clonar-voz / TTS /vc /models /freevc.py
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voice-clone with single audio sample input
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from typing import Dict, List, Optional, Tuple, Union
import librosa
import numpy as np
import torch
from coqpit import Coqpit
from torch import nn
from torch.nn import Conv1d, Conv2d, ConvTranspose1d
from torch.nn import functional as F
from torch.nn.utils import spectral_norm
from torch.nn.utils.parametrizations import weight_norm
from torch.nn.utils.parametrize import remove_parametrizations
import TTS.vc.modules.freevc.commons as commons
import TTS.vc.modules.freevc.modules as modules
from TTS.tts.utils.speakers import SpeakerManager
from TTS.utils.io import load_fsspec
from TTS.vc.configs.freevc_config import FreeVCConfig
from TTS.vc.models.base_vc import BaseVC
from TTS.vc.modules.freevc.commons import get_padding, init_weights
from TTS.vc.modules.freevc.mel_processing import mel_spectrogram_torch
from TTS.vc.modules.freevc.speaker_encoder.speaker_encoder import SpeakerEncoder as SpeakerEncoderEx
from TTS.vc.modules.freevc.wavlm import get_wavlm
class ResidualCouplingBlock(nn.Module):
def __init__(self, channels, hidden_channels, kernel_size, dilation_rate, n_layers, n_flows=4, gin_channels=0):
super().__init__()
self.channels = channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.n_flows = n_flows
self.gin_channels = gin_channels
self.flows = nn.ModuleList()
for i in range(n_flows):
self.flows.append(
modules.ResidualCouplingLayer(
channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=gin_channels,
mean_only=True,
)
)
self.flows.append(modules.Flip())
def forward(self, x, x_mask, g=None, reverse=False):
if not reverse:
for flow in self.flows:
x, _ = flow(x, x_mask, g=g, reverse=reverse)
else:
for flow in reversed(self.flows):
x = flow(x, x_mask, g=g, reverse=reverse)
return x
class Encoder(nn.Module):
def __init__(
self, in_channels, out_channels, hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=0
):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.pre = nn.Conv1d(in_channels, hidden_channels, 1)
self.enc = modules.WN(hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, x, x_lengths, g=None):
x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype)
x = self.pre(x) * x_mask
x = self.enc(x, x_mask, g=g)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask
return z, m, logs, x_mask
class Generator(torch.nn.Module):
def __init__(
self,
initial_channel,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
gin_channels=0,
):
super(Generator, self).__init__()
self.num_kernels = len(resblock_kernel_sizes)
self.num_upsamples = len(upsample_rates)
self.conv_pre = Conv1d(initial_channel, upsample_initial_channel, 7, 1, padding=3)
resblock = modules.ResBlock1 if resblock == "1" else modules.ResBlock2
self.ups = nn.ModuleList()
for i, (u, k) in enumerate(zip(upsample_rates, upsample_kernel_sizes)):
self.ups.append(
weight_norm(
ConvTranspose1d(
upsample_initial_channel // (2**i),
upsample_initial_channel // (2 ** (i + 1)),
k,
u,
padding=(k - u) // 2,
)
)
)
self.resblocks = nn.ModuleList()
for i in range(len(self.ups)):
ch = upsample_initial_channel // (2 ** (i + 1))
for j, (k, d) in enumerate(zip(resblock_kernel_sizes, resblock_dilation_sizes)):
self.resblocks.append(resblock(ch, k, d))
self.conv_post = Conv1d(ch, 1, 7, 1, padding=3, bias=False)
self.ups.apply(init_weights)
if gin_channels != 0:
self.cond = nn.Conv1d(gin_channels, upsample_initial_channel, 1)
def forward(self, x, g=None):
x = self.conv_pre(x)
if g is not None:
x = x + self.cond(g)
for i in range(self.num_upsamples):
x = F.leaky_relu(x, modules.LRELU_SLOPE)
x = self.ups[i](x)
xs = None
for j in range(self.num_kernels):
if xs is None:
xs = self.resblocks[i * self.num_kernels + j](x)
else:
xs += self.resblocks[i * self.num_kernels + j](x)
x = xs / self.num_kernels
x = F.leaky_relu(x)
x = self.conv_post(x)
x = torch.tanh(x)
return x
def remove_weight_norm(self):
print("Removing weight norm...")
