self.model.to(device) |
|
cents_mapping = 20 * np.arange(360) + 1997.3794084376191 |
|
self.cents_mapping = np.pad(cents_mapping, (4, 4)) |
|
|
|
def mel2hidden(self, mel): |
|
with torch.no_grad(): |
|
n_frames = mel.shape[-1] |
|
mel = F.pad( |
|
mel, (0, 32 * ((n_frames - 1) // 32 + 1) - n_frames), mode="constant" |
|
) |
|
if "privateuseone" in str(self.device): |
|
onnx_input_name = self.model.get_inputs()[0].name |
|
onnx_outputs_names = self.model.get_outputs()[0].name |
|
hidden = self.model.run( |
|
[onnx_outputs_names], |
|
input_feed={onnx_input_name: mel.cpu().numpy()}, |
|
)[0] |
|
else: |
|
hidden = self.model(mel) |
|
return hidden[:, :n_frames] |
|
|
|
def decode(self, hidden, thred=0.03): |
|
cents_pred = self.to_local_average_cents(hidden, thred=thred) |
|
f0 = 10 * (2 ** (cents_pred / 1200)) |
|
f0[f0 == 10] = 0 |
|
|
|
return f0 |
|
|
|
def infer_from_audio(self, audio, thred=0.03): |
|
|
|
t0 = ttime() |
|
mel = self.mel_extractor( |
|
torch.from_numpy(audio).float().to(self.device).unsqueeze(0), center=True |
|
) |
|
|
|
|
|
t1 = ttime() |
|
hidden = self.mel2hidden(mel) |
|
|
|
t2 = ttime() |
|
|
|
if "privateuseone" not in str(self.device): |
|
hidden = hidden.squeeze(0).cpu().numpy() |
|
else: |
|
hidden = hidden[0] |
|
if self.is_half == True: |
|
hidden = hidden.astype("float32") |
|
|
|
f0 = self.decode(hidden, thred=thred) |
|
|
|
t3 = ttime() |
|
|
|
return f0 |
|
|
|
def infer_from_audio_with_pitch(self, audio, thred=0.03, f0_min=50, f0_max=1100): |
|
t0 = ttime() |
|
audio = torch.from_numpy(audio).float().to(self.device).unsqueeze(0) |
|
mel = self.mel_extractor(audio, center=True) |
|
t1 = ttime() |
|
hidden = self.mel2hidden(mel) |
|
t2 = ttime() |
|
if "privateuseone" not in str(self.device): |
|
hidden = hidden.squeeze(0).cpu().numpy() |
|
else: |
|
hidden = hidden[0] |
|
if self.is_half == True: |
|
hidden = hidden.astype("float32") |
|
f0 = self.decode(hidden, thred=thred) |
|
f0[(f0 < f0_min) | (f0 > f0_max)] = 0 |
|
t3 = ttime() |
|
return f0 |
|
|
|
def to_local_average_cents(self, salience, thred=0.05): |
|
|
|
center = np.argmax(salience, axis=1) |
|
salience = np.pad(salience, ((0, 0), (4, 4))) |
|
|
|
center += 4 |
|
todo_salience = [] |
|
todo_cents_mapping = [] |
|
starts = center - 4 |
|
ends = center + 5 |
|
for idx in range(salience.shape[0]): |
|
todo_salience.append(salience[:, starts[idx] : ends[idx]][idx]) |
|
todo_cents_mapping.append(self.cents_mapping[starts[idx] : ends[idx]]) |
|
|
|
todo_salience = np.array(todo_salience) |
|
todo_cents_mapping = np.array(todo_cents_mapping) |
|
product_sum = np.sum(todo_salience * todo_cents_mapping, 1) |
|
weight_sum = np.sum(todo_salience, 1) |
|
devided = product_sum / weight_sum |
|
|
|
maxx = np.max(salience, axis=1) |
|
devided[maxx <= thred] = 0 |
|
|
|
|
|
return devided |
|
|
|
|
|
if __name__ == "__main__": |
|
import librosa |
|
import soundfile as sf |
|
|
|
audio, sampling_rate = sf.read(r"C:\Users\liujing04\Desktop\Z\冬之花clip1.wav") |
|
if len(audio.shape) > 1: |
|
audio = librosa.to_mono(audio.transpose(1, 0)) |
|
audio_bak = audio.copy() |
|
if sampling_rate != 16000: |
|
audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000) |
|
model_path = r"D:\BaiduNetdiskDownload\RVC-beta-v2-0727AMD_realtime\rmvpe.pt" |
|
thred = 0.03 |
|
device = "cuda" if torch.cuda.is_available() else "cpu" |
|
rmvpe = RMVPE(model_path, is_half=False, device=device) |
|
t0 = ttime() |
|
f0 = rmvpe.infer_from_audio(audio, thred=thred) |
|
|
|
|
|
|
|
|
|
t1 = ttime() |
|
logger.info("%s %.2f", f0.shape, t1 - t0) |
|
|