File size: 16,306 Bytes
109bb65 |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 |
# 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.
# author: adefossez
import math
import time
import torch as th
from torch import nn
from torch.nn import functional as F
from .resample import downsample2, upsample2
from .utils import capture_init
class BLSTM(nn.Module):
def __init__(self, dim, layers=2, bi=True):
super().__init__()
klass = nn.LSTM
self.lstm = klass(bidirectional=bi, num_layers=layers, hidden_size=dim, input_size=dim)
self.linear = None
if bi:
self.linear = nn.Linear(2 * dim, dim)
def forward(self, x, hidden=None):
x, hidden = self.lstm(x, hidden)
if self.linear:
x = self.linear(x)
return x, hidden
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)
class Demucs(nn.Module):
"""
Demucs speech enhancement model.
Args:
- chin (int): number of input channels.
- chout (int): number of output channels.
- hidden (int): number of initial hidden channels.
- depth (int): number of layers.
- kernel_size (int): kernel size for each layer.
- stride (int): stride for each layer.
- causal (bool): if false, uses BiLSTM instead of LSTM.
- resample (int): amount of resampling to apply to the input/output.
Can be one of 1, 2 or 4.
- growth (float): number of channels is multiplied by this for every layer.
- max_hidden (int): maximum number of channels. Can be useful to
control the size/speed of the model.
- normalize (bool): if true, normalize the input.
- glu (bool): if true uses GLU instead of ReLU in 1x1 convolutions.
- rescale (float): controls custom weight initialization.
See https://arxiv.org/abs/1911.13254.
- floor (float): stability flooring when normalizing.
"""
@capture_init
def __init__(self,
chin=1,
chout=1,
hidden=48,
depth=5,
kernel_size=8,
stride=4,
causal=True,
resample=4,
growth=2,
max_hidden=10_000,
normalize=True,
glu=True,
rescale=0.1,
floor=1e-3):
super().__init__()
if resample not in [1, 2, 4]:
raise ValueError("Resample should be 1, 2 or 4.")
self.chin = chin
self.chout = chout
self.hidden = hidden
self.depth = depth
self.kernel_size = kernel_size
self.stride = stride
self.causal = causal
self.floor = floor
self.resample = resample
self.normalize = normalize
self.encoder = nn.ModuleList()
self.decoder = nn.ModuleList()
activation = nn.GLU(1) if glu else nn.ReLU()
ch_scale = 2 if glu else 1
for index in range(depth):
encode = []
encode += [
nn.Conv1d(chin, hidden, kernel_size, stride),
nn.ReLU(),
nn.Conv1d(hidden, hidden * ch_scale, 1), activation,
]
self.encoder.append(nn.Sequential(*encode))
decode = []
decode += [
nn.Conv1d(hidden, ch_scale * hidden, 1), activation,
nn.ConvTranspose1d(hidden, chout, kernel_size, stride),
]
if index > 0:
decode.append(nn.ReLU())
self.decoder.insert(0, nn.Sequential(*decode))
chout = hidden
chin = hidden
hidden = min(int(growth * hidden), max_hidden)
self.lstm = BLSTM(chin, bi=not causal)
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.
"""
length = math.ceil(length * self.resample)
for idx in range(self.depth):
length = math.ceil((length - self.kernel_size) / self.stride) + 1
length = max(length, 1)
for idx in range(self.depth):
length = (length - 1) * self.stride + self.kernel_size
length = int(math.ceil(length / self.resample))
return int(length)
@property
def total_stride(self):
return self.stride ** self.depth // self.resample
def forward(self, mix):
if mix.dim() == 2:
mix = mix.unsqueeze(1)
if self.normalize:
mono = mix.mean(dim=1, keepdim=True)
std = mono.std(dim=-1, keepdim=True)
mix = mix / (self.floor + std)
else:
std = 1
length = mix.shape[-1]
x = mix
x = F.pad(x, (0, self.valid_length(length) - length))
if self.resample == 2:
x = upsample2(x)
elif self.resample == 4:
x = upsample2(x)
x = upsample2(x)
skips = []
for encode in self.encoder:
x = encode(x)
skips.append(x)
x = x.permute(2, 0, 1)
x, _ = self.lstm(x)
x = x.permute(1, 2, 0)
for decode in self.decoder:
skip = skips.pop(-1)
x = x + skip[..., :x.shape[-1]]
x = decode(x)
if self.resample == 2:
x = downsample2(x)
elif self.resample == 4:
x = downsample2(x)
x = downsample2(x)
x = x[..., :length]
return std * x
def fast_conv(conv, x):
"""
Faster convolution evaluation if either kernel size is 1
or length of sequence is 1.
