Merge branch 'RVC-Boss:main' into main

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Ναρουσέ·μ·γιουμεμί·Χινακάννα 2025-04-03 16:39:11 +08:00 committed by GitHub
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194 changed files with 205751 additions and 6119 deletions

186
.gitignore vendored
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@ -1,4 +1,5 @@
.DS_Store
.vscode
__pycache__
*.pyc
env
@ -9,6 +10,191 @@ logs
reference
GPT_weights
SoVITS_weights
GPT_weights_v2
SoVITS_weights_v2
GPT_weights_v3
SoVITS_weights_v3
TEMP
weight.json
ffmpeg*
ffprobe*
cfg.json
speakers.json
ref_audios
tools/AP_BWE_main/24kto48k/*
!tools/AP_BWE_main/24kto48k/readme.txt
# Byte-compiled / optimized / DLL files
__pycache__/
*.py[cod]
*$py.class
# C extensions
*.so
# Distribution / packaging
.Python
build/
develop-eggs/
dist/
downloads/
eggs/
.eggs/
lib/
lib64/
parts/
sdist/
var/
wheels/
share/python-wheels/
*.egg-info/
.installed.cfg
*.egg
MANIFEST
# PyInstaller
# Usually these files are written by a python script from a template
# before PyInstaller builds the exe, so as to inject date/other infos into it.
*.manifest
*.spec
# Installer logs
pip-log.txt
pip-delete-this-directory.txt
# Unit test / coverage reports
htmlcov/
.tox/
.nox/
.coverage
.coverage.*
.cache
nosetests.xml
coverage.xml
*.cover
*.py,cover
.hypothesis/
.pytest_cache/
cover/
# Translations
*.mo
*.pot
# Django stuff:
*.log
local_settings.py
db.sqlite3
db.sqlite3-journal
# Flask stuff:
instance/
.webassets-cache
# Scrapy stuff:
.scrapy
# Sphinx documentation
docs/_build/
# PyBuilder
.pybuilder/
target/
# Jupyter Notebook
.ipynb_checkpoints
# IPython
profile_default/
ipython_config.py
# pyenv
# For a library or package, you might want to ignore these files since the code is
# intended to run in multiple environments; otherwise, check them in:
# .python-version
# pipenv
# According to pypa/pipenv#598, it is recommended to include Pipfile.lock in version control.
# However, in case of collaboration, if having platform-specific dependencies or dependencies
# having no cross-platform support, pipenv may install dependencies that don't work, or not
# install all needed dependencies.
#Pipfile.lock
# UV
# Similar to Pipfile.lock, it is generally recommended to include uv.lock in version control.
# This is especially recommended for binary packages to ensure reproducibility, and is more
# commonly ignored for libraries.
#uv.lock
# poetry
# Similar to Pipfile.lock, it is generally recommended to include poetry.lock in version control.
# This is especially recommended for binary packages to ensure reproducibility, and is more
# commonly ignored for libraries.
# https://python-poetry.org/docs/basic-usage/#commit-your-poetrylock-file-to-version-control
#poetry.lock
# pdm
# Similar to Pipfile.lock, it is generally recommended to include pdm.lock in version control.
#pdm.lock
# pdm stores project-wide configurations in .pdm.toml, but it is recommended to not include it
# in version control.
# https://pdm.fming.dev/latest/usage/project/#working-with-version-control
.pdm.toml
.pdm-python
.pdm-build/
# PEP 582; used by e.g. github.com/David-OConnor/pyflow and github.com/pdm-project/pdm
__pypackages__/
# Celery stuff
celerybeat-schedule
celerybeat.pid
# SageMath parsed files
*.sage.py
# Environments
.env
.venv
env/
venv/
ENV/
env.bak/
venv.bak/
# Spyder project settings
.spyderproject
.spyproject
# Rope project settings
.ropeproject
# mkdocs documentation
/site
# mypy
.mypy_cache/
.dmypy.json
dmypy.json
# Pyre type checker
.pyre/
# pytype static type analyzer
.pytype/
# Cython debug symbols
cython_debug/
# PyCharm
# JetBrains specific template is maintained in a separate JetBrains.gitignore that can
# be found at https://github.com/github/gitignore/blob/main/Global/JetBrains.gitignore
# and can be added to the global gitignore or merged into this file. For a more nuclear
# option (not recommended) you can uncomment the following to ignore the entire idea folder.
#.idea/
# Ruff stuff:
.ruff_cache/
# PyPI configuration file
.pypirc

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@ -34,9 +34,6 @@ RUN if [ "$IMAGE_TYPE" != "elite" ]; then \
fi
# Copy the rest of the application
COPY . /workspace
# Copy the rest of the application
COPY . /workspace

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@ -15,6 +15,8 @@ from torch.utils.data import DataLoader
from torch.utils.data import Dataset
from transformers import AutoTokenizer
version = os.environ.get('version',None)
from text import cleaned_text_to_sequence
# from config import exp_dir
@ -64,7 +66,7 @@ class Text2SemanticDataset(Dataset):
# get dict
self.path2 = phoneme_path # "%s/2-name2text.txt"%exp_dir#phoneme_path
self.path3 = "%s/3-bert" % (
os.path.basename(phoneme_path)
os.path.dirname(phoneme_path)
) # "%s/3-bert"%exp_dir#bert_dir
self.path6 = semantic_path # "%s/6-name2semantic.tsv"%exp_dir#semantic_path
assert os.path.exists(self.path2)
@ -149,7 +151,7 @@ class Text2SemanticDataset(Dataset):
phoneme = phoneme.split(" ")
try:
phoneme_ids = cleaned_text_to_sequence(phoneme)
phoneme_ids = cleaned_text_to_sequence(phoneme, version)
except:
traceback.print_exc()
# print(f"{item_name} not in self.phoneme_data !")

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@ -1,16 +1,18 @@
# modified from https://github.com/yangdongchao/SoundStorm/blob/master/soundstorm/s1/AR/models/t2s_model.py
# reference: https://github.com/lifeiteng/vall-e
import math
from typing import List, Optional
import torch
from tqdm import tqdm
from AR.models.utils import make_pad_mask
from AR.models.utils import make_pad_mask, make_pad_mask_left
from AR.models.utils import (
topk_sampling,
sample,
logits_to_probs,
multinomial_sample_one_no_sync,
dpo_loss,
make_reject_y,
make_reject_y,
get_batch_logps
)
from AR.modules.embedding import SinePositionalEmbedding
@ -34,6 +36,200 @@ default_config = {
"EOS": 1024,
}
# @torch.jit.script ## 使用的话首次推理会非常慢,而且推理速度不稳定
# Efficient implementation equivalent to the following:
def scaled_dot_product_attention(query:torch.Tensor, key:torch.Tensor, value:torch.Tensor, attn_mask:Optional[torch.Tensor]=None, scale:Optional[torch.Tensor]=None) -> torch.Tensor:
B, H, L, S =query.size(0), query.size(1), query.size(-2), key.size(-2)
if scale is None:
scale_factor = torch.tensor(1 / math.sqrt(query.size(-1)))
else:
scale_factor = scale
attn_bias = torch.zeros(B, H, L, S, dtype=query.dtype, device=query.device)
if attn_mask is not None:
if attn_mask.dtype == torch.bool:
attn_bias.masked_fill_(attn_mask, float("-inf"))
else:
attn_bias += attn_mask
attn_weight = query @ key.transpose(-2, -1) * scale_factor
attn_weight += attn_bias
attn_weight = torch.softmax(attn_weight, dim=-1)
if attn_mask is not None:
if attn_mask.dtype == torch.bool:
attn_weight.masked_fill_(attn_mask, 0)
else:
attn_mask[attn_mask!=float("-inf")] =0
attn_mask[attn_mask==float("-inf")] =1
attn_weight.masked_fill_(attn_mask, 0)
return attn_weight @ value
@torch.jit.script
class T2SMLP:
def __init__(self, w1, b1, w2, b2):
self.w1 = w1
self.b1 = b1
self.w2 = w2
self.b2 = b2
def forward(self, x):
x = F.relu(F.linear(x, self.w1, self.b1))
x = F.linear(x, self.w2, self.b2)
return x
@torch.jit.script
class T2SBlock:
def __init__(
self,
num_heads,
hidden_dim: int,
mlp: T2SMLP,
qkv_w,
qkv_b,
out_w,
out_b,
norm_w1,
norm_b1,
norm_eps1,
norm_w2,
norm_b2,
norm_eps2,
):
self.num_heads = num_heads
self.mlp = mlp
self.hidden_dim: int = hidden_dim
self.qkv_w = qkv_w
self.qkv_b = qkv_b
self.out_w = out_w
self.out_b = out_b
self.norm_w1 = norm_w1
self.norm_b1 = norm_b1
self.norm_eps1 = norm_eps1
self.norm_w2 = norm_w2
self.norm_b2 = norm_b2
self.norm_eps2 = norm_eps2
self.false = torch.tensor(False, dtype=torch.bool)
@torch.jit.ignore
def to_mask(self, x:torch.Tensor, padding_mask:Optional[torch.Tensor]):
if padding_mask is None:
return x
if padding_mask.dtype == torch.bool:
return x.masked_fill(padding_mask, 0)
else:
return x * padding_mask
def process_prompt(self, x:torch.Tensor, attn_mask : torch.Tensor, padding_mask:Optional[torch.Tensor]=None, torch_sdpa:bool=True):
q, k, v = F.linear(self.to_mask(x, padding_mask), self.qkv_w, self.qkv_b).chunk(3, dim=-1)
batch_size = q.shape[0]
q_len = q.shape[1]
kv_len = k.shape[1]
q = self.to_mask(q, padding_mask)
k_cache = self.to_mask(k, padding_mask)
v_cache = self.to_mask(v, padding_mask)
q = q.view(batch_size, q_len, self.num_heads, -1).transpose(1, 2)
k = k_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
v = v_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
if torch_sdpa:
attn = F.scaled_dot_product_attention(q, k, v, ~attn_mask)
else:
attn = scaled_dot_product_attention(q, k, v, attn_mask)
attn = attn.transpose(1, 2).reshape(batch_size, q_len, -1)
attn = F.linear(self.to_mask(attn, padding_mask), self.out_w, self.out_b)
x = x + attn
x = F.layer_norm(
x, [self.hidden_dim], self.norm_w1, self.norm_b1, self.norm_eps1
)
x = x + self.mlp.forward(x)
x = F.layer_norm(
x,
[self.hidden_dim],
self.norm_w2,
self.norm_b2,
self.norm_eps2,
)
return x, k_cache, v_cache
def decode_next_token(self, x:torch.Tensor, k_cache:torch.Tensor, v_cache:torch.Tensor, attn_mask:torch.Tensor=None, torch_sdpa:bool=True):
q, k, v = F.linear(x, self.qkv_w, self.qkv_b).chunk(3, dim=-1)
k_cache = torch.cat([k_cache, k], dim=1)
v_cache = torch.cat([v_cache, v], dim=1)
batch_size = q.shape[0]
q_len = q.shape[1]
kv_len = k_cache.shape[1]
q = q.view(batch_size, q_len, self.num_heads, -1).transpose(1, 2)
k = k_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
v = v_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
if torch_sdpa:
attn = F.scaled_dot_product_attention(q, k, v, (~attn_mask) if attn_mask is not None else None)
else:
attn = scaled_dot_product_attention(q, k, v, attn_mask)
attn = attn.transpose(1, 2).reshape(batch_size, q_len, -1)
attn = F.linear(attn, self.out_w, self.out_b)
x = x + attn
x = F.layer_norm(
x, [self.hidden_dim], self.norm_w1, self.norm_b1, self.norm_eps1
)
x = x + self.mlp.forward(x)
x = F.layer_norm(
x,
[self.hidden_dim],
self.norm_w2,
self.norm_b2,
self.norm_eps2,
)
return x, k_cache, v_cache
@torch.jit.script
class T2STransformer:
def __init__(self, num_blocks : int, blocks: List[T2SBlock]):
self.num_blocks : int = num_blocks
self.blocks = blocks
def process_prompt(
self, x:torch.Tensor, attn_mask : torch.Tensor,
padding_mask : Optional[torch.Tensor]=None,
torch_sdpa:bool=True
):
k_cache : List[torch.Tensor] = []
v_cache : List[torch.Tensor] = []
for i in range(self.num_blocks):
x, k_cache_, v_cache_ = self.blocks[i].process_prompt(x, attn_mask, padding_mask, torch_sdpa)
k_cache.append(k_cache_)
v_cache.append(v_cache_)
return x, k_cache, v_cache
def decode_next_token(
self, x:torch.Tensor,
k_cache: List[torch.Tensor],
v_cache: List[torch.Tensor],
attn_mask : torch.Tensor=None,
torch_sdpa:bool=True
):
for i in range(self.num_blocks):
x, k_cache[i], v_cache[i] = self.blocks[i].decode_next_token(x, k_cache[i], v_cache[i], attn_mask, torch_sdpa)
return x, k_cache, v_cache
class Text2SemanticDecoder(nn.Module):
def __init__(self, config, norm_first=False, top_k=3):
@ -89,6 +285,37 @@ class Text2SemanticDecoder(nn.Module):
ignore_index=self.EOS,
)
blocks = []
for i in range(self.num_layers):
layer = self.h.layers[i]
t2smlp = T2SMLP(
layer.linear1.weight,
layer.linear1.bias,
layer.linear2.weight,
layer.linear2.bias
)
block = T2SBlock(
self.num_head,
self.model_dim,
t2smlp,
layer.self_attn.in_proj_weight,
layer.self_attn.in_proj_bias,
layer.self_attn.out_proj.weight,
layer.self_attn.out_proj.bias,
layer.norm1.weight,
layer.norm1.bias,
layer.norm1.eps,
layer.norm2.weight,
layer.norm2.bias,
layer.norm2.eps
)
blocks.append(block)
self.t2s_transformer = T2STransformer(self.num_layers, blocks)
def make_input_data(self, x, x_lens, y, y_lens, bert_feature):
x = self.ar_text_embedding(x)
x = x + self.bert_proj(bert_feature.transpose(1, 2))
@ -116,7 +343,7 @@ class Text2SemanticDecoder(nn.Module):
(0, y_len),
value=True,
)
# x_attn_mask[:, x_len]=False
y_attn_mask = F.pad(
torch.triu(
torch.ones(y_len, y_len, dtype=torch.bool, device=x.device),
@ -246,14 +473,14 @@ class Text2SemanticDecoder(nn.Module):
# 需要看下这个函数和 forward 的区别以及没有 semantic 的时候 prompts 输入什么
def infer(
self,
x,
x_lens,
prompts,
bert_feature,
top_k: int = -100,
early_stop_num: int = -1,
temperature: float = 1.0,
self,
x,
x_lens,
prompts,
bert_feature,
top_k: int = -100,
early_stop_num: int = -1,
temperature: float = 1.0,
):
x = self.ar_text_embedding(x)
x = x + self.bert_proj(bert_feature.transpose(1, 2))
@ -321,16 +548,242 @@ class Text2SemanticDecoder(nn.Module):
# 错位
return targets[:, :-1], targets[:, 1:]
def infer_panel(
def infer_panel_batch_infer(
self,
x, #####全部文本token
x_lens,
prompts, ####参考音频token
bert_feature,
x:List[torch.LongTensor], #####全部文本token
x_lens:torch.LongTensor,
prompts:torch.LongTensor, ####参考音频token
bert_feature:List[torch.LongTensor],
top_k: int = -100,
top_p: int = 100,
early_stop_num: int = -1,
temperature: float = 1.0,
repetition_penalty: float = 1.35,
**kwargs,
):
if prompts is None:
print("Warning: Prompt free is not supported batch_infer! switch to naive_infer")
return self.infer_panel_naive_batched(x, x_lens, prompts, bert_feature, top_k=top_k, top_p=top_p, early_stop_num=early_stop_num, temperature=temperature, **kwargs)
max_len = kwargs.get("max_len",x_lens.max())
x_list = []
for x_item, bert_item in zip(x, bert_feature):
# max_len = max(max_len, x_item.shape[0], bert_item.shape[1])
x_item = self.ar_text_embedding(x_item.unsqueeze(0))
x_item = x_item + self.bert_proj(bert_item.transpose(0, 1).unsqueeze(0))
x_item = self.ar_text_position(x_item).squeeze(0)
# x_item = F.pad(x_item,(0,0,0,max_len-x_item.shape[0]),value=0) if x_item.shape[0]<max_len else x_item ### padding right
x_item = F.pad(x_item,(0,0,max_len-x_item.shape[0],0),value=0) if x_item.shape[0]<max_len else x_item ### padding left
x_list.append(x_item)
x:torch.Tensor = torch.stack(x_list, dim=0)
# AR Decoder
y = prompts
x_len = x.shape[1]
stop = False
k_cache = None
v_cache = None
################### first step ##########################
assert y is not None, "Error: Prompt free is not supported batch_infer!"
ref_free = False
y_emb = self.ar_audio_embedding(y)
y_len = y_emb.shape[1]
prefix_len = y.shape[1]
y_lens = torch.LongTensor([y_emb.shape[1]]*y_emb.shape[0]).to(x.device)
y_pos = self.ar_audio_position(y_emb)
xy_pos = torch.concat([x, y_pos], dim=1)
##### create mask #####
bsz = x.shape[0]
src_len = x_len + y_len
y_paddind_mask = make_pad_mask_left(y_lens, y_len)
x_paddind_mask = make_pad_mask_left(x_lens, max_len)
# (bsz, x_len + y_len)
padding_mask = torch.concat([x_paddind_mask, y_paddind_mask], dim=1)
x_mask = F.pad(
torch.zeros(x_len, x_len, dtype=torch.bool, device=x.device),
(0, y_len),
value=True,
)
y_mask = F.pad( ###yy的右上1扩展到左边xy的0,(y,x+y)
torch.triu(torch.ones(y_len, y_len, dtype=torch.bool, device=x.device), diagonal=1),
(x_len, 0),
value=False,
)
causal_mask = torch.concat([x_mask, y_mask], dim=0).view(1 , src_len, src_len).repeat(bsz, 1, 1).to(x.device)
# padding_mask = padding_mask.unsqueeze(1) * padding_mask.unsqueeze(2) ### [b, x+y, x+y]
### 上面是错误的会导致padding的token被"看见"
# 正确的padding_mask应该是
# | pad_len | x_len | y_len |
# [[PAD, PAD, PAD, 1, 2, 3, 4, 5, 6],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6], 前3行按理说也应该被mask掉但是为了防止计算attention时不出现nan还是保留了不影响结果
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6]]
padding_mask = padding_mask.view(bsz, 1, src_len).repeat(1, src_len, 1)
attn_mask:torch.Tensor = causal_mask.logical_or(padding_mask)
attn_mask = attn_mask.unsqueeze(1).expand(-1, self.num_head, -1, -1).bool()
# 正确的attn_mask应该是这样的
# | pad_len | x_len | y_len |
# [[PAD, PAD, PAD, 1, 2, 3, EOS, EOS, EOS],
# [PAD, PAD, PAD, 1, 2, 3, EOS, EOS, EOS],
# [PAD, PAD, PAD, 1, 2, 3, EOS, EOS, EOS], 前3行按理说也应该被mask掉但是为了防止计算attention时不出现nan还是保留了不影响结果
# [PAD, PAD, PAD, 1, 2, 3, EOS, EOS, EOS],
# [PAD, PAD, PAD, 1, 2, 3, EOS, EOS, EOS],
# [PAD, PAD, PAD, 1, 2, 3, EOS, EOS, EOS],
# [PAD, PAD, PAD, 1, 2, 3, 4, EOS, EOS],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, EOS],
# [PAD, PAD, PAD, 1, 2, 3, 4, 5, 6]]
###### decode #####
y_list = [None]*y.shape[0]
batch_idx_map = list(range(y.shape[0]))
idx_list = [None]*y.shape[0]
for idx in tqdm(range(1500)):
if idx == 0:
xy_dec, k_cache, v_cache = self.t2s_transformer.process_prompt(xy_pos, attn_mask, None)
else:
xy_dec, k_cache, v_cache = self.t2s_transformer.decode_next_token(xy_pos, k_cache, v_cache, attn_mask)
logits = self.ar_predict_layer(
xy_dec[:, -1]
)
if idx == 0:
attn_mask = F.pad(attn_mask[:,:,-1].unsqueeze(-2),(0,1),value=False)
logits = logits[:, :-1]
else:
attn_mask = F.pad(attn_mask,(0,1),value=False)
samples = sample(
logits, y, top_k=top_k, top_p=top_p, repetition_penalty=repetition_penalty, temperature=temperature
)[0]
y = torch.concat([y, samples], dim=1)
####### 移除batch中已经生成完毕的序列,进一步优化计算量
tokens = torch.argmax(logits, dim=-1)
reserved_idx_of_batch_for_y = None
if (self.EOS in samples[:, 0]) or \
(self.EOS in tokens): ###如果生成到EOS则停止
l1 = samples[:, 0]==self.EOS
l2 = tokens==self.EOS
l = l1.logical_or(l2)
removed_idx_of_batch_for_y = torch.where(l==True)[0].tolist()
reserved_idx_of_batch_for_y = torch.where(l==False)[0]
# batch_indexs = torch.tensor(batch_idx_map, device=y.device)[removed_idx_of_batch_for_y]
for i in removed_idx_of_batch_for_y:
batch_index = batch_idx_map[i]
idx_list[batch_index] = idx
y_list[batch_index] = y[i, :-1]
batch_idx_map = [batch_idx_map[i] for i in reserved_idx_of_batch_for_y.tolist()]
# 只保留batch中未生成完毕的序列
if reserved_idx_of_batch_for_y is not None:
# index = torch.LongTensor(batch_idx_map).to(y.device)
y = torch.index_select(y, dim=0, index=reserved_idx_of_batch_for_y)
attn_mask = torch.index_select(attn_mask, dim=0, index=reserved_idx_of_batch_for_y)
if k_cache is not None :
for i in range(len(k_cache)):
k_cache[i] = torch.index_select(k_cache[i], dim=0, index=reserved_idx_of_batch_for_y)
v_cache[i] = torch.index_select(v_cache[i], dim=0, index=reserved_idx_of_batch_for_y)
if (early_stop_num != -1 and (y.shape[1] - prefix_len) > early_stop_num) or idx==1499:
print("use early stop num:", early_stop_num)
stop = True
for i, batch_index in enumerate(batch_idx_map):
batch_index = batch_idx_map[i]
idx_list[batch_index] = idx
y_list[batch_index] = y[i, :-1]
if not (None in idx_list):
stop = True
if stop:
if y.shape[1]==0:
y = torch.concat([y, torch.zeros_like(samples)], dim=1)
print("bad zero prediction")
print(f"T2S Decoding EOS [{prefix_len} -> {y.shape[1]}]")
break
####################### update next step ###################################
y_emb = self.ar_audio_embedding(y[:, -1:])
xy_pos = y_emb * self.ar_audio_position.x_scale + self.ar_audio_position.alpha * self.ar_audio_position.pe[:, y_len + idx].to( dtype= y_emb.dtype,device=y_emb.device)
if (None in idx_list):
for i in range(x.shape[0]):
if idx_list[i] is None:
idx_list[i] = 1500-1 ###如果没有生成到EOS就用最大长度代替
if ref_free:
return y_list, [0]*x.shape[0]
# print(idx_list)
return y_list, idx_list
def infer_panel_naive_batched(self,
x:List[torch.LongTensor], #####全部文本token
x_lens:torch.LongTensor,
prompts:torch.LongTensor, ####参考音频token
bert_feature:List[torch.LongTensor],
top_k: int = -100,
top_p: int = 100,
early_stop_num: int = -1,
temperature: float = 1.0,
repetition_penalty: float = 1.35,
**kwargs
):
y_list = []
idx_list = []
for i in range(len(x)):
y, idx = self.infer_panel_naive(x[i].unsqueeze(0),
x_lens[i],
prompts[i].unsqueeze(0) if prompts is not None else None,
bert_feature[i].unsqueeze(0),
top_k,
top_p,
early_stop_num,
temperature,
repetition_penalty,
**kwargs)
y_list.append(y[0])
idx_list.append(idx)
return y_list, idx_list
def infer_panel_naive(
self,
x:torch.LongTensor, #####全部文本token
x_lens:torch.LongTensor,
prompts:torch.LongTensor, ####参考音频token
bert_feature:torch.LongTensor,
top_k: int = -100,
top_p: int = 100,
early_stop_num: int = -1,
temperature: float = 1.0,
repetition_penalty: float = 1.35,
**kwargs
):
x = self.ar_text_embedding(x)
x = x + self.bert_proj(bert_feature.transpose(1, 2))
@ -338,22 +791,14 @@ class Text2SemanticDecoder(nn.Module):
# AR Decoder
y = prompts
x_len = x.shape[1]
x_attn_mask = torch.zeros((x_len, x_len), dtype=torch.bool)
stop = False
# print(1111111,self.num_layers)
cache = {
"all_stage": self.num_layers,
"k": [None] * self.num_layers, ###根据配置自己手写
"v": [None] * self.num_layers,
# "xy_pos":None,##y_pos位置编码每次都不一样的没法缓存每次都要重新拼xy_pos.主要还是写法原因,其实是可以历史统一一样的,但也没啥计算量就不管了
"y_emb": None, ##只需要对最新的samples求emb再拼历史的就行
# "logits":None,###原版就已经只对结尾求再拼接了,不用管
# "xy_dec":None,###不需要本来只需要最后一个做logits
"first_infer": 1,
"stage": 0,
}
k_cache = None
v_cache = None
################### first step ##########################
if y is not None:
y_emb = self.ar_audio_embedding(y)
@ -361,7 +806,6 @@ class Text2SemanticDecoder(nn.Module):
prefix_len = y.shape[1]
y_pos = self.ar_audio_position(y_emb)
xy_pos = torch.concat([x, y_pos], dim=1)
cache["y_emb"] = y_emb
ref_free = False
else:
y_emb = None
@ -372,77 +816,78 @@ class Text2SemanticDecoder(nn.Module):
y = torch.zeros(x.shape[0], 0, dtype=torch.int, device=x.device)
ref_free = True
bsz = x.shape[0]
src_len = x_len + y_len
x_attn_mask_pad = F.pad(
x_attn_mask,
(0, y_len), ###xx的纯0扩展到xx纯0+xy纯1(x,x+y)
value=True,
)
x_attn_mask,
(0, y_len), ###xx的纯0扩展到xx纯0+xy纯1(x,x+y)
value=True,
)
y_attn_mask = F.pad( ###yy的右上1扩展到左边xy的0,(y,x+y)
torch.triu(torch.ones(y_len, y_len, dtype=torch.bool), diagonal=1),
(x_len, 0),
value=False,
)
xy_attn_mask = torch.concat([x_attn_mask_pad, y_attn_mask], dim=0).to(
x.device
)
xy_attn_mask = torch.concat([x_attn_mask_pad, y_attn_mask], dim=0)\
.unsqueeze(0)\
.expand(bsz*self.num_head, -1, -1)\
.view(bsz, self.num_head, src_len, src_len)\
.to(device=x.device, dtype=torch.bool)
for idx in tqdm(range(1500)):
xy_dec, _ = self.h((xy_pos, None), mask=xy_attn_mask, cache=cache)
if xy_attn_mask is not None:
xy_dec, k_cache, v_cache = self.t2s_transformer.process_prompt(xy_pos, xy_attn_mask, None)
else:
xy_dec, k_cache, v_cache = self.t2s_transformer.decode_next_token(xy_pos, k_cache, v_cache)
logits = self.ar_predict_layer(
xy_dec[:, -1]
) ##不用改如果用了cache的默认就是只有一帧取最后一帧一样的
# samples = topk_sampling(logits, top_k=top_k, top_p=1.0, temperature=temperature)
if(idx==0):###第一次跑不能EOS否则没有了
logits = logits[:, :-1] ###刨除1024终止符号的概率
)
if idx == 0:
xy_attn_mask = None
if(idx<11):###至少预测出10个token不然不给停止0.4s
logits = logits[:, :-1]
samples = sample(
logits[0], y, top_k=top_k, top_p=top_p, repetition_penalty=1.35, temperature=temperature
)[0].unsqueeze(0)
# 本次生成的 semantic_ids 和之前的 y 构成新的 y
# print(samples.shape)#[1,1]#第一个1是bs
y = torch.concat([y, samples], dim=1)
logits, y, top_k=top_k, top_p=top_p, repetition_penalty=repetition_penalty, temperature=temperature
)[0]
y = torch.concat([y, samples], dim=1)
if early_stop_num != -1 and (y.shape[1] - prefix_len) > early_stop_num:
print("use early stop num:", early_stop_num)
stop = True
if torch.argmax(logits, dim=-1)[0] == self.EOS or samples[0, 0] == self.EOS:
# print(torch.argmax(logits, dim=-1)[0] == self.EOS, samples[0, 0] == self.EOS)
stop = True
if stop:
# if prompts.shape[1] == y.shape[1]:
# y = torch.concat([y, torch.zeros_like(samples)], dim=1)
# print("bad zero prediction")
if y.shape[1]==0:
if y.shape[1] == 0:
y = torch.concat([y, torch.zeros_like(samples)], dim=1)
print("bad zero prediction")
print(f"T2S Decoding EOS [{prefix_len} -> {y.shape[1]}]")
break
####################### update next step ###################################
cache["first_infer"] = 0
if cache["y_emb"] is not None:
y_emb = torch.cat(
[cache["y_emb"], self.ar_audio_embedding(y[:, -1:])], dim = 1
)
cache["y_emb"] = y_emb
y_pos = self.ar_audio_position(y_emb)
xy_pos = y_pos[:, -1:]
else:
y_emb = self.ar_audio_embedding(y[:, -1:])
cache["y_emb"] = y_emb
y_pos = self.ar_audio_position(y_emb)
xy_pos = y_pos
y_len = y_pos.shape[1]
###最右边一列(是错的)
# xy_attn_mask=torch.ones((1, x_len+y_len), dtype=torch.bool,device=xy_pos.device)
# xy_attn_mask[:,-1]=False
###最下面一行(是对的)
xy_attn_mask = torch.zeros(
(1, x_len + y_len), dtype=torch.bool, device=xy_pos.device
)
####################### update next step ###################################
y_emb = self.ar_audio_embedding(y[:, -1:])
xy_pos = y_emb * self.ar_audio_position.x_scale + self.ar_audio_position.alpha * self.ar_audio_position.pe[:, y_len + idx].to(dtype=y_emb.dtype,device=y_emb.device)
if ref_free:
return y[:, :-1], 0
return y[:, :-1], idx-1
return y[:, :-1], idx
def infer_panel(
self,
x:torch.LongTensor, #####全部文本token
x_lens:torch.LongTensor,
prompts:torch.LongTensor, ####参考音频token
bert_feature:torch.LongTensor,
top_k: int = -100,
top_p: int = 100,
early_stop_num: int = -1,
temperature: float = 1.0,
repetition_penalty: float = 1.35,
**kwargs
):
return self.infer_panel_naive(x, x_lens, prompts, bert_feature, top_k, top_p, early_stop_num, temperature, repetition_penalty, **kwargs)

