Merge pull request #494 from FunAudioLLM/dev/lyuxiang.lx

Dev/lyuxiang.lx
This commit is contained in:
Xiang Lyu
2024-10-16 13:07:25 +08:00
committed by GitHub
17 changed files with 711 additions and 82 deletions

View File

@@ -52,5 +52,5 @@ jobs:
set -eux
pip install flake8==3.8.2 flake8-bugbear flake8-comprehensions flake8-executable flake8-pyi==20.5.0 mccabe pycodestyle==2.6.0 pyflakes==2.2.0
flake8 --version
flake8 --max-line-length 150 --ignore B006,B008,B905,C408,E402,E741,W503,W504 --exclude ./third_party/,./runtime/python/grpc/cosyvoice_pb2*py
flake8 --max-line-length 150 --ignore B006,B008,B905,C408,E402,E731,E741,W503,W504 --exclude ./third_party/,./runtime/python/grpc/cosyvoice_pb2*py
if [ $? != 0 ]; then exit 1; fi

View File

@@ -26,9 +26,7 @@ For `SenseVoice`, visit [SenseVoice repo](https://github.com/FunAudioLLM/SenseVo
- [ ] 25hz llama based llm model which supports lora finetune
- [ ] Support more instruction mode
- [ ] Voice conversion
- [ ] Music generation
- [ ] Training script sample based on Mandarin
- [ ] CosyVoice-500M trained with more multi-lingual data
- [ ] More...
@@ -113,7 +111,7 @@ from cosyvoice.cli.cosyvoice import CosyVoice
from cosyvoice.utils.file_utils import load_wav
import torchaudio
cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M-SFT')
cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M-SFT', load_jit=True, load_onnx=False, fp16=True)
# sft usage
print(cosyvoice.list_avaliable_spks())
# change stream=True for chunk stream inference

View File

@@ -0,0 +1,92 @@
# Copyright (c) 2020 Mobvoi Inc (Di Wu)
# 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.
import os
import argparse
import glob
import yaml
import torch
def get_args():
parser = argparse.ArgumentParser(description='average model')
parser.add_argument('--dst_model', required=True, help='averaged model')
parser.add_argument('--src_path',
required=True,
help='src model path for average')
parser.add_argument('--val_best',
action="store_true",
help='averaged model')
parser.add_argument('--num',
default=5,
type=int,
help='nums for averaged model')
args = parser.parse_args()
print(args)
return args
def main():
args = get_args()
val_scores = []
if args.val_best:
yamls = glob.glob('{}/*.yaml'.format(args.src_path))
yamls = [
f for f in yamls
if not (os.path.basename(f).startswith('train')
or os.path.basename(f).startswith('init'))
]
for y in yamls:
with open(y, 'r') as f:
dic_yaml = yaml.load(f, Loader=yaml.BaseLoader)
loss = float(dic_yaml['loss_dict']['loss'])
epoch = int(dic_yaml['epoch'])
step = int(dic_yaml['step'])
tag = dic_yaml['tag']
val_scores += [[epoch, step, loss, tag]]
sorted_val_scores = sorted(val_scores,
key=lambda x: x[2],
reverse=False)
print("best val (epoch, step, loss, tag) = " +
str(sorted_val_scores[:args.num]))
path_list = [
args.src_path + '/epoch_{}_whole.pt'.format(score[0])
for score in sorted_val_scores[:args.num]
]
print(path_list)
avg = {}
num = args.num
assert num == len(path_list)
for path in path_list:
print('Processing {}'.format(path))
states = torch.load(path, map_location=torch.device('cpu'))
for k in states.keys():
if k not in avg.keys():
avg[k] = states[k].clone()
else:
avg[k] += states[k]
# average
for k in avg.keys():
if avg[k] is not None:
# pytorch 1.6 use true_divide instead of /=
avg[k] = torch.true_divide(avg[k], num)
print('Saving to {}'.format(args.dst_model))
torch.save(avg, args.dst_model)
if __name__ == '__main__':
main()

