ultimatevocalremovergui/demucs/htdemucs.py

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# Copyright (c) Meta, 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.
# First author is Simon Rouard.
"""
This code contains the spectrogram and Hybrid version of Demucs.
"""
import math
from .filtering import wiener
import torch
from torch import nn
from torch.nn import functional as F
from fractions import Fraction
from einops import rearrange
from .transformer import CrossTransformerEncoder
from .demucs import rescale_module
from .states import capture_init
from .spec import spectro, ispectro
from .hdemucs import pad1d, ScaledEmbedding, HEncLayer, MultiWrap, HDecLayer
class HTDemucs(nn.Module):
"""
Spectrogram and hybrid Demucs model.
The spectrogram model has the same structure as Demucs, except the first few layers are over the
frequency axis, until there is only 1 frequency, and then it moves to time convolutions.
Frequency layers can still access information across time steps thanks to the DConv residual.
Hybrid model have a parallel time branch. At some layer, the time branch has the same stride
as the frequency branch and then the two are combined. The opposite happens in the decoder.
Models can either use naive iSTFT from masking, Wiener filtering ([Ulhih et al. 2017]),
or complex as channels (CaC) [Choi et al. 2020]. Wiener filtering is based on
Open Unmix implementation [Stoter et al. 2019].
The loss is always on the temporal domain, by backpropagating through the above
output methods and iSTFT. This allows to define hybrid models nicely. However, this breaks
a bit Wiener filtering, as doing more iteration at test time will change the spectrogram
contribution, without changing the one from the waveform, which will lead to worse performance.
I tried using the residual option in OpenUnmix Wiener implementation, but it didn't improve.
CaC on the other hand provides similar performance for hybrid, and works naturally with
hybrid models.
This model also uses frequency embeddings are used to improve efficiency on convolutions
over the freq. axis, following [Isik et al. 2020] (https://arxiv.org/pdf/2008.04470.pdf).
Unlike classic Demucs, there is no resampling here, and normalization is always applied.
"""
@capture_init
def __init__(
self,
sources,
# Channels
audio_channels=2,
channels=48,
channels_time=None,
growth=2,
# STFT
nfft=4096,
wiener_iters=0,
end_iters=0,
wiener_residual=False,
cac=True,
# Main structure
depth=4,
rewrite=True,
# Frequency branch
multi_freqs=None,
multi_freqs_depth=3,
freq_emb=0.2,
emb_scale=10,
emb_smooth=True,
# Convolutions
kernel_size=8,
time_stride=2,
stride=4,
context=1,
context_enc=0,
# Normalization
norm_starts=4,
norm_groups=4,
# DConv residual branch
dconv_mode=1,
dconv_depth=2,
dconv_comp=8,
dconv_init=1e-3,
# Before the Transformer
bottom_channels=0,
# Transformer
t_layers=5,
t_emb="sin",
t_hidden_scale=4.0,
t_heads=8,
t_dropout=0.0,
t_max_positions=10000,
t_norm_in=True,
t_norm_in_group=False,
t_group_norm=False,
t_norm_first=True,
t_norm_out=True,
t_max_period=10000.0,
t_weight_decay=0.0,
t_lr=None,
t_layer_scale=True,
t_gelu=True,
t_weight_pos_embed=1.0,
t_sin_random_shift=0,
t_cape_mean_normalize=True,
t_cape_augment=True,
t_cape_glob_loc_scale=[5000.0, 1.0, 1.4],
t_sparse_self_attn=False,
t_sparse_cross_attn=False,
t_mask_type="diag",
t_mask_random_seed=42,
t_sparse_attn_window=500,
t_global_window=100,
t_sparsity=0.95,
t_auto_sparsity=False,
# ------ Particuliar parameters
t_cross_first=False,
# Weight init
rescale=0.1,
# Metadata
samplerate=44100,
segment=10,
use_train_segment=True,
):
"""
Args:
sources (list[str]): list of source names.
audio_channels (int): input/output audio channels.
channels (int): initial number of hidden channels.
channels_time: if not None, use a different `channels` value for the time branch.
growth: increase the number of hidden channels by this factor at each layer.
nfft: number of fft bins. Note that changing this require careful computation of
various shape parameters and will not work out of the box for hybrid models.
wiener_iters: when using Wiener filtering, number of iterations at test time.
end_iters: same but at train time. For a hybrid model, must be equal to `wiener_iters`.
wiener_residual: add residual source before wiener filtering.
