feat: Integrate IndexTTS2 model and update related schemas and frontend components

This commit is contained in:
2026-03-12 13:30:53 +08:00
parent e5b5a16364
commit 8aec4f6f44
151 changed files with 40077 additions and 85 deletions

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from .base import CodecMixin
from .base import DACFile
from .dac import DAC
from .discriminator import Discriminator

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import math
from dataclasses import dataclass
from pathlib import Path
from typing import Union
import numpy as np
import torch
import tqdm
from audiotools import AudioSignal
from torch import nn
SUPPORTED_VERSIONS = ["1.0.0"]
@dataclass
class DACFile:
codes: torch.Tensor
# Metadata
chunk_length: int
original_length: int
input_db: float
channels: int
sample_rate: int
padding: bool
dac_version: str
def save(self, path):
artifacts = {
"codes": self.codes.numpy().astype(np.uint16),
"metadata": {
"input_db": self.input_db.numpy().astype(np.float32),
"original_length": self.original_length,
"sample_rate": self.sample_rate,
"chunk_length": self.chunk_length,
"channels": self.channels,
"padding": self.padding,
"dac_version": SUPPORTED_VERSIONS[-1],
},
}
path = Path(path).with_suffix(".dac")
with open(path, "wb") as f:
np.save(f, artifacts)
return path
@classmethod
def load(cls, path):
artifacts = np.load(path, allow_pickle=True)[()]
codes = torch.from_numpy(artifacts["codes"].astype(int))
if artifacts["metadata"].get("dac_version", None) not in SUPPORTED_VERSIONS:
raise RuntimeError(
f"Given file {path} can't be loaded with this version of descript-audio-codec."
)
return cls(codes=codes, **artifacts["metadata"])
class CodecMixin:
@property
def padding(self):
if not hasattr(self, "_padding"):
self._padding = True
return self._padding
@padding.setter
def padding(self, value):
assert isinstance(value, bool)
layers = [
l for l in self.modules() if isinstance(l, (nn.Conv1d, nn.ConvTranspose1d))
]
for layer in layers:
if value:
if hasattr(layer, "original_padding"):
layer.padding = layer.original_padding
else:
layer.original_padding = layer.padding
layer.padding = tuple(0 for _ in range(len(layer.padding)))
self._padding = value
def get_delay(self):
# Any number works here, delay is invariant to input length
l_out = self.get_output_length(0)
L = l_out
layers = []
for layer in self.modules():
if isinstance(layer, (nn.Conv1d, nn.ConvTranspose1d)):
layers.append(layer)
for layer in reversed(layers):
d = layer.dilation[0]
k = layer.kernel_size[0]
s = layer.stride[0]
if isinstance(layer, nn.ConvTranspose1d):
L = ((L - d * (k - 1) - 1) / s) + 1
elif isinstance(layer, nn.Conv1d):
L = (L - 1) * s + d * (k - 1) + 1
L = math.ceil(L)
l_in = L
return (l_in - l_out) // 2
def get_output_length(self, input_length):
L = input_length
# Calculate output length
for layer in self.modules():
if isinstance(layer, (nn.Conv1d, nn.ConvTranspose1d)):
d = layer.dilation[0]
k = layer.kernel_size[0]
s = layer.stride[0]
if isinstance(layer, nn.Conv1d):
L = ((L - d * (k - 1) - 1) / s) + 1
elif isinstance(layer, nn.ConvTranspose1d):
L = (L - 1) * s + d * (k - 1) + 1
L = math.floor(L)
return L
@torch.no_grad()
def compress(
self,
audio_path_or_signal: Union[str, Path, AudioSignal],
win_duration: float = 1.0,
verbose: bool = False,
normalize_db: float = -16,
n_quantizers: int = None,
) -> DACFile:
"""Processes an audio signal from a file or AudioSignal object into
discrete codes. This function processes the signal in short windows,
using constant GPU memory.