View File

@ -39,6 +39,39 @@ def make_pad_mask(lengths: torch.Tensor, max_len: int = 0) -> torch.Tensor:
return expaned_lengths >= lengths.unsqueeze(-1)
def make_pad_mask_left(lengths: torch.Tensor, max_len: int = 0) -> torch.Tensor:
"""
Args:
lengths:
A 1-D tensor containing sentence lengths.
max_len:
The length of masks.
Returns:
Return a 2-D bool tensor, where masked positions
are filled with `True` and non-masked positions are
filled with `False`.
#>>> lengths = torch.tensor([1, 3, 2, 5])
#>>> make_pad_mask(lengths)
tensor(
[
[True, True, False],
[True, False, False],
[True, True, False],
...
]
)
"""
assert lengths.ndim == 1, lengths.ndim
max_len = max(max_len, lengths.max())
n = lengths.size(0)
seq_range = torch.arange(0, max_len, device=lengths.device)
expaned_lengths = seq_range.unsqueeze(0).repeat(n, 1)
expaned_lengths -= (max_len-lengths).unsqueeze(-1)
return expaned_lengths<0
# https://github.com/microsoft/unilm/blob/master/xtune/src/transformers/modeling_utils.py
def top_k_top_p_filtering(
logits, top_k=0, top_p=1.0, filter_value=-float("Inf"), min_tokens_to_keep=1
@ -115,17 +148,17 @@ def logits_to_probs(
top_p: Optional[int] = None,
repetition_penalty: float = 1.0,
):
if previous_tokens is not None:
previous_tokens = previous_tokens.squeeze()
# if previous_tokens is not None:
# previous_tokens = previous_tokens.squeeze()
# print(logits.shape,previous_tokens.shape)
# pdb.set_trace()
if previous_tokens is not None and repetition_penalty != 1.0:
previous_tokens = previous_tokens.long()
score = torch.gather(logits, dim=0, index=previous_tokens)
score = torch.gather(logits, dim=1, index=previous_tokens)
score = torch.where(
score < 0, score * repetition_penalty, score / repetition_penalty
)
logits.scatter_(dim=0, index=previous_tokens, src=score)
logits.scatter_(dim=1, index=previous_tokens, src=score)
if top_p is not None and top_p < 1.0:
sorted_logits, sorted_indices = torch.sort(logits, descending=True)
@ -133,9 +166,9 @@ def logits_to_probs(
torch.nn.functional.softmax(sorted_logits, dim=-1), dim=-1
)
sorted_indices_to_remove = cum_probs > top_p
sorted_indices_to_remove[0] = False # keep at least one option
sorted_indices_to_remove[:, 0] = False # keep at least one option
indices_to_remove = sorted_indices_to_remove.scatter(
dim=0, index=sorted_indices, src=sorted_indices_to_remove
dim=1, index=sorted_indices, src=sorted_indices_to_remove
)
logits = logits.masked_fill(indices_to_remove, -float("Inf"))
@ -143,7 +176,7 @@ def logits_to_probs(
if top_k is not None:
v, _ = torch.topk(logits, min(top_k, logits.size(-1)))
pivot = v.select(-1, -1).unsqueeze(-1)
pivot = v[: , -1].unsqueeze(-1)
logits = torch.where(logits < pivot, -float("Inf"), logits)
probs = torch.nn.functional.softmax(logits, dim=-1)

View File

@ -12,33 +12,33 @@ import torch
def multi_head_attention_forward_patched(
query: Tensor,
key: Tensor,
value: Tensor,
embed_dim_to_check: int,
num_heads: int,
in_proj_weight: Optional[Tensor],
in_proj_bias: Optional[Tensor],
bias_k: Optional[Tensor],
bias_v: Optional[Tensor],
add_zero_attn: bool,
query,
key,
value,
embed_dim_to_check,
num_heads,
in_proj_weight,
in_proj_bias,
bias_k,
bias_v,
add_zero_attn,
dropout_p: float,
out_proj_weight: Tensor,
out_proj_bias: Optional[Tensor],
training: bool = True,
key_padding_mask: Optional[Tensor] = None,
need_weights: bool = True,
attn_mask: Optional[Tensor] = None,
use_separate_proj_weight: bool = False,
q_proj_weight: Optional[Tensor] = None,
k_proj_weight: Optional[Tensor] = None,
v_proj_weight: Optional[Tensor] = None,
static_k: Optional[Tensor] = None,
static_v: Optional[Tensor] = None,
average_attn_weights: bool = True,
is_causal: bool = False,
out_proj_weight,
out_proj_bias,
training = True,
key_padding_mask = None,
need_weights = True,
attn_mask = None,
use_separate_proj_weight = False,
q_proj_weight = None,
k_proj_weight = None,
v_proj_weight = None,
static_k = None,
static_v = None,
average_attn_weights = True,
is_causal = False,
cache=None,
) -> Tuple[Tensor, Optional[Tensor]]:
):
r"""
Args:
query, key, value: map a query and a set of key-value pairs to an output.

View File

@ -0,0 +1,21 @@
MIT License
Copyright (c) 2024 NVIDIA CORPORATION.
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.

View File

@ -0,0 +1,266 @@
## BigVGAN: A Universal Neural Vocoder with Large-Scale Training
#### Sang-gil Lee, Wei Ping, Boris Ginsburg, Bryan Catanzaro, Sungroh Yoon
[[Paper]](https://arxiv.org/abs/2206.04658) - [[Code]](https://github.com/NVIDIA/BigVGAN) - [[Showcase]](https://bigvgan-demo.github.io/) - [[Project Page]](https://research.nvidia.com/labs/adlr/projects/bigvgan/) - [[Weights]](https://huggingface.co/collections/nvidia/bigvgan-66959df3d97fd7d98d97dc9a) - [[Demo]](https://huggingface.co/spaces/nvidia/BigVGAN)
[![PWC](https://img.shields.io/endpoint.svg?url=https://paperswithcode.com/badge/bigvgan-a-universal-neural-vocoder-with-large/speech-synthesis-on-libritts)](https://paperswithcode.com/sota/speech-synthesis-on-libritts?p=bigvgan-a-universal-neural-vocoder-with-large)
<center><img src="https://user-images.githubusercontent.com/15963413/218609148-881e39df-33af-4af9-ab95-1427c4ebf062.png" width="800"></center>
## News
- **Sep 2024 (v2.4):**
- We have updated the pretrained checkpoints trained for 5M steps. This is final release of the BigVGAN-v2 checkpoints.
- **Jul 2024 (v2.3):**
- General refactor and code improvements for improved readability.
- Fully fused CUDA kernel of anti-alised activation (upsampling + activation + downsampling) with inference speed benchmark.
- **Jul 2024 (v2.2):** The repository now includes an interactive local demo using gradio.
- **Jul 2024 (v2.1):** BigVGAN is now integrated with 🤗 Hugging Face Hub with easy access to inference using pretrained checkpoints. We also provide an interactive demo on Hugging Face Spaces.
- **Jul 2024 (v2):** We release BigVGAN-v2 along with pretrained checkpoints. Below are the highlights:
- Custom CUDA kernel for inference: we provide a fused upsampling + activation kernel written in CUDA for accelerated inference speed. Our test shows 1.5 - 3x faster speed on a single A100 GPU.
- Improved discriminator and loss: BigVGAN-v2 is trained using a [multi-scale sub-band CQT discriminator](https://arxiv.org/abs/2311.14957) and a [multi-scale mel spectrogram loss](https://arxiv.org/abs/2306.06546).
- Larger training data: BigVGAN-v2 is trained using datasets containing diverse audio types, including speech in multiple languages, environmental sounds, and instruments.
- We provide pretrained checkpoints of BigVGAN-v2 using diverse audio configurations, supporting up to 44 kHz sampling rate and 512x upsampling ratio.
## Installation
The codebase has been tested on Python `3.10` and PyTorch `2.3.1` conda packages with either `pytorch-cuda=12.1` or `pytorch-cuda=11.8`. Below is an example command to create the conda environment:
```shell
conda create -n bigvgan python=3.10 pytorch torchvision torchaudio pytorch-cuda=12.1 -c pytorch -c nvidia
conda activate bigvgan
```
Clone the repository and install dependencies:
```shell
git clone https://github.com/NVIDIA/BigVGAN
cd BigVGAN
pip install -r requirements.txt
```
## Inference Quickstart using 🤗 Hugging Face Hub
Below example describes how you can use BigVGAN: load the pretrained BigVGAN generator from Hugging Face Hub, compute mel spectrogram from input waveform, and generate synthesized waveform using the mel spectrogram as the model's input.
```python
device = 'cuda'
import torch
import bigvgan
import librosa
from meldataset import get_mel_spectrogram
# instantiate the model. You can optionally set use_cuda_kernel=True for faster inference.
model = bigvgan.BigVGAN.from_pretrained('nvidia/bigvgan_v2_24khz_100band_256x', use_cuda_kernel=False)
# remove weight norm in the model and set to eval mode
model.remove_weight_norm()
model = model.eval().to(device)
# load wav file and compute mel spectrogram
wav_path = '/path/to/your/audio.wav'
wav, sr = librosa.load(wav_path, sr=model.h.sampling_rate, mono=True) # wav is np.ndarray with shape [T_time] and values in [-1, 1]
wav = torch.FloatTensor(wav).unsqueeze(0) # wav is FloatTensor with shape [B(1), T_time]
# compute mel spectrogram from the ground truth audio
mel = get_mel_spectrogram(wav, model.h).to(device) # mel is FloatTensor with shape [B(1), C_mel, T_frame]
# generate waveform from mel
with torch.inference_mode():
wav_gen = model(mel) # wav_gen is FloatTensor with shape [B(1), 1, T_time] and values in [-1, 1]
wav_gen_float = wav_gen.squeeze(0).cpu() # wav_gen is FloatTensor with shape [1, T_time]
# you can convert the generated waveform to 16 bit linear PCM
wav_gen_int16 = (wav_gen_float * 32767.0).numpy().astype('int16') # wav_gen is now np.ndarray with shape [1, T_time] and int16 dtype
```
## Local gradio demo <a href='https://github.com/gradio-app/gradio'><img src='https://img.shields.io/github/stars/gradio-app/gradio'></a>
You can run a local gradio demo using below command:
```python
pip install -r demo/requirements.txt
python demo/app.py
```
## Training
Create symbolic link to the root of the dataset. The codebase uses filelist with the relative path from the dataset. Below are the example commands for LibriTTS dataset:
```shell
cd filelists/LibriTTS && \
ln -s /path/to/your/LibriTTS/train-clean-100 train-clean-100 && \
ln -s /path/to/your/LibriTTS/train-clean-360 train-clean-360 && \
ln -s /path/to/your/LibriTTS/train-other-500 train-other-500 && \
ln -s /path/to/your/LibriTTS/dev-clean dev-clean && \
ln -s /path/to/your/LibriTTS/dev-other dev-other && \
ln -s /path/to/your/LibriTTS/test-clean test-clean && \
ln -s /path/to/your/LibriTTS/test-other test-other && \
cd ../..
```
Train BigVGAN model. Below is an example command for training BigVGAN-v2 using LibriTTS dataset at 24kHz with a full 100-band mel spectrogram as input:
```shell
python train.py \
--config configs/bigvgan_v2_24khz_100band_256x.json \
--input_wavs_dir filelists/LibriTTS \
--input_training_file filelists/LibriTTS/train-full.txt \
--input_validation_file filelists/LibriTTS/val-full.txt \
--list_input_unseen_wavs_dir filelists/LibriTTS filelists/LibriTTS \
--list_input_unseen_validation_file filelists/LibriTTS/dev-clean.txt filelists/LibriTTS/dev-other.txt \
--checkpoint_path exp/bigvgan_v2_24khz_100band_256x
```
## Synthesis
Synthesize from BigVGAN model. Below is an example command for generating audio from the model.
It computes mel spectrograms using wav files from `--input_wavs_dir` and saves the generated audio to `--output_dir`.
```shell
python inference.py \
--checkpoint_file /path/to/your/bigvgan_v2_24khz_100band_256x/bigvgan_generator.pt \
--input_wavs_dir /path/to/your/input_wav \
--output_dir /path/to/your/output_wav
```
`inference_e2e.py` supports synthesis directly from the mel spectrogram saved in `.npy` format, with shapes `[1, channel, frame]` or `[channel, frame]`.
It loads mel spectrograms from `--input_mels_dir` and saves the generated audio to `--output_dir`.
Make sure that the STFT hyperparameters for mel spectrogram are the same as the model, which are defined in `config.json` of the corresponding model.
```shell
python inference_e2e.py \
--checkpoint_file /path/to/your/bigvgan_v2_24khz_100band_256x/bigvgan_generator.pt \
--input_mels_dir /path/to/your/input_mel \
--output_dir /path/to/your/output_wav
```
## Using Custom CUDA Kernel for Synthesis
You can apply the fast CUDA inference kernel by using a parameter `use_cuda_kernel` when instantiating BigVGAN:
```python
generator = BigVGAN(h, use_cuda_kernel=True)
```
You can also pass `--use_cuda_kernel` to `inference.py` and `inference_e2e.py` to enable this feature.
When applied for the first time, it builds the kernel using `nvcc` and `ninja`. If the build succeeds, the kernel is saved to `alias_free_activation/cuda/build` and the model automatically loads the kernel. The codebase has been tested using CUDA `12.1`.
Please make sure that both are installed in your system and `nvcc` installed in your system matches the version your PyTorch build is using.
We recommend running `test_cuda_vs_torch_model.py` first to build and check the correctness of the CUDA kernel. See below example command and its output, where it returns `[Success] test CUDA fused vs. plain torch BigVGAN inference`:
```python
python tests/test_cuda_vs_torch_model.py \
--checkpoint_file /path/to/your/bigvgan_generator.pt
```
```shell
loading plain Pytorch BigVGAN
...
loading CUDA kernel BigVGAN with auto-build
Detected CUDA files, patching ldflags
Emitting ninja build file /path/to/your/BigVGAN/alias_free_activation/cuda/build/build.ninja..
Building extension module anti_alias_activation_cuda...
...
Loading extension module anti_alias_activation_cuda...
...
Loading '/path/to/your/bigvgan_generator.pt'
...
[Success] test CUDA fused vs. plain torch BigVGAN inference
> mean_difference=0.0007238413265440613
...
```
If you see `[Fail] test CUDA fused vs. plain torch BigVGAN inference`, it means that the CUDA kernel inference is incorrect. Please check if `nvcc` installed in your system is compatible with your PyTorch version.
## Pretrained Models
We provide the [pretrained models on Hugging Face Collections](https://huggingface.co/collections/nvidia/bigvgan-66959df3d97fd7d98d97dc9a).
One can download the checkpoints of the generator weight (named `bigvgan_generator.pt`) and its discriminator/optimizer states (named `bigvgan_discriminator_optimizer.pt`) within the listed model repositories.
| Model Name | Sampling Rate | Mel band | fmax | Upsampling Ratio | Params | Dataset | Steps | Fine-Tuned |
|:--------------------------------------------------------------------------------------------------------:|:-------------:|:--------:|:-----:|:----------------:|:------:|:--------------------------:|:-----:|:----------:|
| [bigvgan_v2_44khz_128band_512x](https://huggingface.co/nvidia/bigvgan_v2_44khz_128band_512x) | 44 kHz | 128 | 22050 | 512 | 122M | Large-scale Compilation | 5M | No |
| [bigvgan_v2_44khz_128band_256x](https://huggingface.co/nvidia/bigvgan_v2_44khz_128band_256x) | 44 kHz | 128 | 22050 | 256 | 112M | Large-scale Compilation | 5M | No |
| [bigvgan_v2_24khz_100band_256x](https://huggingface.co/nvidia/bigvgan_v2_24khz_100band_256x) | 24 kHz | 100 | 12000 | 256 | 112M | Large-scale Compilation | 5M | No |
| [bigvgan_v2_22khz_80band_256x](https://huggingface.co/nvidia/bigvgan_v2_22khz_80band_256x) | 22 kHz | 80 | 11025 | 256 | 112M | Large-scale Compilation | 5M | No |
| [bigvgan_v2_22khz_80band_fmax8k_256x](https://huggingface.co/nvidia/bigvgan_v2_22khz_80band_fmax8k_256x) | 22 kHz | 80 | 8000 | 256 | 112M | Large-scale Compilation | 5M | No |
| [bigvgan_24khz_100band](https://huggingface.co/nvidia/bigvgan_24khz_100band) | 24 kHz | 100 | 12000 | 256 | 112M | LibriTTS | 5M | No |
| [bigvgan_base_24khz_100band](https://huggingface.co/nvidia/bigvgan_base_24khz_100band) | 24 kHz | 100 | 12000 | 256 | 14M | LibriTTS | 5M | No |
| [bigvgan_22khz_80band](https://huggingface.co/nvidia/bigvgan_22khz_80band) | 22 kHz | 80 | 8000 | 256 | 112M | LibriTTS + VCTK + LJSpeech | 5M | No |
| [bigvgan_base_22khz_80band](https://huggingface.co/nvidia/bigvgan_base_22khz_80band) | 22 kHz | 80 | 8000 | 256 | 14M | LibriTTS + VCTK + LJSpeech | 5M | No |
The paper results are based on the original 24kHz BigVGAN models (`bigvgan_24khz_100band` and `bigvgan_base_24khz_100band`) trained on LibriTTS dataset.
We also provide 22kHz BigVGAN models with band-limited setup (i.e., fmax=8000) for TTS applications.
Note that the checkpoints use `snakebeta` activation with log scale parameterization, which have the best overall quality.
You can fine-tune the models by:
1. downloading the checkpoints (both the generator weight and its discriminator/optimizer states)
2. resuming training using your audio dataset by specifying `--checkpoint_path` that includes the checkpoints when launching `train.py`
## Training Details of BigVGAN-v2
Comapred to the original BigVGAN, the pretrained checkpoints of BigVGAN-v2 used `batch_size=32` with a longer `segment_size=65536` and are trained using 8 A100 GPUs.
Note that the BigVGAN-v2 `json` config files in `./configs` use `batch_size=4` as default to fit in a single A100 GPU for training. You can fine-tune the models adjusting `batch_size` depending on your GPUs.
When training BigVGAN-v2 from scratch with small batch size, it can potentially encounter the early divergence problem mentioned in the paper. In such case, we recommend lowering the `clip_grad_norm` value (e.g. `100`) for the early training iterations (e.g. 20000 steps) and increase the value to the default `500`.
## Evaluation Results of BigVGAN-v2
Below are the objective results of the 24kHz model (`bigvgan_v2_24khz_100band_256x`) obtained from the LibriTTS `dev` sets. BigVGAN-v2 shows noticeable improvements of the metrics. The model also exhibits reduced perceptual artifacts, especially for non-speech audio.
| Model | Dataset | Steps | PESQ(↑) | M-STFT(↓) | MCD(↓) | Periodicity(↓) | V/UV F1(↑) |
|:----------:|:-----------------------:|:-----:|:---------:|:----------:|:----------:|:--------------:|:----------:|
| BigVGAN | LibriTTS | 1M | 4.027 | 0.7997 | 0.3745 | 0.1018 | 0.9598 |
| BigVGAN | LibriTTS | 5M | 4.256 | 0.7409 | 0.2988 | 0.0809 | 0.9698 |
| BigVGAN-v2 | Large-scale Compilation | 3M | 4.359 | 0.7134 | 0.3060 | 0.0621 | 0.9777 |
| BigVGAN-v2 | Large-scale Compilation | 5M | **4.362** | **0.7026** | **0.2903** | **0.0593** | **0.9793** |
## Speed Benchmark
Below are the speed and VRAM usage benchmark results of BigVGAN from `tests/test_cuda_vs_torch_model.py`, using `bigvgan_v2_24khz_100band_256x` as a reference model.
| GPU | num_mel_frame | use_cuda_kernel | Speed (kHz) | Real-time Factor | VRAM (GB) |
|:--------------------------:|:-------------:|:---------------:|:-----------:|:----------------:|:---------:|
| NVIDIA A100 | 256 | False | 1672.1 | 69.7x | 1.3 |
| | | True | 3916.5 | 163.2x | 1.3 |
| | 2048 | False | 1899.6 | 79.2x | 1.7 |
| | | True | 5330.1 | 222.1x | 1.7 |
| | 16384 | False | 1973.8 | 82.2x | 5.0 |
| | | True | 5761.7 | 240.1x | 4.4 |
| NVIDIA GeForce RTX 3080 | 256 | False | 841.1 | 35.0x | 1.3 |
| | | True | 1598.1 | 66.6x | 1.3 |
| | 2048 | False | 929.9 | 38.7x | 1.7 |
| | | True | 1971.3 | 82.1x | 1.6 |
| | 16384 | False | 943.4 | 39.3x | 5.0 |
| | | True | 2026.5 | 84.4x | 3.9 |
| NVIDIA GeForce RTX 2080 Ti | 256 | False | 515.6 | 21.5x | 1.3 |
| | | True | 811.3 | 33.8x | 1.3 |
| | 2048 | False | 576.5 | 24.0x | 1.7 |
| | | True | 1023.0 | 42.6x | 1.5 |
| | 16384 | False | 589.4 | 24.6x | 5.0 |
| | | True | 1068.1 | 44.5x | 3.2 |
## Acknowledgements
We thank Vijay Anand Korthikanti and Kevin J. Shih for their generous support in implementing the CUDA kernel for inference.
## References
- [HiFi-GAN](https://github.com/jik876/hifi-gan) (for generator and multi-period discriminator)
- [Snake](https://github.com/EdwardDixon/snake) (for periodic activation)
- [Alias-free-torch](https://github.com/junjun3518/alias-free-torch) (for anti-aliasing)
- [Julius](https://github.com/adefossez/julius) (for low-pass filter)
- [UnivNet](https://github.com/mindslab-ai/univnet) (for multi-resolution discriminator)
- [descript-audio-codec](https://github.com/descriptinc/descript-audio-codec) and [vocos](https://github.com/gemelo-ai/vocos) (for multi-band multi-scale STFT discriminator and multi-scale mel spectrogram loss)
- [Amphion](https://github.com/open-mmlab/Amphion) (for multi-scale sub-band CQT discriminator)

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# Implementation adapted from https://github.com/EdwardDixon/snake under the MIT license.
# LICENSE is in incl_licenses directory.
import torch
from torch import nn, sin, pow
from torch.nn import Parameter
class Snake(nn.Module):
"""
Implementation of a sine-based periodic activation function
Shape:
- Input: (B, C, T)
- Output: (B, C, T), same shape as the input
Parameters:
- alpha - trainable parameter
References:
- This activation function is from this paper by Liu Ziyin, Tilman Hartwig, Masahito Ueda:
https://arxiv.org/abs/2006.08195
Examples:
>>> a1 = snake(256)
>>> x = torch.randn(256)
>>> x = a1(x)
"""
def __init__(
self, in_features, alpha=1.0, alpha_trainable=True, alpha_logscale=False
):
"""
Initialization.
INPUT:
- in_features: shape of the input
- alpha: trainable parameter
alpha is initialized to 1 by default, higher values = higher-frequency.
alpha will be trained along with the rest of your model.
"""
super(Snake, self).__init__()
self.in_features = in_features
# Initialize alpha
self.alpha_logscale = alpha_logscale
if self.alpha_logscale: # Log scale alphas initialized to zeros
self.alpha = Parameter(torch.zeros(in_features) * alpha)
else: # Linear scale alphas initialized to ones
self.alpha = Parameter(torch.ones(in_features) * alpha)
self.alpha.requires_grad = alpha_trainable
self.no_div_by_zero = 0.000000001
def forward(self, x):
"""
Forward pass of the function.
Applies the function to the input elementwise.
Snake = x + 1/a * sin^2 (xa)
"""
alpha = self.alpha.unsqueeze(0).unsqueeze(-1) # Line up with x to [B, C, T]
if self.alpha_logscale:
alpha = torch.exp(alpha)
x = x + (1.0 / (alpha + self.no_div_by_zero)) * pow(sin(x * alpha), 2)
return x
class SnakeBeta(nn.Module):
"""
A modified Snake function which uses separate parameters for the magnitude of the periodic components
Shape:
- Input: (B, C, T)
- Output: (B, C, T), same shape as the input
Parameters:
- alpha - trainable parameter that controls frequency
- beta - trainable parameter that controls magnitude
References:
- This activation function is a modified version based on this paper by Liu Ziyin, Tilman Hartwig, Masahito Ueda:
https://arxiv.org/abs/2006.08195
Examples:
>>> a1 = snakebeta(256)
>>> x = torch.randn(256)
>>> x = a1(x)
"""
def __init__(
self, in_features, alpha=1.0, alpha_trainable=True, alpha_logscale=False
):
"""
Initialization.
INPUT:
- in_features: shape of the input
- alpha - trainable parameter that controls frequency
- beta - trainable parameter that controls magnitude
alpha is initialized to 1 by default, higher values = higher-frequency.
beta is initialized to 1 by default, higher values = higher-magnitude.
alpha will be trained along with the rest of your model.
"""
super(SnakeBeta, self).__init__()
self.in_features = in_features
# Initialize alpha
self.alpha_logscale = alpha_logscale
if self.alpha_logscale: # Log scale alphas initialized to zeros
self.alpha = Parameter(torch.zeros(in_features) * alpha)
self.beta = Parameter(torch.zeros(in_features) * alpha)
else: # Linear scale alphas initialized to ones
self.alpha = Parameter(torch.ones(in_features) * alpha)
self.beta = Parameter(torch.ones(in_features) * alpha)
self.alpha.requires_grad = alpha_trainable
self.beta.requires_grad = alpha_trainable
self.no_div_by_zero = 0.000000001
def forward(self, x):
"""
Forward pass of the function.
Applies the function to the input elementwise.
SnakeBeta = x + 1/b * sin^2 (xa)
"""
alpha = self.alpha.unsqueeze(0).unsqueeze(-1) # Line up with x to [B, C, T]
beta = self.beta.unsqueeze(0).unsqueeze(-1)
if self.alpha_logscale:
alpha = torch.exp(alpha)
beta = torch.exp(beta)
x = x + (1.0 / (beta + self.no_div_by_zero)) * pow(sin(x * alpha), 2)
return x