View File

@@ -18,6 +18,7 @@ import datetime
import logging
logging.getLogger('matplotlib').setLevel(logging.WARNING)
from copy import deepcopy
import os
import torch
import torch.distributed as dist
import deepspeed
@@ -73,7 +74,7 @@ def get_args():
choices=['model_only', 'model+optimizer'],
help='save model/optimizer states')
parser.add_argument('--timeout',
default=30,
default=60,
type=int,
help='timeout (in seconds) of cosyvoice_join.')
parser = deepspeed.add_config_arguments(parser)
@@ -86,8 +87,12 @@ def main():
args = get_args()
logging.basicConfig(level=logging.DEBUG,
format='%(asctime)s %(levelname)s %(message)s')
# gan train has some special initialization logic
gan = True if args.model == 'hifigan' else False
override_dict = {k: None for k in ['llm', 'flow', 'hift'] if k != args.model}
override_dict = {k: None for k in ['llm', 'flow', 'hift', 'hifigan'] if k != args.model}
if gan is True:
override_dict.pop('hift')
with open(args.config, 'r') as f:
configs = load_hyperpyyaml(f, overrides=override_dict)
configs['train_conf'].update(vars(args))
@@ -97,7 +102,7 @@ def main():
# Get dataset & dataloader
train_dataset, cv_dataset, train_data_loader, cv_data_loader = \
init_dataset_and_dataloader(args, configs)
init_dataset_and_dataloader(args, configs, gan)
# Do some sanity checks and save config to arsg.model_dir
configs = check_modify_and_save_config(args, configs)
@@ -108,20 +113,23 @@ def main():
# load checkpoint
model = configs[args.model]
if args.checkpoint is not None:
model.load_state_dict(torch.load(args.checkpoint, map_location='cpu'))
if os.path.exists(args.checkpoint):
model.load_state_dict(torch.load(args.checkpoint, map_location='cpu'), strict=False)
else:
logging.warning('checkpoint {} do not exsist!'.format(args.checkpoint))
# Dispatch model from cpu to gpu
model = wrap_cuda_model(args, model)
# Get optimizer & scheduler
model, optimizer, scheduler = init_optimizer_and_scheduler(args, configs, model)
model, optimizer, scheduler, optimizer_d, scheduler_d = init_optimizer_and_scheduler(args, configs, model, gan)
# Save init checkpoints
info_dict = deepcopy(configs['train_conf'])
save_model(model, 'init', info_dict)
# Get executor
executor = Executor()
executor = Executor(gan=gan)
# Start training loop
for epoch in range(info_dict['max_epoch']):
@@ -129,7 +137,11 @@ def main():
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, train_data_loader, cv_data_loader, writer, info_dict, group_join)
if gan is True:
executor.train_one_epoc_gan(model, optimizer, scheduler, optimizer_d, scheduler_d, train_data_loader, cv_data_loader,
writer, info_dict, group_join)
else:
executor.train_one_epoc(model, optimizer, scheduler, train_data_loader, cv_data_loader, writer, info_dict, group_join)
dist.destroy_process_group(group_join)

View File

@@ -23,7 +23,7 @@ from cosyvoice.utils.file_utils import logging
class CosyVoice:
def __init__(self, model_dir, load_jit=True, load_onnx=False):
def __init__(self, model_dir, load_jit=True, load_onnx=False, fp16=True):
instruct = True if '-Instruct' in model_dir else False
self.model_dir = model_dir
if not os.path.exists(model_dir):
@@ -37,7 +37,7 @@ class CosyVoice:
'{}/spk2info.pt'.format(model_dir),
instruct,
configs['allowed_special'])
self.model = CosyVoiceModel(configs['llm'], configs['flow'], configs['hift'])
self.model = CosyVoiceModel(configs['llm'], configs['flow'], configs['hift'], fp16)
self.model.load('{}/llm.pt'.format(model_dir),
'{}/flow.pt'.format(model_dir),
'{}/hift.pt'.format(model_dir))