Parameters
----------
audio_path_or_signal : Union[str, Path, AudioSignal]
audio signal to reconstruct
win_duration : float, optional
window duration in seconds, by default 5.0
verbose : bool, optional
by default False
normalize_db : float, optional
normalize db, by default -16
Returns
-------
DACFile
Object containing compressed codes and metadata
required for decompression
"""
audio_signal = audio_path_or_signal
if isinstance(audio_signal, (str, Path)):
audio_signal = AudioSignal.load_from_file_with_ffmpeg(str(audio_signal))
self.eval()
original_padding = self.padding
original_device = audio_signal.device
audio_signal = audio_signal.clone()
original_sr = audio_signal.sample_rate
resample_fn = audio_signal.resample
loudness_fn = audio_signal.loudness
# If audio is > 10 minutes long, use the ffmpeg versions
if audio_signal.signal_duration >= 10 * 60 * 60:
resample_fn = audio_signal.ffmpeg_resample
loudness_fn = audio_signal.ffmpeg_loudness
original_length = audio_signal.signal_length
resample_fn(self.sample_rate)
input_db = loudness_fn()
if normalize_db is not None:
audio_signal.normalize(normalize_db)
audio_signal.ensure_max_of_audio()
nb, nac, nt = audio_signal.audio_data.shape
audio_signal.audio_data = audio_signal.audio_data.reshape(nb * nac, 1, nt)
win_duration = (
audio_signal.signal_duration if win_duration is None else win_duration
)
if audio_signal.signal_duration <= win_duration:
# Unchunked compression (used if signal length < win duration)
self.padding = True
n_samples = nt
hop = nt
else:
# Chunked inference
self.padding = False
# Zero-pad signal on either side by the delay
audio_signal.zero_pad(self.delay, self.delay)
n_samples = int(win_duration * self.sample_rate)
# Round n_samples to nearest hop length multiple
n_samples = int(math.ceil(n_samples / self.hop_length) * self.hop_length)
hop = self.get_output_length(n_samples)
codes = []
range_fn = range if not verbose else tqdm.trange
for i in range_fn(0, nt, hop):
x = audio_signal[..., i : i + n_samples]
x = x.zero_pad(0, max(0, n_samples - x.shape[-1]))
audio_data = x.audio_data.to(self.device)
audio_data = self.preprocess(audio_data, self.sample_rate)
_, c, _, _, _ = self.encode(audio_data, n_quantizers)
codes.append(c.to(original_device))
chunk_length = c.shape[-1]
codes = torch.cat(codes, dim=-1)
dac_file = DACFile(
codes=codes,
chunk_length=chunk_length,
original_length=original_length,
input_db=input_db,
channels=nac,
sample_rate=original_sr,
padding=self.padding,
dac_version=SUPPORTED_VERSIONS[-1],
)
if n_quantizers is not None:
codes = codes[:, :n_quantizers, :]
self.padding = original_padding
return dac_file
@torch.no_grad()
def decompress(
self,
obj: Union[str, Path, DACFile],
verbose: bool = False,
) -> AudioSignal:
"""Reconstruct audio from a given .dac file
Parameters
----------
obj : Union[str, Path, DACFile]
.dac file location or corresponding DACFile object.
verbose : bool, optional
Prints progress if True, by default False
Returns
-------
AudioSignal
Object with the reconstructed audio
"""
self.eval()
if isinstance(obj, (str, Path)):
obj = DACFile.load(obj)
original_padding = self.padding
self.padding = obj.padding
range_fn = range if not verbose else tqdm.trange
codes = obj.codes
original_device = codes.device
chunk_length = obj.chunk_length
recons = []
for i in range_fn(0, codes.shape[-1], chunk_length):
c = codes[..., i : i + chunk_length].to(self.device)
z = self.quantizer.from_codes(c)[0]
r = self.decode(z)
recons.append(r.to(original_device))
recons = torch.cat(recons, dim=-1)
recons = AudioSignal(recons, self.sample_rate)
resample_fn = recons.resample
loudness_fn = recons.loudness
# If audio is > 10 minutes long, use the ffmpeg versions
if recons.signal_duration >= 10 * 60 * 60:
resample_fn = recons.ffmpeg_resample
loudness_fn = recons.ffmpeg_loudness
recons.normalize(obj.input_db)
resample_fn(obj.sample_rate)
recons = recons[..., : obj.original_length]
loudness_fn()
recons.audio_data = recons.audio_data.reshape(
-1, obj.channels, obj.original_length
)
self.padding = original_padding
return recons

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import math
from typing import List
from typing import Union
import numpy as np
import torch
from audiotools import AudioSignal
from audiotools.ml import BaseModel
from torch import nn
from .base import CodecMixin
from indextts.s2mel.dac.nn.layers import Snake1d
from indextts.s2mel.dac.nn.layers import WNConv1d
from indextts.s2mel.dac.nn.layers import WNConvTranspose1d
from indextts.s2mel.dac.nn.quantize import ResidualVectorQuantize
from .encodec import SConv1d, SConvTranspose1d, SLSTM
def init_weights(m):
if isinstance(m, nn.Conv1d):
nn.init.trunc_normal_(m.weight, std=0.02)
nn.init.constant_(m.bias, 0)
class ResidualUnit(nn.Module):
def __init__(self, dim: int = 16, dilation: int = 1, causal: bool = False):
super().__init__()
conv1d_type = SConv1d# if causal else WNConv1d
pad = ((7 - 1) * dilation) // 2
self.block = nn.Sequential(
Snake1d(dim),
conv1d_type(dim, dim, kernel_size=7, dilation=dilation, padding=pad, causal=causal, norm='weight_norm'),
Snake1d(dim),
conv1d_type(dim, dim, kernel_size=1, causal=causal, norm='weight_norm'),
)
def forward(self, x):