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
import torch
import torch.nn as nn
from alias_free_activation.torch.resample import UpSample1d, DownSample1d
# load fused CUDA kernel: this enables importing anti_alias_activation_cuda
from alias_free_activation.cuda import load
anti_alias_activation_cuda = load.load()
class FusedAntiAliasActivation(torch.autograd.Function):
"""
Assumes filter size 12, replication padding on upsampling/downsampling, and logscale alpha/beta parameters as inputs.
The hyperparameters are hard-coded in the kernel to maximize speed.
NOTE: The fused kenrel is incorrect for Activation1d with different hyperparameters.
"""
@staticmethod
def forward(ctx, inputs, up_ftr, down_ftr, alpha, beta):
activation_results = anti_alias_activation_cuda.forward(
inputs, up_ftr, down_ftr, alpha, beta
)
return activation_results
@staticmethod
def backward(ctx, output_grads):
raise NotImplementedError
return output_grads, None, None
class Activation1d(nn.Module):
def __init__(
self,
activation,
up_ratio: int = 2,
down_ratio: int = 2,
up_kernel_size: int = 12,
down_kernel_size: int = 12,
fused: bool = True,
):
super().__init__()
self.up_ratio = up_ratio
self.down_ratio = down_ratio
self.act = activation
self.upsample = UpSample1d(up_ratio, up_kernel_size)
self.downsample = DownSample1d(down_ratio, down_kernel_size)
self.fused = fused # Whether to use fused CUDA kernel or not
def forward(self, x):
if not self.fused:
x = self.upsample(x)
x = self.act(x)
x = self.downsample(x)
return x
else:
if self.act.__class__.__name__ == "Snake":
beta = self.act.alpha.data # Snake uses same params for alpha and beta
else:
beta = (
self.act.beta.data
) # Snakebeta uses different params for alpha and beta
alpha = self.act.alpha.data
if (
not self.act.alpha_logscale
): # Exp baked into cuda kernel, cancel it out with a log
alpha = torch.log(alpha)
beta = torch.log(beta)
x = FusedAntiAliasActivation.apply(
x, self.upsample.filter, self.downsample.lowpass.filter, alpha, beta
)
return x

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/* coding=utf-8
* Copyright (c) 2024, NVIDIA CORPORATION. All rights reserved.
*
* 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.
*/
#include <torch/extension.h>
extern "C" torch::Tensor fwd_cuda(torch::Tensor const &input, torch::Tensor const &up_filter, torch::Tensor const &down_filter, torch::Tensor const &alpha, torch::Tensor const &beta);
PYBIND11_MODULE(TORCH_EXTENSION_NAME, m) {
m.def("forward", &fwd_cuda, "Anti-Alias Activation forward (CUDA)");
}

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/* coding=utf-8
* Copyright (c) 2024, NVIDIA CORPORATION. All rights reserved.
*
* 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.
*/
#include <ATen/ATen.h>
#include <cuda.h>
#include <cuda_runtime.h>
#include <cuda_fp16.h>
#include <cuda_profiler_api.h>
#include <ATen/cuda/CUDAContext.h>
#include <torch/extension.h>
#include "type_shim.h"
#include <assert.h>
#include <cfloat>
#include <limits>
#include <stdint.h>
#include <c10/macros/Macros.h>
namespace
{
// Hard-coded hyperparameters
// WARP_SIZE and WARP_BATCH must match the return values batches_per_warp and
constexpr int ELEMENTS_PER_LDG_STG = 1; //(WARP_ITERATIONS < 4) ? 1 : 4;
constexpr int BUFFER_SIZE = 32;
constexpr int FILTER_SIZE = 12;
constexpr int HALF_FILTER_SIZE = 6;
constexpr int UPSAMPLE_REPLICATION_PAD = 5; // 5 on each side, matching torch impl
constexpr int DOWNSAMPLE_REPLICATION_PAD_LEFT = 5; // matching torch impl
constexpr int DOWNSAMPLE_REPLICATION_PAD_RIGHT = 6; // matching torch impl
template <typename input_t, typename output_t, typename acc_t>
__global__ void anti_alias_activation_forward(
output_t *dst,
const input_t *src,
const input_t *up_ftr,
const input_t *down_ftr,
const input_t *alpha,
const input_t *beta,
int batch_size,
int channels,
int seq_len)
{
// Up and downsample filters
input_t up_filter[FILTER_SIZE];
input_t down_filter[FILTER_SIZE];
// Load data from global memory including extra indices reserved for replication paddings
input_t elements[2 * FILTER_SIZE + 2 * BUFFER_SIZE + 2 * UPSAMPLE_REPLICATION_PAD] = {0};
input_t intermediates[2 * FILTER_SIZE + 2 * BUFFER_SIZE + DOWNSAMPLE_REPLICATION_PAD_LEFT + DOWNSAMPLE_REPLICATION_PAD_RIGHT] = {0};
// Output stores downsampled output before writing to dst
output_t output[BUFFER_SIZE];
// blockDim/threadIdx = (128, 1, 1)
// gridDim/blockIdx = (seq_blocks, channels, batches)
int block_offset = (blockIdx.x * 128 * BUFFER_SIZE + seq_len * (blockIdx.y + gridDim.y * blockIdx.z));
int local_offset = threadIdx.x * BUFFER_SIZE;
int seq_offset = blockIdx.x * 128 * BUFFER_SIZE + local_offset;
// intermediate have double the seq_len
int intermediate_local_offset = threadIdx.x * BUFFER_SIZE * 2;
int intermediate_seq_offset = blockIdx.x * 128 * BUFFER_SIZE * 2 + intermediate_local_offset;
// Get values needed for replication padding before moving pointer
const input_t *right_most_pntr = src + (seq_len * (blockIdx.y + gridDim.y * blockIdx.z));
input_t seq_left_most_value = right_most_pntr[0];
input_t seq_right_most_value = right_most_pntr[seq_len - 1];
// Move src and dst pointers
src += block_offset + local_offset;
dst += block_offset + local_offset;
// Alpha and beta values for snake activatons. Applies exp by default
alpha = alpha + blockIdx.y;
input_t alpha_val = expf(alpha[0]);
beta = beta + blockIdx.y;
input_t beta_val = expf(beta[0]);
#pragma unroll
for (int it = 0; it < FILTER_SIZE; it += 1)
{
up_filter[it] = up_ftr[it];
down_filter[it] = down_ftr[it];
}
// Apply replication padding for upsampling, matching torch impl
#pragma unroll
for (int it = -HALF_FILTER_SIZE; it < BUFFER_SIZE + HALF_FILTER_SIZE; it += 1)
{
int element_index = seq_offset + it; // index for element
if ((element_index < 0) && (element_index >= -UPSAMPLE_REPLICATION_PAD))
{
elements[2 * (HALF_FILTER_SIZE + it)] = 2 * seq_left_most_value;
}
if ((element_index >= seq_len) && (element_index < seq_len + UPSAMPLE_REPLICATION_PAD))
{
elements[2 * (HALF_FILTER_SIZE + it)] = 2 * seq_right_most_value;
}
if ((element_index >= 0) && (element_index < seq_len))
{
elements[2 * (HALF_FILTER_SIZE + it)] = 2 * src[it];
}
}
// Apply upsampling strided convolution and write to intermediates. It reserves DOWNSAMPLE_REPLICATION_PAD_LEFT for replication padding of the downsampilng conv later
#pragma unroll
for (int it = 0; it < (2 * BUFFER_SIZE + 2 * FILTER_SIZE); it += 1)
{
input_t acc = 0.0;
int element_index = intermediate_seq_offset + it; // index for intermediate
#pragma unroll
for (int f_idx = 0; f_idx < FILTER_SIZE; f_idx += 1)
{
if ((element_index + f_idx) >= 0)
{
acc += up_filter[f_idx] * elements[it + f_idx];
}
}
intermediates[it + DOWNSAMPLE_REPLICATION_PAD_LEFT] = acc;
}
// Apply activation function. It reserves DOWNSAMPLE_REPLICATION_PAD_LEFT and DOWNSAMPLE_REPLICATION_PAD_RIGHT for replication padding of the downsampilng conv later
double no_div_by_zero = 0.000000001;
#pragma unroll
for (int it = 0; it < 2 * BUFFER_SIZE + 2 * FILTER_SIZE; it += 1)
{
intermediates[it + DOWNSAMPLE_REPLICATION_PAD_LEFT] += (1.0 / (beta_val + no_div_by_zero)) * sinf(intermediates[it + DOWNSAMPLE_REPLICATION_PAD_LEFT] * alpha_val) * sinf(intermediates[it + DOWNSAMPLE_REPLICATION_PAD_LEFT] * alpha_val);
}
// Apply replication padding before downsampling conv from intermediates
#pragma unroll
for (int it = 0; it < DOWNSAMPLE_REPLICATION_PAD_LEFT; it += 1)
{
intermediates[it] = intermediates[DOWNSAMPLE_REPLICATION_PAD_LEFT];
}
#pragma unroll
for (int it = DOWNSAMPLE_REPLICATION_PAD_LEFT + 2 * BUFFER_SIZE + 2 * FILTER_SIZE; it < DOWNSAMPLE_REPLICATION_PAD_LEFT + 2 * BUFFER_SIZE + 2 * FILTER_SIZE + DOWNSAMPLE_REPLICATION_PAD_RIGHT; it += 1)
{
intermediates[it] = intermediates[DOWNSAMPLE_REPLICATION_PAD_LEFT + 2 * BUFFER_SIZE + 2 * FILTER_SIZE - 1];
}
// Apply downsample strided convolution (assuming stride=2) from intermediates
#pragma unroll
for (int it = 0; it < BUFFER_SIZE; it += 1)
{
input_t acc = 0.0;
#pragma unroll
for (int f_idx = 0; f_idx < FILTER_SIZE; f_idx += 1)
{
// Add constant DOWNSAMPLE_REPLICATION_PAD_RIGHT to match torch implementation
acc += down_filter[f_idx] * intermediates[it * 2 + f_idx + DOWNSAMPLE_REPLICATION_PAD_RIGHT];
}
output[it] = acc;
}
// Write output to dst
#pragma unroll
for (int it = 0; it < BUFFER_SIZE; it += ELEMENTS_PER_LDG_STG)
{
int element_index = seq_offset + it;
if (element_index < seq_len)
{
dst[it] = output[it];
}
}
}
template <typename input_t, typename output_t, typename acc_t>
void dispatch_anti_alias_activation_forward(
output_t *dst,
const input_t *src,
const input_t *up_ftr,
const input_t *down_ftr,
const input_t *alpha,
const input_t *beta,
int batch_size,
int channels,
int seq_len)
{
if (seq_len == 0)
{
return;
}
else
{
// Use 128 threads per block to maximimize gpu utilization
constexpr int threads_per_block = 128;
constexpr int seq_len_per_block = 4096;
int blocks_per_seq_len = (seq_len + seq_len_per_block - 1) / seq_len_per_block;
dim3 blocks(blocks_per_seq_len, channels, batch_size);
dim3 threads(threads_per_block, 1, 1);
anti_alias_activation_forward<input_t, output_t, acc_t>
<<<blocks, threads, 0, at::cuda::getCurrentCUDAStream()>>>(dst, src, up_ftr, down_ftr, alpha, beta, batch_size, channels, seq_len);
}
}
}
extern "C" torch::Tensor fwd_cuda(torch::Tensor const &input, torch::Tensor const &up_filter, torch::Tensor const &down_filter, torch::Tensor const &alpha, torch::Tensor const &beta)
{
// Input is a 3d tensor with dimensions [batches, channels, seq_len]
const int batches = input.size(0);
const int channels = input.size(1);
const int seq_len = input.size(2);
// Output
auto act_options = input.options().requires_grad(false);
torch::Tensor anti_alias_activation_results =
torch::empty({batches, channels, seq_len}, act_options);
void *input_ptr = static_cast<void *>(input.data_ptr());
void *up_filter_ptr = static_cast<void *>(up_filter.data_ptr());
void *down_filter_ptr = static_cast<void *>(down_filter.data_ptr());
void *alpha_ptr = static_cast<void *>(alpha.data_ptr());
void *beta_ptr = static_cast<void *>(beta.data_ptr());
void *anti_alias_activation_results_ptr = static_cast<void *>(anti_alias_activation_results.data_ptr());
DISPATCH_FLOAT_HALF_AND_BFLOAT(
input.scalar_type(),
"dispatch anti alias activation_forward",
dispatch_anti_alias_activation_forward<scalar_t, scalar_t, float>(
reinterpret_cast<scalar_t *>(anti_alias_activation_results_ptr),
reinterpret_cast<const scalar_t *>(input_ptr),
reinterpret_cast<const scalar_t *>(up_filter_ptr),
reinterpret_cast<const scalar_t *>(down_filter_ptr),
reinterpret_cast<const scalar_t *>(alpha_ptr),
reinterpret_cast<const scalar_t *>(beta_ptr),
batches,
channels,
seq_len););
return anti_alias_activation_results;
}

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/* coding=utf-8
* Copyright (c) 2020, NVIDIA CORPORATION. All rights reserved.
*
* 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.
*/
/*This code is copied fron NVIDIA apex:
* https://github.com/NVIDIA/apex
* with minor changes. */
#ifndef TORCH_CHECK
#define TORCH_CHECK AT_CHECK
#endif
#ifdef VERSION_GE_1_3
#define DATA_PTR data_ptr
#else
#define DATA_PTR data
#endif

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
import os
import pathlib
import subprocess
from torch.utils import cpp_extension
"""
Setting this param to a list has a problem of generating different compilation commands (with diferent order of architectures) and leading to recompilation of fused kernels.
Set it to empty stringo avoid recompilation and assign arch flags explicity in extra_cuda_cflags below
"""
os.environ["TORCH_CUDA_ARCH_LIST"] = ""
def load():
# Check if cuda 11 is installed for compute capability 8.0
cc_flag = []
_, bare_metal_major, _ = _get_cuda_bare_metal_version(cpp_extension.CUDA_HOME)
if int(bare_metal_major) >= 11:
cc_flag.append("-gencode")
cc_flag.append("arch=compute_80,code=sm_80")
# Build path
srcpath = pathlib.Path(__file__).parent.absolute()
buildpath = srcpath / "build"
_create_build_dir(buildpath)
# Helper function to build the kernels.
def _cpp_extention_load_helper(name, sources, extra_cuda_flags):
return cpp_extension.load(
name=name,
sources=sources,
build_directory=buildpath,
extra_cflags=[
"-O3",
],
extra_cuda_cflags=[
"-O3",
"-gencode",
"arch=compute_70,code=sm_70",
"--use_fast_math",
]
+ extra_cuda_flags
+ cc_flag,
verbose=True,
)
extra_cuda_flags = [
"-U__CUDA_NO_HALF_OPERATORS__",
"-U__CUDA_NO_HALF_CONVERSIONS__",
"--expt-relaxed-constexpr",
"--expt-extended-lambda",
]
sources = [
srcpath / "anti_alias_activation.cpp",
srcpath / "anti_alias_activation_cuda.cu",
]
anti_alias_activation_cuda = _cpp_extention_load_helper(
"anti_alias_activation_cuda", sources, extra_cuda_flags
)
return anti_alias_activation_cuda
def _get_cuda_bare_metal_version(cuda_dir):
raw_output = subprocess.check_output(
[cuda_dir + "/bin/nvcc", "-V"], universal_newlines=True
)
output = raw_output.split()
release_idx = output.index("release") + 1
release = output[release_idx].split(".")
bare_metal_major = release[0]
bare_metal_minor = release[1][0]
return raw_output, bare_metal_major, bare_metal_minor
def _create_build_dir(buildpath):
try:
os.mkdir(buildpath)
except OSError:
if not os.path.isdir(buildpath):
print(f"Creation of the build directory {buildpath} failed")

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/* coding=utf-8
* Copyright (c) 2020, NVIDIA CORPORATION. All rights reserved.
*
* 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.
*/
#include <ATen/ATen.h>
#include "compat.h"
#define DISPATCH_FLOAT_HALF_AND_BFLOAT(TYPE, NAME, ...) \
switch (TYPE) \
{ \
case at::ScalarType::Float: \
{ \
using scalar_t = float; \
__VA_ARGS__; \
break; \
} \
case at::ScalarType::Half: \
{ \
using scalar_t = at::Half; \
__VA_ARGS__; \
break; \
} \
case at::ScalarType::BFloat16: \
{ \
using scalar_t = at::BFloat16; \
__VA_ARGS__; \
break; \
} \
default: \
AT_ERROR(#NAME, " not implemented for '", toString(TYPE), "'"); \
}
#define DISPATCH_FLOAT_HALF_AND_BFLOAT_INOUT_TYPES(TYPEIN, TYPEOUT, NAME, ...) \
switch (TYPEIN) \
{ \
case at::ScalarType::Float: \
{ \
using scalar_t_in = float; \
switch (TYPEOUT) \
{ \
case at::ScalarType::Float: \
{ \
using scalar_t_out = float; \
__VA_ARGS__; \
break; \
} \
case at::ScalarType::Half: \
{ \
using scalar_t_out = at::Half; \
__VA_ARGS__; \
break; \
} \
case at::ScalarType::BFloat16: \
{ \
using scalar_t_out = at::BFloat16; \
__VA_ARGS__; \
break; \
} \
default: \
AT_ERROR(#NAME, " not implemented for '", toString(TYPEOUT), "'"); \
} \
break; \
} \
case at::ScalarType::Half: \
{ \
using scalar_t_in = at::Half; \
using scalar_t_out = at::Half; \
__VA_ARGS__; \
break; \
} \
case at::ScalarType::BFloat16: \
{ \
using scalar_t_in = at::BFloat16; \
using scalar_t_out = at::BFloat16; \
__VA_ARGS__; \
break; \
} \
default: \
AT_ERROR(#NAME, " not implemented for '", toString(TYPEIN), "'"); \
}

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# Adapted from https://github.com/junjun3518/alias-free-torch under the Apache License 2.0
# LICENSE is in incl_licenses directory.
from .filter import *
from .resample import *
from .act import *

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# Adapted from https://github.com/junjun3518/alias-free-torch under the Apache License 2.0
# LICENSE is in incl_licenses directory.
import torch.nn as nn
from .resample import UpSample1d, DownSample1d
class Activation1d(nn.Module):
def __init__(
self,
activation,
up_ratio: int = 2,
down_ratio: int = 2,
up_kernel_size: int = 12,
down_kernel_size: int = 12,
):
super().__init__()
self.up_ratio = up_ratio
self.down_ratio = down_ratio
self.act = activation
self.upsample = UpSample1d(up_ratio, up_kernel_size)
self.downsample = DownSample1d(down_ratio, down_kernel_size)
# x: [B,C,T]
def forward(self, x):
x = self.upsample(x)
x = self.act(x)
x = self.downsample(x)
return x

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# Adapted from https://github.com/junjun3518/alias-free-torch under the Apache License 2.0
# LICENSE is in incl_licenses directory.
import torch
import torch.nn as nn
import torch.nn.functional as F
import math
if "sinc" in dir(torch):
sinc = torch.sinc
else:
# This code is adopted from adefossez's julius.core.sinc under the MIT License
# https://adefossez.github.io/julius/julius/core.html
# LICENSE is in incl_licenses directory.
def sinc(x: torch.Tensor):
"""
Implementation of sinc, i.e. sin(pi * x) / (pi * x)
__Warning__: Different to julius.sinc, the input is multiplied by `pi`!
"""
return torch.where(
x == 0,
torch.tensor(1.0, device=x.device, dtype=x.dtype),
torch.sin(math.pi * x) / math.pi / x,
)
# This code is adopted from adefossez's julius.lowpass.LowPassFilters under the MIT License
# https://adefossez.github.io/julius/julius/lowpass.html
# LICENSE is in incl_licenses directory.
def kaiser_sinc_filter1d(
cutoff, half_width, kernel_size
): # return filter [1,1,kernel_size]
even = kernel_size % 2 == 0
half_size = kernel_size // 2
# For kaiser window
delta_f = 4 * half_width
A = 2.285 * (half_size - 1) * math.pi * delta_f + 7.95
if A > 50.0:
beta = 0.1102 * (A - 8.7)
elif A >= 21.0:
beta = 0.5842 * (A - 21) ** 0.4 + 0.07886 * (A - 21.0)
else:
beta = 0.0
window = torch.kaiser_window(kernel_size, beta=beta, periodic=False)
# ratio = 0.5/cutoff -> 2 * cutoff = 1 / ratio
if even:
time = torch.arange(-half_size, half_size) + 0.5
else:
time = torch.arange(kernel_size) - half_size
if cutoff == 0:
filter_ = torch.zeros_like(time)
else:
filter_ = 2 * cutoff * window * sinc(2 * cutoff * time)
"""
Normalize filter to have sum = 1, otherwise we will have a small leakage of the constant component in the input signal.
"""
filter_ /= filter_.sum()
filter = filter_.view(1, 1, kernel_size)
return filter
class LowPassFilter1d(nn.Module):
def __init__(
self,
cutoff=0.5,
half_width=0.6,
stride: int = 1,
padding: bool = True,
padding_mode: str = "replicate",
kernel_size: int = 12,
):
"""
kernel_size should be even number for stylegan3 setup, in this implementation, odd number is also possible.
"""
super().__init__()
if cutoff < -0.0:
raise ValueError("Minimum cutoff must be larger than zero.")
if cutoff > 0.5:
raise ValueError("A cutoff above 0.5 does not make sense.")
self.kernel_size = kernel_size
self.even = kernel_size % 2 == 0
self.pad_left = kernel_size // 2 - int(self.even)
self.pad_right = kernel_size // 2
self.stride = stride
self.padding = padding
self.padding_mode = padding_mode
filter = kaiser_sinc_filter1d(cutoff, half_width, kernel_size)
self.register_buffer("filter", filter)
# Input [B, C, T]
def forward(self, x):
_, C, _ = x.shape
if self.padding:
x = F.pad(x, (self.pad_left, self.pad_right), mode=self.padding_mode)
out = F.conv1d(x, self.filter.expand(C, -1, -1), stride=self.stride, groups=C)
return out

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# Adapted from https://github.com/junjun3518/alias-free-torch under the Apache License 2.0
# LICENSE is in incl_licenses directory.
import torch.nn as nn
from torch.nn import functional as F
from .filter import LowPassFilter1d
from .filter import kaiser_sinc_filter1d
class UpSample1d(nn.Module):
def __init__(self, ratio=2, kernel_size=None):
super().__init__()
self.ratio = ratio
self.kernel_size = (
int(6 * ratio // 2) * 2 if kernel_size is None else kernel_size
)
self.stride = ratio
self.pad = self.kernel_size // ratio - 1
self.pad_left = self.pad * self.stride + (self.kernel_size - self.stride) // 2
self.pad_right = (
self.pad * self.stride + (self.kernel_size - self.stride + 1) // 2
)
filter = kaiser_sinc_filter1d(
cutoff=0.5 / ratio, half_width=0.6 / ratio, kernel_size=self.kernel_size
)
self.register_buffer("filter", filter)
# x: [B, C, T]
def forward(self, x):
_, C, _ = x.shape
x = F.pad(x, (self.pad, self.pad), mode="replicate")
x = self.ratio * F.conv_transpose1d(
x, self.filter.expand(C, -1, -1), stride=self.stride, groups=C
)
x = x[..., self.pad_left : -self.pad_right]
return x
class DownSample1d(nn.Module):
def __init__(self, ratio=2, kernel_size=None):
super().__init__()
self.ratio = ratio
self.kernel_size = (
int(6 * ratio // 2) * 2 if kernel_size is None else kernel_size
)
self.lowpass = LowPassFilter1d(
cutoff=0.5 / ratio,
half_width=0.6 / ratio,
stride=ratio,
kernel_size=self.kernel_size,
)
def forward(self, x):
xx = self.lowpass(x)
return xx

