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| # Copyright (c) Facebook, Inc. and its affiliates. | |
| # All rights reserved. | |
| # | |
| # This source code is licensed under the license found in the | |
| # LICENSE file in the root directory of this source tree. | |
| import math | |
| import julius | |
| from torch import nn | |
| from .tasnet_v2 import ConvTasNet | |
| from .utils import capture_init, center_trim | |
| class BLSTM(nn.Module): | |
| def __init__(self, dim, layers=1): | |
| super().__init__() | |
| self.lstm = nn.LSTM(bidirectional=True, num_layers=layers, hidden_size=dim, input_size=dim) | |
| self.linear = nn.Linear(2 * dim, dim) | |
| def forward(self, x): | |
| x = x.permute(2, 0, 1) | |
| x = self.lstm(x)[0] | |
| x = self.linear(x) | |
| x = x.permute(1, 2, 0) | |
| return x | |
| def rescale_conv(conv, reference): | |
| std = conv.weight.std().detach() | |
| scale = (std / reference)**0.5 | |
| conv.weight.data /= scale | |
| if conv.bias is not None: | |
| conv.bias.data /= scale | |
| def rescale_module(module, reference): | |
| for sub in module.modules(): | |
| if isinstance(sub, (nn.Conv1d, nn.ConvTranspose1d)): | |
| rescale_conv(sub, reference) | |
| def auto_load_demucs_model_v2(sources, demucs_model_name): | |
| if '48' in demucs_model_name: | |
| channels=48 | |
| elif 'unittest' in demucs_model_name: | |
| channels=4 | |
| else: | |
| channels=64 | |
| if 'tasnet' in demucs_model_name: | |
| init_demucs_model = ConvTasNet(sources, X=10) | |
| else: | |
| init_demucs_model = Demucs(sources, channels=channels) | |
| return init_demucs_model | |
| class Demucs(nn.Module): | |
| def __init__(self, | |
| sources, | |
| audio_channels=2, | |
| channels=64, | |
| depth=6, | |
| rewrite=True, | |
| glu=True, | |
| rescale=0.1, | |
| resample=True, | |
| kernel_size=8, | |
| stride=4, | |
| growth=2., | |
| lstm_layers=2, | |
| context=3, | |
| normalize=False, | |
| samplerate=44100, | |
| segment_length=4 * 10 * 44100): | |
| """ | |
| Args: | |
| sources (list[str]): list of source names | |
| audio_channels (int): stereo or mono | |
| channels (int): first convolution channels | |
| depth (int): number of encoder/decoder layers | |
| rewrite (bool): add 1x1 convolution to each encoder layer | |
| and a convolution to each decoder layer. | |
| For the decoder layer, `context` gives the kernel size. | |
| glu (bool): use glu instead of ReLU | |
| resample_input (bool): upsample x2 the input and downsample /2 the output. | |
| rescale (int): rescale initial weights of convolutions | |
| to get their standard deviation closer to `rescale` | |
| kernel_size (int): kernel size for convolutions | |
| stride (int): stride for convolutions | |
| growth (float): multiply (resp divide) number of channels by that | |
| for each layer of the encoder (resp decoder) | |
| lstm_layers (int): number of lstm layers, 0 = no lstm | |
| context (int): kernel size of the convolution in the | |
| decoder before the transposed convolution. If > 1, | |
| will provide some context from neighboring time | |
| steps. | |
| samplerate (int): stored as meta information for easing | |
| future evaluations of the model. | |
| segment_length (int): stored as meta information for easing | |
| future evaluations of the model. Length of the segments on which | |
| the model was trained. | |
| """ | |
| super().__init__() | |
| self.audio_channels = audio_channels | |
| self.sources = sources | |
| self.kernel_size = kernel_size | |
| self.context = context | |
| self.stride = stride | |
| self.depth = depth | |
| self.resample = resample | |
| self.channels = channels | |
| self.normalize = normalize | |
| self.samplerate = samplerate | |
| self.segment_length = segment_length | |
| self.encoder = nn.ModuleList() | |
| self.decoder = nn.ModuleList() | |
| if glu: | |
| activation = nn.GLU(dim=1) | |
| ch_scale = 2 | |
| else: | |
| activation = nn.ReLU() | |
| ch_scale = 1 | |
| in_channels = audio_channels | |
| for index in range(depth): | |
| encode = [] | |
| encode += [nn.Conv1d(in_channels, channels, kernel_size, stride), nn.ReLU()] | |
| if rewrite: | |
| encode += [nn.Conv1d(channels, ch_scale * channels, 1), 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, context), activation] | |
| decode += [nn.ConvTranspose1d(channels, out_channels, kernel_size, stride)] | |
| if index > 0: | |
| decode.append(nn.ReLU()) | |
| 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 convolutions, e.g. for all | |
| layers, size of the input - kernel_size % stride = 0. | |
| If the mixture has a valid length, the estimated sources | |
| will have exactly the same length when context = 1. If context > 1, | |
| the two signals can be center trimmed to match. | |
| For training, extracts should have a valid length.For evaluation | |
| on full tracks we recommend passing `pad = True` to :method:`forward`. | |
| """ | |
| if self.resample: | |
| length *= 2 | |
| for _ in range(self.depth): | |
| length = math.ceil((length - self.kernel_size) / self.stride) + 1 | |
| length = max(1, length) | |
| length += self.context - 1 | |
| for _ 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 | |
| if self.normalize: | |
| mono = mix.mean(dim=1, keepdim=True) | |
| mean = mono.mean(dim=-1, keepdim=True) | |
| std = mono.std(dim=-1, keepdim=True) | |
| else: | |
| mean = 0 | |
| std = 1 | |
| x = (x - mean) / (1e-5 + std) | |
| 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 = center_trim(saved.pop(-1), x) | |
| x = x + skip | |
| x = decode(x) | |
| if self.resample: | |
| x = julius.resample_frac(x, 2, 1) | |
| x = x * std + mean | |
| x = x.view(x.size(0), len(self.sources), self.audio_channels, x.size(-1)) | |
| return x | |