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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() | |
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) | |
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() | |
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(): | |
... | |
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(): | |
... | |