y = self.block(x)
pad = (x.shape[-1] - y.shape[-1]) // 2
if pad > 0:
x = x[..., pad:-pad]
return x + y
class EncoderBlock(nn.Module):
def __init__(self, dim: int = 16, stride: int = 1, causal: bool = False):
super().__init__()
conv1d_type = SConv1d# if causal else WNConv1d
self.block = nn.Sequential(
ResidualUnit(dim // 2, dilation=1, causal=causal),
ResidualUnit(dim // 2, dilation=3, causal=causal),
ResidualUnit(dim // 2, dilation=9, causal=causal),
Snake1d(dim // 2),
conv1d_type(
dim // 2,
dim,
kernel_size=2 * stride,
stride=stride,
padding=math.ceil(stride / 2),
causal=causal,
norm='weight_norm',
),
)
def forward(self, x):
return self.block(x)
class Encoder(nn.Module):
def __init__(
self,
d_model: int = 64,
strides: list = [2, 4, 8, 8],
d_latent: int = 64,
causal: bool = False,
lstm: int = 2,
):
super().__init__()
conv1d_type = SConv1d# if causal else WNConv1d
# Create first convolution
self.block = [conv1d_type(1, d_model, kernel_size=7, padding=3, causal=causal, norm='weight_norm')]
# Create EncoderBlocks that double channels as they downsample by `stride`
for stride in strides:
d_model *= 2
self.block += [EncoderBlock(d_model, stride=stride, causal=causal)]
# Add LSTM if needed
self.use_lstm = lstm
if lstm:
self.block += [SLSTM(d_model, lstm)]
# Create last convolution
self.block += [
Snake1d(d_model),
conv1d_type(d_model, d_latent, kernel_size=3, padding=1, causal=causal, norm='weight_norm'),
]
# Wrap black into nn.Sequential
self.block = nn.Sequential(*self.block)
self.enc_dim = d_model
def forward(self, x):
return self.block(x)
def reset_cache(self):
# recursively find all submodules named SConv1d in self.block and use their reset_cache method
def reset_cache(m):
if isinstance(m, SConv1d) or isinstance(m, SLSTM):
m.reset_cache()
return
for child in m.children():
reset_cache(child)
reset_cache(self.block)
class DecoderBlock(nn.Module):
def __init__(self, input_dim: int = 16, output_dim: int = 8, stride: int = 1, causal: bool = False):
super().__init__()
conv1d_type = SConvTranspose1d #if causal else WNConvTranspose1d
self.block = nn.Sequential(
Snake1d(input_dim),
conv1d_type(
input_dim,
output_dim,
kernel_size=2 * stride,
stride=stride,
padding=math.ceil(stride / 2),
causal=causal,
norm='weight_norm'
),
ResidualUnit(output_dim, dilation=1, causal=causal),
ResidualUnit(output_dim, dilation=3, causal=causal),
ResidualUnit(output_dim, dilation=9, causal=causal),
)
def forward(self, x):
return self.block(x)
class Decoder(nn.Module):
def __init__(
self,
input_channel,
channels,
rates,
d_out: int = 1,
causal: bool = False,
lstm: int = 2,
):
super().__init__()
conv1d_type = SConv1d# if causal else WNConv1d
# Add first conv layer
layers = [conv1d_type(input_channel, channels, kernel_size=7, padding=3, causal=causal, norm='weight_norm')]
if lstm:
layers += [SLSTM(channels, num_layers=lstm)]
# Add upsampling + MRF blocks
for i, stride in enumerate(rates):
input_dim = channels // 2**i
output_dim = channels // 2 ** (i + 1)
layers += [DecoderBlock(input_dim, output_dim, stride, causal=causal)]
# Add final conv layer
layers += [
Snake1d(output_dim),
conv1d_type(output_dim, d_out, kernel_size=7, padding=3, causal=causal, norm='weight_norm'),
nn.Tanh(),
]
self.model = nn.Sequential(*layers)
def forward(self, x):
return self.model(x)
class DAC(BaseModel, CodecMixin):
def __init__(
self,
encoder_dim: int = 64,
encoder_rates: List[int] = [2, 4, 8, 8],
latent_dim: int = None,
decoder_dim: int = 1536,
decoder_rates: List[int] = [8, 8, 4, 2],
n_codebooks: int = 9,
codebook_size: int = 1024,
codebook_dim: Union[int, list] = 8,
quantizer_dropout: bool = False,
sample_rate: int = 44100,
lstm: int = 2,
causal: bool = False,
):
super().__init__()
self.encoder_dim = encoder_dim
self.encoder_rates = encoder_rates
self.decoder_dim = decoder_dim
self.decoder_rates = decoder_rates
self.sample_rate = sample_rate
if latent_dim is None:
latent_dim = encoder_dim * (2 ** len(encoder_rates))
self.latent_dim = latent_dim
self.hop_length = np.prod(encoder_rates)
self.encoder = Encoder(encoder_dim, encoder_rates, latent_dim, causal=causal, lstm=lstm)
self.n_codebooks = n_codebooks
self.codebook_size = codebook_size
self.codebook_dim = codebook_dim
self.quantizer = ResidualVectorQuantize(
input_dim=latent_dim,
n_codebooks=n_codebooks,
codebook_size=codebook_size,
codebook_dim=codebook_dim,
quantizer_dropout=quantizer_dropout,
)
self.decoder = Decoder(
latent_dim,
decoder_dim,
decoder_rates,
lstm=lstm,
causal=causal,
)
self.sample_rate = sample_rate
self.apply(init_weights)
self.delay = self.get_delay()
def preprocess(self, audio_data, sample_rate):
if sample_rate is None:
sample_rate = self.sample_rate
assert sample_rate == self.sample_rate
length = audio_data.shape[-1]
right_pad = math.ceil(length / self.hop_length) * self.hop_length - length
audio_data = nn.functional.pad(audio_data, (0, right_pad))
return audio_data
def encode(
self,
audio_data: torch.Tensor,
n_quantizers: int = None,
):
"""Encode given audio data and return quantized latent codes
Parameters
----------
audio_data : Tensor[B x 1 x T]
Audio data to encode
n_quantizers : int, optional
Number of quantizers to use, by default None
If None, all quantizers are used.
