Tsukasa_Speech / train_first.py
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import os
import os.path as osp
import re
import sys
import yaml
import shutil
import numpy as np
import torch
import click
import warnings
warnings.simplefilter('ignore')
# load packages
import random
import yaml
from munch import Munch
import numpy as np
import torch
from torch import nn
import torch.nn.functional as F
import torchaudio
import librosa
from models import *
from meldataset import build_dataloader
from utils import *
from losses import *
from optimizers import build_optimizer
import time
from accelerate import Accelerator
from accelerate.utils import LoggerType
from accelerate import DistributedDataParallelKwargs
from torch.utils.tensorboard import SummaryWriter
import logging
from accelerate.logging import get_logger
logger = get_logger(__name__, log_level="DEBUG")
@click.command()
@click.option('-p', '--config_path', default='Configs/config.yml', type=str)
def main(config_path):
config = yaml.safe_load(open(config_path))
save_iter = 10500
log_dir = config['log_dir']
if not osp.exists(log_dir): os.makedirs(log_dir, exist_ok=True)
shutil.copy(config_path, osp.join(log_dir, osp.basename(config_path)))
ddp_kwargs = DistributedDataParallelKwargs(find_unused_parameters=True)
accelerator = Accelerator(project_dir=log_dir, split_batches=True, kwargs_handlers=[ddp_kwargs], mixed_precision='bf16')
if accelerator.is_main_process:
writer = SummaryWriter(log_dir + "/tensorboard")
# write logs
file_handler = logging.FileHandler(osp.join(log_dir, 'train.log'))
file_handler.setLevel(logging.DEBUG)
file_handler.setFormatter(logging.Formatter('%(levelname)s:%(asctime)s: %(message)s'))
logger.logger.addHandler(file_handler)
batch_size = config.get('batch_size', 10)
device = accelerator.device
epochs = config.get('epochs_1st', 200)
save_freq = config.get('save_freq', 2)
log_interval = config.get('log_interval', 10)
saving_epoch = config.get('save_freq', 2)
data_params = config.get('data_params', None)
sr = config['preprocess_params'].get('sr', 24000)
train_path = data_params['train_data']
val_path = data_params['val_data']
root_path = data_params['root_path']
min_length = data_params['min_length']
OOD_data = data_params['OOD_data']
max_len = config.get('max_len', 200)
# load data
train_list, val_list = get_data_path_list(train_path, val_path)
train_dataloader = build_dataloader(train_list,
root_path,
OOD_data=OOD_data,
min_length=min_length,
batch_size=batch_size,
num_workers=2,
dataset_config={},
device=device)
val_dataloader = build_dataloader(val_list,
root_path,
OOD_data=OOD_data,
min_length=min_length,
batch_size=batch_size,
validation=True,
num_workers=0,
device=device,
dataset_config={})
with accelerator.main_process_first():
# load pretrained ASR model
ASR_config = config.get('ASR_config', False)
ASR_path = config.get('ASR_path', False)
text_aligner = load_ASR_models(ASR_path, ASR_config)
# load pretrained F0 model
F0_path = config.get('F0_path', False)
pitch_extractor = load_F0_models(F0_path)
# load BERT model
from Utils.PLBERT.util import load_plbert
BERT_path = config.get('PLBERT_dir', False)
plbert = load_plbert(BERT_path)
scheduler_params = {
"max_lr": float(config['optimizer_params'].get('lr', 1e-4)),