"""
batch, chin, length = x.shape
chout, chin, kernel = conv.weight.shape
assert batch == 1
if kernel == 1:
x = x.view(chin, length)
out = th.addmm(conv.bias.view(-1, 1),
conv.weight.view(chout, chin), x)
elif length == kernel:
x = x.view(chin * kernel, 1)
out = th.addmm(conv.bias.view(-1, 1),
conv.weight.view(chout, chin * kernel), x)
else:
out = conv(x)
return out.view(batch, chout, -1)
class DemucsStreamer:
"""
Streaming implementation for Demucs. It supports being fed with any amount
of audio at a time. You will get back as much audio as possible at that
point.
Args:
- demucs (Demucs): Demucs model.
- dry (float): amount of dry (e.g. input) signal to keep. 0 is maximum
noise removal, 1 just returns the input signal. Small values > 0
allows to limit distortions.
- num_frames (int): number of frames to process at once. Higher values
will increase overall latency but improve the real time factor.
- resample_lookahead (int): extra lookahead used for the resampling.
- resample_buffer (int): size of the buffer of previous inputs/outputs
kept for resampling.
"""
def __init__(self, demucs,
dry=0,
num_frames=1,
resample_lookahead=64,
resample_buffer=256):
device = next(iter(demucs.parameters())).device
self.demucs = demucs
self.lstm_state = None
self.conv_state = None
self.dry = dry
self.resample_lookahead = resample_lookahead
self.resample_buffer = resample_buffer
self.frame_length = demucs.valid_length(1) + demucs.total_stride * (num_frames - 1)
self.total_length = self.frame_length + self.resample_lookahead
self.stride = demucs.total_stride * num_frames
self.resample_in = th.zeros(demucs.chin, resample_buffer, device=device)
self.resample_out = th.zeros(demucs.chin, resample_buffer, device=device)
self.frames = 0
self.total_time = 0
self.variance = 0
self.pending = th.zeros(demucs.chin, 0, device=device)
bias = demucs.decoder[0][2].bias
weight = demucs.decoder[0][2].weight
chin, chout, kernel = weight.shape
self._bias = bias.view(-1, 1).repeat(1, kernel).view(-1, 1)
self._weight = weight.permute(1, 2, 0).contiguous()
def reset_time_per_frame(self):
self.total_time = 0
self.frames = 0
@property
def time_per_frame(self):
return self.total_time / self.frames
def flush(self):
"""
Flush remaining audio by padding it with zero. Call this
when you have no more input and want to get back the last chunk of audio.
"""
pending_length = self.pending.shape[1]
padding = th.zeros(self.demucs.chin, self.total_length, device=self.pending.device)
out = self.feed(padding)
return out[:, :pending_length]
def feed(self, wav):
"""
Apply the model to mix using true real time evaluation.
Normalization is done online as is the resampling.
"""
begin = time.time()
demucs = self.demucs
resample_buffer = self.resample_buffer
stride = self.stride
resample = demucs.resample
if wav.dim() != 2:
raise ValueError("input wav should be two dimensional.")