Returns
-------
dict
A dictionary with the following keys:
"z" : Tensor[B x D x T]
Quantized continuous representation of input
"codes" : Tensor[B x N x T]
Codebook indices for each codebook
(quantized discrete representation of input)
"latents" : Tensor[B x N*D x T]
Projected latents (continuous representation of input before quantization)
"vq/commitment_loss" : Tensor[1]
Commitment loss to train encoder to predict vectors closer to codebook
entries
"vq/codebook_loss" : Tensor[1]
Codebook loss to update the codebook
"length" : int
Number of samples in input audio
"""
z = self.encoder(audio_data)
z, codes, latents, commitment_loss, codebook_loss = self.quantizer(
z, n_quantizers
)
return z, codes, latents, commitment_loss, codebook_loss
def decode(self, z: torch.Tensor):
"""Decode given latent codes and return audio data
Parameters
----------
z : Tensor[B x D x T]
Quantized continuous representation of input
length : int, optional
Number of samples in output audio, by default None
Returns
-------
dict
A dictionary with the following keys:
"audio" : Tensor[B x 1 x length]
Decoded audio data.
"""
return self.decoder(z)
def forward(
self,
audio_data: torch.Tensor,
sample_rate: int = None,
n_quantizers: int = None,
):
"""Model forward pass
Parameters
----------
audio_data : Tensor[B x 1 x T]
Audio data to encode
sample_rate : int, optional
Sample rate of audio data in Hz, by default None
If None, defaults to `self.sample_rate`
n_quantizers : int, optional
Number of quantizers to use, by default None.
If None, all quantizers are used.
Returns
-------
dict
A dictionary with the following keys:
"z" : Tensor[B x D x T]
Quantized continuous representation of input
"codes" : Tensor[B x N x T]
Codebook indices for each codebook
(quantized discrete representation of input)
"latents" : Tensor[B x N*D x T]
Projected latents (continuous representation of input before quantization)
"vq/commitment_loss" : Tensor[1]
Commitment loss to train encoder to predict vectors closer to codebook
entries
"vq/codebook_loss" : Tensor[1]
Codebook loss to update the codebook
"length" : int
Number of samples in input audio
"audio" : Tensor[B x 1 x length]
Decoded audio data.
"""
length = audio_data.shape[-1]
audio_data = self.preprocess(audio_data, sample_rate)
z, codes, latents, commitment_loss, codebook_loss = self.encode(
audio_data, n_quantizers
)
x = self.decode(z)
return {
"audio": x[..., :length],
"z": z,
"codes": codes,
"latents": latents,
"vq/commitment_loss": commitment_loss,
"vq/codebook_loss": codebook_loss,
}
if __name__ == "__main__":
import numpy as np
from functools import partial
model = DAC().to("cpu")
for n, m in model.named_modules():
o = m.extra_repr()
p = sum([np.prod(p.size()) for p in m.parameters()])
fn = lambda o, p: o + f" {p/1e6:<.3f}M params."