cac: uses complex as channels, i.e. complex numbers are 2 channels each
in input and output. no further processing is done before ISTFT.
depth (int): number of layers in the encoder and in the decoder.
rewrite (bool): add 1x1 convolution to each layer.
multi_freqs: list of frequency ratios for splitting frequency bands with `MultiWrap`.
multi_freqs_depth: how many layers to wrap with `MultiWrap`. Only the outermost
layers will be wrapped.
freq_emb: add frequency embedding after the first frequency layer if > 0,
the actual value controls the weight of the embedding.
emb_scale: equivalent to scaling the embedding learning rate
emb_smooth: initialize the embedding with a smooth one (with respect to frequencies).
kernel_size: kernel_size for encoder and decoder layers.
stride: stride for encoder and decoder layers.
time_stride: stride for the final time layer, after the merge.
context: context for 1x1 conv in the decoder.
context_enc: context for 1x1 conv in the encoder.
norm_starts: layer at which group norm starts being used.
decoder layers are numbered in reverse order.
norm_groups: number of groups for group norm.
dconv_mode: if 1: dconv in encoder only, 2: decoder only, 3: both.
dconv_depth: depth of residual DConv branch.
dconv_comp: compression of DConv branch.
dconv_attn: adds attention layers in DConv branch starting at this layer.
dconv_lstm: adds a LSTM layer in DConv branch starting at this layer.
dconv_init: initial scale for the DConv branch LayerScale.
bottom_channels: if >0 it adds a linear layer (1x1 Conv) before and after the
transformer in order to change the number of channels
t_layers: number of layers in each branch (waveform and spec) of the transformer
t_emb: "sin", "cape" or "scaled"
t_hidden_scale: the hidden scale of the Feedforward parts of the transformer
for instance if C = 384 (the number of channels in the transformer) and
t_hidden_scale = 4.0 then the intermediate layer of the FFN has dimension
384 * 4 = 1536
t_heads: number of heads for the transformer
t_dropout: dropout in the transformer
t_max_positions: max_positions for the "scaled" positional embedding, only
useful if t_emb="scaled"
t_norm_in: (bool) norm before addinf positional embedding and getting into the
transformer layers
t_norm_in_group: (bool) if True while t_norm_in=True, the norm is on all the
timesteps (GroupNorm with group=1)
t_group_norm: (bool) if True, the norms of the Encoder Layers are on all the
timesteps (GroupNorm with group=1)
t_norm_first: (bool) if True the norm is before the attention and before the FFN
t_norm_out: (bool) if True, there is a GroupNorm (group=1) at the end of each layer
t_max_period: (float) denominator in the sinusoidal embedding expression
t_weight_decay: (float) weight decay for the transformer
t_lr: (float) specific learning rate for the transformer
t_layer_scale: (bool) Layer Scale for the transformer
t_gelu: (bool) activations of the transformer are GeLU if True, ReLU else
t_weight_pos_embed: (float) weighting of the positional embedding
t_cape_mean_normalize: (bool) if t_emb="cape", normalisation of positional embeddings
see: https://arxiv.org/abs/2106.03143
t_cape_augment: (bool) if t_emb="cape", must be True during training and False
during the inference, see: https://arxiv.org/abs/2106.03143
t_cape_glob_loc_scale: (list of 3 floats) if t_emb="cape", CAPE parameters
see: https://arxiv.org/abs/2106.03143
t_sparse_self_attn: (bool) if True, the self attentions are sparse
t_sparse_cross_attn: (bool) if True, the cross-attentions are sparse (don't use it
unless you designed really specific masks)
t_mask_type: (str) can be "diag", "jmask", "random", "global" or any combination
with '_' between: i.e. "diag_jmask_random" (note that this is permutation
invariant i.e. "diag_jmask_random" is equivalent to "jmask_random_diag")
t_mask_random_seed: (int) if "random" is in t_mask_type, controls the seed
that generated the random part of the mask
t_sparse_attn_window: (int) if "diag" is in t_mask_type, for a query (i), and
a key (j), the mask is True id |i-j|<=t_sparse_attn_window
t_global_window: (int) if "global" is in t_mask_type, mask[:t_global_window, :]
and mask[:, :t_global_window] will be True
t_sparsity: (float) if "random" is in t_mask_type, t_sparsity is the sparsity
level of the random part of the mask.
t_cross_first: (bool) if True cross attention is the first layer of the
transformer (False seems to be better)
rescale: weight rescaling trick
use_train_segment: (bool) if True, the actual size that is used during the
training is used during inference.