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
import os
import json
from pathlib import Path
from typing import Optional, Union, Dict
import torch
import torch.nn as nn
from torch.nn import Conv1d, ConvTranspose1d
from torch.nn.utils import weight_norm, remove_weight_norm
from . import activations
from .utils0 import init_weights, get_padding
from .alias_free_activation.torch.act import Activation1d as TorchActivation1d
from .env import AttrDict
from huggingface_hub import PyTorchModelHubMixin, hf_hub_download
def load_hparams_from_json(path) -> AttrDict:
with open(path) as f:
data = f.read()
return AttrDict(json.loads(data))
class AMPBlock1(torch.nn.Module):
"""
AMPBlock applies Snake / SnakeBeta activation functions with trainable parameters that control periodicity, defined for each layer.
AMPBlock1 has additional self.convs2 that contains additional Conv1d layers with a fixed dilation=1 followed by each layer in self.convs1
Args:
h (AttrDict): Hyperparameters.
channels (int): Number of convolution channels.
kernel_size (int): Size of the convolution kernel. Default is 3.
dilation (tuple): Dilation rates for the convolutions. Each dilation layer has two convolutions. Default is (1, 3, 5).
activation (str): Activation function type. Should be either 'snake' or 'snakebeta'. Default is None.
"""
def __init__(
self,
h: AttrDict,
channels: int,
kernel_size: int = 3,
dilation: tuple = (1, 3, 5),
activation: str = None,
):
super().__init__()
self.h = h
self.convs1 = nn.ModuleList(
[
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
stride=1,
dilation=d,
padding=get_padding(kernel_size, d),
)
)
for d in dilation
]
)
self.convs1.apply(init_weights)
self.convs2 = nn.ModuleList(
[
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
stride=1,
dilation=1,
padding=get_padding(kernel_size, 1),
)
)
for _ in range(len(dilation))
]
)
self.convs2.apply(init_weights)
self.num_layers = len(self.convs1) + len(
self.convs2
) # Total number of conv layers
# Select which Activation1d, lazy-load cuda version to ensure backward compatibility
if self.h.get("use_cuda_kernel", False):
from .alias_free_activation.cuda.activation1d import (
Activation1d as CudaActivation1d,
)
Activation1d = CudaActivation1d
else:
Activation1d = TorchActivation1d
# Activation functions
if activation == "snake":
self.activations = nn.ModuleList(
[
Activation1d(
activation=activations.Snake(
channels, alpha_logscale=h.snake_logscale
)
)
for _ in range(self.num_layers)
]
)
elif activation == "snakebeta":
self.activations = nn.ModuleList(
[
Activation1d(
activation=activations.SnakeBeta(
channels, alpha_logscale=h.snake_logscale
)
)
for _ in range(self.num_layers)
]
)
else:
raise NotImplementedError(
"activation incorrectly specified. check the config file and look for 'activation'."
)
def forward(self, x):
acts1, acts2 = self.activations[::2], self.activations[1::2]
for c1, c2, a1, a2 in zip(self.convs1, self.convs2, acts1, acts2):
xt = a1(x)
xt = c1(xt)
xt = a2(xt)
xt = c2(xt)
x = xt + x
return x
def remove_weight_norm(self):
for l in self.convs1:
remove_weight_norm(l)
for l in self.convs2:
remove_weight_norm(l)
class AMPBlock2(torch.nn.Module):
"""
AMPBlock applies Snake / SnakeBeta activation functions with trainable parameters that control periodicity, defined for each layer.
Unlike AMPBlock1, AMPBlock2 does not contain extra Conv1d layers with fixed dilation=1
Args:
h (AttrDict): Hyperparameters.
channels (int): Number of convolution channels.
kernel_size (int): Size of the convolution kernel. Default is 3.
dilation (tuple): Dilation rates for the convolutions. Each dilation layer has two convolutions. Default is (1, 3, 5).
activation (str): Activation function type. Should be either 'snake' or 'snakebeta'. Default is None.
"""
def __init__(
self,
h: AttrDict,
channels: int,
kernel_size: int = 3,
dilation: tuple = (1, 3, 5),
activation: str = None,
):
super().__init__()
self.h = h
self.convs = nn.ModuleList(
[
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
stride=1,
dilation=d,
padding=get_padding(kernel_size, d),
)
)
for d in dilation
]
)
self.convs.apply(init_weights)
self.num_layers = len(self.convs) # Total number of conv layers
# Select which Activation1d, lazy-load cuda version to ensure backward compatibility
if self.h.get("use_cuda_kernel", False):
from .alias_free_activation.cuda.activation1d import (
Activation1d as CudaActivation1d,
)
Activation1d = CudaActivation1d
else:
Activation1d = TorchActivation1d
# Activation functions
if activation == "snake":
self.activations = nn.ModuleList(
[
Activation1d(
activation=activations.Snake(
channels, alpha_logscale=h.snake_logscale
)
)
for _ in range(self.num_layers)
]
)
elif activation == "snakebeta":
self.activations = nn.ModuleList(
[
Activation1d(
activation=activations.SnakeBeta(
channels, alpha_logscale=h.snake_logscale
)
)
for _ in range(self.num_layers)
]
)
else:
raise NotImplementedError(
"activation incorrectly specified. check the config file and look for 'activation'."
)
def forward(self, x):
for c, a in zip(self.convs, self.activations):
xt = a(x)
xt = c(xt)
x = xt + x
return x
def remove_weight_norm(self):
for l in self.convs:
remove_weight_norm(l)
class BigVGAN(
torch.nn.Module,
PyTorchModelHubMixin,
# library_name="bigvgan",
# repo_url="https://github.com/NVIDIA/BigVGAN",
# docs_url="https://github.com/NVIDIA/BigVGAN/blob/main/README.md",
# pipeline_tag="audio-to-audio",
# license="mit",
# tags=["neural-vocoder", "audio-generation", "arxiv:2206.04658"],
):
"""
BigVGAN is a neural vocoder model that applies anti-aliased periodic activation for residual blocks (resblocks).
New in BigVGAN-v2: it can optionally use optimized CUDA kernels for AMP (anti-aliased multi-periodicity) blocks.
Args:
h (AttrDict): Hyperparameters.
use_cuda_kernel (bool): If set to True, loads optimized CUDA kernels for AMP. This should be used for inference only, as training is not supported with CUDA kernels.
Note:
- The `use_cuda_kernel` parameter should be used for inference only, as training with CUDA kernels is not supported.
- Ensure that the activation function is correctly specified in the hyperparameters (h.activation).
"""
def __init__(self, h: AttrDict, use_cuda_kernel: bool = False):
super().__init__()
self.h = h
self.h["use_cuda_kernel"] = use_cuda_kernel
# Select which Activation1d, lazy-load cuda version to ensure backward compatibility
if self.h.get("use_cuda_kernel", False):
from .alias_free_activation.cuda.activation1d import (
Activation1d as CudaActivation1d,
)
Activation1d = CudaActivation1d
else:
Activation1d = TorchActivation1d
self.num_kernels = len(h.resblock_kernel_sizes)
self.num_upsamples = len(h.upsample_rates)
# Pre-conv
self.conv_pre = weight_norm(
Conv1d(h.num_mels, h.upsample_initial_channel, 7, 1, padding=3)
)
# Define which AMPBlock to use. BigVGAN uses AMPBlock1 as default
if h.resblock == "1":
resblock_class = AMPBlock1
elif h.resblock == "2":
resblock_class = AMPBlock2
else:
raise ValueError(
f"Incorrect resblock class specified in hyperparameters. Got {h.resblock}"
)
# Transposed conv-based upsamplers. does not apply anti-aliasing
self.ups = nn.ModuleList()
for i, (u, k) in enumerate(zip(h.upsample_rates, h.upsample_kernel_sizes)):
self.ups.append(
nn.ModuleList(
[
weight_norm(
ConvTranspose1d(
h.upsample_initial_channel // (2**i),
h.upsample_initial_channel // (2 ** (i + 1)),
k,
u,
padding=(k - u) // 2,
)
)
]
)
)
# Residual blocks using anti-aliased multi-periodicity composition modules (AMP)
self.resblocks = nn.ModuleList()
for i in range(len(self.ups)):
ch = h.upsample_initial_channel // (2 ** (i + 1))
for j, (k, d) in enumerate(
zip(h.resblock_kernel_sizes, h.resblock_dilation_sizes)
):
self.resblocks.append(
resblock_class(h, ch, k, d, activation=h.activation)
)
# Post-conv
activation_post = (
activations.Snake(ch, alpha_logscale=h.snake_logscale)
if h.activation == "snake"
else (
activations.SnakeBeta(ch, alpha_logscale=h.snake_logscale)
if h.activation == "snakebeta"
else None
)
)
if activation_post is None:
raise NotImplementedError(
"activation incorrectly specified. check the config file and look for 'activation'."
)
self.activation_post = Activation1d(activation=activation_post)
# Whether to use bias for the final conv_post. Default to True for backward compatibility
self.use_bias_at_final = h.get("use_bias_at_final", True)
self.conv_post = weight_norm(
Conv1d(ch, 1, 7, 1, padding=3, bias=self.use_bias_at_final)
)
# Weight initialization
for i in range(len(self.ups)):
self.ups[i].apply(init_weights)
self.conv_post.apply(init_weights)
# Final tanh activation. Defaults to True for backward compatibility
self.use_tanh_at_final = h.get("use_tanh_at_final", True)
def forward(self, x):
# Pre-conv
x = self.conv_pre(x)
for i in range(self.num_upsamples):
# Upsampling
for i_up in range(len(self.ups[i])):
x = self.ups[i][i_up](x)
# AMP blocks
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
# Post-conv
x = self.activation_post(x)
x = self.conv_post(x)
# Final tanh activation
if self.use_tanh_at_final:
x = torch.tanh(x)
else:
x = torch.clamp(x, min=-1.0, max=1.0) # Bound the output to [-1, 1]
return x
def remove_weight_norm(self):
try:
# print("Removing weight norm...")
for l in self.ups:
for l_i in l:
remove_weight_norm(l_i)
for l in self.resblocks:
l.remove_weight_norm()
remove_weight_norm(self.conv_pre)
remove_weight_norm(self.conv_post)
except ValueError:
print("[INFO] Model already removed weight norm. Skipping!")
pass
# Additional methods for huggingface_hub support
def _save_pretrained(self, save_directory: Path) -> None:
"""Save weights and config.json from a Pytorch model to a local directory."""
model_path = save_directory / "bigvgan_generator.pt"
torch.save({"generator": self.state_dict()}, model_path)
config_path = save_directory / "config.json"
with open(config_path, "w") as config_file:
json.dump(self.h, config_file, indent=4)
@classmethod
def _from_pretrained(
cls,
*,
model_id: str,
revision: str,
cache_dir: str,
force_download: bool,
proxies: Optional[Dict],
resume_download: bool,
local_files_only: bool,
token: Union[str, bool, None],
map_location: str = "cpu", # Additional argument
strict: bool = False, # Additional argument
use_cuda_kernel: bool = False,
**model_kwargs,
):
"""Load Pytorch pretrained weights and return the loaded model."""
# Download and load hyperparameters (h) used by BigVGAN
if os.path.isdir(model_id):
# print("Loading config.json from local directory")
config_file = os.path.join(model_id, "config.json")
else:
config_file = hf_hub_download(
repo_id=model_id,
filename="config.json",
revision=revision,
cache_dir=cache_dir,
force_download=force_download,
proxies=proxies,
resume_download=resume_download,
token=token,
local_files_only=local_files_only,
)
h = load_hparams_from_json(config_file)
# instantiate BigVGAN using h
if use_cuda_kernel:
print(
f"[WARNING] You have specified use_cuda_kernel=True during BigVGAN.from_pretrained(). Only inference is supported (training is not implemented)!"
)
print(
f"[WARNING] You need nvcc and ninja installed in your system that matches your PyTorch build is using to build the kernel. If not, the model will fail to initialize or generate incorrect waveform!"
)
print(
f"[WARNING] For detail, see the official GitHub repository: https://github.com/NVIDIA/BigVGAN?tab=readme-ov-file#using-custom-cuda-kernel-for-synthesis"
)
model = cls(h, use_cuda_kernel=use_cuda_kernel)
# Download and load pretrained generator weight
if os.path.isdir(model_id):
# print("Loading weights from local directory")
model_file = os.path.join(model_id, "bigvgan_generator.pt")
else:
# print(f"Loading weights from {model_id}")
model_file = hf_hub_download(
repo_id=model_id,
filename="bigvgan_generator.pt",
revision=revision,
cache_dir=cache_dir,
force_download=force_download,
proxies=proxies,
resume_download=resume_download,
token=token,
local_files_only=local_files_only,
)
checkpoint_dict = torch.load(model_file, map_location=map_location)
try:
model.load_state_dict(checkpoint_dict["generator"])
except RuntimeError:
print(
f"[INFO] the pretrained checkpoint does not contain weight norm. Loading the checkpoint after removing weight norm!"
)
model.remove_weight_norm()
model.load_state_dict(checkpoint_dict["generator"])
return model

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{
"resblock": "1",
"num_gpus": 0,
"batch_size": 32,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [4,4,2,2,2,2],
"upsample_kernel_sizes": [8,8,4,4,4,4],
"upsample_initial_channel": 1536,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"activation": "snakebeta",
"snake_logscale": true,
"resolutions": [[1024, 120, 600], [2048, 240, 1200], [512, 50, 240]],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"segment_size": 8192,
"num_mels": 80,
"num_freq": 1025,
"n_fft": 1024,
"hop_size": 256,
"win_size": 1024,
"sampling_rate": 22050,
"fmin": 0,
"fmax": 8000,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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@ -0,0 +1,45 @@
{
"resblock": "1",
"num_gpus": 0,
"batch_size": 32,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [4,4,2,2,2,2],
"upsample_kernel_sizes": [8,8,4,4,4,4],
"upsample_initial_channel": 1536,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"activation": "snakebeta",
"snake_logscale": true,
"resolutions": [[1024, 120, 600], [2048, 240, 1200], [512, 50, 240]],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"segment_size": 8192,
"num_mels": 100,
"num_freq": 1025,
"n_fft": 1024,
"hop_size": 256,
"win_size": 1024,
"sampling_rate": 24000,
"fmin": 0,
"fmax": 12000,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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@ -0,0 +1,45 @@
{
"resblock": "1",
"num_gpus": 0,
"batch_size": 32,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [8,8,2,2],
"upsample_kernel_sizes": [16,16,4,4],
"upsample_initial_channel": 512,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"activation": "snakebeta",
"snake_logscale": true,
"resolutions": [[1024, 120, 600], [2048, 240, 1200], [512, 50, 240]],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"segment_size": 8192,
"num_mels": 80,
"num_freq": 1025,
"n_fft": 1024,
"hop_size": 256,
"win_size": 1024,
"sampling_rate": 22050,
"fmin": 0,
"fmax": 8000,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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@ -0,0 +1,45 @@
{
"resblock": "1",
"num_gpus": 0,
"batch_size": 32,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [8,8,2,2],
"upsample_kernel_sizes": [16,16,4,4],
"upsample_initial_channel": 512,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"activation": "snakebeta",
"snake_logscale": true,
"resolutions": [[1024, 120, 600], [2048, 240, 1200], [512, 50, 240]],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"segment_size": 8192,
"num_mels": 100,
"num_freq": 1025,
"n_fft": 1024,
"hop_size": 256,
"win_size": 1024,
"sampling_rate": 24000,
"fmin": 0,
"fmax": 12000,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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@ -0,0 +1,61 @@
{
"resblock": "1",
"num_gpus": 0,
"batch_size": 4,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [4,4,2,2,2,2],
"upsample_kernel_sizes": [8,8,4,4,4,4],
"upsample_initial_channel": 1536,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"use_tanh_at_final": false,
"use_bias_at_final": false,
"activation": "snakebeta",
"snake_logscale": true,
"use_cqtd_instead_of_mrd": true,
"cqtd_filters": 128,
"cqtd_max_filters": 1024,
"cqtd_filters_scale": 1,
"cqtd_dilations": [1, 2, 4],
"cqtd_hop_lengths": [512, 256, 256],
"cqtd_n_octaves": [9, 9, 9],
"cqtd_bins_per_octaves": [24, 36, 48],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"use_multiscale_melloss": true,
"lambda_melloss": 15,
"clip_grad_norm": 500,
"segment_size": 65536,
"num_mels": 80,
"num_freq": 1025,
"n_fft": 1024,
"hop_size": 256,
"win_size": 1024,
"sampling_rate": 22050,
"fmin": 0,
"fmax": null,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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@ -0,0 +1,61 @@
{
"resblock": "1",
"num_gpus": 0,
"batch_size": 4,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [4,4,2,2,2,2],
"upsample_kernel_sizes": [8,8,4,4,4,4],
"upsample_initial_channel": 1536,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"use_tanh_at_final": false,
"use_bias_at_final": false,
"activation": "snakebeta",
"snake_logscale": true,
"use_cqtd_instead_of_mrd": true,
"cqtd_filters": 128,
"cqtd_max_filters": 1024,
"cqtd_filters_scale": 1,
"cqtd_dilations": [1, 2, 4],
"cqtd_hop_lengths": [512, 256, 256],
"cqtd_n_octaves": [9, 9, 9],
"cqtd_bins_per_octaves": [24, 36, 48],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"use_multiscale_melloss": true,
"lambda_melloss": 15,
"clip_grad_norm": 500,
"segment_size": 65536,
"num_mels": 80,
"num_freq": 1025,
"n_fft": 1024,
"hop_size": 256,
"win_size": 1024,
"sampling_rate": 22050,
"fmin": 0,
"fmax": 8000,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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@ -0,0 +1,61 @@
{
"resblock": "1",
"num_gpus": 0,
"batch_size": 4,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [4,4,2,2,2,2],
"upsample_kernel_sizes": [8,8,4,4,4,4],
"upsample_initial_channel": 1536,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"use_tanh_at_final": false,
"use_bias_at_final": false,
"activation": "snakebeta",
"snake_logscale": true,
"use_cqtd_instead_of_mrd": true,
"cqtd_filters": 128,
"cqtd_max_filters": 1024,
"cqtd_filters_scale": 1,
"cqtd_dilations": [1, 2, 4],
"cqtd_hop_lengths": [512, 256, 256],
"cqtd_n_octaves": [9, 9, 9],
"cqtd_bins_per_octaves": [24, 36, 48],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"use_multiscale_melloss": true,
"lambda_melloss": 15,
"clip_grad_norm": 500,
"segment_size": 65536,
"num_mels": 100,
"num_freq": 1025,
"n_fft": 1024,
"hop_size": 256,
"win_size": 1024,
"sampling_rate": 24000,
"fmin": 0,
"fmax": null,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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@ -0,0 +1,61 @@
{
"resblock": "1",
"num_gpus": 0,
"batch_size": 4,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [4,4,2,2,2,2],
"upsample_kernel_sizes": [8,8,4,4,4,4],
"upsample_initial_channel": 1536,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"use_tanh_at_final": false,
"use_bias_at_final": false,
"activation": "snakebeta",
"snake_logscale": true,
"use_cqtd_instead_of_mrd": true,
"cqtd_filters": 128,
"cqtd_max_filters": 1024,
"cqtd_filters_scale": 1,
"cqtd_dilations": [1, 2, 4],
"cqtd_hop_lengths": [512, 256, 256],
"cqtd_n_octaves": [9, 9, 9],
"cqtd_bins_per_octaves": [24, 36, 48],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"use_multiscale_melloss": true,
"lambda_melloss": 15,
"clip_grad_norm": 500,
"segment_size": 65536,
"num_mels": 128,
"num_freq": 1025,
"n_fft": 1024,
"hop_size": 256,
"win_size": 1024,
"sampling_rate": 44100,
"fmin": 0,
"fmax": null,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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@ -0,0 +1,61 @@
{
"resblock": "1",
"num_gpus": 0,
"batch_size": 4,
"learning_rate": 0.0001,
"adam_b1": 0.8,
"adam_b2": 0.99,
"lr_decay": 0.9999996,
"seed": 1234,
"upsample_rates": [8,4,2,2,2,2],
"upsample_kernel_sizes": [16,8,4,4,4,4],
"upsample_initial_channel": 1536,
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"use_tanh_at_final": false,
"use_bias_at_final": false,
"activation": "snakebeta",
"snake_logscale": true,
"use_cqtd_instead_of_mrd": true,
"cqtd_filters": 128,
"cqtd_max_filters": 1024,
"cqtd_filters_scale": 1,
"cqtd_dilations": [1, 2, 4],
"cqtd_hop_lengths": [512, 256, 256],
"cqtd_n_octaves": [9, 9, 9],
"cqtd_bins_per_octaves": [24, 36, 48],
"mpd_reshapes": [2, 3, 5, 7, 11],
"use_spectral_norm": false,
"discriminator_channel_mult": 1,
"use_multiscale_melloss": true,
"lambda_melloss": 15,
"clip_grad_norm": 500,
"segment_size": 65536,
"num_mels": 128,
"num_freq": 2049,
"n_fft": 2048,
"hop_size": 512,
"win_size": 2048,
"sampling_rate": 44100,
"fmin": 0,
"fmax": null,
"fmax_for_loss": null,
"num_workers": 4,
"dist_config": {
"dist_backend": "nccl",
"dist_url": "tcp://localhost:54321",
"world_size": 1
}
}

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
import torch
import torch.nn.functional as F
import torch.nn as nn
from torch.nn import Conv2d
from torch.nn.utils import weight_norm, spectral_norm
from torchaudio.transforms import Spectrogram, Resample
from env import AttrDict
from utils import get_padding
import typing
from typing import Optional, List, Union, Dict, Tuple
class DiscriminatorP(torch.nn.Module):
def __init__(
self,
h: AttrDict,
period: List[int],
kernel_size: int = 5,
stride: int = 3,
use_spectral_norm: bool = False,
):
super().__init__()
self.period = period
self.d_mult = h.discriminator_channel_mult
norm_f = weight_norm if not use_spectral_norm else spectral_norm
self.convs = nn.ModuleList(
[
norm_f(
Conv2d(
1,
int(32 * self.d_mult),
(kernel_size, 1),
(stride, 1),
padding=(get_padding(5, 1), 0),
)
),
norm_f(
Conv2d(
int(32 * self.d_mult),
int(128 * self.d_mult),
(kernel_size, 1),
(stride, 1),
padding=(get_padding(5, 1), 0),
)
),
norm_f(
Conv2d(
int(128 * self.d_mult),
int(512 * self.d_mult),
(kernel_size, 1),
(stride, 1),
padding=(get_padding(5, 1), 0),
)
),
norm_f(
Conv2d(
int(512 * self.d_mult),
int(1024 * self.d_mult),
(kernel_size, 1),
(stride, 1),
padding=(get_padding(5, 1), 0),
)
),
norm_f(
Conv2d(
int(1024 * self.d_mult),
int(1024 * self.d_mult),
(kernel_size, 1),
1,
padding=(2, 0),
)
),
]
)
self.conv_post = norm_f(
Conv2d(int(1024 * self.d_mult), 1, (3, 1), 1, padding=(1, 0))
)
def forward(self, x: torch.Tensor) -> Tuple[torch.Tensor, List[torch.Tensor]]:
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, 0.1)
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, h: AttrDict):
super().__init__()
self.mpd_reshapes = h.mpd_reshapes
print(f"mpd_reshapes: {self.mpd_reshapes}")
self.discriminators = nn.ModuleList(
[
DiscriminatorP(h, rs, use_spectral_norm=h.use_spectral_norm)
for rs in self.mpd_reshapes
]
)
def forward(self, y: torch.Tensor, y_hat: torch.Tensor) -> 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 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)
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, cfg: AttrDict, resolution: List[List[int]]):
super().__init__()
self.resolution = resolution
assert (
len(self.resolution) == 3
), f"MRD layer requires list with len=3, got {self.resolution}"
self.lrelu_slope = 0.1
norm_f = weight_norm if cfg.use_spectral_norm == False else spectral_norm
if hasattr(cfg, "mrd_use_spectral_norm"):
print(
f"[INFO] overriding MRD use_spectral_norm as {cfg.mrd_use_spectral_norm}"
)
norm_f = (
weight_norm if cfg.mrd_use_spectral_norm == False else spectral_norm
)
self.d_mult = cfg.discriminator_channel_mult
if hasattr(cfg, "mrd_channel_mult"):
print(f"[INFO] overriding mrd channel multiplier as {cfg.mrd_channel_mult}")
self.d_mult = cfg.mrd_channel_mult
self.convs = nn.ModuleList(
[
norm_f(nn.Conv2d(1, int(32 * self.d_mult), (3, 9), padding=(1, 4))),
norm_f(
nn.Conv2d(
int(32 * self.d_mult),
int(32 * self.d_mult),
(3, 9),
stride=(1, 2),
padding=(1, 4),
)
),
norm_f(
nn.Conv2d(
int(32 * self.d_mult),
int(32 * self.d_mult),
(3, 9),
stride=(1, 2),
padding=(1, 4),
)
),
norm_f(
nn.Conv2d(
int(32 * self.d_mult),
int(32 * self.d_mult),
(3, 9),
stride=(1, 2),
padding=(1, 4),
)
),
norm_f(
nn.Conv2d(
int(32 * self.d_mult),
int(32 * self.d_mult),
(3, 3),
padding=(1, 1),
)
),
]
)
self.conv_post = norm_f(
nn.Conv2d(int(32 * self.d_mult), 1, (3, 3), padding=(1, 1))
)
def forward(self, x: torch.Tensor) -> Tuple[torch.Tensor, List[torch.Tensor]]:
fmap = []
x = self.spectrogram(x)
x = x.unsqueeze(1)
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, self.lrelu_slope)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
def spectrogram(self, x: torch.Tensor) -> torch.Tensor:
n_fft, hop_length, win_length = self.resolution
x = F.pad(
x,
(int((n_fft - hop_length) / 2), int((n_fft - hop_length) / 2)),
mode="reflect",
)
x = x.squeeze(1)
x = torch.stft(
x,
n_fft=n_fft,
hop_length=hop_length,
win_length=win_length,
center=False,
return_complex=True,
)
x = torch.view_as_real(x) # [B, F, TT, 2]
mag = torch.norm(x, p=2, dim=-1) # [B, F, TT]
return mag
class MultiResolutionDiscriminator(nn.Module):
def __init__(self, cfg, debug=False):
super().__init__()
self.resolutions = cfg.resolutions
assert (
len(self.resolutions) == 3
), f"MRD requires list of list with len=3, each element having a list with len=3. Got {self.resolutions}"
self.discriminators = nn.ModuleList(
[DiscriminatorR(cfg, resolution) for resolution in self.resolutions]
)
def forward(self, y: torch.Tensor, y_hat: torch.Tensor) -> 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 i, d in enumerate(self.discriminators):
y_d_r, fmap_r = d(x=y)
y_d_g, fmap_g = d(x=y_hat)
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
# Method based on descript-audio-codec: https://github.com/descriptinc/descript-audio-codec
# Modified code adapted from https://github.com/gemelo-ai/vocos under the MIT license.
# LICENSE is in incl_licenses directory.
class DiscriminatorB(nn.Module):
def __init__(
self,
window_length: int,
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))])
self.conv_post = weight_norm(
nn.Conv2d(channels, 1, (3, 3), (1, 1), padding=(1, 1))
)
def spectrogram(self, x: torch.Tensor) -> List[torch.Tensor]:
# 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 = x.permute(0, 3, 2, 1) # [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) -> Tuple[torch.Tensor, List[torch.Tensor]]:
x_bands = self.spectrogram(x.squeeze(1))
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)
x = self.conv_post(x)
fmap.append(x)
return x, fmap
# Method based on descript-audio-codec: https://github.com/descriptinc/descript-audio-codec
# Modified code adapted from https://github.com/gemelo-ai/vocos under the MIT license.
# LICENSE is in incl_licenses directory.
class MultiBandDiscriminator(nn.Module):
def __init__(
self,
h,
):
"""
Multi-band multi-scale STFT discriminator, with the architecture based on https://github.com/descriptinc/descript-audio-codec.
and the modified code adapted from https://github.com/gemelo-ai/vocos.
"""
super().__init__()
# fft_sizes (list[int]): Tuple of window lengths for FFT. Defaults to [2048, 1024, 512] if not set in h.
self.fft_sizes = h.get("mbd_fft_sizes", [2048, 1024, 512])
self.discriminators = nn.ModuleList(
[DiscriminatorB(window_length=w) for w in self.fft_sizes]
)
def forward(self, y: torch.Tensor, y_hat: torch.Tensor) -> 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)
y_d_g, fmap_g = d(x=y_hat)
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
# Adapted from https://github.com/open-mmlab/Amphion/blob/main/models/vocoders/gan/discriminator/mssbcqtd.py under the MIT license.
# LICENSE is in incl_licenses directory.
class DiscriminatorCQT(nn.Module):
def __init__(self, cfg: AttrDict, hop_length: int, n_octaves:int, bins_per_octave: int):
super().__init__()
self.cfg = cfg
self.filters = cfg["cqtd_filters"]
self.max_filters = cfg["cqtd_max_filters"]
self.filters_scale = cfg["cqtd_filters_scale"]
self.kernel_size = (3, 9)
self.dilations = cfg["cqtd_dilations"]
self.stride = (1, 2)
self.in_channels = cfg["cqtd_in_channels"]
self.out_channels = cfg["cqtd_out_channels"]
self.fs = cfg["sampling_rate"]
self.hop_length = hop_length
self.n_octaves = n_octaves
self.bins_per_octave = bins_per_octave
# Lazy-load
from nnAudio import features
self.cqt_transform = features.cqt.CQT2010v2(
sr=self.fs * 2,
hop_length=self.hop_length,
n_bins=self.bins_per_octave * self.n_octaves,
bins_per_octave=self.bins_per_octave,
output_format="Complex",
pad_mode="constant",
)
self.conv_pres = nn.ModuleList()
for _ in range(self.n_octaves):
self.conv_pres.append(
nn.Conv2d(
self.in_channels * 2,
self.in_channels * 2,
kernel_size=self.kernel_size,
padding=self.get_2d_padding(self.kernel_size),
)
)
self.convs = nn.ModuleList()
self.convs.append(
nn.Conv2d(
self.in_channels * 2,
self.filters,
kernel_size=self.kernel_size,
padding=self.get_2d_padding(self.kernel_size),
)
)
in_chs = min(self.filters_scale * self.filters, self.max_filters)
for i, dilation in enumerate(self.dilations):
out_chs = min(
(self.filters_scale ** (i + 1)) * self.filters, self.max_filters
)
self.convs.append(
weight_norm(
nn.Conv2d(
in_chs,
out_chs,
kernel_size=self.kernel_size,
stride=self.stride,
dilation=(dilation, 1),
padding=self.get_2d_padding(self.kernel_size, (dilation, 1)),
)
)
)
in_chs = out_chs
out_chs = min(
(self.filters_scale ** (len(self.dilations) + 1)) * self.filters,
self.max_filters,
)
self.convs.append(
weight_norm(
nn.Conv2d(
in_chs,
out_chs,
kernel_size=(self.kernel_size[0], self.kernel_size[0]),
padding=self.get_2d_padding(
(self.kernel_size[0], self.kernel_size[0])
),
)
)
)
self.conv_post = weight_norm(
nn.Conv2d(
out_chs,
self.out_channels,
kernel_size=(self.kernel_size[0], self.kernel_size[0]),
padding=self.get_2d_padding((self.kernel_size[0], self.kernel_size[0])),
)
)
self.activation = torch.nn.LeakyReLU(negative_slope=0.1)
self.resample = Resample(orig_freq=self.fs, new_freq=self.fs * 2)
self.cqtd_normalize_volume = self.cfg.get("cqtd_normalize_volume", False)
if self.cqtd_normalize_volume:
print(
f"[INFO] cqtd_normalize_volume set to True. Will apply DC offset removal & peak volume normalization in CQTD!"
)
def get_2d_padding(
self,
kernel_size: typing.Tuple[int, int],
dilation: typing.Tuple[int, int] = (1, 1),
):
return (
((kernel_size[0] - 1) * dilation[0]) // 2,
((kernel_size[1] - 1) * dilation[1]) // 2,
)
def forward(self, x: torch.tensor) -> Tuple[torch.Tensor, List[torch.Tensor]]:
fmap = []
if self.cqtd_normalize_volume:
# 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.resample(x)
z = self.cqt_transform(x)
z_amplitude = z[:, :, :, 0].unsqueeze(1)
z_phase = z[:, :, :, 1].unsqueeze(1)
z = torch.cat([z_amplitude, z_phase], dim=1)
z = torch.permute(z, (0, 1, 3, 2)) # [B, C, W, T] -> [B, C, T, W]
latent_z = []
for i in range(self.n_octaves):
latent_z.append(
self.conv_pres[i](
z[
:,
:,
:,
i * self.bins_per_octave : (i + 1) * self.bins_per_octave,
]
)
)
latent_z = torch.cat(latent_z, dim=-1)
for i, l in enumerate(self.convs):
latent_z = l(latent_z)
latent_z = self.activation(latent_z)
fmap.append(latent_z)
latent_z = self.conv_post(latent_z)
return latent_z, fmap
class MultiScaleSubbandCQTDiscriminator(nn.Module):
def __init__(self, cfg: AttrDict):
super().__init__()
self.cfg = cfg
# Using get with defaults
self.cfg["cqtd_filters"] = self.cfg.get("cqtd_filters", 32)
self.cfg["cqtd_max_filters"] = self.cfg.get("cqtd_max_filters", 1024)
self.cfg["cqtd_filters_scale"] = self.cfg.get("cqtd_filters_scale", 1)
self.cfg["cqtd_dilations"] = self.cfg.get("cqtd_dilations", [1, 2, 4])
self.cfg["cqtd_in_channels"] = self.cfg.get("cqtd_in_channels", 1)
self.cfg["cqtd_out_channels"] = self.cfg.get("cqtd_out_channels", 1)
# Multi-scale params to loop over
self.cfg["cqtd_hop_lengths"] = self.cfg.get("cqtd_hop_lengths", [512, 256, 256])
self.cfg["cqtd_n_octaves"] = self.cfg.get("cqtd_n_octaves", [9, 9, 9])
self.cfg["cqtd_bins_per_octaves"] = self.cfg.get(
"cqtd_bins_per_octaves", [24, 36, 48]
)
self.discriminators = nn.ModuleList(
[
DiscriminatorCQT(
self.cfg,
hop_length=self.cfg["cqtd_hop_lengths"][i],
n_octaves=self.cfg["cqtd_n_octaves"][i],
bins_per_octave=self.cfg["cqtd_bins_per_octaves"][i],
)
for i in range(len(self.cfg["cqtd_hop_lengths"]))
]
)
def forward(self, y: torch.Tensor, y_hat: torch.Tensor) -> 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 disc in self.discriminators:
y_d_r, fmap_r = disc(y)
y_d_g, fmap_g = disc(y_hat)
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 CombinedDiscriminator(nn.Module):
"""
Wrapper of chaining multiple discrimiantor architectures.
Example: combine mbd and cqtd as a single class
"""
def __init__(self, list_discriminator: List[nn.Module]):
super().__init__()
self.discrimiantor = nn.ModuleList(list_discriminator)
def forward(self, y: torch.Tensor, y_hat: torch.Tensor) -> 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 disc in self.discrimiantor:
y_d_r, y_d_g, fmap_r, fmap_g = disc(y, y_hat)
y_d_rs.extend(y_d_r)
fmap_rs.extend(fmap_r)
y_d_gs.extend(y_d_g)
fmap_gs.extend(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs

18
GPT_SoVITS/BigVGAN/env.py Normal file
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# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
import os
import shutil
class AttrDict(dict):
def __init__(self, *args, **kwargs):
super(AttrDict, self).__init__(*args, **kwargs)
self.__dict__ = self
def build_env(config, config_name, path):
t_path = os.path.join(path, config_name)
if config != t_path:
os.makedirs(path, exist_ok=True)
shutil.copyfile(config, os.path.join(path, config_name))

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MIT License
Copyright (c) 2020 Jungil Kong
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.

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MIT License
Copyright (c) 2020 Edward Dixon
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.

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MIT License
Copyright (c) 2023-present, Descript
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.

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MIT License
Copyright (c) 2023 Charactr Inc.
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.

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MIT License
Copyright (c) 2023 Amphion
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.

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# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
from __future__ import absolute_import, division, print_function, unicode_literals
import os
import argparse
import json
import torch
import librosa
from utils import load_checkpoint
from meldataset import get_mel_spectrogram
from scipy.io.wavfile import write
from env import AttrDict
from meldataset import MAX_WAV_VALUE
from bigvgan import BigVGAN as Generator
h = None
device = None
torch.backends.cudnn.benchmark = False
def inference(a, h):
generator = Generator(h, use_cuda_kernel=a.use_cuda_kernel).to(device)
state_dict_g = load_checkpoint(a.checkpoint_file, device)
generator.load_state_dict(state_dict_g["generator"])
filelist = os.listdir(a.input_wavs_dir)
os.makedirs(a.output_dir, exist_ok=True)
generator.eval()
generator.remove_weight_norm()
with torch.no_grad():
for i, filname in enumerate(filelist):
# Load the ground truth audio and resample if necessary
wav, sr = librosa.load(
os.path.join(a.input_wavs_dir, filname), sr=h.sampling_rate, mono=True
)
wav = torch.FloatTensor(wav).to(device)
# Compute mel spectrogram from the ground truth audio
x = get_mel_spectrogram(wav.unsqueeze(0), generator.h)
y_g_hat = generator(x)
audio = y_g_hat.squeeze()
audio = audio * MAX_WAV_VALUE
audio = audio.cpu().numpy().astype("int16")
output_file = os.path.join(
a.output_dir, os.path.splitext(filname)[0] + "_generated.wav"
)
write(output_file, h.sampling_rate, audio)
print(output_file)
def main():
print("Initializing Inference Process..")
parser = argparse.ArgumentParser()
parser.add_argument("--input_wavs_dir", default="test_files")
parser.add_argument("--output_dir", default="generated_files")
parser.add_argument("--checkpoint_file", required=True)
parser.add_argument("--use_cuda_kernel", action="store_true", default=False)
a = parser.parse_args()
config_file = os.path.join(os.path.split(a.checkpoint_file)[0], "config.json")
with open(config_file) as f:
data = f.read()
global h
json_config = json.loads(data)
h = AttrDict(json_config)
torch.manual_seed(h.seed)
global device
if torch.cuda.is_available():
torch.cuda.manual_seed(h.seed)
device = torch.device("cuda")
else:
device = torch.device("cpu")
inference(a, h)
if __name__ == "__main__":
main()

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# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
from __future__ import absolute_import, division, print_function, unicode_literals
import glob
import os
import numpy as np
import argparse
import json
import torch
from scipy.io.wavfile import write
from env import AttrDict
from meldataset import MAX_WAV_VALUE
from bigvgan import BigVGAN as Generator
h = None
device = None
torch.backends.cudnn.benchmark = False
def load_checkpoint(filepath, device):
assert os.path.isfile(filepath)
print(f"Loading '{filepath}'")
checkpoint_dict = torch.load(filepath, map_location=device)
print("Complete.")
return checkpoint_dict
def scan_checkpoint(cp_dir, prefix):
pattern = os.path.join(cp_dir, prefix + "*")
cp_list = glob.glob(pattern)
if len(cp_list) == 0:
return ""
return sorted(cp_list)[-1]
def inference(a, h):
generator = Generator(h, use_cuda_kernel=a.use_cuda_kernel).to(device)
state_dict_g = load_checkpoint(a.checkpoint_file, device)
generator.load_state_dict(state_dict_g["generator"])
filelist = os.listdir(a.input_mels_dir)
os.makedirs(a.output_dir, exist_ok=True)
generator.eval()
generator.remove_weight_norm()
with torch.no_grad():
for i, filname in enumerate(filelist):
# Load the mel spectrogram in .npy format
x = np.load(os.path.join(a.input_mels_dir, filname))
x = torch.FloatTensor(x).to(device)
if len(x.shape) == 2:
x = x.unsqueeze(0)
y_g_hat = generator(x)
audio = y_g_hat.squeeze()
audio = audio * MAX_WAV_VALUE
audio = audio.cpu().numpy().astype("int16")
output_file = os.path.join(
a.output_dir, os.path.splitext(filname)[0] + "_generated_e2e.wav"
)
write(output_file, h.sampling_rate, audio)
print(output_file)
def main():
print("Initializing Inference Process..")
parser = argparse.ArgumentParser()
parser.add_argument("--input_mels_dir", default="test_mel_files")
parser.add_argument("--output_dir", default="generated_files_from_mel")
parser.add_argument("--checkpoint_file", required=True)
parser.add_argument("--use_cuda_kernel", action="store_true", default=False)
a = parser.parse_args()
config_file = os.path.join(os.path.split(a.checkpoint_file)[0], "config.json")
with open(config_file) as f:
data = f.read()
global h
json_config = json.loads(data)
h = AttrDict(json_config)
torch.manual_seed(h.seed)
global device
if torch.cuda.is_available():
torch.cuda.manual_seed(h.seed)
device = torch.device("cuda")
else:
device = torch.device("cpu")
inference(a, h)
if __name__ == "__main__":
main()

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GPT_SoVITS/BigVGAN/loss.py Normal file
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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
import torch
import torch.nn.functional as F
import torch.nn as nn
from librosa.filters import mel as librosa_mel_fn
from scipy import signal
import typing
from typing import Optional, List, Union, Dict, Tuple
from collections import namedtuple
import math
import functools
# Adapted from https://github.com/descriptinc/descript-audio-codec/blob/main/dac/nn/loss.py under the MIT license.
# LICENSE is in incl_licenses directory.
class MultiScaleMelSpectrogramLoss(nn.Module):
"""Compute distance between mel spectrograms. Can be used
in a multi-scale way.
Parameters
----------
n_mels : List[int]
Number of mels per STFT, by default [5, 10, 20, 40, 80, 160, 320],
window_lengths : List[int], optional
Length of each window of each STFT, by default [32, 64, 128, 256, 512, 1024, 2048]
loss_fn : typing.Callable, optional
How to compare each loss, by default nn.L1Loss()
clamp_eps : float, optional
Clamp on the log magnitude, below, by default 1e-5
mag_weight : float, optional
Weight of raw magnitude portion of loss, by default 0.0 (no ampliciation on mag part)
log_weight : float, optional
Weight of log magnitude portion of loss, by default 1.0
pow : float, optional
Power to raise magnitude to before taking log, by default 1.0
weight : float, optional
Weight of this loss, by default 1.0
match_stride : bool, optional
Whether to match the stride of convolutional layers, by default False
Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/spectral.py
Additional code copied and modified from https://github.com/descriptinc/audiotools/blob/master/audiotools/core/audio_signal.py
"""
def __init__(
self,
sampling_rate: int,
n_mels: List[int] = [5, 10, 20, 40, 80, 160, 320],
window_lengths: List[int] = [32, 64, 128, 256, 512, 1024, 2048],
loss_fn: typing.Callable = nn.L1Loss(),
clamp_eps: float = 1e-5,
mag_weight: float = 0.0,
log_weight: float = 1.0,
pow: float = 1.0,
weight: float = 1.0,
match_stride: bool = False,
mel_fmin: List[float] = [0, 0, 0, 0, 0, 0, 0],
mel_fmax: List[float] = [None, None, None, None, None, None, None],
window_type: str = "hann",
):
super().__init__()
self.sampling_rate = sampling_rate
STFTParams = namedtuple(
"STFTParams",
["window_length", "hop_length", "window_type", "match_stride"],
)
self.stft_params = [
STFTParams(
window_length=w,
hop_length=w // 4,
match_stride=match_stride,
window_type=window_type,
)
for w in window_lengths
]
self.n_mels = n_mels
self.loss_fn = loss_fn
self.clamp_eps = clamp_eps
self.log_weight = log_weight
self.mag_weight = mag_weight
self.weight = weight
self.mel_fmin = mel_fmin
self.mel_fmax = mel_fmax
self.pow = pow
@staticmethod
@functools.lru_cache(None)
def get_window(
window_type,
window_length,
):
return signal.get_window(window_type, window_length)
@staticmethod
@functools.lru_cache(None)
def get_mel_filters(sr, n_fft, n_mels, fmin, fmax):
return librosa_mel_fn(sr=sr, n_fft=n_fft, n_mels=n_mels, fmin=fmin, fmax=fmax)
def mel_spectrogram(
self,
wav,
n_mels,
fmin,
fmax,
window_length,
hop_length,
match_stride,
window_type,
):
"""
Mirrors AudioSignal.mel_spectrogram used by BigVGAN-v2 training from:
https://github.com/descriptinc/audiotools/blob/master/audiotools/core/audio_signal.py
"""
B, C, T = wav.shape
if match_stride:
assert (
hop_length == window_length // 4
), "For match_stride, hop must equal n_fft // 4"
right_pad = math.ceil(T / hop_length) * hop_length - T
pad = (window_length - hop_length) // 2
else:
right_pad = 0
pad = 0
wav = torch.nn.functional.pad(wav, (pad, pad + right_pad), mode="reflect")
window = self.get_window(window_type, window_length)
window = torch.from_numpy(window).to(wav.device).float()
stft = torch.stft(
wav.reshape(-1, T),
n_fft=window_length,
hop_length=hop_length,
window=window,
return_complex=True,
center=True,
)
_, nf, nt = stft.shape
stft = stft.reshape(B, C, nf, nt)
if match_stride:
"""
Drop first two and last two frames, which are added, because of padding. Now num_frames * hop_length = num_samples.
"""
stft = stft[..., 2:-2]
magnitude = torch.abs(stft)
nf = magnitude.shape[2]
mel_basis = self.get_mel_filters(
self.sampling_rate, 2 * (nf - 1), n_mels, fmin, fmax
)
mel_basis = torch.from_numpy(mel_basis).to(wav.device)
mel_spectrogram = magnitude.transpose(2, -1) @ mel_basis.T
mel_spectrogram = mel_spectrogram.transpose(-1, 2)
return mel_spectrogram
def forward(self, x: torch.Tensor, y: torch.Tensor) -> torch.Tensor:
"""Computes mel loss between an estimate and a reference
signal.
Parameters
----------
x : torch.Tensor
Estimate signal
y : torch.Tensor
Reference signal
Returns
-------
torch.Tensor
Mel loss.
"""
loss = 0.0
for n_mels, fmin, fmax, s in zip(
self.n_mels, self.mel_fmin, self.mel_fmax, self.stft_params
):
kwargs = {
"n_mels": n_mels,
"fmin": fmin,
"fmax": fmax,
"window_length": s.window_length,
"hop_length": s.hop_length,
"match_stride": s.match_stride,
"window_type": s.window_type,
}
x_mels = self.mel_spectrogram(x, **kwargs)
y_mels = self.mel_spectrogram(y, **kwargs)
x_logmels = torch.log(
x_mels.clamp(min=self.clamp_eps).pow(self.pow)
) / torch.log(torch.tensor(10.0))
y_logmels = torch.log(
y_mels.clamp(min=self.clamp_eps).pow(self.pow)
) / torch.log(torch.tensor(10.0))
loss += self.log_weight * self.loss_fn(x_logmels, y_logmels)
loss += self.mag_weight * self.loss_fn(x_logmels, y_logmels)
return loss
# Loss functions
def feature_loss(
fmap_r: List[List[torch.Tensor]], fmap_g: List[List[torch.Tensor]]
) -> torch.Tensor:
loss = 0
for dr, dg in zip(fmap_r, fmap_g):
for rl, gl in zip(dr, dg):
loss += torch.mean(torch.abs(rl - gl))
return loss * 2 # This equates to lambda=2.0 for the feature matching loss
def discriminator_loss(
disc_real_outputs: List[torch.Tensor], disc_generated_outputs: List[torch.Tensor]
) -> Tuple[torch.Tensor, List[torch.Tensor], List[torch.Tensor]]:
loss = 0
r_losses = []
g_losses = []
for dr, dg in zip(disc_real_outputs, disc_generated_outputs):
r_loss = torch.mean((1 - dr) ** 2)
g_loss = torch.mean(dg**2)
loss += r_loss + g_loss
r_losses.append(r_loss.item())
g_losses.append(g_loss.item())
return loss, r_losses, g_losses
def generator_loss(
disc_outputs: List[torch.Tensor],
) -> Tuple[torch.Tensor, List[torch.Tensor]]:
loss = 0
gen_losses = []
for dg in disc_outputs:
l = torch.mean((1 - dg) ** 2)
gen_losses.append(l)
loss += l
return loss, gen_losses

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
import math
import os
import random
import torch
import torch.utils.data
import numpy as np
import librosa
from librosa.filters import mel as librosa_mel_fn
import pathlib
from tqdm import tqdm
from typing import List, Tuple, Optional
from .env import AttrDict
MAX_WAV_VALUE = 32767.0 # NOTE: 32768.0 -1 to prevent int16 overflow (results in popping sound in corner cases)
def dynamic_range_compression(x, C=1, clip_val=1e-5):
return np.log(np.clip(x, a_min=clip_val, a_max=None) * C)
def dynamic_range_decompression(x, C=1):
return np.exp(x) / C
def dynamic_range_compression_torch(x, C=1, clip_val=1e-5):
return torch.log(torch.clamp(x, min=clip_val) * C)
def dynamic_range_decompression_torch(x, C=1):
return torch.exp(x) / C
def spectral_normalize_torch(magnitudes):
return dynamic_range_compression_torch(magnitudes)
def spectral_de_normalize_torch(magnitudes):
return dynamic_range_decompression_torch(magnitudes)
mel_basis_cache = {}
hann_window_cache = {}
def mel_spectrogram(
y: torch.Tensor,
n_fft: int,
num_mels: int,
sampling_rate: int,
hop_size: int,
win_size: int,
fmin: int,
fmax: int = None,
center: bool = False,
) -> torch.Tensor:
"""
Calculate the mel spectrogram of an input signal.
This function uses slaney norm for the librosa mel filterbank (using librosa.filters.mel) and uses Hann window for STFT (using torch.stft).
Args:
y (torch.Tensor): Input signal.
n_fft (int): FFT size.
num_mels (int): Number of mel bins.
sampling_rate (int): Sampling rate of the input signal.
hop_size (int): Hop size for STFT.
win_size (int): Window size for STFT.
fmin (int): Minimum frequency for mel filterbank.
fmax (int): Maximum frequency for mel filterbank. If None, defaults to half the sampling rate (fmax = sr / 2.0) inside librosa_mel_fn
center (bool): Whether to pad the input to center the frames. Default is False.
Returns:
torch.Tensor: Mel spectrogram.
"""
if torch.min(y) < -1.0:
print(f"[WARNING] Min value of input waveform signal is {torch.min(y)}")
if torch.max(y) > 1.0:
print(f"[WARNING] Max value of input waveform signal is {torch.max(y)}")
device = y.device
key = f"{n_fft}_{num_mels}_{sampling_rate}_{hop_size}_{win_size}_{fmin}_{fmax}_{device}"
if key not in mel_basis_cache:
mel = librosa_mel_fn(
sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax
)
mel_basis_cache[key] = torch.from_numpy(mel).float().to(device)
hann_window_cache[key] = torch.hann_window(win_size).to(device)
mel_basis = mel_basis_cache[key]
hann_window = hann_window_cache[key]
padding = (n_fft - hop_size) // 2
y = torch.nn.functional.pad(
y.unsqueeze(1), (padding, padding), mode="reflect"
).squeeze(1)
spec = torch.stft(
y,
n_fft,
hop_length=hop_size,
win_length=win_size,
window=hann_window,
center=center,
pad_mode="reflect",
normalized=False,
onesided=True,
return_complex=True,
)
spec = torch.sqrt(torch.view_as_real(spec).pow(2).sum(-1) + 1e-9)
mel_spec = torch.matmul(mel_basis, spec)
mel_spec = spectral_normalize_torch(mel_spec)
return mel_spec
def get_mel_spectrogram(wav, h):
"""
Generate mel spectrogram from a waveform using given hyperparameters.
Args:
wav (torch.Tensor): Input waveform.
h: Hyperparameters object with attributes n_fft, num_mels, sampling_rate, hop_size, win_size, fmin, fmax.
Returns:
torch.Tensor: Mel spectrogram.
"""
return mel_spectrogram(
wav,
h.n_fft,
h.num_mels,
h.sampling_rate,
h.hop_size,
h.win_size,
h.fmin,
h.fmax,
)
def get_dataset_filelist(a):
training_files = []
validation_files = []
list_unseen_validation_files = []
with open(a.input_training_file, "r", encoding="utf-8") as fi:
training_files = [
os.path.join(a.input_wavs_dir, x.split("|")[0] + ".wav")
for x in fi.read().split("\n")
if len(x) > 0
]
print(f"first training file: {training_files[0]}")
with open(a.input_validation_file, "r", encoding="utf-8") as fi:
validation_files = [
os.path.join(a.input_wavs_dir, x.split("|")[0] + ".wav")
for x in fi.read().split("\n")
if len(x) > 0
]
print(f"first validation file: {validation_files[0]}")
for i in range(len(a.list_input_unseen_validation_file)):
with open(a.list_input_unseen_validation_file[i], "r", encoding="utf-8") as fi:
unseen_validation_files = [
os.path.join(a.list_input_unseen_wavs_dir[i], x.split("|")[0] + ".wav")
for x in fi.read().split("\n")
if len(x) > 0
]
print(
f"first unseen {i}th validation fileset: {unseen_validation_files[0]}"
)
list_unseen_validation_files.append(unseen_validation_files)
return training_files, validation_files, list_unseen_validation_files
class MelDataset(torch.utils.data.Dataset):
def __init__(
self,
training_files: List[str],
hparams: AttrDict,
segment_size: int,
n_fft: int,
num_mels: int,
hop_size: int,
win_size: int,
sampling_rate: int,
fmin: int,
fmax: Optional[int],
split: bool = True,
shuffle: bool = True,
device: str = None,
fmax_loss: Optional[int] = None,
fine_tuning: bool = False,
base_mels_path: str = None,
is_seen: bool = True,
):
self.audio_files = training_files
random.seed(1234)
if shuffle:
random.shuffle(self.audio_files)
self.hparams = hparams
self.is_seen = is_seen
if self.is_seen:
self.name = pathlib.Path(self.audio_files[0]).parts[0]
else:
self.name = "-".join(pathlib.Path(self.audio_files[0]).parts[:2]).strip("/")
self.segment_size = segment_size
self.sampling_rate = sampling_rate
self.split = split
self.n_fft = n_fft
self.num_mels = num_mels
self.hop_size = hop_size
self.win_size = win_size
self.fmin = fmin
self.fmax = fmax
self.fmax_loss = fmax_loss
self.device = device
self.fine_tuning = fine_tuning
self.base_mels_path = base_mels_path
print("[INFO] checking dataset integrity...")
for i in tqdm(range(len(self.audio_files))):
assert os.path.exists(
self.audio_files[i]
), f"{self.audio_files[i]} not found"
def __getitem__(
self, index: int
) -> Tuple[torch.Tensor, torch.Tensor, str, torch.Tensor]:
try:
filename = self.audio_files[index]
# Use librosa.load that ensures loading waveform into mono with [-1, 1] float values
# Audio is ndarray with shape [T_time]. Disable auto-resampling here to minimize overhead
# The on-the-fly resampling during training will be done only for the obtained random chunk
audio, source_sampling_rate = librosa.load(filename, sr=None, mono=True)
# Main logic that uses <mel, audio> pair for training BigVGAN
if not self.fine_tuning:
if self.split: # Training step
# Obtain randomized audio chunk
if source_sampling_rate != self.sampling_rate:
# Adjust segment size to crop if the source sr is different
target_segment_size = math.ceil(
self.segment_size
* (source_sampling_rate / self.sampling_rate)
)
else:
target_segment_size = self.segment_size
# Compute upper bound index for the random chunk
random_chunk_upper_bound = max(
0, audio.shape[0] - target_segment_size
)
# Crop or pad audio to obtain random chunk with target_segment_size
if audio.shape[0] >= target_segment_size:
audio_start = random.randint(0, random_chunk_upper_bound)
audio = audio[audio_start : audio_start + target_segment_size]
else:
audio = np.pad(
audio,
(0, target_segment_size - audio.shape[0]),
mode="constant",
)
# Resample audio chunk to self.sampling rate
if source_sampling_rate != self.sampling_rate:
audio = librosa.resample(
audio,
orig_sr=source_sampling_rate,
target_sr=self.sampling_rate,
)
if audio.shape[0] > self.segment_size:
# trim last elements to match self.segment_size (e.g., 16385 for 44khz downsampled to 24khz -> 16384)
audio = audio[: self.segment_size]
else: # Validation step
# Resample full audio clip to target sampling rate
if source_sampling_rate != self.sampling_rate:
audio = librosa.resample(
audio,
orig_sr=source_sampling_rate,
target_sr=self.sampling_rate,
)
# Trim last elements to match audio length to self.hop_size * n for evaluation
if (audio.shape[0] % self.hop_size) != 0:
audio = audio[: -(audio.shape[0] % self.hop_size)]
# BigVGAN is trained using volume-normalized waveform
audio = librosa.util.normalize(audio) * 0.95
# Cast ndarray to torch tensor
audio = torch.FloatTensor(audio)
audio = audio.unsqueeze(0) # [B(1), self.segment_size]
# Compute mel spectrogram corresponding to audio
mel = mel_spectrogram(
audio,
self.n_fft,
self.num_mels,
self.sampling_rate,
self.hop_size,
self.win_size,
self.fmin,
self.fmax,
center=False,
) # [B(1), self.num_mels, self.segment_size // self.hop_size]
# Fine-tuning logic that uses pre-computed mel. Example: Using TTS model-generated mel as input
else:
# For fine-tuning, assert that the waveform is in the defined sampling_rate
# Fine-tuning won't support on-the-fly resampling to be fool-proof (the dataset should have been prepared properly)
assert (
source_sampling_rate == self.sampling_rate
), f"For fine_tuning, waveform must be in the spcified sampling rate {self.sampling_rate}, got {source_sampling_rate}"
# Cast ndarray to torch tensor
audio = torch.FloatTensor(audio)
audio = audio.unsqueeze(0) # [B(1), T_time]
# Load pre-computed mel from disk
mel = np.load(
os.path.join(
self.base_mels_path,
os.path.splitext(os.path.split(filename)[-1])[0] + ".npy",
)
)
mel = torch.from_numpy(mel)
if len(mel.shape) < 3:
mel = mel.unsqueeze(0) # ensure [B, C, T]
if self.split:
frames_per_seg = math.ceil(self.segment_size / self.hop_size)
if audio.size(1) >= self.segment_size:
mel_start = random.randint(0, mel.size(2) - frames_per_seg - 1)
mel = mel[:, :, mel_start : mel_start + frames_per_seg]
audio = audio[
:,
mel_start
* self.hop_size : (mel_start + frames_per_seg)
* self.hop_size,
]
# Pad pre-computed mel and audio to match length to ensuring fine-tuning without error.
# NOTE: this may introduce a single-frame misalignment of the <pre-computed mel, audio>
# To remove possible misalignment, it is recommended to prepare the <pre-computed mel, audio> pair where the audio length is the integer multiple of self.hop_size
mel = torch.nn.functional.pad(
mel, (0, frames_per_seg - mel.size(2)), "constant"
)
audio = torch.nn.functional.pad(
audio, (0, self.segment_size - audio.size(1)), "constant"
)
# Compute mel_loss used by spectral regression objective. Uses self.fmax_loss instead (usually None)
mel_loss = mel_spectrogram(
audio,
self.n_fft,
self.num_mels,
self.sampling_rate,
self.hop_size,
self.win_size,
self.fmin,
self.fmax_loss,
center=False,
) # [B(1), self.num_mels, self.segment_size // self.hop_size]
# Shape sanity checks
assert (
audio.shape[1] == mel.shape[2] * self.hop_size
and audio.shape[1] == mel_loss.shape[2] * self.hop_size
), f"Audio length must be mel frame length * hop_size. Got audio shape {audio.shape} mel shape {mel.shape} mel_loss shape {mel_loss.shape}"
return (mel.squeeze(), audio.squeeze(0), filename, mel_loss.squeeze())
# If it encounters error during loading the data, skip this sample and load random other sample to the batch
except Exception as e:
if self.fine_tuning:
raise e # Terminate training if it is fine-tuning. The dataset should have been prepared properly.
else:
print(
f"[WARNING] Failed to load waveform, skipping! filename: {filename} Error: {e}"
)
return self[random.randrange(len(self))]
def __len__(self):
return len(self.audio_files)