setattr(m, "extra_repr", partial(fn, o=o, p=p))
print(model)
print("Total # of params: ", sum([np.prod(p.size()) for p in model.parameters()]))
length = 88200 * 2
x = torch.randn(1, 1, length).to(model.device)
x.requires_grad_(True)
x.retain_grad()
# Make a forward pass
out = model(x)["audio"]
print("Input shape:", x.shape)
print("Output shape:", out.shape)
# Create gradient variable
grad = torch.zeros_like(out)
grad[:, :, grad.shape[-1] // 2] = 1
# Make a backward pass
out.backward(grad)
# Check non-zero values
gradmap = x.grad.squeeze(0)
gradmap = (gradmap != 0).sum(0) # sum across features
rf = (gradmap != 0).sum()
print(f"Receptive field: {rf.item()}")
x = AudioSignal(torch.randn(1, 1, 44100 * 60), 44100)
model.decompress(model.compress(x, verbose=True), verbose=True)

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import torch
import torch.nn as nn
import torch.nn.functional as F
from audiotools import AudioSignal
from audiotools import ml
from audiotools import STFTParams
from einops import rearrange
from torch.nn.utils import weight_norm
def WNConv1d(*args, **kwargs):
act = kwargs.pop("act", True)
conv = weight_norm(nn.Conv1d(*args, **kwargs))
if not act:
return conv
return nn.Sequential(conv, nn.LeakyReLU(0.1))
def WNConv2d(*args, **kwargs):
act = kwargs.pop("act", True)
conv = weight_norm(nn.Conv2d(*args, **kwargs))
if not act:
return conv
return nn.Sequential(conv, nn.LeakyReLU(0.1))
class MPD(nn.Module):
def __init__(self, period):
super().__init__()
self.period = period
self.convs = nn.ModuleList(
[
WNConv2d(1, 32, (5, 1), (3, 1), padding=(2, 0)),
WNConv2d(32, 128, (5, 1), (3, 1), padding=(2, 0)),
WNConv2d(128, 512, (5, 1), (3, 1), padding=(2, 0)),
WNConv2d(512, 1024, (5, 1), (3, 1), padding=(2, 0)),
WNConv2d(1024, 1024, (5, 1), 1, padding=(2, 0)),
]
)
self.conv_post = WNConv2d(
1024, 1, kernel_size=(3, 1), padding=(1, 0), act=False
)
def pad_to_period(self, x):
t = x.shape[-1]
x = F.pad(x, (0, self.period - t % self.period), mode="reflect")
return x
def forward(self, x):
fmap = []
x = self.pad_to_period(x)
x = rearrange(x, "b c (l p) -> b c l p", p=self.period)
for layer in self.convs:
x = layer(x)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
return fmap
class MSD(nn.Module):
def __init__(self, rate: int = 1, sample_rate: int = 44100):
super().__init__()
self.convs = nn.ModuleList(
[
WNConv1d(1, 16, 15, 1, padding=7),
WNConv1d(16, 64, 41, 4, groups=4, padding=20),
WNConv1d(64, 256, 41, 4, groups=16, padding=20),
WNConv1d(256, 1024, 41, 4, groups=64, padding=20),
WNConv1d(1024, 1024, 41, 4, groups=256, padding=20),
WNConv1d(1024, 1024, 5, 1, padding=2),
]
)
self.conv_post = WNConv1d(1024, 1, 3, 1, padding=1, act=False)
self.sample_rate = sample_rate
self.rate = rate
def forward(self, x):
x = AudioSignal(x, self.sample_rate)
x.resample(self.sample_rate // self.rate)
x = x.audio_data
fmap = []
for l in self.convs:
x = l(x)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
return fmap
BANDS = [(0.0, 0.1), (0.1, 0.25), (0.25, 0.5), (0.5, 0.75), (0.75, 1.0)]
class MRD(nn.Module):
def __init__(
self,
window_length: int,
hop_factor: float = 0.25,
sample_rate: int = 44100,
bands: list = BANDS,
):
"""Complex multi-band spectrogram discriminator.
Parameters
----------
window_length : int
Window length of STFT.
hop_factor : float, optional
Hop factor of the STFT, defaults to ``0.25 * window_length``.
sample_rate : int, optional
Sampling rate of audio in Hz, by default 44100
bands : list, optional
Bands to run discriminator over.
