add hifigan train code

This commit is contained in:
lyuxiang.lx
2024-10-09 17:36:42 +08:00
parent 67f298d94a
commit cb200b21c5
10 changed files with 768 additions and 40 deletions

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@@ -87,7 +87,7 @@ def main():
logging.basicConfig(level=logging.DEBUG,
format='%(asctime)s %(levelname)s %(message)s')
override_dict = {k: None for k in ['llm', 'flow', 'hift'] if k != args.model}
override_dict = {k: None for k in ['llm', 'flow', 'hifigan'] if k != args.model}
with open(args.config, 'r') as f:
configs = load_hyperpyyaml(f, overrides=override_dict)
configs['train_conf'].update(vars(args))

137
cosyvoice/bin/train_gan.py Normal file
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@@ -0,0 +1,137 @@
# Copyright (c) 2024 Alibaba Inc (authors: Xiang Lyu)
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
from __future__ import print_function
import argparse
import datetime
import logging
logging.getLogger('matplotlib').setLevel(logging.WARNING)
from copy import deepcopy
import torch
import torch.distributed as dist
import deepspeed
from hyperpyyaml import load_hyperpyyaml
from torch.distributed.elastic.multiprocessing.errors import record
from cosyvoice.utils.executor_gan import Executor
from cosyvoice.utils.train_utils import (
init_distributed,
init_dataset_and_dataloader,
init_optimizer_and_scheduler_gan,
init_summarywriter, save_model,
wrap_cuda_model, check_modify_and_save_config)
def get_args():
parser = argparse.ArgumentParser(description='training your network')
parser.add_argument('--train_engine',
default='torch_ddp',
choices=['torch_ddp', 'deepspeed'],
help='Engine for paralleled training')
parser.add_argument('--model', required=True, help='model which will be trained')
parser.add_argument('--config', required=True, help='config file')
parser.add_argument('--train_data', required=True, help='train data file')
parser.add_argument('--cv_data', required=True, help='cv data file')
parser.add_argument('--checkpoint', help='checkpoint model')
parser.add_argument('--model_dir', required=True, help='save model dir')
parser.add_argument('--tensorboard_dir',
default='tensorboard',
help='tensorboard log dir')
parser.add_argument('--ddp.dist_backend',
dest='dist_backend',
default='nccl',
choices=['nccl', 'gloo'],
help='distributed backend')
parser.add_argument('--num_workers',
default=0,
type=int,
help='num of subprocess workers for reading')
parser.add_argument('--prefetch',
default=100,
type=int,
help='prefetch number')
parser.add_argument('--pin_memory',
action='store_true',
default=False,
help='Use pinned memory buffers used for reading')
parser.add_argument('--deepspeed.save_states',
dest='save_states',
default='model_only',
choices=['model_only', 'model+optimizer'],
help='save model/optimizer states')
parser.add_argument('--timeout',
default=30,
type=int,
help='timeout (in seconds) of cosyvoice_join.')
parser = deepspeed.add_config_arguments(parser)
args = parser.parse_args()
return args
@record
def main():
args = get_args()
logging.basicConfig(level=logging.DEBUG,
format='%(asctime)s %(levelname)s %(message)s')
override_dict = {k: None for k in ['llm', 'flow', 'hifigan'] if k != args.model}
with open(args.config, 'r') as f:
configs = load_hyperpyyaml(f, overrides=override_dict, overrides_must_match=False)
configs['train_conf'].update(vars(args))
# Init env for ddp
init_distributed(args)
# Get dataset & dataloader
train_dataset, cv_dataset, train_data_loader, cv_data_loader = \
init_dataset_and_dataloader(args, configs)
# Do some sanity checks and save config to arsg.model_dir
configs = check_modify_and_save_config(args, configs)
# Tensorboard summary
writer = init_summarywriter(args)
# load checkpoint
model = configs[args.model]
if args.checkpoint is not None:
model.load_state_dict(torch.load(args.checkpoint, map_location='cpu'))
# Dispatch model from cpu to gpu
model = wrap_cuda_model(args, model)
# Get optimizer & scheduler
model, optimizer, scheduler, optimizer_d, scheduler_d = init_optimizer_and_scheduler_gan(args, configs, model)
# Save init checkpoints
info_dict = deepcopy(configs['train_conf'])
save_model(model, 'init', info_dict)
# Get executor
executor = Executor()
# Start training loop
for epoch in range(info_dict['max_epoch']):
executor.epoch = epoch
train_dataset.set_epoch(epoch)
dist.barrier()
group_join = dist.new_group(backend="gloo", timeout=datetime.timedelta(seconds=args.timeout))
executor.train_one_epoc(model, optimizer, scheduler, optimizer_d, scheduler_d, train_data_loader, cv_data_loader, writer, info_dict, group_join)
dist.destroy_process_group(group_join)
if __name__ == '__main__':
main()

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@@ -85,6 +85,7 @@ def filter(data,
"""
for sample in data:
sample['speech'], sample['sample_rate'] = torchaudio.load(BytesIO(sample['audio_data']))
sample['speech'] = sample['speech'].mean(dim=0, keepdim=True)
del sample['audio_data']
# sample['wav'] is torch.Tensor, we have 100 frames every second
num_frames = sample['speech'].size(1) / sample['sample_rate'] * 100
@@ -134,6 +135,27 @@ def resample(data, resample_rate=22050, min_sample_rate=16000, mode='train'):
yield sample
def truncate(data, truncate_length=24576, mode='train'):
""" Truncate data.