View File

@@ -26,11 +26,13 @@ class CosyVoiceModel:
def __init__(self,
llm: torch.nn.Module,
flow: torch.nn.Module,
hift: torch.nn.Module):
hift: torch.nn.Module,
fp16: bool):
self.device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
self.llm = llm
self.flow = flow
self.hift = hift
self.fp16 = fp16
self.token_min_hop_len = 2 * self.flow.input_frame_rate
self.token_max_hop_len = 4 * self.flow.input_frame_rate
self.token_overlap_len = 20
@@ -56,13 +58,17 @@ class CosyVoiceModel:
def load(self, llm_model, flow_model, hift_model):
self.llm.load_state_dict(torch.load(llm_model, map_location=self.device))
self.llm.to(self.device).eval()
self.llm.half()
if self.fp16 is True:
self.llm.half()
self.flow.load_state_dict(torch.load(flow_model, map_location=self.device))
self.flow.to(self.device).eval()
self.hift.load_state_dict(torch.load(hift_model, map_location=self.device))
# in case hift_model is a hifigan model
hift_state_dict = {k.replace('generator.', ''): v for k, v in torch.load(hift_model, map_location=self.device)}
self.hift.load_state_dict(hift_state_dict, strict=False)
self.hift.to(self.device).eval()
def load_jit(self, llm_text_encoder_model, llm_llm_model, flow_encoder_model):
assert self.fp16 is True, "we only provide fp16 jit model, set fp16=True if you want to use jit model"
llm_text_encoder = torch.jit.load(llm_text_encoder_model, map_location=self.device)
self.llm.text_encoder = llm_text_encoder
llm_llm = torch.jit.load(llm_llm_model, map_location=self.device)
@@ -80,6 +86,8 @@ class CosyVoiceModel:
self.flow.decoder.estimator = onnxruntime.InferenceSession(flow_decoder_estimator_model, sess_options=option, providers=providers)
def llm_job(self, text, prompt_text, llm_prompt_speech_token, llm_embedding, uuid):
if self.fp16 is True:
llm_embedding = llm_embedding.half()
with self.llm_context:
for i in self.llm.inference(text=text.to(self.device),
text_len=torch.tensor([text.shape[1]], dtype=torch.int32).to(self.device),
@@ -87,7 +95,7 @@ class CosyVoiceModel:
prompt_text_len=torch.tensor([prompt_text.shape[1]], dtype=torch.int32).to(self.device),
prompt_speech_token=llm_prompt_speech_token.to(self.device),
prompt_speech_token_len=torch.tensor([llm_prompt_speech_token.shape[1]], dtype=torch.int32).to(self.device),
embedding=llm_embedding.to(self.device).half()):
embedding=llm_embedding.to(self.device)):
self.tts_speech_token_dict[uuid].append(i)
self.llm_end_dict[uuid] = True
@@ -123,7 +131,7 @@ class CosyVoiceModel:
if speed != 1.0:
assert self.hift_cache_dict[uuid] is None, 'speed change only support non-stream inference mode'
tts_mel = F.interpolate(tts_mel, size=int(tts_mel.shape[2] / speed), mode='linear')
tts_speech, tts_source = self.hift.inference(mel=tts_mel, cache_source=hift_cache_source)
tts_speech, tts_source = self.hift.inference(speech_feat=tts_mel, cache_source=hift_cache_source)
if self.hift_cache_dict[uuid] is not None:
tts_speech = fade_in_out(tts_speech, self.hift_cache_dict[uuid]['speech'], self.speech_window)
return tts_speech