"""
super().__init__()
self.cac = cac
self.wiener_residual = wiener_residual
self.audio_channels = audio_channels
self.sources = sources
self.kernel_size = kernel_size
self.context = context
self.stride = stride
self.depth = depth
self.bottom_channels = bottom_channels
self.channels = channels
self.samplerate = samplerate
self.segment = segment
self.use_train_segment = use_train_segment
self.nfft = nfft
self.hop_length = nfft // 4
self.wiener_iters = wiener_iters
self.end_iters = end_iters
self.freq_emb = None
assert wiener_iters == end_iters
self.encoder = nn.ModuleList()
self.decoder = nn.ModuleList()
self.tencoder = nn.ModuleList()
self.tdecoder = nn.ModuleList()
chin = audio_channels
chin_z = chin # number of channels for the freq branch
if self.cac:
chin_z *= 2
chout = channels_time or channels
chout_z = channels
freqs = nfft // 2
for index in range(depth):
norm = index >= norm_starts
freq = freqs > 1
stri = stride
ker = kernel_size
if not freq:
assert freqs == 1
ker = time_stride * 2
stri = time_stride
pad = True
last_freq = False
if freq and freqs <= kernel_size:
ker = freqs
pad = False
last_freq = True
kw = {
"kernel_size": ker,
"stride": stri,
"freq": freq,
"pad": pad,
"norm": norm,
"rewrite": rewrite,
"norm_groups": norm_groups,
"dconv_kw": {
"depth": dconv_depth,
"compress": dconv_comp,
"init": dconv_init,
"gelu": True,
},
}
kwt = dict(kw)
kwt["freq"] = 0
kwt["kernel_size"] = kernel_size
kwt["stride"] = stride
kwt["pad"] = True
kw_dec = dict(kw)
multi = False
if multi_freqs and index < multi_freqs_depth:
multi = True
kw_dec["context_freq"] = False
if last_freq:
chout_z = max(chout, chout_z)
chout = chout_z
enc = HEncLayer(
chin_z, chout_z, dconv=dconv_mode & 1, context=context_enc, **kw
)
if freq:
tenc = HEncLayer(
chin,
chout,
dconv=dconv_mode & 1,
context=context_enc,
empty=last_freq,
**kwt
)
self.tencoder.append(tenc)
if multi:
enc = MultiWrap(enc, multi_freqs)
self.encoder.append(enc)
if index == 0:
chin = self.audio_channels * len(self.sources)
chin_z = chin
if self.cac:
chin_z *= 2
dec = HDecLayer(
chout_z,
chin_z,
dconv=dconv_mode & 2,
last=index == 0,
context=context,
**kw_dec
)
if multi:
dec = MultiWrap(dec, multi_freqs)
if freq:
tdec = HDecLayer(
chout,
chin,
dconv=dconv_mode & 2,
empty=last_freq,
last=index == 0,
context=context,
**kwt
)
self.tdecoder.insert(0, tdec)
self.decoder.insert(0, dec)
chin = chout
chin_z = chout_z
chout = int(growth * chout)
chout_z = int(growth * chout_z)
if freq:
if freqs <= kernel_size:
freqs = 1
else:
freqs //= stride
if index == 0 and freq_emb:
self.freq_emb = ScaledEmbedding(
freqs, chin_z, smooth=emb_smooth, scale=emb_scale
)
self.freq_emb_scale = freq_emb
if rescale:
rescale_module(self, reference=rescale)
transformer_channels = channels * growth ** (depth - 1)
if bottom_channels:
self.channel_upsampler = nn.Conv1d(transformer_channels, bottom_channels, 1)
self.channel_downsampler = nn.Conv1d(
bottom_channels, transformer_channels, 1
)
self.channel_upsampler_t = nn.Conv1d(
transformer_channels, bottom_channels, 1
)
self.channel_downsampler_t = nn.Conv1d(
bottom_channels, transformer_channels, 1
)
transformer_channels = bottom_channels
if t_layers > 0:
self.crosstransformer = CrossTransformerEncoder(
dim=transformer_channels,
emb=t_emb,
hidden_scale=t_hidden_scale,
num_heads=t_heads,
num_layers=t_layers,
cross_first=t_cross_first,
dropout=t_dropout,
max_positions=t_max_positions,
norm_in=t_norm_in,
norm_in_group=t_norm_in_group,
group_norm=t_group_norm,
norm_first=t_norm_first,
norm_out=t_norm_out,
max_period=t_max_period,
weight_decay=t_weight_decay,
lr=t_lr,
layer_scale=t_layer_scale,
gelu=t_gelu,
sin_random_shift=t_sin_random_shift,
weight_pos_embed=t_weight_pos_embed,
cape_mean_normalize=t_cape_mean_normalize,
cape_augment=t_cape_augment,
cape_glob_loc_scale=t_cape_glob_loc_scale,
sparse_self_attn=t_sparse_self_attn,
sparse_cross_attn=t_sparse_cross_attn,
mask_type=t_mask_type,
mask_random_seed=t_mask_random_seed,
sparse_attn_window=t_sparse_attn_window,
global_window=t_global_window,
sparsity=t_sparsity,
auto_sparsity=t_auto_sparsity,
)
else:
self.crosstransformer = None
def _spec(self, x):