"pct_start": float(config['optimizer_params'].get('pct_start', 0.0)),
"epochs": epochs,
"steps_per_epoch": len(train_dataloader),
}
model_params = recursive_munch(config['model_params'])
multispeaker = model_params.multispeaker
model = build_model(model_params, text_aligner, pitch_extractor, plbert)
best_loss = float('inf') # best test loss
loss_train_record = list([])
loss_test_record = list([])
loss_params = Munch(config['loss_params'])
TMA_epoch = loss_params.TMA_epoch
for k in model:
model[k] = accelerator.prepare(model[k])
train_dataloader, val_dataloader = accelerator.prepare(
train_dataloader, val_dataloader
)
_ = [model[key].to(device) for key in model]
# initialize optimizers after preparing models for compatibility with FSDP
optimizer = build_optimizer({key: model[key].parameters() for key in model},
scheduler_params_dict= {key: scheduler_params.copy() for key in model},
lr=float(config['optimizer_params'].get('lr', 1e-4)))
for k, v in optimizer.optimizers.items():
optimizer.optimizers[k] = accelerator.prepare(optimizer.optimizers[k])
optimizer.schedulers[k] = accelerator.prepare(optimizer.schedulers[k])
with accelerator.main_process_first():
if config.get('pretrained_model', '') != '':
model, optimizer, start_epoch, iters = load_checkpoint(model, optimizer, config['pretrained_model'],
load_only_params=config.get('load_only_params', True))
else:
start_epoch = 0
iters = 0
# in case not distributed
try:
n_down = model.text_aligner.module.n_down
except:
n_down = model.text_aligner.n_down
# wrapped losses for compatibility with mixed precision
stft_loss = MultiResolutionSTFTLoss().to(device)
gl = GeneratorLoss(model.mpd, model.msd).to(device)
dl = DiscriminatorLoss(model.mpd, model.msd).to(device)
wl = WavLMLoss(model_params.slm.model,
model.wd,
sr,
model_params.slm.sr).to(device)
for epoch in range(start_epoch, epochs):
running_loss = 0
start_time = time.time()
_ = [model[key].train() for key in model]
for i, batch in enumerate(train_dataloader):
waves = batch[0]
batch = [b.to(device) for b in batch[1:]]
texts, input_lengths, _, _, mels, mel_input_length, _ = batch
with torch.no_grad():
mask = length_to_mask(mel_input_length // (2 ** n_down)).to('cuda')
text_mask = length_to_mask(input_lengths).to(texts.device)
ppgs, s2s_pred, s2s_attn = model.text_aligner(mels, mask, texts)
s2s_attn = s2s_attn.transpose(-1, -2)
s2s_attn = s2s_attn[..., 1:]
s2s_attn = s2s_attn.transpose(-1, -2)
with torch.no_grad():
attn_mask = (~mask).unsqueeze(-1).expand(mask.shape[0], mask.shape[1], text_mask.shape[-1]).float().transpose(-1, -2)
attn_mask = attn_mask.float() * (~text_mask).unsqueeze(-1).expand(text_mask.shape[0], text_mask.shape[1], mask.shape[-1]).float()
attn_mask = (attn_mask < 1)
s2s_attn.masked_fill_(attn_mask, 0.0)
with torch.no_grad():
mask_ST = mask_from_lens(s2s_attn, input_lengths, mel_input_length // (2 ** n_down))
s2s_attn_mono = maximum_path(s2s_attn, mask_ST)
# encode
t_en = model.text_encoder(texts, input_lengths, text_mask)
# 50% of chance of using monotonic version
if bool(random.getrandbits(1)):
asr = (t_en @ s2s_attn)
else:
asr = (t_en @ s2s_attn_mono)
# get clips
mel_input_length_all = accelerator.gather(mel_input_length) # for balanced load
mel_len = min([int(mel_input_length_all.min().item() / 2 - 1), max_len // 2])
