mirror of
https://github.com/Anjok07/ultimatevocalremovergui.git
synced 2024-11-14 19:07:37 +01:00
460 lines
17 KiB
Python
460 lines
17 KiB
Python
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# Copyright (c) Facebook, Inc. and its affiliates.
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# All rights reserved.
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#
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# This source code is licensed under the license found in the
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# LICENSE file in the root directory of this source tree.
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import math
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import typing as tp
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import julius
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import torch
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from torch import nn
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from torch.nn import functional as F
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from .states import capture_init
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from .utils import center_trim, unfold
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class BLSTM(nn.Module):
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"""
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BiLSTM with same hidden units as input dim.
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If `max_steps` is not None, input will be splitting in overlapping
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chunks and the LSTM applied separately on each chunk.
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"""
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def __init__(self, dim, layers=1, max_steps=None, skip=False):
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super().__init__()
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assert max_steps is None or max_steps % 4 == 0
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self.max_steps = max_steps
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self.lstm = nn.LSTM(bidirectional=True, num_layers=layers, hidden_size=dim, input_size=dim)
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self.linear = nn.Linear(2 * dim, dim)
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self.skip = skip
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def forward(self, x):
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B, C, T = x.shape
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y = x
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framed = False
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if self.max_steps is not None and T > self.max_steps:
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width = self.max_steps
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stride = width // 2
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frames = unfold(x, width, stride)
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nframes = frames.shape[2]
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framed = True
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x = frames.permute(0, 2, 1, 3).reshape(-1, C, width)
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x = x.permute(2, 0, 1)
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x = self.lstm(x)[0]
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x = self.linear(x)
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x = x.permute(1, 2, 0)
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if framed:
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out = []
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frames = x.reshape(B, -1, C, width)
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limit = stride // 2
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for k in range(nframes):
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if k == 0:
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out.append(frames[:, k, :, :-limit])
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elif k == nframes - 1:
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out.append(frames[:, k, :, limit:])
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else:
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out.append(frames[:, k, :, limit:-limit])
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out = torch.cat(out, -1)
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out = out[..., :T]
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x = out
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if self.skip:
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x = x + y
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return x
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def rescale_conv(conv, reference):
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"""Rescale initial weight scale. It is unclear why it helps but it certainly does.
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"""
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std = conv.weight.std().detach()
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scale = (std / reference)**0.5
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conv.weight.data /= scale
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if conv.bias is not None:
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conv.bias.data /= scale
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def rescale_module(module, reference):
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for sub in module.modules():
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if isinstance(sub, (nn.Conv1d, nn.ConvTranspose1d, nn.Conv2d, nn.ConvTranspose2d)):
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rescale_conv(sub, reference)
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class LayerScale(nn.Module):
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"""Layer scale from [Touvron et al 2021] (https://arxiv.org/pdf/2103.17239.pdf).
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This rescales diagonaly residual outputs close to 0 initially, then learnt.
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"""
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def __init__(self, channels: int, init: float = 0):
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super().__init__()
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self.scale = nn.Parameter(torch.zeros(channels, requires_grad=True))
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self.scale.data[:] = init
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def forward(self, x):
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return self.scale[:, None] * x
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class DConv(nn.Module):
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"""
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New residual branches in each encoder layer.
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This alternates dilated convolutions, potentially with LSTMs and attention.
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Also before entering each residual branch, dimension is projected on a smaller subspace,
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e.g. of dim `channels // compress`.
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"""
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def __init__(self, channels: int, compress: float = 4, depth: int = 2, init: float = 1e-4,
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norm=True, attn=False, heads=4, ndecay=4, lstm=False, gelu=True,
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kernel=3, dilate=True):
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"""
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Args:
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channels: input/output channels for residual branch.
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compress: amount of channel compression inside the branch.
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depth: number of layers in the residual branch. Each layer has its own
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projection, and potentially LSTM and attention.
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init: initial scale for LayerNorm.
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norm: use GroupNorm.
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attn: use LocalAttention.
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heads: number of heads for the LocalAttention.
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ndecay: number of decay controls in the LocalAttention.
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lstm: use LSTM.
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gelu: Use GELU activation.
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kernel: kernel size for the (dilated) convolutions.
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dilate: if true, use dilation, increasing with the depth.