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| Field | Response |
| :--------------------------------------------------------------------------------------------------------- | :--------------------------------------------------- |
| Participation considerations from adversely impacted groups protected classes in model design and testing: | None |
| Measures taken to mitigate against unwanted bias: | No measures taken to mitigate against unwanted bias. |

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| Field | Response |
| :---------------------------------------------------------------------------------------------------- | :--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| Intended Application & Domain: | Generating waveform from mel spectrogram. |
| Model Type: | Convolutional Neural Network (CNN) |
| Intended Users: | This model is intended for developers to synthesize and generate waveforms from the AI-generated mel spectrograms. |
| Output: | Audio Waveform |
| Describe how the model works: | Model generates audio waveform corresponding to the input mel spectrogram. |
| Name the adversely impacted groups this has been tested to deliver comparable outcomes regardless of: | Not Applicable |
| Technical Limitations: | This may not perform well on synthetically-generated mel spectrograms that deviate significantly from the profile of mel spectrograms on which this was trained. |
| Verified to have met prescribed NVIDIA quality standards: | Yes |
| Performance Metrics: | Perceptual Evaluation of Speech Quality (PESQ), Virtual Speech Quality Objective Listener (VISQOL), Multi-resolution STFT (MRSTFT), Mel cepstral distortion (MCD), Periodicity RMSE, Voice/Unvoiced F1 Score (V/UV F1) |
| Potential Known Risks: | This model may generate low-quality or distorted soundwaves. |
| Licensing: | https://github.com/NVIDIA/BigVGAN/blob/main/LICENSE |

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# Model Overview
## Description:
BigVGAN is a generative AI model specialized in synthesizing audio waveforms using Mel spectrogram as inputs.
<center><img src="https://user-images.githubusercontent.com/15963413/218609148-881e39df-33af-4af9-ab95-1427c4ebf062.png" width="800"></center>
BigVGAN is a fully convolutional architecture with several upsampling blocks using transposed convolution followed by multiple residual dilated convolution layers.
BigVGAN consists of a novel module, called anti-aliased multi-periodicity composition (AMP), which is specifically designed for generating waveforms. AMP is specialized in synthesizing high-frequency and periodic soundwaves drawing inspiration from audio signal processing principles.
It applies a periodic activation function, called Snake, which provides an inductive bias to the architecture in generating periodic soundwaves. It also applies anti-aliasing filters to reduce undesired artifacts in the generated waveforms. <br>
This model is ready for commercial use.<br>
## References(s):
- [BigVGAN: A Universal Neural Vocoder with Large-Scale Training](https://arxiv.org/abs/2206.04658) <br>
- [Project Page](https://research.nvidia.com/labs/adlr/projects/bigvgan/) <br>
- [Audio Demo](https://bigvgan-demo.github.io/) <br>
## Model Architecture:
**Architecture Type:** Convolution Neural Network (CNN) <br>
**Network Architecture:** You can see the details of this model on this link: https://github.com/NVIDIA/BigVGAN and the related paper can be found here: https://arxiv.org/abs/2206.04658<br>
**Model Version:** 2.0 <br>
## Input:
**Input Type:** Audio <br>
**Input Format:** Mel Spectrogram <br>
**Input Parameters:** None <br>
**Other Properties Related to Input:** The input mel spectrogram has shape `[batch, channels, frames]`, where `channels` refers to the number of mel bands defined by the model and `frames` refers to the temporal length. The model supports arbitrary long `frames` that fits into the GPU memory.
## Output:
**Input Type:** Audio <br>
**Output Format:** Audio Waveform <br>
**Output Parameters:** None <br>
**Other Properties Related to Output:** The output audio waveform has shape `[batch, 1, time]`, where `1` refers to the mono audio channels and `time` refers to the temporal length. `time` is defined as a fixed integer multiple of input `frames`, which is an upsampling ratio of the model (`time = upsampling ratio * frames`). The output audio waveform consitutes float values with a range of `[-1, 1]`.
## Software Integration:
**Runtime Engine(s):** PyTorch
**Supported Hardware Microarchitecture Compatibility:** NVIDIA Ampere, NVIDIA Hopper, NVIDIA Lovelace, NVIDIA Turing, NVIDIA Volta <br>
## Preferred/Supported Operating System(s):
Linux
## Model Version(s):
v2.0
## Training, Testing, and Evaluation Datasets:
### Training Dataset:
The dataset contains diverse audio types, including speech in multiple languages, environmental sounds, and instruments.
**Links:**
- [AAM: Artificial Audio Multitracks Dataset](https://zenodo.org/records/5794629)
- [AudioCaps](https://audiocaps.github.io/)
- [AudioSet](https://research.google.com/audioset/index.html)
- [common-accent](https://huggingface.co/datasets/DTU54DL/common-accent)
- [Crowd Sourced Emotional Multimodal Actors Dataset (CREMA-D)](https://ieeexplore.ieee.org/document/6849440)
- [DCASE2017 Challenge, Task 4: Large-scale weakly supervised sound event detection for smart cars](https://dcase.community/challenge2017/task-large-scale-sound-event-detection)
- [FSDnoisy18k](https://zenodo.org/records/2529934)
- [Free Universal Sound Separation Dataset](https://zenodo.org/records/3694384)
- [Greatest Hits dataset](https://andrewowens.com/vis/)
- [GTZAN](https://ieeexplore.ieee.org/document/1021072)
- [JL corpus](https://www.kaggle.com/datasets/tli725/jl-corpus)
- [Medley-solos-DB: a cross-collection dataset for musical instrument recognition](https://zenodo.org/records/3464194)
- [MUSAN: A Music, Speech, and Noise Corpus](https://www.openslr.org/17/)
- [MusicBench](https://huggingface.co/datasets/amaai-lab/MusicBench)
- [MusicCaps](https://www.kaggle.com/datasets/googleai/musiccaps)
- [MusicNet](https://www.kaggle.com/datasets/imsparsh/musicnet-dataset)
- [NSynth](https://magenta.tensorflow.org/datasets/nsynth)
- [OnAir-Music-Dataset](https://github.com/sevagh/OnAir-Music-Dataset)
- [Audio Piano Triads Dataset](https://zenodo.org/records/4740877)
- [Pitch Audio Dataset (Surge synthesizer)](https://zenodo.org/records/4677097)
- [SONYC Urban Sound Tagging (SONYC-UST): a multilabel dataset from an urban acoustic sensor network](https://zenodo.org/records/3966543)
- [VocalSound: A Dataset for Improving Human Vocal Sounds Recognition](https://arxiv.org/abs/2205.03433)
- [WavText5K](https://github.com/microsoft/WavText5K)
- [CSS10: A Collection of Single Speaker Speech Datasets for 10 Languages](https://github.com/Kyubyong/css10)
- [Hi-Fi Multi-Speaker English TTS Dataset (Hi-Fi TTS)](https://www.openslr.org/109/)
- [IIIT-H Indic Speech Databases](http://festvox.org/databases/iiit_voices/)
- [Libri-Light: A Benchmark for ASR with Limited or No Supervision](https://arxiv.org/abs/1912.07875)
- [LibriTTS: A Corpus Derived from LibriSpeech for Text-to-Speech](https://www.openslr.org/60)
- [LibriTTS-R: A Restored Multi-Speaker Text-to-Speech Corpus](https://www.openslr.org/141/)
- [The SIWIS French Speech Synthesis Database](https://datashare.ed.ac.uk/handle/10283/2353)
- [Crowdsourced high-quality Colombian Spanish speech data set](https://openslr.org/72/)
- [TTS-Portuguese Corpus](https://github.com/Edresson/TTS-Portuguese-Corpus)
- [CSTR VCTK Corpus: English Multi-speaker Corpus for CSTR Voice Cloning Toolkit](https://datashare.ed.ac.uk/handle/10283/3443)
\*\* Data Collection Method by dataset <br>
- Human <br>
\*\* Labeling Method by dataset (for those with labels) <br>
- Hybrid: Automated, Human, Unknown <br>
### Evaluating Dataset:
Properties: The audio generation quality of BigVGAN is evaluated using `dev` splits of the [LibriTTS dataset](https://www.openslr.org/60/) and [Hi-Fi TTS dataset](https://www.openslr.org/109/). The datasets include speech in English language with equal balance of genders.
\*\* Data Collection Method by dataset <br>
- Human <br>
\*\* Labeling Method by dataset <br>
- Automated <br>
## Inference:
**Engine:** PyTorch <br>
**Test Hardware:** NVIDIA A100 GPU <br>
## Ethical Considerations:
NVIDIA believes Trustworthy AI is a shared responsibility and we have established policies and practices to enable development for a wide array of AI applications. When downloaded or used in accordance with our terms of service, developers should work with their internal model team to ensure this model meets requirements for the relevant industry and use case and addresses unforeseen product misuse. For more detailed information on ethical considerations for this model, please see the Model Card++ Explainability, Bias, Safety & Security, and Privacy Subcards. Please report security vulnerabilities or NVIDIA AI Concerns [here](https://www.nvidia.com/en-us/support/submit-security-vulnerability/).

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| Field | Response |
| :------------------------------------------------------------------------------------------------------------------------------------- | :--------------------------------------------- |
| Generatable or reverse engineerable personal information? | None |
| Protected class data used to create this model? | None |
| Was consent obtained for any personal data used? | Not Applicable (No Personal Data) |
| How often is dataset reviewed? | Before Release |
| Is a mechanism in place to honor data subject right of access or deletion of personal data? | Not Applicable |
| If personal collected for the development of the model, was it collected directly by NVIDIA? | Not Applicable |
| If personal collected for the development of the model by NVIDIA, do you maintain or have access to disclosures made to data subjects? | Not Applicable |
| If personal collected for the development of this AI model, was it minimized to only what was required? | Not Applicable |
| Is data in dataset traceable? | Yes |
| Is there provenance for all datasets used in training? | Yes |
| Does data labeling (annotation, metadata) comply with privacy laws? | Yes |
| Is data compliant with data subject requests for data correction or removal, if such a request was made? | No, not possible with externally-sourced data. |

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| Field | Response |
| :---------------------------------------------- | :---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| Model Application(s): | Synethic Audio Generation |
| Describe the life critical impact (if present). | Not Applicable |
| Use Case Restrictions: | None |
| Model and dataset restrictions: | The Principle of least privilege (PoLP) is applied limiting access for dataset generation and model development. Restrictions enforce dataset access during training, and dataset license constraints adhered to. |

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torch
numpy
librosa>=0.8.1
scipy
tensorboard
soundfile
matplotlib
pesq
auraloss
tqdm
nnAudio
ninja
huggingface_hub>=0.23.4

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
import os
import sys
# to import modules from parent_dir
parent_dir = os.path.abspath(os.path.join(os.path.dirname(__file__), ".."))
sys.path.append(parent_dir)
import torch
from alias_free_activation.cuda import activation1d
from activations import Snake
def test_load_fused_kernels():
try:
print("[Success] load_fused_kernels")
except ImportError as e:
print("[Fail] load_fused_kernels")
raise e
def test_anti_alias_activation():
data = torch.rand((10, 10, 200), device="cuda")
# Check activations.Snake cuda vs. torch
fused_anti_alias_activation = activation1d.Activation1d(
activation=Snake(10), fused=True
).cuda()
fused_activation_output = fused_anti_alias_activation(data)
torch_anti_alias_activation = activation1d.Activation1d(
activation=Snake(10), fused=False
).cuda()
torch_activation_output = torch_anti_alias_activation(data)
test_result = (fused_activation_output - torch_activation_output).abs()
while test_result.dim() != 1:
test_result = test_result.mean(dim=-1)
diff = test_result.mean(dim=-1)
if diff <= 1e-3:
print(
f"\n[Success] test_fused_anti_alias_activation"
f"\n > mean_difference={diff}"
f"\n > fused_values={fused_activation_output[-1][-1][:].tolist()}"
f"\n > torch_values={torch_activation_output[-1][-1][:].tolist()}"
)
else:
print(
f"\n[Fail] test_fused_anti_alias_activation"
f"\n > mean_difference={diff}, "
f"\n > fused_values={fused_activation_output[-1][-1][:].tolist()}, "
f"\n > torch_values={torch_activation_output[-1][-1][:].tolist()}"
)
if __name__ == "__main__":
from alias_free_activation.cuda import load
load.load()
test_load_fused_kernels()
test_anti_alias_activation()

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
import os
import sys
# to import modules from parent_dir
parent_dir = os.path.abspath(os.path.join(os.path.dirname(__file__), ".."))
sys.path.append(parent_dir)
import torch
from alias_free_activation.cuda import activation1d
from activations import SnakeBeta
def test_load_fused_kernels():
try:
print("[Success] load_fused_kernels")
except ImportError as e:
print("[Fail] load_fused_kernels")
raise e
def test_anti_alias_activation():
data = torch.rand((10, 10, 200), device="cuda")
# Check activations, Snake CUDA vs. Torch
fused_anti_alias_activation = activation1d.Activation1d(
activation=SnakeBeta(10), fused=True
).cuda()
fused_activation_output = fused_anti_alias_activation(data)
torch_anti_alias_activation = activation1d.Activation1d(
activation=SnakeBeta(10), fused=False
).cuda()
torch_activation_output = torch_anti_alias_activation(data)
test_result = (fused_activation_output - torch_activation_output).abs()
while test_result.dim() != 1:
test_result = test_result.mean(dim=-1)
diff = test_result.mean(dim=-1)
if diff <= 1e-3:
print(
f"\n[Success] test_fused_anti_alias_activation"
f"\n > mean_difference={diff}"
f"\n > fused_values={fused_activation_output[-1][-1][:].tolist()}"
f"\n > torch_values={torch_activation_output[-1][-1][:].tolist()}"
)
else:
print(
f"\n[Fail] test_fused_anti_alias_activation"
f"\n > mean_difference={diff}, "
f"\n > fused_values={fused_activation_output[-1][-1][:].tolist()}, "
f"\n > torch_values={torch_activation_output[-1][-1][:].tolist()}"
)
if __name__ == "__main__":
from alias_free_activation.cuda import load
load.load()
test_load_fused_kernels()
test_anti_alias_activation()

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
import os
import sys
# to import modules from parent_dir
parent_dir = os.path.abspath(os.path.join(os.path.dirname(__file__), ".."))
sys.path.append(parent_dir)
import torch
import json
from env import AttrDict
from bigvgan import BigVGAN
from time import time
from tqdm import tqdm
from meldataset import mel_spectrogram, MAX_WAV_VALUE
from scipy.io.wavfile import write
import numpy as np
import argparse
torch.backends.cudnn.benchmark = True
# For easier debugging
torch.set_printoptions(linewidth=200, threshold=10_000)
def generate_soundwave(duration=5.0, sr=24000):
t = np.linspace(0, duration, int(sr * duration), False, dtype=np.float32)
modulation = np.sin(2 * np.pi * t / duration)
min_freq = 220
max_freq = 1760
frequencies = min_freq + (max_freq - min_freq) * (modulation + 1) / 2
soundwave = np.sin(2 * np.pi * frequencies * t)
soundwave = soundwave / np.max(np.abs(soundwave)) * 0.95
return soundwave, sr
def get_mel(x, h):
return mel_spectrogram(
x, h.n_fft, h.num_mels, h.sampling_rate, h.hop_size, h.win_size, h.fmin, h.fmax
)
def load_checkpoint(filepath, device):
assert os.path.isfile(filepath)
print(f"Loading '{filepath}'")
checkpoint_dict = torch.load(filepath, map_location=device)
print("Complete.")
return checkpoint_dict
if __name__ == "__main__":
parser = argparse.ArgumentParser(
description="Test script to check CUDA kernel correctness."
)
parser.add_argument(
"--checkpoint_file",
type=str,
required=True,
help="Path to the checkpoint file. Assumes config.json exists in the directory.",
)
args = parser.parse_args()
config_file = os.path.join(os.path.split(args.checkpoint_file)[0], "config.json")
with open(config_file) as f:
config = f.read()
json_config = json.loads(config)
h = AttrDict({**json_config})
print("loading plain Pytorch BigVGAN")
generator_original = BigVGAN(h).to("cuda")
print("loading CUDA kernel BigVGAN with auto-build")
generator_cuda_kernel = BigVGAN(h, use_cuda_kernel=True).to("cuda")
state_dict_g = load_checkpoint(args.checkpoint_file, "cuda")
generator_original.load_state_dict(state_dict_g["generator"])
generator_cuda_kernel.load_state_dict(state_dict_g["generator"])
generator_original.remove_weight_norm()
generator_original.eval()
generator_cuda_kernel.remove_weight_norm()
generator_cuda_kernel.eval()
# define number of samples and length of mel frame to benchmark
num_sample = 10
num_mel_frame = 16384
# CUDA kernel correctness check
diff = 0.0
for i in tqdm(range(num_sample)):
# Random mel
data = torch.rand((1, h.num_mels, num_mel_frame), device="cuda")
with torch.inference_mode():
audio_original = generator_original(data)
with torch.inference_mode():
audio_cuda_kernel = generator_cuda_kernel(data)
# Both outputs should be (almost) the same
test_result = (audio_original - audio_cuda_kernel).abs()
diff += test_result.mean(dim=-1).item()
diff /= num_sample
if (
diff <= 2e-3
): # We can expect a small difference (~1e-3) which does not affect perceptual quality
print(
f"\n[Success] test CUDA fused vs. plain torch BigVGAN inference"
f"\n > mean_difference={diff}"
f"\n > fused_values={audio_cuda_kernel[-1][-1][-30:].tolist()}"
f"\n > torch_values={audio_original[-1][-1][-30:].tolist()}"
)
else:
print(
f"\n[Fail] test CUDA fused vs. plain torch BigVGAN inference"
f"\n > mean_difference={diff}"
f"\n > fused_values={audio_cuda_kernel[-1][-1][-30:].tolist()}, "
f"\n > torch_values={audio_original[-1][-1][-30:].tolist()}"
)
del data, audio_original, audio_cuda_kernel
# Variables for tracking total time and VRAM usage
toc_total_original = 0
toc_total_cuda_kernel = 0
vram_used_original_total = 0
vram_used_cuda_kernel_total = 0
audio_length_total = 0
# Measure Original inference in isolation
for i in tqdm(range(num_sample)):
torch.cuda.reset_peak_memory_stats(device="cuda")
data = torch.rand((1, h.num_mels, num_mel_frame), device="cuda")
torch.cuda.synchronize()
tic = time()
with torch.inference_mode():
audio_original = generator_original(data)
torch.cuda.synchronize()
toc = time() - tic
toc_total_original += toc
vram_used_original_total += torch.cuda.max_memory_allocated(device="cuda")
del data, audio_original
torch.cuda.empty_cache()
# Measure CUDA kernel inference in isolation
for i in tqdm(range(num_sample)):
torch.cuda.reset_peak_memory_stats(device="cuda")
data = torch.rand((1, h.num_mels, num_mel_frame), device="cuda")
torch.cuda.synchronize()
tic = time()
with torch.inference_mode():
audio_cuda_kernel = generator_cuda_kernel(data)
torch.cuda.synchronize()
toc = time() - tic
toc_total_cuda_kernel += toc
audio_length_total += audio_cuda_kernel.shape[-1]
vram_used_cuda_kernel_total += torch.cuda.max_memory_allocated(device="cuda")
del data, audio_cuda_kernel
torch.cuda.empty_cache()
# Calculate metrics
audio_second = audio_length_total / h.sampling_rate
khz_original = audio_length_total / toc_total_original / 1000
khz_cuda_kernel = audio_length_total / toc_total_cuda_kernel / 1000
vram_used_original_gb = vram_used_original_total / num_sample / (1024 ** 3)
vram_used_cuda_kernel_gb = vram_used_cuda_kernel_total / num_sample / (1024 ** 3)
# Print results
print(
f"Original BigVGAN: took {toc_total_original:.2f} seconds to generate {audio_second:.2f} seconds of audio, {khz_original:.1f}kHz, {audio_second / toc_total_original:.1f} faster than realtime, VRAM used {vram_used_original_gb:.1f} GB"
)
print(
f"CUDA kernel BigVGAN: took {toc_total_cuda_kernel:.2f} seconds to generate {audio_second:.2f} seconds of audio, {khz_cuda_kernel:.1f}kHz, {audio_second / toc_total_cuda_kernel:.1f} faster than realtime, VRAM used {vram_used_cuda_kernel_gb:.1f} GB"
)
print(f"speedup of CUDA kernel: {khz_cuda_kernel / khz_original}")
print(f"VRAM saving of CUDA kernel: {vram_used_original_gb / vram_used_cuda_kernel_gb}")
# Use artificial sine waves for inference test
audio_real, sr = generate_soundwave(duration=5.0, sr=h.sampling_rate)
audio_real = torch.tensor(audio_real).to("cuda")
# Compute mel spectrogram from the ground truth audio
x = get_mel(audio_real.unsqueeze(0), h)
with torch.inference_mode():
y_g_hat_original = generator_original(x)
y_g_hat_cuda_kernel = generator_cuda_kernel(x)
audio_real = audio_real.squeeze()
audio_real = audio_real * MAX_WAV_VALUE
audio_real = audio_real.cpu().numpy().astype("int16")
audio_original = y_g_hat_original.squeeze()
audio_original = audio_original * MAX_WAV_VALUE
audio_original = audio_original.cpu().numpy().astype("int16")
audio_cuda_kernel = y_g_hat_cuda_kernel.squeeze()
audio_cuda_kernel = audio_cuda_kernel * MAX_WAV_VALUE
audio_cuda_kernel = audio_cuda_kernel.cpu().numpy().astype("int16")
os.makedirs("tmp", exist_ok=True)
output_file_real = os.path.join("tmp", "audio_real.wav")
output_file_original = os.path.join("tmp", "audio_generated_original.wav")
output_file_cuda_kernel = os.path.join("tmp", "audio_generated_cuda_kernel.wav")
write(output_file_real, h.sampling_rate, audio_real)
write(output_file_original, h.sampling_rate, audio_original)
write(output_file_cuda_kernel, h.sampling_rate, audio_cuda_kernel)
print("Example generated audios of original vs. fused CUDA kernel written to tmp!")
print("Done")