"""
super().__init__()
self.window_length = window_length
self.hop_factor = hop_factor
self.sample_rate = sample_rate
self.stft_params = STFTParams(
window_length=window_length,
hop_length=int(window_length * hop_factor),
match_stride=True,
)
n_fft = window_length // 2 + 1
bands = [(int(b[0] * n_fft), int(b[1] * n_fft)) for b in bands]
self.bands = bands
ch = 32
convs = lambda: nn.ModuleList(
[
WNConv2d(2, ch, (3, 9), (1, 1), padding=(1, 4)),
WNConv2d(ch, ch, (3, 9), (1, 2), padding=(1, 4)),
WNConv2d(ch, ch, (3, 9), (1, 2), padding=(1, 4)),
WNConv2d(ch, ch, (3, 9), (1, 2), padding=(1, 4)),
WNConv2d(ch, ch, (3, 3), (1, 1), padding=(1, 1)),
]
)
self.band_convs = nn.ModuleList([convs() for _ in range(len(self.bands))])
self.conv_post = WNConv2d(ch, 1, (3, 3), (1, 1), padding=(1, 1), act=False)
def spectrogram(self, x):
x = AudioSignal(x, self.sample_rate, stft_params=self.stft_params)
x = torch.view_as_real(x.stft())
x = rearrange(x, "b 1 f t c -> (b 1) c t f")
# Split into bands
x_bands = [x[..., b[0] : b[1]] for b in self.bands]
return x_bands
def forward(self, x):
x_bands = self.spectrogram(x)
fmap = []
x = []
for band, stack in zip(x_bands, self.band_convs):
for layer in stack:
band = layer(band)
fmap.append(band)
x.append(band)
x = torch.cat(x, dim=-1)
x = self.conv_post(x)
fmap.append(x)
return fmap
class Discriminator(nn.Module):
def __init__(
self,
rates: list = [],
periods: list = [2, 3, 5, 7, 11],
fft_sizes: list = [2048, 1024, 512],
sample_rate: int = 44100,
bands: list = BANDS,
):
"""Discriminator that combines multiple discriminators.
Parameters
----------
rates : list, optional
sampling rates (in Hz) to run MSD at, by default []
If empty, MSD is not used.
periods : list, optional
periods (of samples) to run MPD at, by default [2, 3, 5, 7, 11]
fft_sizes : list, optional
Window sizes of the FFT to run MRD at, by default [2048, 1024, 512]
sample_rate : int, optional
Sampling rate of audio in Hz, by default 44100
bands : list, optional
Bands to run MRD at, by default `BANDS`
"""
super().__init__()
discs = []
discs += [MPD(p) for p in periods]
discs += [MSD(r, sample_rate=sample_rate) for r in rates]
discs += [MRD(f, sample_rate=sample_rate, bands=bands) for f in fft_sizes]
self.discriminators = nn.ModuleList(discs)
def preprocess(self, y):
# Remove DC offset
y = y - y.mean(dim=-1, keepdims=True)
# Peak normalize the volume of input audio
y = 0.8 * y / (y.abs().max(dim=-1, keepdim=True)[0] + 1e-9)
return y
def forward(self, x):
x = self.preprocess(x)
fmaps = [d(x) for d in self.discriminators]
return fmaps
if __name__ == "__main__":
disc = Discriminator()
x = torch.zeros(1, 1, 44100)
results = disc(x)
for i, result in enumerate(results):
print(f"disc{i}")
for i, r in enumerate(result):
print(r.shape, r.mean(), r.min(), r.max())
print()

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# Copyright (c) Meta Platforms, Inc. and 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.
"""Convolutional layers wrappers and utilities."""
import math
import typing as tp
import warnings
import torch
from torch import nn
from torch.nn import functional as F
from torch.nn.utils import spectral_norm, weight_norm
import typing as tp
import einops
class ConvLayerNorm(nn.LayerNorm):
"""
Convolution-friendly LayerNorm that moves channels to last dimensions
before running the normalization and moves them back to original position right after.
"""
def __init__(self, normalized_shape: tp.Union[int, tp.List[int], torch.Size], **kwargs):
super().__init__(normalized_shape, **kwargs)
def forward(self, x):
x = einops.rearrange(x, 'b ... t -> b t ...')
x = super().forward(x)
x = einops.rearrange(x, 'b t ... -> b ... t')
return
CONV_NORMALIZATIONS = frozenset(['none', 'weight_norm', 'spectral_norm',
'time_layer_norm', 'layer_norm', 'time_group_norm'])
def apply_parametrization_norm(module: nn.Module, norm: str = 'none') -> nn.Module:
assert norm in CONV_NORMALIZATIONS
if norm == 'weight_norm':
return weight_norm(module)
elif norm == 'spectral_norm':
return spectral_norm(module)
else:
# We already check was in CONV_NORMALIZATION, so any other choice
# doesn't need reparametrization.
return module
def get_norm_module(module: nn.Module, causal: bool = False, norm: str = 'none', **norm_kwargs) -> nn.Module:
"""Return the proper normalization module. If causal is True, this will ensure the returned
module is causal, or return an error if the normalization doesn't support causal evaluation.
"""
assert norm in CONV_NORMALIZATIONS
if norm == 'layer_norm':
assert isinstance(module, nn.modules.conv._ConvNd)
return ConvLayerNorm(module.out_channels, **norm_kwargs)
elif norm == 'time_group_norm':
if causal:
raise ValueError("GroupNorm doesn't support causal evaluation.")
assert isinstance(module, nn.modules.conv._ConvNd)
return nn.GroupNorm(1, module.out_channels, **norm_kwargs)
else:
return nn.Identity()
def get_extra_padding_for_conv1d(x: torch.Tensor, kernel_size: int, stride: int,
padding_total: int = 0) -> int:
"""See `pad_for_conv1d`.