View File

@@ -126,6 +126,7 @@ class DataList(IterableDataset):
def Dataset(data_list_file,
data_pipeline,
mode='train',
gan=False,
shuffle=True,
partition=True,
tts_file='',
@@ -153,8 +154,11 @@ def Dataset(data_list_file,
shuffle=shuffle,
partition=partition)
if mode == 'inference':
# map partial arg tts_data in inference mode
# map partial arg to parquet_opener func in inference mode
data_pipeline[0] = partial(data_pipeline[0], tts_data=tts_data)
if gan is True:
# map partial arg to padding func in gan mode
data_pipeline[-1] = partial(data_pipeline[-1], gan=gan)
for func in data_pipeline:
dataset = Processor(dataset, func, mode=mode)
return dataset

View File

@@ -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 +175,27 @@ 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
@@ -309,7 +351,7 @@ def batch(data, batch_type='static', batch_size=16, max_frames_in_batch=12000, m
logging.fatal('Unsupported batch type {}'.format(batch_type))
def padding(data, use_spk_embedding, mode='train'):
def padding(data, use_spk_embedding, mode='train', gan=False):
""" Padding the data into training data
Args:
@@ -325,6 +367,9 @@ 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,
@@ -343,6 +388,8 @@ 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,
@@ -353,6 +400,19 @@ def padding(data, use_spk_embedding, mode='train'):
"utt_embedding": utt_embedding,
"spk_embedding": spk_embedding,
}
if gan is True:
# in gan train, we need pitch_feat
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)
batch["pitch_feat"] = pitch_feat
batch["pitch_feat_len"] = pitch_feat_len
else:
# only gan train needs speech, delete it to save memory
del batch["speech"]
del batch["speech_len"]
if mode == 'inference':
tts_text = [sample[i]['tts_text'] for i in order]
tts_index = [sample[i]['tts_index'] for i in order]

View File

@@ -0,0 +1,140 @@
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

View File

@@ -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

View File

@@ -0,0 +1,69 @@
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}

View File

@@ -25,7 +25,8 @@ from cosyvoice.utils.train_utils import update_parameter_and_lr, log_per_step, l
class Executor:
def __init__(self):
def __init__(self, gan: bool = False):
self.gan = gan
self.step = 0
self.epoch = 0
self.rank = int(os.environ.get('RANK', 0))
@@ -80,6 +81,64 @@ class Executor:
dist.barrier()
self.cv(model, cv_data_loader, writer, info_dict, on_batch_end=True)
def train_one_epoc_gan(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)
optimizer.zero_grad()
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)
optimizer_d.zero_grad()
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
@@ -96,6 +155,8 @@ class Executor:
num_utts = len(batch_dict["utts"])
total_num_utts += num_utts
if self.gan is True:
batch_dict['turn'] = 'generator'
info_dict = batch_forward(model, batch_dict, info_dict)
for k, v in info_dict['loss_dict'].items():