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"""
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super().__init__()
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assert kernel % 2 == 1
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self.channels = channels
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self.compress = compress
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self.depth = abs(depth)
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dilate = depth > 0
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norm_fn: tp.Callable[[int], nn.Module]
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norm_fn = lambda d: nn.Identity() # noqa
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if norm:
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norm_fn = lambda d: nn.GroupNorm(1, d) # noqa
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hidden = int(channels / compress)
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act: tp.Type[nn.Module]
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if gelu:
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act = nn.GELU
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else:
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act = nn.ReLU
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self.layers = nn.ModuleList([])
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for d in range(self.depth):
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dilation = 2 ** d if dilate else 1
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padding = dilation * (kernel // 2)
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mods = [
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nn.Conv1d(channels, hidden, kernel, dilation=dilation, padding=padding),
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norm_fn(hidden), act(),
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nn.Conv1d(hidden, 2 * channels, 1),
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norm_fn(2 * channels), nn.GLU(1),
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LayerScale(channels, init),
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]
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if attn:
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mods.insert(3, LocalState(hidden, heads=heads, ndecay=ndecay))
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if lstm:
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mods.insert(3, BLSTM(hidden, layers=2, max_steps=200, skip=True))
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layer = nn.Sequential(*mods)
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self.layers.append(layer)
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def forward(self, x):
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for layer in self.layers:
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x = x + layer(x)
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return x
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class LocalState(nn.Module):
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"""Local state allows to have attention based only on data (no positional embedding),
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but while setting a constraint on the time window (e.g. decaying penalty term).
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Also a failed experiments with trying to provide some frequency based attention.
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"""
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def __init__(self, channels: int, heads: int = 4, nfreqs: int = 0, ndecay: int = 4):
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super().__init__()
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assert channels % heads == 0, (channels, heads)
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self.heads = heads
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self.nfreqs = nfreqs
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self.ndecay = ndecay
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self.content = nn.Conv1d(channels, channels, 1)
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self.query = nn.Conv1d(channels, channels, 1)
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self.key = nn.Conv1d(channels, channels, 1)
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if nfreqs:
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self.query_freqs = nn.Conv1d(channels, heads * nfreqs, 1)
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if ndecay:
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self.query_decay = nn.Conv1d(channels, heads * ndecay, 1)
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# Initialize decay close to zero (there is a sigmoid), for maximum initial window.
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self.query_decay.weight.data *= 0.01
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assert self.query_decay.bias is not None # stupid type checker
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self.query_decay.bias.data[:] = -2
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self.proj = nn.Conv1d(channels + heads * nfreqs, channels, 1)
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def forward(self, x):
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B, C, T = x.shape
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heads = self.heads
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indexes = torch.arange(T, device=x.device, dtype=x.dtype)
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# left index are keys, right index are queries
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delta = indexes[:, None] - indexes[None, :]
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queries = self.query(x).view(B, heads, -1, T)
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keys = self.key(x).view(B, heads, -1, T)
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# t are keys, s are queries
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dots = torch.einsum("bhct,bhcs->bhts", keys, queries)
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dots /= keys.shape[2]**0.5
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if self.nfreqs:
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periods = torch.arange(1, self.nfreqs + 1, device=x.device, dtype=x.dtype)
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freq_kernel = torch.cos(2 * math.pi * delta / periods.view(-1, 1, 1))
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freq_q = self.query_freqs(x).view(B, heads, -1, T) / self.nfreqs ** 0.5
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dots += torch.einsum("fts,bhfs->bhts", freq_kernel, freq_q)
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if self.ndecay:
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decays = torch.arange(1, self.ndecay + 1, device=x.device, dtype=x.dtype)
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decay_q = self.query_decay(x).view(B, heads, -1, T)
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decay_q = torch.sigmoid(decay_q) / 2
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decay_kernel = - decays.view(-1, 1, 1) * delta.abs() / self.ndecay**0.5
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dots += torch.einsum("fts,bhfs->bhts", decay_kernel, decay_q)
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# Kill self reference.