Args:
data: Iterable[{key, wav, label, sample_rate}]
truncate_length: truncate length
Returns:
Iterable[{key, wav, label, sample_rate}]
"""
for sample in data:
waveform = sample['speech']
if waveform.shape[1] > truncate_length:
start = random.randint(0, waveform.shape[1] - truncate_length)
waveform = waveform[:, start: start + truncate_length]
else:
waveform = torch.concat([waveform, torch.zeros(1, truncate_length - waveform.shape[1])], dim=1)
sample['speech'] = waveform
yield sample
def compute_fbank(data,
feat_extractor,
mode='train'):
@@ -153,7 +175,26 @@ def compute_fbank(data,
waveform = sample['speech']
mat = feat_extractor(waveform).squeeze(dim=0).transpose(0, 1)
sample['speech_feat'] = mat
del sample['speech']
yield sample
def compute_f0(data, pitch_extractor, mode='train'):
""" Extract f0
Args:
data: Iterable[{key, wav, label, sample_rate}]
Returns:
Iterable[{key, feat, label}]
"""
for sample in data:
assert 'sample_rate' in sample
assert 'speech' in sample
assert 'utt' in sample
assert 'text_token' in sample
waveform = sample['speech']
mat = pitch_extractor(waveform).transpose(1, 2)
mat = F.interpolate(mat, size=sample['speech_feat'].shape[0], mode='linear')
sample['pitch_feat'] = mat[0, 0]
yield sample
@@ -325,6 +366,9 @@ def padding(data, use_spk_embedding, mode='train'):
order = torch.argsort(speech_feat_len, descending=True)
utts = [sample[i]['utt'] for i in order]
speech = [sample[i]['speech'].squeeze(dim=0) for i in order]
speech_len = torch.tensor([i.size(0) for i in speech], dtype=torch.int32)
speech = pad_sequence(speech, batch_first=True, padding_value=0)
speech_token = [torch.tensor(sample[i]['speech_token']) for i in order]
speech_token_len = torch.tensor([i.size(0) for i in speech_token], dtype=torch.int32)
speech_token = pad_sequence(speech_token,
@@ -335,6 +379,11 @@ def padding(data, use_spk_embedding, mode='train'):
speech_feat = pad_sequence(speech_feat,
batch_first=True,
padding_value=0)
pitch_feat = [sample[i]['pitch_feat'] for i in order]
pitch_feat_len = torch.tensor([i.size(0) for i in pitch_feat], dtype=torch.int32)
pitch_feat = pad_sequence(pitch_feat,
batch_first=True,
padding_value=0)
text = [sample[i]['text'] for i in order]
text_token = [torch.tensor(sample[i]['text_token']) for i in order]
text_token_len = torch.tensor([i.size(0) for i in text_token], dtype=torch.int32)
@@ -343,10 +392,14 @@ def padding(data, use_spk_embedding, mode='train'):
spk_embedding = torch.stack([sample[i]['spk_embedding'] for i in order], dim=0)
batch = {
"utts": utts,
"speech": speech,
"speech_len": speech_len,
"speech_token": speech_token,
"speech_token_len": speech_token_len,
"speech_feat": speech_feat,
"speech_feat_len": speech_feat_len,
"pitch_feat": pitch_feat,
"pitch_feat_len": pitch_feat_len,
"text": text,
"text_token": text_token,
"text_token_len": text_token_len,

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@@ -0,0 +1,139 @@
from typing import List
import torch
import torch.nn as nn
from torch.nn.utils import weight_norm
from typing import List, Optional, Tuple
from einops import rearrange
from torchaudio.transforms import Spectrogram
class MultipleDiscriminator(nn.Module):
def __init__(
self, mpd: nn.Module, mrd: nn.Module
):
super().__init__()
self.mpd = mpd
self.mrd = mrd
def forward(self, y: torch.Tensor, y_hat: torch.Tensor):
y_d_rs, y_d_gs, fmap_rs, fmap_gs = [], [], [], []
this_y_d_rs, this_y_d_gs, this_fmap_rs, this_fmap_gs = self.mpd(y.unsqueeze(dim=1), y_hat.unsqueeze(dim=1))
y_d_rs += this_y_d_rs
y_d_gs += this_y_d_gs
fmap_rs += this_fmap_rs
fmap_gs += this_fmap_gs
this_y_d_rs, this_y_d_gs, this_fmap_rs, this_fmap_gs = self.mrd(y, y_hat)
y_d_rs += this_y_d_rs
y_d_gs += this_y_d_gs
fmap_rs += this_fmap_rs
fmap_gs += this_fmap_gs
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class MultiResolutionDiscriminator(nn.Module):
def __init__(
self,
fft_sizes: Tuple[int, ...] = (2048, 1024, 512),
num_embeddings: Optional[int] = None,
):
"""
Multi-Resolution Discriminator module adapted from https://github.com/descriptinc/descript-audio-codec.