20
cosyvoice/utils/losses.py Normal file
View File

@@ -0,0 +1,20 @@
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

@@ -51,9 +51,10 @@ def init_distributed(args):
return world_size, local_rank, rank
def init_dataset_and_dataloader(args, configs):
train_dataset = Dataset(args.train_data, data_pipeline=configs['data_pipeline'], mode='train', shuffle=True, partition=True)
cv_dataset = Dataset(args.cv_data, data_pipeline=configs['data_pipeline'], mode='train', shuffle=False, partition=False)
def init_dataset_and_dataloader(args, configs, gan):
data_pipeline = configs['data_pipeline_gan'] if gan is True else configs['data_pipeline']
train_dataset = Dataset(args.train_data, data_pipeline=data_pipeline, mode='train', gan=gan, shuffle=True, partition=True)
cv_dataset = Dataset(args.cv_data, data_pipeline=data_pipeline, mode='train', gan=gan, shuffle=False, partition=False)
# do not use persistent_workers=True, as whisper tokenizer opens tiktoken file each time when the for loop starts
train_data_loader = DataLoader(train_dataset,
@@ -108,30 +109,31 @@ def wrap_cuda_model(args, model):
return model
def init_optimizer_and_scheduler(args, configs, model):
if configs['train_conf']['optim'] == 'adam':
optimizer = optim.Adam(model.parameters(), **configs['train_conf']['optim_conf'])
elif configs['train_conf']['optim'] == 'adamw':
optimizer = optim.AdamW(model.parameters(), **configs['train_conf']['optim_conf'])
def init_optimizer_and_scheduler(args, configs, model, gan):
key = 'train_conf_gan' if gan is True else 'train_conf'
if configs[key]['optim'] == 'adam':
optimizer = optim.Adam(model.parameters(), **configs[key]['optim_conf'])
elif configs[key]['optim'] == 'adamw':
optimizer = optim.AdamW(model.parameters(), **configs[key]['optim_conf'])
else:
raise ValueError("unknown optimizer: " + configs['train_conf'])
raise ValueError("unknown optimizer: " + configs[key])
if configs['train_conf']['scheduler'] == 'warmuplr':
if configs[key]['scheduler'] == 'warmuplr':
scheduler_type = WarmupLR
scheduler = WarmupLR(optimizer, **configs['train_conf']['scheduler_conf'])
elif configs['train_conf']['scheduler'] == 'NoamHoldAnnealing':
scheduler = WarmupLR(optimizer, **configs[key]['scheduler_conf'])
elif configs[key]['scheduler'] == 'NoamHoldAnnealing':
scheduler_type = NoamHoldAnnealing
scheduler = NoamHoldAnnealing(optimizer, **configs['train_conf']['scheduler_conf'])
elif configs['train_conf']['scheduler'] == 'constantlr':
scheduler = NoamHoldAnnealing(optimizer, **configs[key]['scheduler_conf'])
elif configs[key]['scheduler'] == 'constantlr':
scheduler_type = ConstantLR
scheduler = ConstantLR(optimizer)
else:
raise ValueError("unknown scheduler: " + configs['train_conf'])
raise ValueError("unknown scheduler: " + configs[key])
# use deepspeed optimizer for speedup
if args.train_engine == "deepspeed":
def scheduler(opt):
return scheduler_type(opt, **configs['train_conf']['scheduler_conf'])
return scheduler_type(opt, **configs[key]['scheduler_conf'])
model, optimizer, _, scheduler = deepspeed.initialize(
args=args,
model=model,
@@ -139,7 +141,29 @@ def init_optimizer_and_scheduler(args, configs, model):
lr_scheduler=scheduler,
model_parameters=model.parameters())
return model, optimizer, scheduler
# currently we wrap generator and discriminator in one model, so we cannot use deepspeed
if gan is True:
if configs[key]['optim_d'] == 'adam':
optimizer_d = optim.Adam(model.module.discriminator.parameters(), **configs[key]['optim_conf'])
elif configs[key]['optim_d'] == 'adamw':
optimizer_d = optim.AdamW(model.module.discriminator.parameters(), **configs[key]['optim_conf'])
else:
raise ValueError("unknown optimizer: " + configs[key])
if configs[key]['scheduler_d'] == 'warmuplr':
scheduler_type = WarmupLR
scheduler_d = WarmupLR(optimizer_d, **configs[key]['scheduler_conf'])
elif configs[key]['scheduler_d'] == 'NoamHoldAnnealing':
scheduler_type = NoamHoldAnnealing
scheduler_d = NoamHoldAnnealing(optimizer_d, **configs[key]['scheduler_conf'])
elif configs[key]['scheduler'] == 'constantlr':
scheduler_type = ConstantLR
scheduler_d = ConstantLR(optimizer_d)
else:
raise ValueError("unknown scheduler: " + configs[key])
else:
optimizer_d, scheduler_d = None, None
return model, optimizer, scheduler, optimizer_d, scheduler_d
def init_summarywriter(args):