for l in self.ups:
remove_parametrizations(l, "weight")
for l in self.resblocks:
remove_parametrizations(l, "weight")
class DiscriminatorP(torch.nn.Module):
def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False):
super(DiscriminatorP, self).__init__()
self.period = period
self.use_spectral_norm = use_spectral_norm
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList(
[
norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(get_padding(kernel_size, 1), 0))),
]
)
self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0)))
def forward(self, x):
fmap = []
# 1d to 2d
b, c, t = x.shape
if t % self.period != 0: # pad first
n_pad = self.period - (t % self.period)
x = F.pad(x, (0, n_pad), "reflect")
t = t + n_pad
x = x.view(b, c, t // self.period, self.period)
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class DiscriminatorS(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(DiscriminatorS, self).__init__()
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList(
[
norm_f(Conv1d(1, 16, 15, 1, padding=7)),
norm_f(Conv1d(16, 64, 41, 4, groups=4, padding=20)),
norm_f(Conv1d(64, 256, 41, 4, groups=16, padding=20)),
norm_f(Conv1d(256, 1024, 41, 4, groups=64, padding=20)),
norm_f(Conv1d(1024, 1024, 41, 4, groups=256, padding=20)),
norm_f(Conv1d(1024, 1024, 5, 1, padding=2)),
]
)
self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1))
def forward(self, x):
fmap = []
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class MultiPeriodDiscriminator(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(MultiPeriodDiscriminator, self).__init__()
periods = [2, 3, 5, 7, 11]
discs = [DiscriminatorS(use_spectral_norm=use_spectral_norm)]
discs = discs + [DiscriminatorP(i, use_spectral_norm=use_spectral_norm) for i in periods]
self.discriminators = nn.ModuleList(discs)
def forward(self, y, y_hat):
y_d_rs = []
y_d_gs = []
fmap_rs = []
fmap_gs = []
for i, d in enumerate(self.discriminators):
y_d_r, fmap_r = d(y)
y_d_g, fmap_g = d(y_hat)
y_d_rs.append(y_d_r)
y_d_gs.append(y_d_g)
fmap_rs.append(fmap_r)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class SpeakerEncoder(torch.nn.Module):
def __init__(self, mel_n_channels=80, model_num_layers=3, model_hidden_size=256, model_embedding_size=256):
super(SpeakerEncoder, self).__init__()
self.lstm = nn.LSTM(mel_n_channels, model_hidden_size, model_num_layers, batch_first=True)
self.linear = nn.Linear(model_hidden_size, model_embedding_size)
self.relu = nn.ReLU()
def forward(self, mels):
self.lstm.flatten_parameters()
_, (hidden, _) = self.lstm(mels)
embeds_raw = self.relu(self.linear(hidden[-1]))
return embeds_raw / torch.norm(embeds_raw, dim=1, keepdim=True)
def compute_partial_slices(self, total_frames, partial_frames, partial_hop):
mel_slices = []
for i in range(0, total_frames - partial_frames, partial_hop):
mel_range = torch.arange(i, i + partial_frames)
mel_slices.append(mel_range)
return mel_slices
def embed_utterance(self, mel, partial_frames=128, partial_hop=64):
mel_len = mel.size(1)
last_mel = mel[:, -partial_frames:]
if mel_len > partial_frames:
mel_slices = self.compute_partial_slices(mel_len, partial_frames, partial_hop)
mels = list(mel[:, s] for s in mel_slices)
mels.append(last_mel)
mels = torch.stack(tuple(mels), 0).squeeze(1)
with torch.no_grad():
partial_embeds = self(mels)
embed = torch.mean(partial_embeds, axis=0).unsqueeze(0)
# embed = embed / torch.linalg.norm(embed, 2)
else:
with torch.no_grad():
embed = self(last_mel)
return embed
class FreeVC(BaseVC):
"""
Papaer::
https://arxiv.org/abs/2210.15418#
Paper Abstract::
Voice conversion (VC) can be achieved by first extracting source content information and target speaker
information, and then reconstructing waveform with these information. However, current approaches normally
either extract dirty content information with speaker information leaked in, or demand a large amount of
annotated data for training. Besides, the quality of reconstructed waveform can be degraded by the
mismatch between conversion model and vocoder. In this paper, we adopt the end-to-end framework of VITS for
high-quality waveform reconstruction, and propose strategies for clean content information extraction without
text annotation. We disentangle content information by imposing an information bottleneck to WavLM features,
and propose the spectrogram-resize based data augmentation to improve the purity of extracted content
information. Experimental results show that the proposed method outperforms the latest VC models trained with
annotated data and has greater robustness.