Additionally, it allows incorporating conditional information with a learned embeddings table.
Args:
fft_sizes (tuple[int]): Tuple of window lengths for FFT. Defaults to (2048, 1024, 512).
num_embeddings (int, optional): Number of embeddings. None means non-conditional discriminator.
Defaults to None.
"""
super().__init__()
self.discriminators = nn.ModuleList(
[DiscriminatorR(window_length=w, num_embeddings=num_embeddings) for w in fft_sizes]
)
def forward(
self, y: torch.Tensor, y_hat: torch.Tensor, bandwidth_id: torch.Tensor = None
) -> Tuple[List[torch.Tensor], List[torch.Tensor], List[List[torch.Tensor]], List[List[torch.Tensor]]]:
y_d_rs = []
y_d_gs = []
fmap_rs = []
fmap_gs = []
for d in self.discriminators:
y_d_r, fmap_r = d(x=y, cond_embedding_id=bandwidth_id)
y_d_g, fmap_g = d(x=y_hat, cond_embedding_id=bandwidth_id)
y_d_rs.append(y_d_r)
fmap_rs.append(fmap_r)
y_d_gs.append(y_d_g)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class DiscriminatorR(nn.Module):
def __init__(
self,
window_length: int,
num_embeddings: Optional[int] = None,
channels: int = 32,
hop_factor: float = 0.25,
bands: Tuple[Tuple[float, float], ...] = ((0.0, 0.1), (0.1, 0.25), (0.25, 0.5), (0.5, 0.75), (0.75, 1.0)),
):
super().__init__()
self.window_length = window_length
self.hop_factor = hop_factor
self.spec_fn = Spectrogram(
n_fft=window_length, hop_length=int(window_length * hop_factor), win_length=window_length, power=None
)
n_fft = window_length // 2 + 1
bands = [(int(b[0] * n_fft), int(b[1] * n_fft)) for b in bands]
self.bands = bands
convs = lambda: nn.ModuleList(
[
weight_norm(nn.Conv2d(2, channels, (3, 9), (1, 1), padding=(1, 4))),
weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
weight_norm(nn.Conv2d(channels, channels, (3, 3), (1, 1), padding=(1, 1))),
]
)
self.band_convs = nn.ModuleList([convs() for _ in range(len(self.bands))])
if num_embeddings is not None:
self.emb = torch.nn.Embedding(num_embeddings=num_embeddings, embedding_dim=channels)
torch.nn.init.zeros_(self.emb.weight)
self.conv_post = weight_norm(nn.Conv2d(channels, 1, (3, 3), (1, 1), padding=(1, 1)))
def spectrogram(self, x):
# Remove DC offset
x = x - x.mean(dim=-1, keepdims=True)
# Peak normalize the volume of input audio
x = 0.8 * x / (x.abs().max(dim=-1, keepdim=True)[0] + 1e-9)
x = self.spec_fn(x)
x = torch.view_as_real(x)
x = rearrange(x, "b f t c -> b 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: torch.Tensor, cond_embedding_id: torch.Tensor = None):
x_bands = self.spectrogram(x)
fmap = []
x = []
for band, stack in zip(x_bands, self.band_convs):
for i, layer in enumerate(stack):
band = layer(band)
band = torch.nn.functional.leaky_relu(band, 0.1)
if i > 0:
fmap.append(band)
x.append(band)
x = torch.cat(x, dim=-1)
if cond_embedding_id is not None:
emb = self.emb(cond_embedding_id)
h = (emb.view(1, -1, 1, 1) * x).sum(dim=1, keepdims=True)
else:
h = 0
x = self.conv_post(x)
fmap.append(x)
x += h
return x, fmap

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@@ -14,7 +14,7 @@
"""HIFI-GAN"""