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dots.masked_fill_(torch.eye(T, device=dots.device, dtype=torch.bool), -100)
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weights = torch.softmax(dots, dim=2)
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content = self.content(x).view(B, heads, -1, T)
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result = torch.einsum("bhts,bhct->bhcs", weights, content)
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if self.nfreqs:
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time_sig = torch.einsum("bhts,fts->bhfs", weights, freq_kernel)
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result = torch.cat([result, time_sig], 2)
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result = result.reshape(B, -1, T)
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return x + self.proj(result)
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class Demucs(nn.Module):
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@capture_init
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def __init__(self,
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sources,
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# Channels
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audio_channels=2,
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channels=64,
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growth=2.,
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# Main structure
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depth=6,
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rewrite=True,
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lstm_layers=0,
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# Convolutions
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kernel_size=8,
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stride=4,
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context=1,
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# Activations
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gelu=True,
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glu=True,
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# Normalization
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norm_starts=4,
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norm_groups=4,
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# DConv residual branch
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dconv_mode=1,
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dconv_depth=2,
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dconv_comp=4,
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dconv_attn=4,
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dconv_lstm=4,
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dconv_init=1e-4,
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# Pre/post processing
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normalize=True,
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resample=True,
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# Weight init
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rescale=0.1,
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# Metadata
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samplerate=44100,
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segment=4 * 10):
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"""
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Args:
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sources (list[str]): list of source names
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audio_channels (int): stereo or mono
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channels (int): first convolution channels
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depth (int): number of encoder/decoder layers
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growth (float): multiply (resp divide) number of channels by that
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for each layer of the encoder (resp decoder)
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depth (int): number of layers in the encoder and in the decoder.
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rewrite (bool): add 1x1 convolution to each layer.
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lstm_layers (int): number of lstm layers, 0 = no lstm. Deactivated
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by default, as this is now replaced by the smaller and faster small LSTMs
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in the DConv branches.
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kernel_size (int): kernel size for convolutions
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stride (int): stride for convolutions
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context (int): kernel size of the convolution in the
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decoder before the transposed convolution. If > 1,
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will provide some context from neighboring time steps.
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gelu: use GELU activation function.
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glu (bool): use glu instead of ReLU for the 1x1 rewrite conv.
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norm_starts: layer at which group norm starts being used.
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decoder layers are numbered in reverse order.
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norm_groups: number of groups for group norm.
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dconv_mode: if 1: dconv in encoder only, 2: decoder only, 3: both.
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dconv_depth: depth of residual DConv branch.
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dconv_comp: compression of DConv branch.
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dconv_attn: adds attention layers in DConv branch starting at this layer.
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dconv_lstm: adds a LSTM layer in DConv branch starting at this layer.
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dconv_init: initial scale for the DConv branch LayerScale.
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normalize (bool): normalizes the input audio on the fly, and scales back
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the output by the same amount.
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resample (bool): upsample x2 the input and downsample /2 the output.
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rescale (int): rescale initial weights of convolutions
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to get their standard deviation closer to `rescale`.
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samplerate (int): stored as meta information for easing
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future evaluations of the model.
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segment (float): duration of the chunks of audio to ideally evaluate the model on.