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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
import warnings
warnings.simplefilter(action="ignore", category=FutureWarning)
import itertools
import os
import time
import argparse
import json
import torch
import torch.nn.functional as F
from torch.utils.tensorboard import SummaryWriter
from torch.utils.data import DistributedSampler, DataLoader
import torch.multiprocessing as mp
from torch.distributed import init_process_group
from torch.nn.parallel import DistributedDataParallel
from env import AttrDict, build_env
from meldataset import MelDataset, mel_spectrogram, get_dataset_filelist, MAX_WAV_VALUE
from bigvgan import BigVGAN
from discriminators import (
MultiPeriodDiscriminator,
MultiResolutionDiscriminator,
MultiBandDiscriminator,
MultiScaleSubbandCQTDiscriminator,
)
from loss import (
feature_loss,
generator_loss,
discriminator_loss,
MultiScaleMelSpectrogramLoss,
)
from utils import (
plot_spectrogram,
plot_spectrogram_clipped,
scan_checkpoint,
load_checkpoint,
save_checkpoint,
save_audio,
)
import torchaudio as ta
from pesq import pesq
from tqdm import tqdm
import auraloss
torch.backends.cudnn.benchmark = False
def train(rank, a, h):
if h.num_gpus > 1:
# initialize distributed
init_process_group(
backend=h.dist_config["dist_backend"],
init_method=h.dist_config["dist_url"],
world_size=h.dist_config["world_size"] * h.num_gpus,
rank=rank,
)
# Set seed and device
torch.cuda.manual_seed(h.seed)
torch.cuda.set_device(rank)
device = torch.device(f"cuda:{rank:d}")
# Define BigVGAN generator
generator = BigVGAN(h).to(device)
# Define discriminators. MPD is used by default
mpd = MultiPeriodDiscriminator(h).to(device)
# Define additional discriminators. BigVGAN-v1 uses UnivNet's MRD as default
# New in BigVGAN-v2: option to switch to new discriminators: MultiBandDiscriminator / MultiScaleSubbandCQTDiscriminator
if h.get("use_mbd_instead_of_mrd", False): # Switch to MBD
print(
"[INFO] using MultiBandDiscriminator of BigVGAN-v2 instead of MultiResolutionDiscriminator"
)
# Variable name is kept as "mrd" for backward compatibility & minimal code change
mrd = MultiBandDiscriminator(h).to(device)
elif h.get("use_cqtd_instead_of_mrd", False): # Switch to CQTD
print(
"[INFO] using MultiScaleSubbandCQTDiscriminator of BigVGAN-v2 instead of MultiResolutionDiscriminator"
)
mrd = MultiScaleSubbandCQTDiscriminator(h).to(device)
else: # Fallback to original MRD in BigVGAN-v1
mrd = MultiResolutionDiscriminator(h).to(device)
# New in BigVGAN-v2: option to switch to multi-scale L1 mel loss
if h.get("use_multiscale_melloss", False):
print(
"[INFO] using multi-scale Mel l1 loss of BigVGAN-v2 instead of the original single-scale loss"
)
fn_mel_loss_multiscale = MultiScaleMelSpectrogramLoss(
sampling_rate=h.sampling_rate
) # NOTE: accepts waveform as input
else:
fn_mel_loss_singlescale = F.l1_loss
# Print the model & number of parameters, and create or scan the latest checkpoint from checkpoints directory
if rank == 0:
print(generator)
print(mpd)
print(mrd)
print(f"Generator params: {sum(p.numel() for p in generator.parameters())}")
print(f"Discriminator mpd params: {sum(p.numel() for p in mpd.parameters())}")
print(f"Discriminator mrd params: {sum(p.numel() for p in mrd.parameters())}")
os.makedirs(a.checkpoint_path, exist_ok=True)
print(f"Checkpoints directory: {a.checkpoint_path}")
if os.path.isdir(a.checkpoint_path):
# New in v2.1: If the step prefix pattern-based checkpoints are not found, also check for renamed files in Hugging Face Hub to resume training
cp_g = scan_checkpoint(
a.checkpoint_path, prefix="g_", renamed_file="bigvgan_generator.pt"
)
cp_do = scan_checkpoint(
a.checkpoint_path,
prefix="do_",
renamed_file="bigvgan_discriminator_optimizer.pt",
)
# Load the latest checkpoint if exists
steps = 0
if cp_g is None or cp_do is None:
state_dict_do = None
last_epoch = -1
else:
state_dict_g = load_checkpoint(cp_g, device)
state_dict_do = load_checkpoint(cp_do, device)
generator.load_state_dict(state_dict_g["generator"])
mpd.load_state_dict(state_dict_do["mpd"])
mrd.load_state_dict(state_dict_do["mrd"])
steps = state_dict_do["steps"] + 1
last_epoch = state_dict_do["epoch"]
# Initialize DDP, optimizers, and schedulers
if h.num_gpus > 1:
generator = DistributedDataParallel(generator, device_ids=[rank]).to(device)
mpd = DistributedDataParallel(mpd, device_ids=[rank]).to(device)
mrd = DistributedDataParallel(mrd, device_ids=[rank]).to(device)
optim_g = torch.optim.AdamW(
generator.parameters(), h.learning_rate, betas=[h.adam_b1, h.adam_b2]
)
optim_d = torch.optim.AdamW(
itertools.chain(mrd.parameters(), mpd.parameters()),
h.learning_rate,
betas=[h.adam_b1, h.adam_b2],
)
if state_dict_do is not None:
optim_g.load_state_dict(state_dict_do["optim_g"])
optim_d.load_state_dict(state_dict_do["optim_d"])
scheduler_g = torch.optim.lr_scheduler.ExponentialLR(
optim_g, gamma=h.lr_decay, last_epoch=last_epoch
)
scheduler_d = torch.optim.lr_scheduler.ExponentialLR(
optim_d, gamma=h.lr_decay, last_epoch=last_epoch
)
# Define training and validation datasets
"""
unseen_validation_filelist will contain sample filepaths outside the seen training & validation dataset
Example: trained on LibriTTS, validate on VCTK
"""
training_filelist, validation_filelist, list_unseen_validation_filelist = (
get_dataset_filelist(a)
)
trainset = MelDataset(
training_filelist,
h,
h.segment_size,
h.n_fft,
h.num_mels,
h.hop_size,
h.win_size,
h.sampling_rate,
h.fmin,
h.fmax,
shuffle=False if h.num_gpus > 1 else True,
fmax_loss=h.fmax_for_loss,
device=device,
fine_tuning=a.fine_tuning,
base_mels_path=a.input_mels_dir,
is_seen=True,
)
train_sampler = DistributedSampler(trainset) if h.num_gpus > 1 else None
train_loader = DataLoader(
trainset,
num_workers=h.num_workers,
shuffle=False,
sampler=train_sampler,
batch_size=h.batch_size,
pin_memory=True,
drop_last=True,
)
if rank == 0:
validset = MelDataset(
validation_filelist,
h,
h.segment_size,
h.n_fft,
h.num_mels,
h.hop_size,
h.win_size,
h.sampling_rate,
h.fmin,
h.fmax,
False,
False,
fmax_loss=h.fmax_for_loss,
device=device,
fine_tuning=a.fine_tuning,
base_mels_path=a.input_mels_dir,
is_seen=True,
)
validation_loader = DataLoader(
validset,
num_workers=1,
shuffle=False,
sampler=None,
batch_size=1,
pin_memory=True,
drop_last=True,
)
list_unseen_validset = []
list_unseen_validation_loader = []
for i in range(len(list_unseen_validation_filelist)):
unseen_validset = MelDataset(
list_unseen_validation_filelist[i],
h,
h.segment_size,
h.n_fft,
h.num_mels,
h.hop_size,
h.win_size,
h.sampling_rate,
h.fmin,
h.fmax,
False,
False,
fmax_loss=h.fmax_for_loss,
device=device,
fine_tuning=a.fine_tuning,
base_mels_path=a.input_mels_dir,
is_seen=False,
)
unseen_validation_loader = DataLoader(
unseen_validset,
num_workers=1,
shuffle=False,
sampler=None,
batch_size=1,
pin_memory=True,
drop_last=True,
)
list_unseen_validset.append(unseen_validset)
list_unseen_validation_loader.append(unseen_validation_loader)
# Tensorboard logger
sw = SummaryWriter(os.path.join(a.checkpoint_path, "logs"))
if a.save_audio: # Also save audio to disk if --save_audio is set to True
os.makedirs(os.path.join(a.checkpoint_path, "samples"), exist_ok=True)
"""
Validation loop, "mode" parameter is automatically defined as (seen or unseen)_(name of the dataset).
If the name of the dataset contains "nonspeech", it skips PESQ calculation to prevent errors
"""
def validate(rank, a, h, loader, mode="seen"):
assert rank == 0, "validate should only run on rank=0"
generator.eval()
torch.cuda.empty_cache()
val_err_tot = 0
val_pesq_tot = 0
val_mrstft_tot = 0
# Modules for evaluation metrics
pesq_resampler = ta.transforms.Resample(h.sampling_rate, 16000).cuda()
loss_mrstft = auraloss.freq.MultiResolutionSTFTLoss(device="cuda")
if a.save_audio: # Also save audio to disk if --save_audio is set to True
os.makedirs(
os.path.join(a.checkpoint_path, "samples", f"gt_{mode}"),
exist_ok=True,
)
os.makedirs(
os.path.join(a.checkpoint_path, "samples", f"{mode}_{steps:08d}"),
exist_ok=True,
)
with torch.no_grad():
print(f"step {steps} {mode} speaker validation...")
# Loop over validation set and compute metrics
for j, batch in enumerate(tqdm(loader)):
x, y, _, y_mel = batch
y = y.to(device)
if hasattr(generator, "module"):
y_g_hat = generator.module(x.to(device))
else:
y_g_hat = generator(x.to(device))
y_mel = y_mel.to(device, non_blocking=True)
y_g_hat_mel = mel_spectrogram(
y_g_hat.squeeze(1),
h.n_fft,
h.num_mels,
h.sampling_rate,
h.hop_size,
h.win_size,
h.fmin,
h.fmax_for_loss,
)
min_t = min(y_mel.size(-1), y_g_hat_mel.size(-1))
val_err_tot += F.l1_loss(y_mel[...,:min_t], y_g_hat_mel[...,:min_t]).item()
# PESQ calculation. only evaluate PESQ if it's speech signal (nonspeech PESQ will error out)
if (
not "nonspeech" in mode
): # Skips if the name of dataset (in mode string) contains "nonspeech"
# Resample to 16000 for pesq
y_16k = pesq_resampler(y)
y_g_hat_16k = pesq_resampler(y_g_hat.squeeze(1))
y_int_16k = (y_16k[0] * MAX_WAV_VALUE).short().cpu().numpy()
y_g_hat_int_16k = (
(y_g_hat_16k[0] * MAX_WAV_VALUE).short().cpu().numpy()
)
val_pesq_tot += pesq(16000, y_int_16k, y_g_hat_int_16k, "wb")
# MRSTFT calculation
min_t = min(y.size(-1), y_g_hat.size(-1))
val_mrstft_tot += loss_mrstft(y_g_hat[...,:min_t], y[...,:min_t]).item()
# Log audio and figures to Tensorboard
if j % a.eval_subsample == 0: # Subsample every nth from validation set
if steps >= 0:
sw.add_audio(f"gt_{mode}/y_{j}", y[0], steps, h.sampling_rate)
if (
a.save_audio
): # Also save audio to disk if --save_audio is set to True
save_audio(
y[0],
os.path.join(
a.checkpoint_path,
"samples",
f"gt_{mode}",
f"{j:04d}.wav",
),
h.sampling_rate,
)
sw.add_figure(
f"gt_{mode}/y_spec_{j}",
plot_spectrogram(x[0]),
steps,
)
sw.add_audio(
f"generated_{mode}/y_hat_{j}",
y_g_hat[0],
steps,
h.sampling_rate,
)
if (
a.save_audio
): # Also save audio to disk if --save_audio is set to True
save_audio(
y_g_hat[0, 0],
os.path.join(
a.checkpoint_path,
"samples",
f"{mode}_{steps:08d}",
f"{j:04d}.wav",
),
h.sampling_rate,
)
# Spectrogram of synthesized audio
y_hat_spec = mel_spectrogram(
y_g_hat.squeeze(1),
h.n_fft,
h.num_mels,
h.sampling_rate,
h.hop_size,
h.win_size,
h.fmin,
h.fmax,
)
sw.add_figure(
f"generated_{mode}/y_hat_spec_{j}",
plot_spectrogram(y_hat_spec.squeeze(0).cpu().numpy()),
steps,
)
"""
Visualization of spectrogram difference between GT and synthesized audio, difference higher than 1 is clipped for better visualization.
"""
spec_delta = torch.clamp(
torch.abs(x[0] - y_hat_spec.squeeze(0).cpu()),
min=1e-6,
max=1.0,
)
sw.add_figure(
f"delta_dclip1_{mode}/spec_{j}",
plot_spectrogram_clipped(spec_delta.numpy(), clip_max=1.0),
steps,
)
val_err = val_err_tot / (j + 1)
val_pesq = val_pesq_tot / (j + 1)
val_mrstft = val_mrstft_tot / (j + 1)
# Log evaluation metrics to Tensorboard
sw.add_scalar(f"validation_{mode}/mel_spec_error", val_err, steps)
sw.add_scalar(f"validation_{mode}/pesq", val_pesq, steps)
sw.add_scalar(f"validation_{mode}/mrstft", val_mrstft, steps)
generator.train()
# If the checkpoint is loaded, start with validation loop
if steps != 0 and rank == 0 and not a.debug:
if not a.skip_seen:
validate(
rank,
a,
h,
validation_loader,
mode=f"seen_{train_loader.dataset.name}",
)
for i in range(len(list_unseen_validation_loader)):
validate(
rank,
a,
h,
list_unseen_validation_loader[i],
mode=f"unseen_{list_unseen_validation_loader[i].dataset.name}",
)
# Exit the script if --evaluate is set to True
if a.evaluate:
exit()
# Main training loop
generator.train()
mpd.train()
mrd.train()
for epoch in range(max(0, last_epoch), a.training_epochs):
if rank == 0:
start = time.time()
print(f"Epoch: {epoch + 1}")
if h.num_gpus > 1:
train_sampler.set_epoch(epoch)
for i, batch in enumerate(train_loader):
if rank == 0:
start_b = time.time()
x, y, _, y_mel = batch
x = x.to(device, non_blocking=True)
y = y.to(device, non_blocking=True)
y_mel = y_mel.to(device, non_blocking=True)
y = y.unsqueeze(1)
y_g_hat = generator(x)
y_g_hat_mel = mel_spectrogram(
y_g_hat.squeeze(1),
h.n_fft,
h.num_mels,
h.sampling_rate,
h.hop_size,
h.win_size,
h.fmin,
h.fmax_for_loss,
)
optim_d.zero_grad()
# MPD
y_df_hat_r, y_df_hat_g, _, _ = mpd(y, y_g_hat.detach())
loss_disc_f, losses_disc_f_r, losses_disc_f_g = discriminator_loss(
y_df_hat_r, y_df_hat_g
)
# MRD
y_ds_hat_r, y_ds_hat_g, _, _ = mrd(y, y_g_hat.detach())
loss_disc_s, losses_disc_s_r, losses_disc_s_g = discriminator_loss(
y_ds_hat_r, y_ds_hat_g
)
loss_disc_all = loss_disc_s + loss_disc_f
# Set clip_grad_norm value
clip_grad_norm = h.get("clip_grad_norm", 1000.0) # Default to 1000
# Whether to freeze D for initial training steps
if steps >= a.freeze_step:
loss_disc_all.backward()
grad_norm_mpd = torch.nn.utils.clip_grad_norm_(
mpd.parameters(), clip_grad_norm
)
grad_norm_mrd = torch.nn.utils.clip_grad_norm_(
mrd.parameters(), clip_grad_norm
)
optim_d.step()
else:
print(
f"[WARNING] skipping D training for the first {a.freeze_step} steps"
)
grad_norm_mpd = 0.0
grad_norm_mrd = 0.0
# Generator
optim_g.zero_grad()
# L1 Mel-Spectrogram Loss
lambda_melloss = h.get(
"lambda_melloss", 45.0
) # Defaults to 45 in BigVGAN-v1 if not set
if h.get("use_multiscale_melloss", False): # uses wav <y, y_g_hat> for loss
loss_mel = fn_mel_loss_multiscale(y, y_g_hat) * lambda_melloss
else: # Uses mel <y_mel, y_g_hat_mel> for loss
loss_mel = fn_mel_loss_singlescale(y_mel, y_g_hat_mel) * lambda_melloss
# MPD loss
y_df_hat_r, y_df_hat_g, fmap_f_r, fmap_f_g = mpd(y, y_g_hat)
loss_fm_f = feature_loss(fmap_f_r, fmap_f_g)
loss_gen_f, losses_gen_f = generator_loss(y_df_hat_g)
# MRD loss
y_ds_hat_r, y_ds_hat_g, fmap_s_r, fmap_s_g = mrd(y, y_g_hat)
loss_fm_s = feature_loss(fmap_s_r, fmap_s_g)
loss_gen_s, losses_gen_s = generator_loss(y_ds_hat_g)
if steps >= a.freeze_step:
loss_gen_all = (
loss_gen_s + loss_gen_f + loss_fm_s + loss_fm_f + loss_mel
)
else:
print(
f"[WARNING] using regression loss only for G for the first {a.freeze_step} steps"
)
loss_gen_all = loss_mel
loss_gen_all.backward()
grad_norm_g = torch.nn.utils.clip_grad_norm_(
generator.parameters(), clip_grad_norm
)
optim_g.step()
if rank == 0:
# STDOUT logging
if steps % a.stdout_interval == 0:
mel_error = (
loss_mel.item() / lambda_melloss
) # Log training mel regression loss to stdout
print(
f"Steps: {steps:d}, "
f"Gen Loss Total: {loss_gen_all:4.3f}, "
f"Mel Error: {mel_error:4.3f}, "
f"s/b: {time.time() - start_b:4.3f} "
f"lr: {optim_g.param_groups[0]['lr']:4.7f} "
f"grad_norm_g: {grad_norm_g:4.3f}"
)
# Checkpointing
if steps % a.checkpoint_interval == 0 and steps != 0:
checkpoint_path = f"{a.checkpoint_path}/g_{steps:08d}"
save_checkpoint(
checkpoint_path,
{
"generator": (
generator.module if h.num_gpus > 1 else generator
).state_dict()
},
)
checkpoint_path = f"{a.checkpoint_path}/do_{steps:08d}"
save_checkpoint(
checkpoint_path,
{
"mpd": (mpd.module if h.num_gpus > 1 else mpd).state_dict(),
"mrd": (mrd.module if h.num_gpus > 1 else mrd).state_dict(),
"optim_g": optim_g.state_dict(),
"optim_d": optim_d.state_dict(),
"steps": steps,
"epoch": epoch,
},
)
# Tensorboard summary logging
if steps % a.summary_interval == 0:
mel_error = (
loss_mel.item() / lambda_melloss
) # Log training mel regression loss to tensorboard
sw.add_scalar("training/gen_loss_total", loss_gen_all.item(), steps)
sw.add_scalar("training/mel_spec_error", mel_error, steps)
sw.add_scalar("training/fm_loss_mpd", loss_fm_f.item(), steps)
sw.add_scalar("training/gen_loss_mpd", loss_gen_f.item(), steps)
sw.add_scalar("training/disc_loss_mpd", loss_disc_f.item(), steps)
sw.add_scalar("training/grad_norm_mpd", grad_norm_mpd, steps)
sw.add_scalar("training/fm_loss_mrd", loss_fm_s.item(), steps)
sw.add_scalar("training/gen_loss_mrd", loss_gen_s.item(), steps)
sw.add_scalar("training/disc_loss_mrd", loss_disc_s.item(), steps)
sw.add_scalar("training/grad_norm_mrd", grad_norm_mrd, steps)
sw.add_scalar("training/grad_norm_g", grad_norm_g, steps)
sw.add_scalar(
"training/learning_rate_d", scheduler_d.get_last_lr()[0], steps
)
sw.add_scalar(
"training/learning_rate_g", scheduler_g.get_last_lr()[0], steps
)
sw.add_scalar("training/epoch", epoch + 1, steps)
# Validation
if steps % a.validation_interval == 0:
# Plot training input x so far used
for i_x in range(x.shape[0]):
sw.add_figure(
f"training_input/x_{i_x}",
plot_spectrogram(x[i_x].cpu()),
steps,
)
sw.add_audio(
f"training_input/y_{i_x}",
y[i_x][0],
steps,
h.sampling_rate,
)
# Seen and unseen speakers validation loops
if not a.debug and steps != 0:
validate(
rank,
a,
h,
validation_loader,
mode=f"seen_{train_loader.dataset.name}",
)
for i in range(len(list_unseen_validation_loader)):
validate(
rank,
a,
h,
list_unseen_validation_loader[i],
mode=f"unseen_{list_unseen_validation_loader[i].dataset.name}",
)
steps += 1
# BigVGAN-v2 learning rate scheduler is changed from epoch-level to step-level
scheduler_g.step()
scheduler_d.step()
if rank == 0:
print(
f"Time taken for epoch {epoch + 1} is {int(time.time() - start)} sec\n"
)
def main():
print("Initializing Training Process..")
parser = argparse.ArgumentParser()
parser.add_argument("--group_name", default=None)
parser.add_argument("--input_wavs_dir", default="LibriTTS")
parser.add_argument("--input_mels_dir", default="ft_dataset")
parser.add_argument(
"--input_training_file", default="tests/LibriTTS/train-full.txt"
)
parser.add_argument(
"--input_validation_file", default="tests/LibriTTS/val-full.txt"
)
parser.add_argument(
"--list_input_unseen_wavs_dir",
nargs="+",
default=["tests/LibriTTS", "tests/LibriTTS"],
)
parser.add_argument(
"--list_input_unseen_validation_file",
nargs="+",
default=["tests/LibriTTS/dev-clean.txt", "tests/LibriTTS/dev-other.txt"],
)
parser.add_argument("--checkpoint_path", default="exp/bigvgan")
parser.add_argument("--config", default="")
parser.add_argument("--training_epochs", default=100000, type=int)
parser.add_argument("--stdout_interval", default=5, type=int)
parser.add_argument("--checkpoint_interval", default=50000, type=int)
parser.add_argument("--summary_interval", default=100, type=int)
parser.add_argument("--validation_interval", default=50000, type=int)
parser.add_argument(
"--freeze_step",
default=0,
type=int,
help="freeze D for the first specified steps. G only uses regression loss for these steps.",
)
parser.add_argument("--fine_tuning", default=False, type=bool)
parser.add_argument(
"--debug",
default=False,
type=bool,
help="debug mode. skips validation loop throughout training",
)
parser.add_argument(
"--evaluate",
default=False,
type=bool,
help="only run evaluation from checkpoint and exit",
)
parser.add_argument(
"--eval_subsample",
default=5,
type=int,
help="subsampling during evaluation loop",
)
parser.add_argument(
"--skip_seen",
default=False,
type=bool,
help="skip seen dataset. useful for test set inference",
)
parser.add_argument(
"--save_audio",
default=False,
type=bool,
help="save audio of test set inference to disk",
)
a = parser.parse_args()
with open(a.config) as f:
data = f.read()
json_config = json.loads(data)
h = AttrDict(json_config)
build_env(a.config, "config.json", a.checkpoint_path)
torch.manual_seed(h.seed)
if torch.cuda.is_available():
torch.cuda.manual_seed(h.seed)
h.num_gpus = torch.cuda.device_count()
h.batch_size = int(h.batch_size / h.num_gpus)
print(f"Batch size per GPU: {h.batch_size}")
else:
pass
if h.num_gpus > 1:
mp.spawn(
train,
nprocs=h.num_gpus,
args=(
a,
h,
),
)
else:
train(0, a, h)
if __name__ == "__main__":
main()

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# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
import glob
import os
import matplotlib
import torch
from torch.nn.utils import weight_norm
matplotlib.use("Agg")
import matplotlib.pylab as plt
from .meldataset import MAX_WAV_VALUE
from scipy.io.wavfile import write
def plot_spectrogram(spectrogram):
fig, ax = plt.subplots(figsize=(10, 2))
im = ax.imshow(spectrogram, aspect="auto", origin="lower", interpolation="none")
plt.colorbar(im, ax=ax)
fig.canvas.draw()
plt.close()
return fig
def plot_spectrogram_clipped(spectrogram, clip_max=2.0):
fig, ax = plt.subplots(figsize=(10, 2))
im = ax.imshow(
spectrogram,
aspect="auto",
origin="lower",
interpolation="none",
vmin=1e-6,
vmax=clip_max,
)
plt.colorbar(im, ax=ax)
fig.canvas.draw()
plt.close()
return fig
def init_weights(m, mean=0.0, std=0.01):
classname = m.__class__.__name__
if classname.find("Conv") != -1:
m.weight.data.normal_(mean, std)
def apply_weight_norm(m):
classname = m.__class__.__name__
if classname.find("Conv") != -1:
weight_norm(m)
def get_padding(kernel_size, dilation=1):
return int((kernel_size * dilation - dilation) / 2)
def load_checkpoint(filepath, device):
assert os.path.isfile(filepath)
print(f"Loading '{filepath}'")
checkpoint_dict = torch.load(filepath, map_location=device)
print("Complete.")
return checkpoint_dict
def save_checkpoint(filepath, obj):
print(f"Saving checkpoint to {filepath}")
torch.save(obj, filepath)
print("Complete.")
def scan_checkpoint(cp_dir, prefix, renamed_file=None):
# Fallback to original scanning logic first
pattern = os.path.join(cp_dir, prefix + "????????")
cp_list = glob.glob(pattern)
if len(cp_list) > 0:
last_checkpoint_path = sorted(cp_list)[-1]
print(f"[INFO] Resuming from checkpoint: '{last_checkpoint_path}'")
return last_checkpoint_path
# If no pattern-based checkpoints are found, check for renamed file
if renamed_file:
renamed_path = os.path.join(cp_dir, renamed_file)
if os.path.isfile(renamed_path):
print(f"[INFO] Resuming from renamed checkpoint: '{renamed_file}'")
return renamed_path
return None
def save_audio(audio, path, sr):
# wav: torch with 1d shape
audio = audio * MAX_WAV_VALUE
audio = audio.cpu().numpy().astype("int16")
write(path, sr, audio)

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import os, sys
import threading
from tqdm import tqdm
now_dir = os.getcwd()
sys.path.append(now_dir)
import re
import torch
from text.LangSegmenter import LangSegmenter
from text import chinese
from typing import Dict, List, Tuple
from text.cleaner import clean_text
from text import cleaned_text_to_sequence
from transformers import AutoModelForMaskedLM, AutoTokenizer
from TTS_infer_pack.text_segmentation_method import split_big_text, splits, get_method as get_seg_method
from tools.i18n.i18n import I18nAuto, scan_language_list
language=os.environ.get("language","Auto")
language=sys.argv[-1] if sys.argv[-1] in scan_language_list() else language
i18n = I18nAuto(language=language)
punctuation = set(['!', '?', '', ',', '.', '-'])
def get_first(text:str) -> str:
pattern = "[" + "".join(re.escape(sep) for sep in splits) + "]"
text = re.split(pattern, text)[0].strip()
return text
def merge_short_text_in_array(texts:str, threshold:int) -> list:
if (len(texts)) < 2:
return texts
result = []
text = ""
for ele in texts:
text += ele
if len(text) >= threshold:
result.append(text)
text = ""
if (len(text) > 0):
if len(result) == 0:
result.append(text)
else:
result[len(result) - 1] += text
return result
class TextPreprocessor:
def __init__(self, bert_model:AutoModelForMaskedLM,
tokenizer:AutoTokenizer, device:torch.device):
self.bert_model = bert_model
self.tokenizer = tokenizer
self.device = device
self.bert_lock = threading.RLock()
def preprocess(self, text:str, lang:str, text_split_method:str, version:str="v2")->List[Dict]:
print(f'############ {i18n("切分文本")} ############')
text = self.replace_consecutive_punctuation(text)
texts = self.pre_seg_text(text, lang, text_split_method)
result = []
print(f'############ {i18n("提取文本Bert特征")} ############')
for text in tqdm(texts):
phones, bert_features, norm_text = self.segment_and_extract_feature_for_text(text, lang, version)
if phones is None or norm_text=="":
continue
res={
"phones": phones,
"bert_features": bert_features,
"norm_text": norm_text,
}
result.append(res)
return result
def pre_seg_text(self, text:str, lang:str, text_split_method:str):
text = text.strip("\n")
if len(text) == 0:
return []
if (text[0] not in splits and len(get_first(text)) < 4):
text = "" + text if lang != "en" else "." + text
print(i18n("实际输入的目标文本:"))
print(text)
seg_method = get_seg_method(text_split_method)
text = seg_method(text)
while "\n\n" in text:
text = text.replace("\n\n", "\n")
_texts = text.split("\n")
_texts = self.filter_text(_texts)
_texts = merge_short_text_in_array(_texts, 5)
texts = []
for text in _texts:
# 解决输入目标文本的空行导致报错的问题
if (len(text.strip()) == 0):
continue
if not re.sub("\W+", "", text):
# 检测一下,如果是纯符号,就跳过。
continue
if (text[-1] not in splits): text += "" if lang != "en" else "."
# 解决句子过长导致Bert报错的问题
if (len(text) > 510):
texts.extend(split_big_text(text))
else:
texts.append(text)
print(i18n("实际输入的目标文本(切句后):"))
print(texts)
return texts
def segment_and_extract_feature_for_text(self, text:str, language:str, version:str="v1")->Tuple[list, torch.Tensor, str]:
return self.get_phones_and_bert(text, language, version)
def get_phones_and_bert(self, text:str, language:str, version:str, final:bool=False):
with self.bert_lock:
if language in {"en", "all_zh", "all_ja", "all_ko", "all_yue"}:
# language = language.replace("all_","")
formattext = text
while " " in formattext:
formattext = formattext.replace(" ", " ")
if language == "all_zh":
if re.search(r'[A-Za-z]', formattext):
formattext = re.sub(r'[a-z]', lambda x: x.group(0).upper(), formattext)
formattext = chinese.mix_text_normalize(formattext)
return self.get_phones_and_bert(formattext,"zh",version)
else:
phones, word2ph, norm_text = self.clean_text_inf(formattext, language, version)
bert = self.get_bert_feature(norm_text, word2ph).to(self.device)
elif language == "all_yue" and re.search(r'[A-Za-z]', formattext):
formattext = re.sub(r'[a-z]', lambda x: x.group(0).upper(), formattext)
formattext = chinese.mix_text_normalize(formattext)
return self.get_phones_and_bert(formattext,"yue",version)
else:
phones, word2ph, norm_text = self.clean_text_inf(formattext, language, version)
bert = torch.zeros(
(1024, len(phones)),
dtype=torch.float32,
).to(self.device)
elif language in {"zh", "ja", "ko", "yue", "auto", "auto_yue"}:
textlist=[]
langlist=[]
if language == "auto":
for tmp in LangSegmenter.getTexts(text):
langlist.append(tmp["lang"])
textlist.append(tmp["text"])
elif language == "auto_yue":
for tmp in LangSegmenter.getTexts(text):
if tmp["lang"] == "zh":
tmp["lang"] = "yue"
langlist.append(tmp["lang"])
textlist.append(tmp["text"])
else:
for tmp in LangSegmenter.getTexts(text):
if tmp["lang"] == "en":
langlist.append(tmp["lang"])
else:
# 因无法区别中日韩文汉字,以用户输入为准
langlist.append(language)
textlist.append(tmp["text"])
# print(textlist)
# print(langlist)
phones_list = []
bert_list = []
norm_text_list = []
for i in range(len(textlist)):
lang = langlist[i]
phones, word2ph, norm_text = self.clean_text_inf(textlist[i], lang, version)
bert = self.get_bert_inf(phones, word2ph, norm_text, lang)
phones_list.append(phones)
norm_text_list.append(norm_text)
bert_list.append(bert)
bert = torch.cat(bert_list, dim=1)
phones = sum(phones_list, [])
norm_text = ''.join(norm_text_list)
if not final and len(phones) < 6:
return self.get_phones_and_bert("." + text,language,version,final=True)
return phones, bert, norm_text
def get_bert_feature(self, text:str, word2ph:list)->torch.Tensor:
with torch.no_grad():
inputs = self.tokenizer(text, return_tensors="pt")
for i in inputs:
inputs[i] = inputs[i].to(self.device)
res = self.bert_model(**inputs, output_hidden_states=True)
res = torch.cat(res["hidden_states"][-3:-2], -1)[0].cpu()[1:-1]
assert len(word2ph) == len(text)
phone_level_feature = []
for i in range(len(word2ph)):
repeat_feature = res[i].repeat(word2ph[i], 1)
phone_level_feature.append(repeat_feature)
phone_level_feature = torch.cat(phone_level_feature, dim=0)
return phone_level_feature.T
def clean_text_inf(self, text:str, language:str, version:str="v2"):
language = language.replace("all_","")
phones, word2ph, norm_text = clean_text(text, language, version)
phones = cleaned_text_to_sequence(phones, version)
return phones, word2ph, norm_text
def get_bert_inf(self, phones:list, word2ph:list, norm_text:str, language:str):
language=language.replace("all_","")
if language == "zh":
feature = self.get_bert_feature(norm_text, word2ph).to(self.device)
else:
feature = torch.zeros(
(1024, len(phones)),
dtype=torch.float32,
).to(self.device)
return feature
def filter_text(self,texts):
_text=[]
if all(text in [None, " ", "\n",""] for text in texts):
raise ValueError(i18n("请输入有效文本"))
for text in texts:
if text in [None, " ", ""]:
pass
else:
_text.append(text)
return _text
def replace_consecutive_punctuation(self,text):
punctuations = ''.join(re.escape(p) for p in punctuation)
pattern = f'([{punctuations}])([{punctuations}])+'
result = re.sub(pattern, r'\1', text)
return result