"""
length = x.shape[-1]
n_frames = (length - kernel_size + padding_total) / stride + 1
ideal_length = (math.ceil(n_frames) - 1) * stride + (kernel_size - padding_total)
return ideal_length - length
def pad_for_conv1d(x: torch.Tensor, kernel_size: int, stride: int, padding_total: int = 0):
"""Pad for a convolution to make sure that the last window is full.
Extra padding is added at the end. This is required to ensure that we can rebuild
an output of the same length, as otherwise, even with padding, some time steps
might get removed.
For instance, with total padding = 4, kernel size = 4, stride = 2:
0 0 1 2 3 4 5 0 0 # (0s are padding)
1 2 3 # (output frames of a convolution, last 0 is never used)
0 0 1 2 3 4 5 0 # (output of tr. conv., but pos. 5 is going to get removed as padding)
1 2 3 4 # once you removed padding, we are missing one time step !
"""
extra_padding = get_extra_padding_for_conv1d(x, kernel_size, stride, padding_total)
return F.pad(x, (0, extra_padding))
def pad1d(x: torch.Tensor, paddings: tp.Tuple[int, int], mode: str = 'zero', value: float = 0.):
"""Tiny wrapper around F.pad, just to allow for reflect padding on small input.
If this is the case, we insert extra 0 padding to the right before the reflection happen.
"""
length = x.shape[-1]
padding_left, padding_right = paddings
assert padding_left >= 0 and padding_right >= 0, (padding_left, padding_right)
if mode == 'reflect':
max_pad = max(padding_left, padding_right)
extra_pad = 0
if length <= max_pad:
extra_pad = max_pad - length + 1
x = F.pad(x, (0, extra_pad))
padded = F.pad(x, paddings, mode, value)
end = padded.shape[-1] - extra_pad
return padded[..., :end]
else:
return F.pad(x, paddings, mode, value)
def unpad1d(x: torch.Tensor, paddings: tp.Tuple[int, int]):
"""Remove padding from x, handling properly zero padding. Only for 1d!"""
padding_left, padding_right = paddings
assert padding_left >= 0 and padding_right >= 0, (padding_left, padding_right)
assert (padding_left + padding_right) <= x.shape[-1]
end = x.shape[-1] - padding_right
return x[..., padding_left: end]
class NormConv1d(nn.Module):
"""Wrapper around Conv1d and normalization applied to this conv
to provide a uniform interface across normalization approaches.
"""
def __init__(self, *args, causal: bool = False, norm: str = 'none',
norm_kwargs: tp.Dict[str, tp.Any] = {}, **kwargs):
super().__init__()
self.conv = apply_parametrization_norm(nn.Conv1d(*args, **kwargs), norm)
self.norm = get_norm_module(self.conv, causal, norm, **norm_kwargs)
self.norm_type = norm
def forward(self, x):
x = self.conv(x)
x = self.norm(x)
return x
class NormConv2d(nn.Module):
"""Wrapper around Conv2d and normalization applied to this conv
to provide a uniform interface across normalization approaches.
"""
def __init__(self, *args, norm: str = 'none',
norm_kwargs: tp.Dict[str, tp.Any] = {}, **kwargs):
super().__init__()
self.conv = apply_parametrization_norm(nn.Conv2d(*args, **kwargs), norm)
self.norm = get_norm_module(self.conv, causal=False, norm=norm, **norm_kwargs)
self.norm_type = norm
def forward(self, x):
x = self.conv(x)
x = self.norm(x)
return x
class NormConvTranspose1d(nn.Module):
"""Wrapper around ConvTranspose1d and normalization applied to this conv
to provide a uniform interface across normalization approaches.
"""
def __init__(self, *args, causal: bool = False, norm: str = 'none',
norm_kwargs: tp.Dict[str, tp.Any] = {}, **kwargs):
super().__init__()
self.convtr = apply_parametrization_norm(nn.ConvTranspose1d(*args, **kwargs), norm)
self.norm = get_norm_module(self.convtr, causal, norm, **norm_kwargs)
self.norm_type = norm
def forward(self, x):
x = self.convtr(x)
x = self.norm(x)
return x
class NormConvTranspose2d(nn.Module):
"""Wrapper around ConvTranspose2d and normalization applied to this conv
to provide a uniform interface across normalization approaches.
"""
def __init__(self, *args, norm: str = 'none',
norm_kwargs: tp.Dict[str, tp.Any] = {}, **kwargs):
super().__init__()
self.convtr = apply_parametrization_norm(nn.ConvTranspose2d(*args, **kwargs), norm)
self.norm = get_norm_module(self.convtr, causal=False, norm=norm, **norm_kwargs)
def forward(self, x):
x = self.convtr(x)
x = self.norm(x)
return x
class SConv1d(nn.Module):
"""Conv1d with some builtin handling of asymmetric or causal padding
and normalization.