chin, _ = wav.shape
if chin != demucs.chin:
raise ValueError(f"Expected {demucs.chin} channels, got {chin}")
self.pending = th.cat([self.pending, wav], dim=1)
outs = []
while self.pending.shape[1] >= self.total_length:
self.frames += 1
frame = self.pending[:, :self.total_length]
dry_signal = frame[:, :stride]
if demucs.normalize:
mono = frame.mean(0)
variance = (mono**2).mean()
self.variance = variance / self.frames + (1 - 1 / self.frames) * self.variance
frame = frame / (demucs.floor + math.sqrt(self.variance))
frame = th.cat([self.resample_in, frame], dim=-1)
self.resample_in[:] = frame[:, stride - resample_buffer:stride]
if resample == 4:
frame = upsample2(upsample2(frame))
elif resample == 2:
frame = upsample2(frame)
frame = frame[:, resample * resample_buffer:] # remove pre sampling buffer
frame = frame[:, :resample * self.frame_length] # remove extra samples after window
out, extra = self._separate_frame(frame)
padded_out = th.cat([self.resample_out, out, extra], 1)
self.resample_out[:] = out[:, -resample_buffer:]
if resample == 4:
out = downsample2(downsample2(padded_out))
elif resample == 2:
out = downsample2(padded_out)
else:
out = padded_out
out = out[:, resample_buffer // resample:]
out = out[:, :stride]
if demucs.normalize:
out *= math.sqrt(self.variance)
out = self.dry * dry_signal + (1 - self.dry) * out
outs.append(out)
self.pending = self.pending[:, stride:]
self.total_time += time.time() - begin
if outs:
out = th.cat(outs, 1)
else:
out = th.zeros(chin, 0, device=wav.device)
return out
def _separate_frame(self, frame):
demucs = self.demucs
skips = []
next_state = []
first = self.conv_state is None
stride = self.stride * demucs.resample
x = frame[None]
for idx, encode in enumerate(demucs.encoder):
stride //= demucs.stride
length = x.shape[2]
if idx == demucs.depth - 1:
# This is sligthly faster for the last conv
x = fast_conv(encode[0], x)
x = encode[1](x)
x = fast_conv(encode[2], x)
x = encode[3](x)
else:
if not first:
prev = self.conv_state.pop(0)
prev = prev[..., stride:]
tgt = (length - demucs.kernel_size) // demucs.stride + 1
missing = tgt - prev.shape[-1]
offset = length - demucs.kernel_size - demucs.stride * (missing - 1)
x = x[..., offset:]
x = encode[1](encode[0](x))
x = fast_conv(encode[2], x)
x = encode[3](x)
if not first:
x = th.cat([prev, x], -1)
next_state.append(x)
skips.append(x)
x = x.permute(2, 0, 1)
x, self.lstm_state = demucs.lstm(x, self.lstm_state)
x = x.permute(1, 2, 0)
# In the following, x contains only correct samples, i.e. the one
# for which each time position is covered by two window of the upper layer.
# extra contains extra samples to the right, and is used only as a
# better padding for the online resampling.
extra = None
for idx, decode in enumerate(demucs.decoder):
skip = skips.pop(-1)
x += skip[..., :x.shape[-1]]
x = fast_conv(decode[0], x)
x = decode[1](x)
if extra is not None:
skip = skip[..., x.shape[-1]:]
extra += skip[..., :extra.shape[-1]]
extra = decode[2](decode[1](decode[0](extra)))
x = decode[2](x)
next_state.append(x[..., -demucs.stride:] - decode[2].bias.view(-1, 1))
if extra is None:
extra = x[..., -demucs.stride:]
else:
extra[..., :demucs.stride] += next_state[-1]
x = x[..., :-demucs.stride]
if not first:
prev = self.conv_state.pop(0)
x[..., :demucs.stride] += prev
if idx != demucs.depth - 1:
x = decode[3](x)
extra = decode[3](extra)
self.conv_state = next_state
return x[0], extra[0]
def test():
import argparse
parser = argparse.ArgumentParser(
"denoiser.demucs",
description="Benchmark the streaming Demucs implementation, "
"as well as checking the delta with the offline implementation.")
parser.add_argument("--resample", default=4, type=int)
parser.add_argument("--hidden", default=48, type=int)
parser.add_argument("--device", default="cpu")
parser.add_argument("-t", "--num_threads", type=int)
parser.add_argument("-f", "--num_frames", type=int, default=1)
args = parser.parse_args()
if args.num_threads:
th.set_num_threads(args.num_threads)
sr = 16_000
sr_ms = sr / 1000
demucs = Demucs(hidden=args.hidden, resample=args.resample).to(args.device)
x = th.randn(1, sr * 4).to(args.device)
out = demucs(x[None])[0]
streamer = DemucsStreamer(demucs, num_frames=args.num_frames)
out_rt = []
frame_size = streamer.total_length
with th.no_grad():
while x.shape[1] > 0:
out_rt.append(streamer.feed(x[:, :frame_size]))
x = x[:, frame_size:]
frame_size = streamer.demucs.total_stride
out_rt.append(streamer.flush())
out_rt = th.cat(out_rt, 1)
print(f"total lag: {streamer.total_length / sr_ms:.1f}ms, ", end='')
print(f"stride: {streamer.stride / sr_ms:.1f}ms, ", end='')
print(f"time per frame: {1000 * streamer.time_per_frame:.1f}ms, ", end='')
print(f"delta: {th.norm(out - out_rt) / th.norm(out):.2%}, ", end='')
print(f"RTF: {((1000 * streamer.time_per_frame) / (streamer.stride / sr_ms)):.1f}")
if __name__ == "__main__":
test()
|