import typing as tp
from typing import Dict, Optional, List
import numpy as np
from scipy.signal import get_window
import torch
@@ -46,7 +46,7 @@ class ResBlock(torch.nn.Module):
self,
channels: int = 512,
kernel_size: int = 3,
dilations: tp.List[int] = [1, 3, 5],
dilations: List[int] = [1, 3, 5],
):
super(ResBlock, self).__init__()
self.convs1 = nn.ModuleList()
@@ -234,13 +234,13 @@ class HiFTGenerator(nn.Module):
nsf_alpha: float = 0.1,
nsf_sigma: float = 0.003,
nsf_voiced_threshold: float = 10,
upsample_rates: tp.List[int] = [8, 8],
upsample_kernel_sizes: tp.List[int] = [16, 16],
istft_params: tp.Dict[str, int] = {"n_fft": 16, "hop_len": 4},
resblock_kernel_sizes: tp.List[int] = [3, 7, 11],
resblock_dilation_sizes: tp.List[tp.List[int]] = [[1, 3, 5], [1, 3, 5], [1, 3, 5]],
source_resblock_kernel_sizes: tp.List[int] = [7, 11],
source_resblock_dilation_sizes: tp.List[tp.List[int]] = [[1, 3, 5], [1, 3, 5]],
upsample_rates: List[int] = [8, 8],
upsample_kernel_sizes: List[int] = [16, 16],
istft_params: Dict[str, int] = {"n_fft": 16, "hop_len": 4},
resblock_kernel_sizes: List[int] = [3, 7, 11],
resblock_dilation_sizes: List[List[int]] = [[1, 3, 5], [1, 3, 5], [1, 3, 5]],
source_resblock_kernel_sizes: List[int] = [7, 11],
source_resblock_dilation_sizes: List[List[int]] = [[1, 3, 5], [1, 3, 5]],
lrelu_slope: float = 0.1,
audio_limit: float = 0.99,
f0_predictor: torch.nn.Module = None,
@@ -316,11 +316,19 @@ class HiFTGenerator(nn.Module):
self.stft_window = torch.from_numpy(get_window("hann", istft_params["n_fft"], fftbins=True).astype(np.float32))
self.f0_predictor = f0_predictor
def _f02source(self, f0: torch.Tensor) -> torch.Tensor:
f0 = self.f0_upsamp(f0[:, None]).transpose(1, 2) # bs,n,t
har_source, _, _ = self.m_source(f0)
return har_source.transpose(1, 2)
def remove_weight_norm(self):
print('Removing weight norm...')
for l in self.ups:
remove_weight_norm(l)
for l in self.resblocks:
l.remove_weight_norm()
remove_weight_norm(self.conv_pre)
remove_weight_norm(self.conv_post)
self.m_source.remove_weight_norm()
for l in self.source_downs:
remove_weight_norm(l)
for l in self.source_resblocks:
l.remove_weight_norm()
def _stft(self, x):
spec = torch.stft(
@@ -338,14 +346,7 @@ class HiFTGenerator(nn.Module):
self.istft_params["n_fft"], window=self.stft_window.to(magnitude.device))
return inverse_transform
def forward(self, x: torch.Tensor, cache_source: torch.Tensor = torch.zeros(1, 1, 0)) -> torch.Tensor:
f0 = self.f0_predictor(x)
s = self._f02source(f0)
# use cache_source to avoid glitch
if cache_source.shape[2] != 0:
s[:, :, :cache_source.shape[2]] = cache_source
def decode(self, x: torch.Tensor, s: torch.Tensor = torch.zeros(1, 1, 0)) -> torch.Tensor:
s_stft_real, s_stft_imag = self._stft(s.squeeze(1))
s_stft = torch.cat([s_stft_real, s_stft_imag], dim=1)
@@ -377,22 +378,34 @@ class HiFTGenerator(nn.Module):
x = self._istft(magnitude, phase)
x = torch.clamp(x, -self.audio_limit, self.audio_limit)
return x, s
return x
def remove_weight_norm(self):
print('Removing weight norm...')