hl = self.hop_length
nfft = self.nfft
x0 = x # noqa
# We re-pad the signal in order to keep the property
# that the size of the output is exactly the size of the input
# divided by the stride (here hop_length), when divisible.
# This is achieved by padding by 1/4th of the kernel size (here nfft).
# which is not supported by torch.stft.
# Having all convolution operations follow this convention allow to easily
# align the time and frequency branches later on.
assert hl == nfft // 4
le = int(math.ceil(x.shape[-1] / hl))
pad = hl // 2 * 3
x = pad1d(x, (pad, pad + le * hl - x.shape[-1]), mode="reflect")
z = spectro(x, nfft, hl)[..., :-1, :]
assert z.shape[-1] == le + 4, (z.shape, x.shape, le)
z = z[..., 2: 2 + le]
return z
def _ispec(self, z, length=None, scale=0):
hl = self.hop_length // (4**scale)
z = F.pad(z, (0, 0, 0, 1))
z = F.pad(z, (2, 2))
pad = hl // 2 * 3
le = hl * int(math.ceil(length / hl)) + 2 * pad
x = ispectro(z, hl, length=le)
x = x[..., pad: pad + length]
return x
def _magnitude(self, z):
# return the magnitude of the spectrogram, except when cac is True,
# in which case we just move the complex dimension to the channel one.
if self.cac:
B, C, Fr, T = z.shape
m = torch.view_as_real(z).permute(0, 1, 4, 2, 3)
m = m.reshape(B, C * 2, Fr, T)
else:
m = z.abs()
return m
def _mask(self, z, m):
# Apply masking given the mixture spectrogram `z` and the estimated mask `m`.
# If `cac` is True, `m` is actually a full spectrogram and `z` is ignored.
niters = self.wiener_iters
if self.cac:
B, S, C, Fr, T = m.shape
out = m.view(B, S, -1, 2, Fr, T).permute(0, 1, 2, 4, 5, 3)
out = torch.view_as_complex(out.contiguous())
return out
if self.training:
niters = self.end_iters
if niters < 0:
z = z[:, None]
return z / (1e-8 + z.abs()) * m
else:
return self._wiener(m, z, niters)
def _wiener(self, mag_out, mix_stft, niters):
# apply wiener filtering from OpenUnmix.
init = mix_stft.dtype
wiener_win_len = 300
residual = self.wiener_residual
B, S, C, Fq, T = mag_out.shape
mag_out = mag_out.permute(0, 4, 3, 2, 1)
mix_stft = torch.view_as_real(mix_stft.permute(0, 3, 2, 1))
outs = []
for sample in range(B):
pos = 0
out = []
for pos in range(0, T, wiener_win_len):
frame = slice(pos, pos + wiener_win_len)
z_out = wiener(
mag_out[sample, frame],
mix_stft[sample, frame],
niters,
residual=residual,
)
out.append(z_out.transpose(-1, -2))
outs.append(torch.cat(out, dim=0))
out = torch.view_as_complex(torch.stack(outs, 0))
out = out.permute(0, 4, 3, 2, 1).contiguous()
if residual:
out = out[:, :-1]
assert list(out.shape) == [B, S, C, Fq, T]
return out.to(init)
def valid_length(self, length: int):
"""
Return a length that is appropriate for evaluation.
In our case, always return the training length, unless
it is smaller than the given length, in which case this
raises an error.