View File

@@ -133,6 +133,25 @@ hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
in_channels: 80
cond_channels: 512
# gan related module
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
@@ -151,6 +170,8 @@ filter: !name:cosyvoice.dataset.processor.filter
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
@@ -162,6 +183,12 @@ feat_extractor: !name:matcha.utils.audio.mel_spectrogram
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
@@ -187,8 +214,22 @@ data_pipeline: [
!ref <batch>,
!ref <padding>,
]
data_pipeline_gan: [
!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
# llm flow train conf
train_conf:
optim: adam
optim_conf:
@@ -200,4 +241,20 @@ train_conf:
grad_clip: 5
accum_grad: 2
log_interval: 100
save_per_step: -1
# gan train conf
train_conf_gan:
optim: adam
optim_conf:
lr: 0.0002 # use small lr for gan training
scheduler: constantlr
optim_d: adam
optim_conf_d:
lr: 0.0002 # use small lr for gan training
scheduler_d: constantlr
max_epoch: 200
grad_clip: 5
accum_grad: 1 # in gan training, accum_grad must be 1
log_interval: 100
save_per_step: -1

View File

@@ -133,6 +133,25 @@ hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
in_channels: 80
cond_channels: 512
# gan related module
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
@@ -151,6 +170,8 @@ filter: !name:cosyvoice.dataset.processor.filter
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
@@ -162,6 +183,12 @@ feat_extractor: !name:matcha.utils.audio.mel_spectrogram
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
@@ -170,7 +197,7 @@ 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: 2000
max_frames_in_batch: 2000 # change to 1400 in gan train on v100 16g
padding: !name:cosyvoice.dataset.processor.padding
use_spk_embedding: False # change to True during sft
@@ -187,8 +214,22 @@ data_pipeline: [
!ref <batch>,
!ref <padding>,
]
data_pipeline_gan: [
!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
# llm flow train conf
train_conf:
optim: adam
optim_conf:
@@ -200,4 +241,20 @@ train_conf:
grad_clip: 5
accum_grad: 2
log_interval: 100
save_per_step: -1
# gan train conf
train_conf_gan:
optim: adam
optim_conf:
lr: 0.0002 # use small lr for gan training
scheduler: constantlr
optim_d: adam
optim_conf_d:
lr: 0.0002 # use small lr for gan training
scheduler_d: constantlr
max_epoch: 200
grad_clip: 5
accum_grad: 1 # in gan training, accum_grad must be 1
log_interval: 100
save_per_step: -1

View File

@@ -83,9 +83,9 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
fi
cat data/{train-clean-100,train-clean-360,train-other-500}/parquet/data.list > data/train.data.list
cat data/{dev-clean,dev-other}/parquet/data.list > data/dev.data.list
for model in llm flow; do
for model in llm flow hifigan; do
torchrun --nnodes=1 --nproc_per_node=$num_gpus \
--rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:0" \
--rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:1234" \
cosyvoice/bin/train.py \
--train_engine $train_engine \
--config conf/cosyvoice.yaml \
@@ -104,7 +104,21 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
done
fi
# average model
average_num=5
if [ ${stage} -le 6 ] && [ ${stop_stage} -ge 6 ]; then
for model in llm flow hifigan; do
decode_checkpoint=`pwd`/exp/cosyvoice/$model/$train_engine/${model}.pt
echo "do model average and final checkpoint is $decode_checkpoint"
python cosyvoice/bin/average_model.py \
--dst_model $decode_checkpoint \
--src_path `pwd`/exp/cosyvoice/$model/$train_engine \
--num ${average_num} \
--val_best
done
fi
if [ ${stage} -le 7 ] && [ ${stop_stage} -ge 7 ]; then
echo "Export your model for inference speedup. Remember copy your llm or flow model to model_dir"
python cosyvoice/bin/export_jit.py --model_dir $pretrained_model_dir
python cosyvoice/bin/export_onnx.py --model_dir $pretrained_model_dir