for l in self.ups:
remove_weight_norm(l)
for l in self.resblocks:
l.remove_weight_norm()
remove_weight_norm(self.conv_pre)
remove_weight_norm(self.conv_post)
self.source_module.remove_weight_norm()
for l in self.source_downs:
remove_weight_norm(l)
for l in self.source_resblocks:
l.remove_weight_norm()
def forward(
self,
batch: dict,
device: torch.device,
) -> Dict[str, Optional[torch.Tensor]]:
speech_feat = batch['speech_feat'].transpose(1, 2).to(device)
# mel->f0
f0 = self.f0_predictor(speech_feat)
# f0->source
s = self.f0_upsamp(f0[:, None]).transpose(1, 2) # bs,n,t
s, _, _ = self.m_source(s)
s = s.transpose(1, 2)
# mel+source->speech
generated_speech = self.decode(x=speech_feat, s=s)
return generated_speech, f0
@torch.inference_mode()
def inference(self, mel: torch.Tensor, cache_source: torch.Tensor = torch.zeros(1, 1, 0)) -> torch.Tensor:
return self.forward(x=mel, cache_source=cache_source)
def inference(self, speech_feat: torch.Tensor, cache_source: torch.Tensor = torch.zeros(1, 1, 0)) -> torch.Tensor:
# mel->f0
f0 = self.f0_predictor(speech_feat)
# f0->source
s = self.f0_upsamp(f0[:, None]).transpose(1, 2) # bs,n,t
s, _, _ = self.m_source(s)
s = s.transpose(1, 2)
# use cache_source to avoid glitch
if cache_source.shape[2] != 0:
s[:, :, :cache_source.shape[2]] = cache_source
generated_speech = self.decode(x=speech_feat, s=s)
return generated_speech, s

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@@ -0,0 +1,66 @@
from typing import Dict, Optional
import torch
import torch.nn as nn
import torch.nn.functional as F
from matcha.hifigan.models import feature_loss, generator_loss, discriminator_loss
from cosyvoice.utils.losses import tpr_loss, mel_loss
class HiFiGan(nn.Module):
def __init__(self, generator, discriminator, mel_spec_transform,
multi_mel_spectral_recon_loss_weight=45, feat_match_loss_weight=2.0,
tpr_loss_weight=1.0, tpr_loss_tau=0.04):
super(HiFiGan, self).__init__()
self.generator = generator
self.discriminator = discriminator
self.mel_spec_transform = mel_spec_transform
self.multi_mel_spectral_recon_loss_weight = multi_mel_spectral_recon_loss_weight
self.feat_match_loss_weight = feat_match_loss_weight
self.tpr_loss_weight = tpr_loss_weight
self.tpr_loss_tau = tpr_loss_tau
def forward(
self,
batch: dict,
device: torch.device,
) -> Dict[str, Optional[torch.Tensor]]:
if batch['turn'] == 'generator':
return self.forward_generator(batch, device)
else:
return self.forward_discriminator(batch, device)
def forward_generator(self, batch, device):
real_speech = batch['speech'].to(device)
pitch_feat = batch['pitch_feat'].to(device)
# 1. calculate generator outputs
generated_speech, generated_f0 = self.generator(batch, device)
# 2. calculate discriminator outputs
y_d_rs, y_d_gs, fmap_rs, fmap_gs = self.discriminator(real_speech, generated_speech)
# 3. calculate generator losses, feature loss, mel loss, tpr losses [Optional]
loss_gen, _ = generator_loss(y_d_gs)
loss_fm = feature_loss(fmap_rs, fmap_gs)
loss_mel = mel_loss(real_speech, generated_speech, self.mel_spec_transform)
if self.tpr_loss_weight != 0:
loss_tpr = tpr_loss(y_d_rs, y_d_gs, self.tpr_loss_tau)
else:
loss_tpr = torch.zeros(1).to(device)
loss_f0 = F.l1_loss(generated_f0, pitch_feat)
loss = loss_gen + self.feat_match_loss_weight * loss_fm + self.multi_mel_spectral_recon_loss_weight * loss_mel + self.tpr_loss_weight * loss_tpr + loss_f0
return {'loss': loss, 'loss_gen': loss_gen, 'loss_fm': loss_fm, 'loss_mel': loss_mel, 'loss_tpr': loss_tpr, 'loss_f0': loss_f0}
def forward_discriminator(self, batch, device):
real_speech = batch['speech'].to(device)
pitch_feat = batch['pitch_feat'].to(device)
# 1. calculate generator outputs
with torch.no_grad():
generated_speech, generated_f0 = self.generator(batch, device)
# 2. calculate discriminator outputs
y_d_rs, y_d_gs, fmap_rs, fmap_gs = self.discriminator(real_speech, generated_speech)
# 3. calculate discriminator losses, tpr losses [Optional]
loss_disc, _, _ = discriminator_loss(y_d_rs, y_d_gs)