mel_len_st = int(mel_input_length.min().item() / 2 - 1)
en = []
gt = []
wav = []
st = []
for bib in range(len(mel_input_length)):
mel_length = int(mel_input_length[bib].item() / 2)
random_start = np.random.randint(0, mel_length - mel_len)
en.append(asr[bib, :, random_start:random_start+mel_len])
gt.append(mels[bib, :, (random_start * 2):((random_start+mel_len) * 2)])
y = waves[bib][(random_start * 2) * 300:((random_start+mel_len) * 2) * 300]
wav.append(torch.from_numpy(y).to(device))
# style reference (better to be different from the GT)
random_start = np.random.randint(0, mel_length - mel_len_st)
st.append(mels[bib, :, (random_start * 2):((random_start+mel_len_st) * 2)])
en = torch.stack(en)
gt = torch.stack(gt).detach()
st = torch.stack(st).detach()
wav = torch.stack(wav).float().detach()
# clip too short to be used by the style encoder
if gt.shape[-1] < 80:
continue
with torch.no_grad():
real_norm = log_norm(gt.unsqueeze(1)).squeeze(1).detach()
F0_real, _, _ = model.pitch_extractor(gt.unsqueeze(1))
s = model.style_encoder(st.unsqueeze(1) if multispeaker else gt.unsqueeze(1))
y_rec = model.decoder(en, F0_real, real_norm, s)
# discriminator loss
if epoch >= TMA_epoch:
optimizer.zero_grad()
d_loss = dl(wav.detach().unsqueeze(1).float(), y_rec.detach()).mean()
accelerator.backward(d_loss)
optimizer.step('msd')
optimizer.step('mpd')
else:
d_loss = 0
# generator loss
optimizer.zero_grad()
loss_mel = stft_loss(y_rec.squeeze(), wav.detach())
if epoch >= TMA_epoch: # start TMA training
loss_s2s = 0
for _s2s_pred, _text_input, _text_length in zip(s2s_pred, texts, input_lengths):
loss_s2s += F.cross_entropy(_s2s_pred[:_text_length], _text_input[:_text_length])
loss_s2s /= texts.size(0)
loss_mono = F.l1_loss(s2s_attn, s2s_attn_mono) * 10
loss_gen_all = gl(wav.detach().unsqueeze(1).float(), y_rec).mean()
loss_slm = wl(wav.detach(), y_rec).mean()
g_loss = loss_params.lambda_mel * loss_mel + \
loss_params.lambda_mono * loss_mono + \
loss_params.lambda_s2s * loss_s2s + \
loss_params.lambda_gen * loss_gen_all + \
loss_params.lambda_slm * loss_slm
else:
loss_s2s = 0
loss_mono = 0
loss_gen_all = 0
loss_slm = 0
g_loss = loss_mel
running_loss += accelerator.gather(loss_mel).mean().item()
accelerator.backward(g_loss)
optimizer.step('text_encoder')
optimizer.step('style_encoder')
optimizer.step('decoder')
if epoch >= TMA_epoch:
optimizer.step('text_aligner')
optimizer.step('pitch_extractor')
iters = iters + 1
if (i+1)%log_interval == 0 and accelerator.is_main_process:
log_print ('Epoch [%d/%d], Step [%d/%d], Mel Loss: %.5f, Gen Loss: %.5f, Disc Loss: %.5f, Mono Loss: %.5f, S2S Loss: %.5f, SLM Loss: %.5f'
%(epoch+1, epochs, i+1, len(train_list)//batch_size, running_loss / log_interval, loss_gen_all, d_loss, loss_mono, loss_s2s, loss_slm), logger)
writer.add_scalar('train/mel_loss', running_loss / log_interval, iters)
writer.add_scalar('train/gen_loss', loss_gen_all, iters)
writer.add_scalar('train/d_loss', d_loss, iters)
writer.add_scalar('train/mono_loss', loss_mono, iters)
writer.add_scalar('train/s2s_loss', loss_s2s, iters)
writer.add_scalar('train/slm_loss', loss_slm, iters)
running_loss = 0
print('Time elasped:', time.time()-start_time)
if (i+1)%save_iter == 0 and accelerator.is_main_process:
print(f'Saving on step {epoch*len(train_dataloader)+i}...')