"""
if not self.use_train_segment:
return length
training_length = int(self.segment * self.samplerate)
if training_length < length:
raise ValueError(
f"Given length {length} is longer than "
f"training length {training_length}")
return training_length
def forward(self, mix):
length = mix.shape[-1]
length_pre_pad = None
if self.use_train_segment:
if self.training:
self.segment = Fraction(mix.shape[-1], self.samplerate)
else:
training_length = int(self.segment * self.samplerate)
if mix.shape[-1] < training_length:
length_pre_pad = mix.shape[-1]
mix = F.pad(mix, (0, training_length - length_pre_pad))
z = self._spec(mix)
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mag = self._magnitude(z).to(mix.device)
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x = mag
B, C, Fq, T = x.shape
# unlike previous Demucs, we always normalize because it is easier.
mean = x.mean(dim=(1, 2, 3), keepdim=True)
std = x.std(dim=(1, 2, 3), keepdim=True)
x = (x - mean) / (1e-5 + std)
# x will be the freq. branch input.
# Prepare the time branch input.
xt = mix
meant = xt.mean(dim=(1, 2), keepdim=True)
stdt = xt.std(dim=(1, 2), keepdim=True)
xt = (xt - meant) / (1e-5 + stdt)
# okay, this is a giant mess I know...
saved = [] # skip connections, freq.
saved_t = [] # skip connections, time.
lengths = [] # saved lengths to properly remove padding, freq branch.
lengths_t = [] # saved lengths for time branch.
for idx, encode in enumerate(self.encoder):
lengths.append(x.shape[-1])
inject = None
if idx < len(self.tencoder):
# we have not yet merged branches.
lengths_t.append(xt.shape[-1])
tenc = self.tencoder[idx]
xt = tenc(xt)
if not tenc.empty:
# save for skip connection
saved_t.append(xt)
else:
# tenc contains just the first conv., so that now time and freq.
# branches have the same shape and can be merged.
inject = xt
x = encode(x, inject)
if idx == 0 and self.freq_emb is not None:
# add frequency embedding to allow for non equivariant convolutions
# over the frequency axis.
frs = torch.arange(x.shape[-2], device=x.device)
emb = self.freq_emb(frs).t()[None, :, :, None].expand_as(x)
x = x + self.freq_emb_scale * emb
saved.append(x)
if self.crosstransformer:
if self.bottom_channels:
b, c, f, t = x.shape
x = rearrange(x, "b c f t-> b c (f t)")
x = self.channel_upsampler(x)
x = rearrange(x, "b c (f t)-> b c f t", f=f)
xt = self.channel_upsampler_t(xt)
x, xt = self.crosstransformer(x, xt)
if self.bottom_channels:
x = rearrange(x, "b c f t-> b c (f t)")
x = self.channel_downsampler(x)
x = rearrange(x, "b c (f t)-> b c f t", f=f)
xt = self.channel_downsampler_t(xt)
for idx, decode in enumerate(self.decoder):
skip = saved.pop(-1)
x, pre = decode(x, skip, lengths.pop(-1))
# `pre` contains the output just before final transposed convolution,
# which is used when the freq. and time branch separate.
offset = self.depth - len(self.tdecoder)
if idx >= offset:
tdec = self.tdecoder[idx - offset]
length_t = lengths_t.pop(-1)
if tdec.empty:
assert pre.shape[2] == 1, pre.shape
pre = pre[:, :, 0]
xt, _ = tdec(pre, None, length_t)
else:
skip = saved_t.pop(-1)
xt, _ = tdec(xt, skip, length_t)
# Let's make sure we used all stored skip connections.
assert len(saved) == 0
assert len(lengths_t) == 0
assert len(saved_t) == 0
S = len(self.sources)
x = x.view(B, S, -1, Fq, T)
x = x * std[:, None] + mean[:, None]
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# to cpu as non-cuda GPUs don't support complex numbers
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# demucs issue #435 ##432
# NOTE: in this case z already is on cpu
# TODO: remove this when mps supports complex numbers
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device_type = x.device.type
device_load = f"{device_type}:{x.device.index}" if not device_type == 'mps' else device_type
x_is_other_gpu = not device_type in ["cuda", "cpu"]
if x_is_other_gpu:
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x = x.cpu()
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zout = self._mask(z, x)
if self.use_train_segment:
if self.training:
x = self._ispec(zout, length)
else:
x = self._ispec(zout, training_length)
else:
x = self._ispec(zout, length)
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# back to other device
if x_is_other_gpu:
x = x.to(device_load)
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if self.use_train_segment:
if self.training:
xt = xt.view(B, S, -1, length)
else:
xt = xt.view(B, S, -1, training_length)
else:
xt = xt.view(B, S, -1, length)
xt = xt * stdt[:, None] + meant[:, None]
x = xt + x
if length_pre_pad:
x = x[..., :length_pre_pad]
return x