Original Code::
https://github.com/OlaWod/FreeVC
Examples:
>>> from TTS.vc.configs.freevc_config import FreeVCConfig
>>> from TTS.vc.models.freevc import FreeVC
>>> config = FreeVCConfig()
>>> model = FreeVC(config)
"""
def __init__(self, config: Coqpit, speaker_manager: SpeakerManager = None):
super().__init__(config, None, speaker_manager, None)
self.init_multispeaker(config)
self.spec_channels = self.args.spec_channels
self.inter_channels = self.args.inter_channels
self.hidden_channels = self.args.hidden_channels
self.filter_channels = self.args.filter_channels
self.n_heads = self.args.n_heads
self.n_layers = self.args.n_layers
self.kernel_size = self.args.kernel_size
self.p_dropout = self.args.p_dropout
self.resblock = self.args.resblock
self.resblock_kernel_sizes = self.args.resblock_kernel_sizes
self.resblock_dilation_sizes = self.args.resblock_dilation_sizes
self.upsample_rates = self.args.upsample_rates
self.upsample_initial_channel = self.args.upsample_initial_channel
self.upsample_kernel_sizes = self.args.upsample_kernel_sizes
self.segment_size = self.args.segment_size
self.gin_channels = self.args.gin_channels
self.ssl_dim = self.args.ssl_dim
self.use_spk = self.args.use_spk
self.enc_p = Encoder(self.args.ssl_dim, self.inter_channels, self.hidden_channels, 5, 1, 16)
self.dec = Generator(
self.inter_channels,
self.resblock,
self.resblock_kernel_sizes,
self.resblock_dilation_sizes,
self.upsample_rates,
self.upsample_initial_channel,
self.upsample_kernel_sizes,
gin_channels=self.gin_channels,
)
self.enc_q = Encoder(
self.spec_channels, self.inter_channels, self.hidden_channels, 5, 1, 16, gin_channels=self.gin_channels
)
self.flow = ResidualCouplingBlock(
self.inter_channels, self.hidden_channels, 5, 1, 4, gin_channels=self.gin_channels
)
if not self.use_spk:
self.enc_spk = SpeakerEncoder(model_hidden_size=self.gin_channels, model_embedding_size=self.gin_channels)
else:
self.load_pretrained_speaker_encoder()
self.wavlm = get_wavlm()
@property
def device(self):
return next(self.parameters()).device
def load_pretrained_speaker_encoder(self):
"""Load pretrained speaker encoder model as mentioned in the paper."""
print(" > Loading pretrained speaker encoder model ...")
self.enc_spk_ex = SpeakerEncoderEx(
"https://github.com/coqui-ai/TTS/releases/download/v0.13.0_models/speaker_encoder.pt"
)
def init_multispeaker(self, config: Coqpit):
"""Initialize multi-speaker modules of a model. A model can be trained either with a speaker embedding layer
or with external `d_vectors` computed from a speaker encoder model.
You must provide a `speaker_manager` at initialization to set up the multi-speaker modules.
Args:
config (Coqpit): Model configuration.
data (List, optional): Dataset items to infer number of speakers. Defaults to None.
"""
self.num_spks = self.args.num_spks
if self.speaker_manager:
self.num_spks = self.speaker_manager.num_spks
def forward(
self,
c: torch.Tensor,
spec: torch.Tensor,
g: Optional[torch.Tensor] = None,
mel: Optional[torch.Tensor] = None,
c_lengths: Optional[torch.Tensor] = None,
spec_lengths: Optional[torch.Tensor] = None,
) -> Tuple[
torch.Tensor,
torch.Tensor,
torch.Tensor,
Tuple[torch.Tensor, torch.Tensor, torch.Tensor, torch.Tensor, torch.Tensor, torch.Tensor],
]:
"""
Forward pass of the model.
Args:
c: WavLM features. Shape: (batch_size, c_seq_len).
spec: The input spectrogram. Shape: (batch_size, spec_seq_len, spec_dim).
g: The speaker embedding. Shape: (batch_size, spk_emb_dim).
mel: The input mel-spectrogram for the speaker encoder. Shape: (batch_size, mel_seq_len, mel_dim).
c_lengths: The lengths of the WavLM features. Shape: (batch_size,).
spec_lengths: The lengths of the spectrogram. Shape: (batch_size,).
Returns:
o: The output spectrogram. Shape: (batch_size, spec_seq_len, spec_dim).
ids_slice: The slice indices. Shape: (batch_size, num_slices).
spec_mask: The spectrogram mask. Shape: (batch_size, spec_seq_len).
(z, z_p, m_p, logs_p, m_q, logs_q): A tuple of latent variables.