if self.tpr_loss_weight != 0:
loss_tpr = tpr_loss(y_d_rs, y_d_gs, self.tpr_loss_tau)
else:
loss_tpr = torch.zeros(1).to(device)
loss_f0 = F.l1_loss(generated_f0, pitch_feat)
loss = loss_disc + self.tpr_loss_weight * loss_tpr + loss_f0
return {'loss': loss, 'loss_disc': loss_disc, 'loss_tpr': loss_tpr, 'loss_f0': loss_f0}

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@@ -0,0 +1,118 @@
# Copyright (c) 2020 Mobvoi Inc (Binbin Zhang)
# 2024 Alibaba Inc (authors: Xiang Lyu)
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import logging
from contextlib import nullcontext
import os
import torch
import torch.distributed as dist
from cosyvoice.utils.train_utils import update_parameter_and_lr, log_per_step, log_per_save, batch_forward, batch_backward, save_model, cosyvoice_join
class Executor:
def __init__(self):
self.step = 0
self.epoch = 0
self.rank = int(os.environ.get('RANK', 0))
self.device = torch.device('cuda:{}'.format(self.rank))
def train_one_epoc(self, model, optimizer, scheduler, optimizer_d, scheduler_d, train_data_loader, cv_data_loader, writer, info_dict, group_join):
''' Train one epoch
'''
lr = optimizer.param_groups[0]['lr']
logging.info('Epoch {} TRAIN info lr {} rank {}'.format(self.epoch, lr, self.rank))
logging.info('using accumulate grad, new batch size is {} times'
' larger than before'.format(info_dict['accum_grad']))
# A context manager to be used in conjunction with an instance of
# torch.nn.parallel.DistributedDataParallel to be able to train
# with uneven inputs across participating processes.
model.train()
model_context = model.join if info_dict['train_engine'] == 'torch_ddp' else nullcontext
with model_context():
for batch_idx, batch_dict in enumerate(train_data_loader):
info_dict["tag"] = "TRAIN"
info_dict["step"] = self.step
info_dict["epoch"] = self.epoch
info_dict["batch_idx"] = batch_idx
if cosyvoice_join(group_join, info_dict):
break
# Disable gradient synchronizations across DDP processes.
# Within this context, gradients will be accumulated on module
# variables, which will later be synchronized.
if info_dict['train_engine'] == 'torch_ddp' and (batch_idx + 1) % info_dict["accum_grad"] != 0:
context = model.no_sync
# Used for single gpu training and DDP gradient synchronization
# processes.
else:
context = nullcontext
with context():
batch_dict['turn'] = 'discriminator'
info_dict = batch_forward(model, batch_dict, info_dict)
info_dict = batch_backward(model, info_dict)
info_dict = update_parameter_and_lr(model, optimizer_d, scheduler_d, info_dict)
log_per_step(writer, info_dict)
with context():
batch_dict['turn'] = 'generator'
info_dict = batch_forward(model, batch_dict, info_dict)
info_dict = batch_backward(model, info_dict)
info_dict = update_parameter_and_lr(model, optimizer, scheduler, info_dict)
log_per_step(writer, info_dict)
# NOTE specify save_per_step in cosyvoice.yaml if you want to enable step save
if info_dict['save_per_step'] > 0 and (self.step + 1) % info_dict['save_per_step'] == 0 and \
(batch_idx + 1) % info_dict["accum_grad"] == 0:
dist.barrier()
self.cv(model, cv_data_loader, writer, info_dict, on_batch_end=False)
model.train()
if (batch_idx + 1) % info_dict["accum_grad"] == 0:
self.step += 1
dist.barrier()
self.cv(model, cv_data_loader, writer, info_dict, on_batch_end=True)
@torch.inference_mode()
def cv(self, model, cv_data_loader, writer, info_dict, on_batch_end=True):
''' Cross validation on
'''
logging.info('Epoch {} Step {} on_batch_end {} CV rank {}'.format(self.epoch, self.step + 1, on_batch_end, self.rank))
model.eval()
total_num_utts, total_loss_dict = 0, {} # avoid division by 0
for batch_idx, batch_dict in enumerate(cv_data_loader):
info_dict["tag"] = "CV"
info_dict["step"] = self.step
info_dict["epoch"] = self.epoch
info_dict["batch_idx"] = batch_idx
num_utts = len(batch_dict["utts"])
total_num_utts += num_utts
batch_dict['turn'] = 'generator'
info_dict = batch_forward(model, batch_dict, info_dict)