"""
def __init__(self, in_channels: int, out_channels: int,
kernel_size: int, stride: int = 1, dilation: int = 1,
groups: int = 1, bias: bool = True, causal: bool = False,
norm: str = 'none', norm_kwargs: tp.Dict[str, tp.Any] = {},
pad_mode: str = 'reflect', **kwargs):
super().__init__()
# warn user on unusual setup between dilation and stride
if stride > 1 and dilation > 1:
warnings.warn('SConv1d has been initialized with stride > 1 and dilation > 1'
f' (kernel_size={kernel_size} stride={stride}, dilation={dilation}).')
self.conv = NormConv1d(in_channels, out_channels, kernel_size, stride,
dilation=dilation, groups=groups, bias=bias, causal=causal,
norm=norm, norm_kwargs=norm_kwargs)
self.causal = causal
self.pad_mode = pad_mode
self.cache_enabled = False
def reset_cache(self):
"""Reset the cache when starting a new stream."""
self.cache = None
self.cache_enabled = True
def forward(self, x):
B, C, T = x.shape
kernel_size = self.conv.conv.kernel_size[0]
stride = self.conv.conv.stride[0]
dilation = self.conv.conv.dilation[0]
kernel_size = (kernel_size - 1) * dilation + 1 # effective kernel size with dilations
padding_total = kernel_size - stride
extra_padding = get_extra_padding_for_conv1d(x, kernel_size, stride, padding_total)
if self.causal:
# Left padding for causal
if self.cache_enabled and self.cache is not None:
# Concatenate the cache (previous inputs) with the new input for streaming
x = torch.cat([self.cache, x], dim=2)
else:
x = pad1d(x, (padding_total, extra_padding), mode=self.pad_mode)
else:
# Asymmetric padding required for odd strides
padding_right = padding_total // 2
padding_left = padding_total - padding_right
x = pad1d(x, (padding_left, padding_right + extra_padding), mode=self.pad_mode)
# Store the most recent input frames for future cache use
if self.cache_enabled:
if self.cache is None:
# Initialize cache with zeros (at the start of streaming)
self.cache = torch.zeros(B, C, kernel_size - 1, device=x.device)
# Update the cache by storing the latest input frames
if kernel_size > 1:
self.cache = x[:, :, -kernel_size + 1:].detach() # Only store the necessary frames
return self.conv(x)
class SConvTranspose1d(nn.Module):
"""ConvTranspose1d with some builtin handling of asymmetric or causal padding
and normalization.
"""
def __init__(self, in_channels: int, out_channels: int,
kernel_size: int, stride: int = 1, causal: bool = False,
norm: str = 'none', trim_right_ratio: float = 1.,
norm_kwargs: tp.Dict[str, tp.Any] = {}, **kwargs):
super().__init__()
self.convtr = NormConvTranspose1d(in_channels, out_channels, kernel_size, stride,
causal=causal, norm=norm, norm_kwargs=norm_kwargs)
self.causal = causal
self.trim_right_ratio = trim_right_ratio
assert self.causal or self.trim_right_ratio == 1., \
"`trim_right_ratio` != 1.0 only makes sense for causal convolutions"
assert self.trim_right_ratio >= 0. and self.trim_right_ratio <= 1.
def forward(self, x):
kernel_size = self.convtr.convtr.kernel_size[0]
stride = self.convtr.convtr.stride[0]
padding_total = kernel_size - stride
y = self.convtr(x)
# We will only trim fixed padding. Extra padding from `pad_for_conv1d` would be
# removed at the very end, when keeping only the right length for the output,
# as removing it here would require also passing the length at the matching layer
# in the encoder.
if self.causal:
# Trim the padding on the right according to the specified ratio
# if trim_right_ratio = 1.0, trim everything from right
padding_right = math.ceil(padding_total * self.trim_right_ratio)
padding_left = padding_total - padding_right
y = unpad1d(y, (padding_left, padding_right))
else:
# Asymmetric padding required for odd strides
padding_right = padding_total // 2
padding_left = padding_total - padding_right
y = unpad1d(y, (padding_left, padding_right))
return y
class SLSTM(nn.Module):
"""
LSTM without worrying about the hidden state, nor the layout of the data.
Expects input as convolutional layout.
"""
def __init__(self, dimension: int, num_layers: int = 2, skip: bool = True):
super().__init__()
self.skip = skip
self.lstm = nn.LSTM(dimension, dimension, num_layers)
self.hidden = None
self.cache_enabled = False
def forward(self, x):
x = x.permute(2, 0, 1)
if self.training or not self.cache_enabled:
y, _ = self.lstm(x)
else:
y, self.hidden = self.lstm(x, self.hidden)
if self.skip:
y = y + x
y = y.permute(1, 2, 0)
return y
def reset_cache(self):
self.hidden = None
self.cache_enabled = True