"""
# If c_lengths is None, set it to the length of the last dimension of c
if c_lengths is None:
c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device)
# If spec_lengths is None, set it to the length of the last dimension of spec
if spec_lengths is None:
spec_lengths = (torch.ones(spec.size(0)) * spec.size(-1)).to(spec.device)
# If use_spk is False, compute g from mel using enc_spk
g = None
if not self.use_spk:
g = self.enc_spk(mel).unsqueeze(-1)
# Compute m_p, logs_p, z, m_q, logs_q, and spec_mask using enc_p and enc_q
_, m_p, logs_p, _ = self.enc_p(c, c_lengths)
z, m_q, logs_q, spec_mask = self.enc_q(spec.transpose(1, 2), spec_lengths, g=g)
# Compute z_p using flow
z_p = self.flow(z, spec_mask, g=g)
# Randomly slice z and compute o using dec
z_slice, ids_slice = commons.rand_slice_segments(z, spec_lengths, self.segment_size)
o = self.dec(z_slice, g=g)
return o, ids_slice, spec_mask, (z, z_p, m_p, logs_p, m_q, logs_q)
@torch.no_grad()
def inference(self, c, g=None, mel=None, c_lengths=None):
"""
Inference pass of the model
Args:
c (torch.Tensor): Input tensor. Shape: (batch_size, c_seq_len).
g (torch.Tensor): Speaker embedding tensor. Shape: (batch_size, spk_emb_dim).
mel (torch.Tensor): Mel-spectrogram tensor. Shape: (batch_size, mel_seq_len, mel_dim).
c_lengths (torch.Tensor): Lengths of the input tensor. Shape: (batch_size,).
Returns:
torch.Tensor: Output tensor.
"""
if c_lengths == None:
c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device)
if not self.use_spk:
g = self.enc_spk.embed_utterance(mel)
g = g.unsqueeze(-1)
z_p, m_p, logs_p, c_mask = self.enc_p(c, c_lengths)
z = self.flow(z_p, c_mask, g=g, reverse=True)
o = self.dec(z * c_mask, g=g)
return o
def extract_wavlm_features(self, y):
"""Extract WavLM features from an audio tensor.
Args:
y (torch.Tensor): Audio tensor. Shape: (batch_size, audio_seq_len).
"""
with torch.no_grad():
c = self.wavlm.extract_features(y)[0]
c = c.transpose(1, 2)
return c
def load_audio(self, wav):
"""Read and format the input audio."""
if isinstance(wav, str):
wav, _ = librosa.load(wav, sr=self.config.audio.input_sample_rate)
if isinstance(wav, np.ndarray):
wav = torch.from_numpy(wav).to(self.device)
if isinstance(wav, torch.Tensor):
wav = wav.to(self.device)
if isinstance(wav, list):
wav = torch.from_numpy(np.array(wav)).to(self.device)
return wav.float()
@torch.inference_mode()
def voice_conversion(self, src, tgt):
"""
Voice conversion pass of the model.
Args:
src (str or torch.Tensor): Source utterance.
tgt (str or torch.Tensor): Target utterance.
Returns:
torch.Tensor: Output tensor.
"""
wav_tgt = self.load_audio(tgt).cpu().numpy()
wav_tgt, _ = librosa.effects.trim(wav_tgt, top_db=20)
if self.config.model_args.use_spk:
g_tgt = self.enc_spk_ex.embed_utterance(wav_tgt)
g_tgt = torch.from_numpy(g_tgt)[None, :, None].to(self.device)
else:
wav_tgt = torch.from_numpy(wav_tgt).unsqueeze(0).to(self.device)
mel_tgt = mel_spectrogram_torch(
wav_tgt,
self.config.audio.filter_length,
self.config.audio.n_mel_channels,
self.config.audio.input_sample_rate,
self.config.audio.hop_length,
self.config.audio.win_length,
self.config.audio.mel_fmin,
self.config.audio.mel_fmax,
)
# src
wav_src = self.load_audio(src)
c = self.extract_wavlm_features(wav_src[None, :])
if self.config.model_args.use_spk:
audio = self.inference(c, g=g_tgt)
else:
audio = self.inference(c, mel=mel_tgt.transpose(1, 2))
audio = audio[0][0].data.cpu().float().numpy()
return audio
def eval_step():
...
@staticmethod
def init_from_config(config: FreeVCConfig, samples: Union[List[List], List[Dict]] = None, verbose=True):
model = FreeVC(config)
return model
def load_checkpoint(self, config, checkpoint_path, eval=False, strict=True, cache=False):
state = load_fsspec(checkpoint_path, map_location=torch.device("cpu"), cache=cache)
self.load_state_dict(state["model"], strict=strict)
if eval:
self.eval()
def train_step():
...