for k, v in info_dict['loss_dict'].items():
if k not in total_loss_dict:
total_loss_dict[k] = []
total_loss_dict[k].append(v.item() * num_utts)
log_per_step(None, info_dict)
for k, v in total_loss_dict.items():
total_loss_dict[k] = sum(v) / total_num_utts
info_dict['loss_dict'] = total_loss_dict
log_per_save(writer, info_dict)
model_name = 'epoch_{}_whole'.format(self.epoch) if on_batch_end else 'epoch_{}_step_{}'.format(self.epoch, self.step + 1)
save_model(model, model_name, info_dict)

18
cosyvoice/utils/losses.py Normal file
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@@ -0,0 +1,18 @@
import torch
import torch.nn.functional as F
def tpr_loss(disc_real_outputs, disc_generated_outputs, tau):
loss = 0
for dr, dg in zip(disc_real_outputs, disc_generated_outputs):
m_DG = torch.median((dr - dg))
L_rel = torch.mean((((dr - dg) - m_DG) ** 2)[dr < dg + m_DG])
loss += tau - F.relu(tau - L_rel)
return loss
def mel_loss(real_speech, generated_speech, mel_transforms):
loss = 0
for transform in mel_transforms:
mel_r = transform(real_speech)
mel_g = transform(generated_speech)
loss += F.l1_loss(mel_g, mel_r)
return loss

View File

@@ -142,6 +142,49 @@ def init_optimizer_and_scheduler(args, configs, model):
return model, optimizer, scheduler
def init_optimizer_and_scheduler_gan(args, configs, model):
if configs['train_conf']['optim'] == 'adam':
optimizer = optim.Adam(model.module.generator.parameters(), **configs['train_conf']['optim_conf'])
elif configs['train_conf']['optim'] == 'adamw':
optimizer = optim.AdamW(model.module.generator.parameters(), **configs['train_conf']['optim_conf'])
else:
raise ValueError("unknown optimizer: " + configs['train_conf'])
if configs['train_conf']['scheduler'] == 'warmuplr':
scheduler_type = WarmupLR
scheduler = WarmupLR(optimizer, **configs['train_conf']['scheduler_conf'])
elif configs['train_conf']['scheduler'] == 'NoamHoldAnnealing':
scheduler_type = NoamHoldAnnealing
scheduler = NoamHoldAnnealing(optimizer, **configs['train_conf']['scheduler_conf'])
elif configs['train_conf']['scheduler'] == 'constantlr':
scheduler_type = ConstantLR
scheduler = ConstantLR(optimizer)
else:
raise ValueError("unknown scheduler: " + configs['train_conf'])
if configs['train_conf']['optim_d'] == 'adam':
optimizer_d = optim.Adam(model.module.discriminator.parameters(), **configs['train_conf']['optim_conf'])
elif configs['train_conf']['optim_d'] == 'adamw':
optimizer_d = optim.AdamW(model.module.discriminator.parameters(), **configs['train_conf']['optim_conf'])
else:
raise ValueError("unknown optimizer: " + configs['train_conf'])
if configs['train_conf']['scheduler_d'] == 'warmuplr':
scheduler_type = WarmupLR
scheduler_d = WarmupLR(optimizer_d, **configs['train_conf']['scheduler_conf'])
elif configs['train_conf']['scheduler_d'] == 'NoamHoldAnnealing':
scheduler_type = NoamHoldAnnealing
scheduler_d = NoamHoldAnnealing(optimizer_d, **configs['train_conf']['scheduler_conf'])
elif configs['train_conf']['scheduler'] == 'constantlr':
scheduler_type = ConstantLR
scheduler_d = ConstantLR(optimizer_d)
else:
raise ValueError("unknown scheduler: " + configs['train_conf'])
# currently we wrap generator and discriminator in one model, so we cannot use deepspeed
return model, optimizer, scheduler, optimizer_d, scheduler_d
def init_summarywriter(args):
writer = None
if int(os.environ.get('RANK', 0)) == 0:

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@@ -0,0 +1,141 @@
# set random seed, so that you may reproduce your result.
__set_seed1: !apply:random.seed [1986]
__set_seed2: !apply:numpy.random.seed [1986]
__set_seed3: !apply:torch.manual_seed [1986]
__set_seed4: !apply:torch.cuda.manual_seed_all [1986]
# fixed params
sample_rate: 22050
text_encoder_input_size: 512
llm_input_size: 1024
llm_output_size: 1024
spk_embed_dim: 192
# model params
# for all class/function included in this repo, we use !<name> or !<new> for intialization, so that user may find all corresponding class/function according to one single yaml.
