|
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 = [] |
|
|
|
|
|
b, c, t = x.shape |
|
if t % self.period != 0: |
|
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) |
|
|
|
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: |
|
c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device) |
|
|
|
|
|
if spec_lengths is None: |
|
spec_lengths = (torch.ones(spec.size(0)) * spec.size(-1)).to(spec.device) |
|
|
|
|
|
g = None |
|
if not self.use_spk: |
|
g = self.enc_spk(mel).unsqueeze(-1) |
|
|
|
|
|
_, 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) |
|
|
|
|
|
z_p = self.flow(z, spec_mask, g=g) |
|
|
|
|
|
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, |
|
) |
|
|
|
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(): |
|
... |
|
|