state = {
'net': {key: model[key].state_dict() for key in model},
'optimizer': optimizer.state_dict(),
'iters': iters,
'epoch': epoch,
}
save_path = osp.join(log_dir, f'2nd_phase_{epoch*len(train_dataloader)+i}.pth')
torch.save(state, save_path)
loss_test = 0
_ = [model[key].eval() for key in model]
with torch.no_grad():
iters_test = 0
for batch_idx, batch in enumerate(val_dataloader):
optimizer.zero_grad()
waves = batch[0]
batch = [b.to(device) for b in batch[1:]]
texts, input_lengths, _, _, mels, mel_input_length, _ = batch
with torch.no_grad():
mask = length_to_mask(mel_input_length // (2 ** n_down)).to('cuda')
ppgs, s2s_pred, s2s_attn = model.text_aligner(mels, mask, texts)
s2s_attn = s2s_attn.transpose(-1, -2)
s2s_attn = s2s_attn[..., 1:]
s2s_attn = s2s_attn.transpose(-1, -2)
text_mask = length_to_mask(input_lengths).to(texts.device)
attn_mask = (~mask).unsqueeze(-1).expand(mask.shape[0], mask.shape[1], text_mask.shape[-1]).float().transpose(-1, -2)
attn_mask = attn_mask.float() * (~text_mask).unsqueeze(-1).expand(text_mask.shape[0], text_mask.shape[1], mask.shape[-1]).float()
attn_mask = (attn_mask < 1)
s2s_attn.masked_fill_(attn_mask, 0.0)
# encode
t_en = model.text_encoder(texts, input_lengths, text_mask)
asr = (t_en @ s2s_attn)
# get clips
mel_input_length_all = accelerator.gather(mel_input_length) # for balanced load
mel_len = min([int(mel_input_length.min().item() / 2 - 1), max_len // 2])
en = []
gt = []
wav = []
for bib in range(len(mel_input_length)):
mel_length = int(mel_input_length[bib].item() / 2)
random_start = np.random.randint(0, mel_length - mel_len)
en.append(asr[bib, :, random_start:random_start+mel_len])
gt.append(mels[bib, :, (random_start * 2):((random_start+mel_len) * 2)])
y = waves[bib][(random_start * 2) * 300:((random_start+mel_len) * 2) * 300]
wav.append(torch.from_numpy(y).to('cuda'))
wav = torch.stack(wav).float().detach()
en = torch.stack(en)
gt = torch.stack(gt).detach()
F0_real, _, F0 = model.pitch_extractor(gt.unsqueeze(1))
s = model.style_encoder(gt.unsqueeze(1))
real_norm = log_norm(gt.unsqueeze(1)).squeeze(1)
y_rec = model.decoder(en, F0_real, real_norm, s)
loss_mel = stft_loss(y_rec.squeeze(), wav.detach())
loss_test += accelerator.gather(loss_mel).mean().item()
iters_test += 1
if accelerator.is_main_process:
print('Epochs:', epoch + 1)
log_print('Validation loss: %.3f' % (loss_test / iters_test) + '\n\n\n\n', logger)
print('\n\n\n')
writer.add_scalar('eval/mel_loss', loss_test / iters_test, epoch + 1)
attn_image = get_image(s2s_attn[0].cpu().numpy().squeeze())
writer.add_figure('eval/attn', attn_image, epoch)
with torch.no_grad():
for bib in range(len(asr)):
mel_length = int(mel_input_length[bib].item())
gt = mels[bib, :, :mel_length].unsqueeze(0)
en = asr[bib, :, :mel_length // 2].unsqueeze(0)
F0_real, _, _ = model.pitch_extractor(gt.unsqueeze(1))
F0_real = F0_real.unsqueeze(0)
s = model.style_encoder(gt.unsqueeze(1))
real_norm = log_norm(gt.unsqueeze(1)).squeeze(1)
y_rec = model.decoder(en, F0_real, real_norm, s)
writer.add_audio('eval/y' + str(bib), y_rec.cpu().numpy().squeeze(), epoch, sample_rate=sr)
if epoch == 0:
writer.add_audio('gt/y' + str(bib), waves[bib].squeeze(), epoch, sample_rate=sr)
if bib >= 15:
break
if epoch % saving_epoch == 0:
if (loss_test / iters_test) < best_loss:
best_loss = loss_test / iters_test
print('Saving..')
state = {
'net': {key: model[key].state_dict() for key in model},
'optimizer': optimizer.state_dict(),
'iters': iters,
'val_loss': loss_test / iters_test,
'epoch': epoch,
}
save_path = osp.join(log_dir, 'epoch_1st_%05d.pth' % epoch)
torch.save(state, save_path)
if accelerator.is_main_process:
print('Saving..')
state = {
'net': {key: model[key].state_dict() for key in model},
'optimizer': optimizer.state_dict(),
'iters': iters,
'val_loss': loss_test / iters_test,
'epoch': epoch,
}
save_path = osp.join(log_dir, config.get('first_stage_path', 'first_stage.pth'))
torch.save(state, save_path)
if __name__=="__main__":
main()