# for system/third_party class/function, we do not require this.
hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
in_channels: 80
base_channels: 512
nb_harmonics: 8
sampling_rate: !ref <sample_rate>
nsf_alpha: 0.1
nsf_sigma: 0.003
nsf_voiced_threshold: 10
upsample_rates: [8, 8]
upsample_kernel_sizes: [16, 16]
istft_params:
n_fft: 16
hop_len: 4
resblock_kernel_sizes: [3, 7, 11]
resblock_dilation_sizes: [[1, 3, 5], [1, 3, 5], [1, 3, 5]]
source_resblock_kernel_sizes: [7, 11]
source_resblock_dilation_sizes: [[1, 3, 5], [1, 3, 5]]
lrelu_slope: 0.1
audio_limit: 0.99
f0_predictor: !new:cosyvoice.hifigan.f0_predictor.ConvRNNF0Predictor
num_class: 1
in_channels: 80
cond_channels: 512
mel_spec_transform1: !name:matcha.utils.audio.mel_spectrogram
n_fft: 1024
num_mels: 80
sampling_rate: !ref <sample_rate>
hop_size: 256
win_size: 1024
fmin: 0
fmax: 8000
center: False
hifigan: !new:cosyvoice.hifigan.hifigan.HiFiGan
generator: !ref <hift>
discriminator: !new:cosyvoice.hifigan.discriminator.MultipleDiscriminator
mpd: !new:matcha.hifigan.models.MultiPeriodDiscriminator
mrd: !new:cosyvoice.hifigan.discriminator.MultiResolutionDiscriminator
mel_spec_transform: [
!ref <mel_spec_transform1>
]
# processor functions
parquet_opener: !name:cosyvoice.dataset.processor.parquet_opener
get_tokenizer: !name:whisper.tokenizer.get_tokenizer # change to !name:cosyvoice.tokenizer.tokenizer.get_tokenizer if you want to train with CosyVoice-300M-25Hz recipe
multilingual: True
num_languages: 100
language: 'en'
task: 'transcribe'
tokenize: !name:cosyvoice.dataset.processor.tokenize
get_tokenizer: !ref <get_tokenizer>
allowed_special: 'all'
filter: !name:cosyvoice.dataset.processor.filter
max_length: 40960
min_length: 0
token_max_length: 200
token_min_length: 1
resample: !name:cosyvoice.dataset.processor.resample
resample_rate: !ref <sample_rate>
truncate: !name:cosyvoice.dataset.processor.truncate
truncate_length: 24576 # must be a multiplier of hop_size
feat_extractor: !name:matcha.utils.audio.mel_spectrogram
n_fft: 1024
num_mels: 80
sampling_rate: !ref <sample_rate>
hop_size: 256
win_size: 1024
fmin: 0
fmax: 8000
center: False
compute_fbank: !name:cosyvoice.dataset.processor.compute_fbank
feat_extractor: !ref <feat_extractor>
pitch_extractor: !name:torchaudio.functional.compute_kaldi_pitch
sample_rate: !ref <sample_rate>
frame_length: 46.4 # match feat_extractor win_size/sampling_rate
frame_shift: 11.6 # match feat_extractor hop_size/sampling_rate
compute_f0: !name:cosyvoice.dataset.processor.compute_f0
pitch_extractor: !ref <pitch_extractor>
parse_embedding: !name:cosyvoice.dataset.processor.parse_embedding
normalize: True
shuffle: !name:cosyvoice.dataset.processor.shuffle
shuffle_size: 1000
sort: !name:cosyvoice.dataset.processor.sort
sort_size: 500 # sort_size should be less than shuffle_size
batch: !name:cosyvoice.dataset.processor.batch
batch_type: 'dynamic'
max_frames_in_batch: 1200
padding: !name:cosyvoice.dataset.processor.padding
use_spk_embedding: False # change to True during sft
# dataset processor pipeline
data_pipeline: [
!ref <parquet_opener>,
!ref <tokenize>,
!ref <filter>,
!ref <resample>,
!ref <truncate>,
!ref <compute_fbank>,
!ref <compute_f0>,
!ref <parse_embedding>,
!ref <shuffle>,
!ref <sort>,
!ref <batch>,
!ref <padding>,
]
# train conf
train_conf:
optim: adam
optim_conf:
lr: 0.002 # change to 0.001 if you want to train flow from scratch
scheduler: warmuplr
scheduler_conf:
warmup_steps: 25000
optim_d: adam
optim_conf_d:
lr: 0.002 # change to 0.001 if you want to train flow from scratch
scheduler_d: warmuplr
scheduler_conf_d:
warmup_steps: 25000
max_epoch: 200
grad_clip: 5
accum_grad: 2
log_interval: 100
save_per_step: -1