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This is used by `demucs.apply.apply_model`.
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"""
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super().__init__()
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self.audio_channels = audio_channels
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self.sources = sources
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self.kernel_size = kernel_size
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self.context = context
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self.stride = stride
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self.depth = depth
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self.resample = resample
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self.channels = channels
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self.normalize = normalize
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self.samplerate = samplerate
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self.segment = segment
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self.encoder = nn.ModuleList()
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self.decoder = nn.ModuleList()
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self.skip_scales = nn.ModuleList()
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if glu:
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activation = nn.GLU(dim=1)
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ch_scale = 2
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else:
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activation = nn.ReLU()
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ch_scale = 1
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if gelu:
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act2 = nn.GELU
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else:
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act2 = nn.ReLU
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in_channels = audio_channels
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padding = 0
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for index in range(depth):
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norm_fn = lambda d: nn.Identity() # noqa
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if index >= norm_starts:
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norm_fn = lambda d: nn.GroupNorm(norm_groups, d) # noqa
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encode = []
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encode += [
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nn.Conv1d(in_channels, channels, kernel_size, stride),
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norm_fn(channels),
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act2(),
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]
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attn = index >= dconv_attn
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lstm = index >= dconv_lstm
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if dconv_mode & 1:
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encode += [DConv(channels, depth=dconv_depth, init=dconv_init,
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compress=dconv_comp, attn=attn, lstm=lstm)]
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if rewrite:
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|
encode += [
|
||
|
|
nn.Conv1d(channels, ch_scale * channels, 1),
|
||
|
|
norm_fn(ch_scale * channels), activation]
|
||
|
|
self.encoder.append(nn.Sequential(*encode))
|
||
|
|
|
||
|
|
decode = []
|
||
|
|
if index > 0:
|
||
|
|
out_channels = in_channels
|
||
|
|
else:
|
||
|
|
out_channels = len(self.sources) * audio_channels
|
||
|
|
if rewrite:
|
||
|
|
decode += [
|
||
|
|
nn.Conv1d(channels, ch_scale * channels, 2 * context + 1, padding=context),
|
||
|
|
norm_fn(ch_scale * channels), activation]
|
||
|
|
if dconv_mode & 2:
|
||
|
|
decode += [DConv(channels, depth=dconv_depth, init=dconv_init,
|
||
|
|
compress=dconv_comp, attn=attn, lstm=lstm)]
|
||
|
|
decode += [nn.ConvTranspose1d(channels, out_channels,
|
||
|
|
kernel_size, stride, padding=padding)]
|
||
|
|
if index > 0:
|
||
|
|
decode += [norm_fn(out_channels), act2()]
|
||
|
|
self.decoder.insert(0, nn.Sequential(*decode))
|
||
|
|
in_channels = channels
|
||
|
|
channels = int(growth * channels)
|
||
|
|
|
||
|
|
channels = in_channels
|
||
|
|
if lstm_layers:
|
||
|
|
self.lstm = BLSTM(channels, lstm_layers)
|
||
|
|
else:
|
||
|
|
self.lstm = None
|
||
|
|
|
||
|
|
if rescale:
|
||
|
|
rescale_module(self, reference=rescale)
|
||
|
|
|
||
|
|
def valid_length(self, length):
|
||
|
|
"""
|
||
|
|
Return the nearest valid length to use with the model so that
|
||
|
|
there is no time steps left over in a convolution, e.g. for all
|
||
|
|
layers, size of the input - kernel_size % stride = 0.
|
||
|
|
|
||
|
|
Note that input are automatically padded if necessary to ensure that the output
|
||
|
|
has the same length as the input.
|
||
|
|
"""
|
||
|
|
if self.resample:
|
||
|
|
length *= 2
|
||
|
|
|
||
|
|
for _ in range(self.depth):
|
||
|
|
length = math.ceil((length - self.kernel_size) / self.stride) + 1
|
||
|
|
length = max(1, length)
|
||
|
|
|
||
|
|
for idx in range(self.depth):
|
||
|
|
length = (length - 1) * self.stride + self.kernel_size
|
||
|
|
|
||
|
|
if self.resample:
|
||
|
|
length = math.ceil(length / 2)
|
||
|
|
return int(length)
|
||
|
|
|
||
|
|
def forward(self, mix):
|
||
|
|
x = mix
|
||
|
|
length = x.shape[-1]
|
||
|
|
|
||
|
|
if self.normalize:
|
||
|
|
mono = mix.mean(dim=1, keepdim=True)
|
||
|
|
mean = mono.mean(dim=-1, keepdim=True)
|
||
|
|
std = mono.std(dim=-1, keepdim=True)
|
||
|
|
x = (x - mean) / (1e-5 + std)
|
||
|
|
else:
|
||
|
|
mean = 0
|
||
|
|
std = 1
|
||
|
|
|
||
|
|
delta = self.valid_length(length) - length
|
||
|
|
x = F.pad(x, (delta // 2, delta - delta // 2))
|
||
|
|
|
||
|
|
if self.resample:
|
||
|
|
x = julius.resample_frac(x, 1, 2)
|
||
|
|
|
||
|
|
saved = []
|
||
|
|
for encode in self.encoder:
|
||
|
|
x = encode(x)
|
||
|
|
saved.append(x)
|
||
|
|
|
||
|
|
if self.lstm:
|
||
|
|
x = self.lstm(x)
|
||
|
|
|
||
|
|
for decode in self.decoder:
|
||
|
|
skip = saved.pop(-1)
|
||
|
|
skip = center_trim(skip, x)
|
||
|
|
x = decode(x + skip)
|
||
|
|
|
||
|
|
if self.resample:
|
||
|
|
x = julius.resample_frac(x, 2, 1)
|
||
|
|
x = x * std + mean
|
||
|
|
x = center_trim(x, length)
|
||
|
|
x = x.view(x.size(0), len(self.sources), self.audio_channels, x.size(-1))
|
||
|
|
return x
|
||
|
|
|
||
|
|
def load_state_dict(self, state, strict=True):
|
||
|
|
# fix a mismatch with previous generation Demucs models.
|
||
|
|
for idx in range(self.depth):
|
||
|
|
for a in ['encoder', 'decoder']:
|
||
|
|
for b in ['bias', 'weight']:
|
||
|
|
new = f'{a}.{idx}.3.{b}'
|
||
|
|
old = f'{a}.{idx}.2.{b}'
|
||
|
|
if old in state and new not in state:
|
||
|
|
state[new] = state.pop(old)
|
||
|
|
super().load_state_dict(state, strict=strict)
|