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from . import TTS, text_segmentation_method

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import re
from typing import Callable
punctuation = set(['!', '?', '', ',', '.', '-'," "])
METHODS = dict()
def get_method(name:str)->Callable:
method = METHODS.get(name, None)
if method is None:
raise ValueError(f"Method {name} not found")
return method
def get_method_names()->list:
return list(METHODS.keys())
def register_method(name):
def decorator(func):
METHODS[name] = func
return func
return decorator
splits = {"", "", "", "", ",", ".", "?", "!", "~", ":", "", "", "", }
def split_big_text(text, max_len=510):
# 定义全角和半角标点符号
punctuation = "".join(splits)
# 切割文本
segments = re.split('([' + punctuation + '])', text)
# 初始化结果列表和当前片段
result = []
current_segment = ''
for segment in segments:
# 如果当前片段加上新的片段长度超过max_len就将当前片段加入结果列表并重置当前片段
if len(current_segment + segment) > max_len:
result.append(current_segment)
current_segment = segment
else:
current_segment += segment
# 将最后一个片段加入结果列表
if current_segment:
result.append(current_segment)
return result
def split(todo_text):
todo_text = todo_text.replace("……", "").replace("——", "")
if todo_text[-1] not in splits:
todo_text += ""
i_split_head = i_split_tail = 0
len_text = len(todo_text)
todo_texts = []
while 1:
if i_split_head >= len_text:
break # 结尾一定有标点,所以直接跳出即可,最后一段在上次已加入
if todo_text[i_split_head] in splits:
i_split_head += 1
todo_texts.append(todo_text[i_split_tail:i_split_head])
i_split_tail = i_split_head
else:
i_split_head += 1
return todo_texts
# 不切
@register_method("cut0")
def cut0(inp):
if not set(inp).issubset(punctuation):
return inp
else:
return "/n"
# 凑四句一切
@register_method("cut1")
def cut1(inp):
inp = inp.strip("\n")
inps = split(inp)
split_idx = list(range(0, len(inps), 4))
split_idx[-1] = None
if len(split_idx) > 1:
opts = []
for idx in range(len(split_idx) - 1):
opts.append("".join(inps[split_idx[idx]: split_idx[idx + 1]]))
else:
opts = [inp]
opts = [item for item in opts if not set(item).issubset(punctuation)]
return "\n".join(opts)
# 凑50字一切
@register_method("cut2")
def cut2(inp):
inp = inp.strip("\n")
inps = split(inp)
if len(inps) < 2:
return inp
opts = []
summ = 0
tmp_str = ""
for i in range(len(inps)):
summ += len(inps[i])
tmp_str += inps[i]
if summ > 50:
summ = 0
opts.append(tmp_str)
tmp_str = ""
if tmp_str != "":
opts.append(tmp_str)
# print(opts)
if len(opts) > 1 and len(opts[-1]) < 50: ##如果最后一个太短了,和前一个合一起
opts[-2] = opts[-2] + opts[-1]
opts = opts[:-1]
opts = [item for item in opts if not set(item).issubset(punctuation)]
return "\n".join(opts)
# 按中文句号。切
@register_method("cut3")
def cut3(inp):
inp = inp.strip("\n")
opts = ["%s" % item for item in inp.strip("").split("")]
opts = [item for item in opts if not set(item).issubset(punctuation)]
return "\n".join(opts)
#按英文句号.切
@register_method("cut4")
def cut4(inp):
inp = inp.strip("\n")
opts = re.split(r'(?<!\d)\.(?!\d)', inp.strip("."))
opts = [item for item in opts if not set(item).issubset(punctuation)]
return "\n".join(opts)
# 按标点符号切
# contributed by https://github.com/AI-Hobbyist/GPT-SoVITS/blob/main/GPT_SoVITS/inference_webui.py
@register_method("cut5")
def cut5(inp):
inp = inp.strip("\n")
punds = {',', '.', ';', '?', '!', '', '', '', '', '', ';', '', ''}
mergeitems = []
items = []
for i, char in enumerate(inp):
if char in punds:
if char == '.' and i > 0 and i < len(inp) - 1 and inp[i - 1].isdigit() and inp[i + 1].isdigit():
items.append(char)
else:
items.append(char)
mergeitems.append("".join(items))
items = []
else:
items.append(char)
if items:
mergeitems.append("".join(items))
opt = [item for item in mergeitems if not set(item).issubset(punds)]
return "\n".join(opt)
if __name__ == '__main__':
method = get_method("cut5")
print(method("你好,我是小明。你好,我是小红。你好,我是小刚。你好,我是小张。"))

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GPT_SoVITS/configs/.gitignore vendored Normal file
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*.yaml

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train:
seed: 1234
epochs: 20
batch_size: 8
save_every_n_epoch: 1
precision: 16-mixed
gradient_clip: 1.0
optimizer:
lr: 0.01
lr_init: 0.00001
lr_end: 0.0001
warmup_steps: 2000
decay_steps: 40000
data:
max_eval_sample: 8
max_sec: 54
num_workers: 4
pad_val: 1024 # same with EOS in model
model:
vocab_size: 1025
phoneme_vocab_size: 732
embedding_dim: 512
hidden_dim: 512
head: 16
linear_units: 2048
n_layer: 24
dropout: 0
EOS: 1024
random_bert: 0
inference:
top_k: 15

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@ -18,7 +18,8 @@
"warmup_epochs": 0,
"c_mel": 45,
"c_kl": 1.0,
"text_low_lr_rate": 0.4
"text_low_lr_rate": 0.4,
"grad_ckpt": false
},
"data": {
"max_wav_value": 32768.0,

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custom:
bert_base_path: GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large
cnhuhbert_base_path: GPT_SoVITS/pretrained_models/chinese-hubert-base
device: cuda
is_half: true
t2s_weights_path: GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s1bert25hz-5kh-longer-epoch=12-step=369668.ckpt
version: v2
vits_weights_path: GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s2G2333k.pth
v1:
bert_base_path: GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large
cnhuhbert_base_path: GPT_SoVITS/pretrained_models/chinese-hubert-base
device: cpu
is_half: false
t2s_weights_path: GPT_SoVITS/pretrained_models/s1bert25hz-2kh-longer-epoch=68e-step=50232.ckpt
version: v1
vits_weights_path: GPT_SoVITS/pretrained_models/s2G488k.pth
v2:
bert_base_path: GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large
cnhuhbert_base_path: GPT_SoVITS/pretrained_models/chinese-hubert-base
device: cpu
is_half: false
t2s_weights_path: GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s1bert25hz-5kh-longer-epoch=12-step=369668.ckpt
version: v2
vits_weights_path: GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s2G2333k.pth
v3:
bert_base_path: GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large
cnhuhbert_base_path: GPT_SoVITS/pretrained_models/chinese-hubert-base
device: cpu
is_half: false
t2s_weights_path: GPT_SoVITS/pretrained_models/s1v3.ckpt
version: v3
vits_weights_path: GPT_SoVITS/pretrained_models/s2Gv3.pth

5
GPT_SoVITS/download.py Normal file
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import os, sys
now_dir = os.getcwd()
sys.path.insert(0, now_dir)
from text.g2pw import G2PWPinyin
g2pw = G2PWPinyin(model_dir="GPT_SoVITS/text/G2PWModel",model_source="GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large",v_to_u=False, neutral_tone_with_five=True)

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# modified from https://github.com/yangdongchao/SoundStorm/blob/master/soundstorm/s1/AR/models/t2s_model.py
# reference: https://github.com/lifeiteng/vall-e
import argparse
from typing import Optional
from my_utils import load_audio
from text import cleaned_text_to_sequence
import torch
import torchaudio
from torch import IntTensor, LongTensor, Tensor, nn
from torch.nn import functional as F
from transformers import AutoModelForMaskedLM, AutoTokenizer
from feature_extractor import cnhubert
from AR.models.t2s_lightning_module import Text2SemanticLightningModule
from module.models_onnx import SynthesizerTrn
from inference_webui import get_phones_and_bert
import os
import soundfile
default_config = {
"embedding_dim": 512,
"hidden_dim": 512,
"num_head": 8,
"num_layers": 12,
"num_codebook": 8,
"p_dropout": 0.0,
"vocab_size": 1024 + 1,
"phoneme_vocab_size": 512,
"EOS": 1024,
}
def get_raw_t2s_model(dict_s1) -> Text2SemanticLightningModule:
config = dict_s1["config"]
config["model"]["dropout"] = float(config["model"]["dropout"])
t2s_model = Text2SemanticLightningModule(config, "****", is_train=False)
t2s_model.load_state_dict(dict_s1["weight"])
t2s_model = t2s_model.eval()
return t2s_model
@torch.jit.script
def logits_to_probs(
logits,
previous_tokens: Optional[torch.Tensor] = None,
temperature: float = 1.0,
top_k: Optional[int] = None,
top_p: Optional[int] = None,
repetition_penalty: float = 1.0,
):
# if previous_tokens is not None:
# previous_tokens = previous_tokens.squeeze()
# print(logits.shape,previous_tokens.shape)
# pdb.set_trace()
if previous_tokens is not None and repetition_penalty != 1.0:
previous_tokens = previous_tokens.long()
score = torch.gather(logits, dim=1, index=previous_tokens)
score = torch.where(
score < 0, score * repetition_penalty, score / repetition_penalty
)
logits.scatter_(dim=1, index=previous_tokens, src=score)
if top_p is not None and top_p < 1.0:
sorted_logits, sorted_indices = torch.sort(logits, descending=True)
cum_probs = torch.cumsum(
torch.nn.functional.softmax(sorted_logits, dim=-1), dim=-1
)
sorted_indices_to_remove = cum_probs > top_p
sorted_indices_to_remove[:, 0] = False # keep at least one option
indices_to_remove = sorted_indices_to_remove.scatter(
dim=1, index=sorted_indices, src=sorted_indices_to_remove
)
logits = logits.masked_fill(indices_to_remove, -float("Inf"))
logits = logits / max(temperature, 1e-5)
if top_k is not None:
v, _ = torch.topk(logits, min(top_k, logits.size(-1)))
pivot = v[: , -1].unsqueeze(-1)
logits = torch.where(logits < pivot, -float("Inf"), logits)
probs = torch.nn.functional.softmax(logits, dim=-1)
return probs
@torch.jit.script
def multinomial_sample_one_no_sync(probs_sort):
# Does multinomial sampling without a cuda synchronization
q = torch.randn_like(probs_sort)
return torch.argmax(probs_sort / q, dim=-1, keepdim=True).to(dtype=torch.int)
@torch.jit.script
def sample(
logits,
previous_tokens,
temperature: float = 1.0,
top_k: Optional[int] = None,
top_p: Optional[int] = None,
repetition_penalty: float = 1.0,
):
probs = logits_to_probs(
logits=logits, previous_tokens=previous_tokens, temperature=temperature, top_k=top_k, top_p=top_p, repetition_penalty=repetition_penalty
)
idx_next = multinomial_sample_one_no_sync(probs)
return idx_next, probs
@torch.jit.script
def spectrogram_torch(y:Tensor, n_fft:int, sampling_rate:int, hop_size:int, win_size:int, center:bool=False):
hann_window = torch.hann_window(win_size,device=y.device,dtype=y.dtype)
y = torch.nn.functional.pad(
y.unsqueeze(1),
(int((n_fft - hop_size) / 2), int((n_fft - hop_size) / 2)),
mode="reflect",
)
y = y.squeeze(1)
spec = torch.stft(
y,
n_fft,
hop_length=hop_size,
win_length=win_size,
window=hann_window,
center=center,
pad_mode="reflect",
normalized=False,
onesided=True,
return_complex=False,
)
spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-6)
return spec
class DictToAttrRecursive(dict):
def __init__(self, input_dict):
super().__init__(input_dict)
for key, value in input_dict.items():
if isinstance(value, dict):
value = DictToAttrRecursive(value)
self[key] = value
setattr(self, key, value)
def __getattr__(self, item):
try:
return self[item]
except KeyError:
raise AttributeError(f"Attribute {item} not found")
def __setattr__(self, key, value):
if isinstance(value, dict):
value = DictToAttrRecursive(value)
super(DictToAttrRecursive, self).__setitem__(key, value)
super().__setattr__(key, value)
def __delattr__(self, item):
try:
del self[item]
except KeyError:
raise AttributeError(f"Attribute {item} not found")
@torch.jit.script
class T2SMLP:
def __init__(self, w1, b1, w2, b2):
self.w1 = w1
self.b1 = b1
self.w2 = w2
self.b2 = b2
def forward(self, x):
x = F.relu(F.linear(x, self.w1, self.b1))
x = F.linear(x, self.w2, self.b2)
return x
@torch.jit.script
class T2SBlock:
def __init__(
self,
num_heads: int,
hidden_dim: int,
mlp: T2SMLP,
qkv_w,
qkv_b,
out_w,
out_b,
norm_w1,
norm_b1,
norm_eps1: float,
norm_w2,
norm_b2,
norm_eps2: float,
):
self.num_heads = num_heads
self.mlp = mlp
self.hidden_dim: int = hidden_dim
self.qkv_w = qkv_w
self.qkv_b = qkv_b
self.out_w = out_w
self.out_b = out_b
self.norm_w1 = norm_w1
self.norm_b1 = norm_b1
self.norm_eps1 = norm_eps1
self.norm_w2 = norm_w2
self.norm_b2 = norm_b2
self.norm_eps2 = norm_eps2
self.false = torch.tensor(False, dtype=torch.bool)
@torch.jit.ignore
def to_mask(self, x:torch.Tensor, padding_mask:Optional[torch.Tensor]):
if padding_mask is None:
return x
if padding_mask.dtype == torch.bool:
return x.masked_fill(padding_mask, 0)
else:
return x * padding_mask
def process_prompt(self, x:torch.Tensor, attn_mask : torch.Tensor, padding_mask:Optional[torch.Tensor]=None):
q, k, v = F.linear(self.to_mask(x, padding_mask), self.qkv_w, self.qkv_b).chunk(3, dim=-1)
batch_size = q.shape[0]
q_len = q.shape[1]
kv_len = k.shape[1]
q = self.to_mask(q, padding_mask)
k_cache = self.to_mask(k, padding_mask)
v_cache = self.to_mask(v, padding_mask)
q = q.view(batch_size, q_len, self.num_heads, -1).transpose(1, 2)
k = k_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
v = v_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
attn = F.scaled_dot_product_attention(q, k, v, ~attn_mask)
attn = attn.permute(2, 0, 1, 3).reshape(batch_size*q_len, self.hidden_dim)
attn = attn.view(q_len, batch_size, self.hidden_dim).transpose(1, 0)
attn = F.linear(self.to_mask(attn, padding_mask), self.out_w, self.out_b)
if padding_mask is not None:
for i in range(batch_size):
# mask = padding_mask[i,:,0]
if self.false.device!= padding_mask.device:
self.false = self.false.to(padding_mask.device)
idx = torch.where(padding_mask[i,:,0]==self.false)[0]
x_item = x[i,idx,:].unsqueeze(0)
attn_item = attn[i,idx,:].unsqueeze(0)
x_item = x_item + attn_item
x_item = F.layer_norm(
x_item, [self.hidden_dim], self.norm_w1, self.norm_b1, self.norm_eps1
)
x_item = x_item + self.mlp.forward(x_item)
x_item = F.layer_norm(
x_item,
[self.hidden_dim],
self.norm_w2,
self.norm_b2,
self.norm_eps2,
)
x[i,idx,:] = x_item.squeeze(0)
x = self.to_mask(x, padding_mask)
else:
x = x + attn
x = F.layer_norm(
x, [self.hidden_dim], self.norm_w1, self.norm_b1, self.norm_eps1
)
x = x + self.mlp.forward(x)
x = F.layer_norm(
x,
[self.hidden_dim],
self.norm_w2,
self.norm_b2,
self.norm_eps2,
)
return x, k_cache, v_cache
def decode_next_token(self, x:torch.Tensor, k_cache:torch.Tensor, v_cache:torch.Tensor):
q, k, v = F.linear(x, self.qkv_w, self.qkv_b).chunk(3, dim=-1)
k_cache = torch.cat([k_cache, k], dim=1)
v_cache = torch.cat([v_cache, v], dim=1)
batch_size = q.shape[0]
q_len = q.shape[1]
kv_len = k_cache.shape[1]
q = q.view(batch_size, q_len, self.num_heads, -1).transpose(1, 2)
k = k_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
v = v_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
attn = F.scaled_dot_product_attention(q, k, v)
attn = attn.permute(2, 0, 1, 3).reshape(batch_size*q_len, self.hidden_dim)
attn = attn.view(q_len, batch_size, self.hidden_dim).transpose(1, 0)
attn = F.linear(attn, self.out_w, self.out_b)
x = x + attn
x = F.layer_norm(
x, [self.hidden_dim], self.norm_w1, self.norm_b1, self.norm_eps1
)
x = x + self.mlp.forward(x)
x = F.layer_norm(
x,
[self.hidden_dim],
self.norm_w2,
self.norm_b2,
self.norm_eps2,
)
return x, k_cache, v_cache
@torch.jit.script
class T2STransformer:
def __init__(self, num_blocks : int, blocks: list[T2SBlock]):
self.num_blocks : int = num_blocks
self.blocks = blocks
def process_prompt(
self, x:torch.Tensor, attn_mask : torch.Tensor,padding_mask : Optional[torch.Tensor]=None):
k_cache : list[torch.Tensor] = []
v_cache : list[torch.Tensor] = []
for i in range(self.num_blocks):
x, k_cache_, v_cache_ = self.blocks[i].process_prompt(x, attn_mask, padding_mask)
k_cache.append(k_cache_)
v_cache.append(v_cache_)
return x, k_cache, v_cache
def decode_next_token(
self, x:torch.Tensor,
k_cache: list[torch.Tensor],
v_cache: list[torch.Tensor]):
for i in range(self.num_blocks):
x, k_cache[i], v_cache[i] = self.blocks[i].decode_next_token(x, k_cache[i], v_cache[i])
return x, k_cache, v_cache
class VitsModel(nn.Module):
def __init__(self, vits_path):
super().__init__()
# dict_s2 = torch.load(vits_path,map_location="cpu")
dict_s2 = torch.load(vits_path)
self.hps = dict_s2["config"]
if dict_s2['weight']['enc_p.text_embedding.weight'].shape[0] == 322:
self.hps["model"]["version"] = "v1"
else:
self.hps["model"]["version"] = "v2"
self.hps = DictToAttrRecursive(self.hps)
self.hps.model.semantic_frame_rate = "25hz"
self.vq_model = SynthesizerTrn(
self.hps.data.filter_length // 2 + 1,
self.hps.train.segment_size // self.hps.data.hop_length,
n_speakers=self.hps.data.n_speakers,
**self.hps.model
)
self.vq_model.eval()
self.vq_model.load_state_dict(dict_s2["weight"], strict=False)
def forward(self, text_seq, pred_semantic, ref_audio, speed=1.0):
refer = spectrogram_torch(
ref_audio,
self.hps.data.filter_length,
self.hps.data.sampling_rate,
self.hps.data.hop_length,
self.hps.data.win_length,
center=False
)
return self.vq_model(pred_semantic, text_seq, refer, speed)[0, 0]
class T2SModel(nn.Module):
def __init__(self,raw_t2s:Text2SemanticLightningModule):
super(T2SModel, self).__init__()
self.model_dim = raw_t2s.model.model_dim
self.embedding_dim = raw_t2s.model.embedding_dim
self.num_head = raw_t2s.model.num_head
self.num_layers = raw_t2s.model.num_layers
self.vocab_size = raw_t2s.model.vocab_size
self.phoneme_vocab_size = raw_t2s.model.phoneme_vocab_size
# self.p_dropout = float(raw_t2s.model.p_dropout)
self.EOS:int = int(raw_t2s.model.EOS)
self.norm_first = raw_t2s.model.norm_first
assert self.EOS == self.vocab_size - 1
self.hz = 50
self.bert_proj = raw_t2s.model.bert_proj
self.ar_text_embedding = raw_t2s.model.ar_text_embedding
self.ar_text_position = raw_t2s.model.ar_text_position
self.ar_audio_embedding = raw_t2s.model.ar_audio_embedding
self.ar_audio_position = raw_t2s.model.ar_audio_position
# self.t2s_transformer = T2STransformer(self.num_layers, blocks)
# self.t2s_transformer = raw_t2s.model.t2s_transformer
blocks = []
h = raw_t2s.model.h
for i in range(self.num_layers):
layer = h.layers[i]
t2smlp = T2SMLP(
layer.linear1.weight,
layer.linear1.bias,
layer.linear2.weight,
layer.linear2.bias
)
block = T2SBlock(
self.num_head,
self.model_dim,
t2smlp,
layer.self_attn.in_proj_weight,
layer.self_attn.in_proj_bias,
layer.self_attn.out_proj.weight,
layer.self_attn.out_proj.bias,
layer.norm1.weight,
layer.norm1.bias,
layer.norm1.eps,
layer.norm2.weight,
layer.norm2.bias,
layer.norm2.eps
)
blocks.append(block)
self.t2s_transformer = T2STransformer(self.num_layers, blocks)
# self.ar_predict_layer = nn.Linear(self.model_dim, self.vocab_size, bias=False)
self.ar_predict_layer = raw_t2s.model.ar_predict_layer
# self.loss_fct = nn.CrossEntropyLoss(reduction="sum")
self.max_sec = raw_t2s.config["data"]["max_sec"]
self.top_k = int(raw_t2s.config["inference"]["top_k"])
self.early_stop_num = torch.LongTensor([self.hz * self.max_sec])
def forward(self,prompts:LongTensor, ref_seq:LongTensor, text_seq:LongTensor, ref_bert:torch.Tensor, text_bert:torch.Tensor,top_k:LongTensor):
bert = torch.cat([ref_bert.T, text_bert.T], 1)
all_phoneme_ids = torch.cat([ref_seq, text_seq], 1)
bert = bert.unsqueeze(0)
x = self.ar_text_embedding(all_phoneme_ids)
x = x + self.bert_proj(bert.transpose(1, 2))
x:torch.Tensor = self.ar_text_position(x)
early_stop_num = self.early_stop_num
#[1,N,512] [1,N]
# y, k, v, y_emb, x_example = self.first_stage_decoder(x, prompts)
y = prompts
# x_example = x[:,:,0] * 0.0
x_len = x.shape[1]
x_attn_mask = torch.zeros((x_len, x_len), dtype=torch.bool)
y_emb = self.ar_audio_embedding(y)
y_len = y_emb.shape[1]
prefix_len = y.shape[1]
y_pos = self.ar_audio_position(y_emb)
xy_pos = torch.concat([x, y_pos], dim=1)
bsz = x.shape[0]
src_len = x_len + y_len
x_attn_mask_pad = F.pad(
x_attn_mask,
(0, y_len), ###xx的纯0扩展到xx纯0+xy纯1(x,x+y)
value=True,
)
y_attn_mask = F.pad( ###yy的右上1扩展到左边xy的0,(y,x+y)
torch.triu(torch.ones(y_len, y_len, dtype=torch.bool), diagonal=1),
(x_len, 0),
value=False,
)
xy_attn_mask = torch.concat([x_attn_mask_pad, y_attn_mask], dim=0)\
.unsqueeze(0)\
.expand(bsz*self.num_head, -1, -1)\
.view(bsz, self.num_head, src_len, src_len)\
.to(device=x.device, dtype=torch.bool)
idx = 0
top_k = int(top_k)
xy_dec, k_cache, v_cache = self.t2s_transformer.process_prompt(xy_pos, xy_attn_mask, None)
logits = self.ar_predict_layer(xy_dec[:, -1])
logits = logits[:, :-1]
samples = sample(logits, y, top_k=top_k, top_p=1, repetition_penalty=1.35, temperature=1.0)[0]
y = torch.concat([y, samples], dim=1)
y_emb = self.ar_audio_embedding(y[:, -1:])
xy_pos = y_emb * self.ar_audio_position.x_scale + self.ar_audio_position.alpha * self.ar_audio_position.pe[:, y_len + idx].to(dtype=y_emb.dtype,device=y_emb.device)
stop = False
# for idx in range(1, 50):
for idx in range(1, 1500):
#[1, N] [N_layer, N, 1, 512] [N_layer, N, 1, 512] [1, N, 512] [1] [1, N, 512] [1, N]
# y, k, v, y_emb, logits, samples = self.stage_decoder(y, k, v, y_emb, x_example)
xy_dec, k_cache, v_cache = self.t2s_transformer.decode_next_token(xy_pos, k_cache, v_cache)
logits = self.ar_predict_layer(xy_dec[:, -1])
if(idx<11):###至少预测出10个token不然不给停止0.4s
logits = logits[:, :-1]
samples = sample(logits, y, top_k=top_k, top_p=1, repetition_penalty=1.35, temperature=1.0)[0]
y = torch.concat([y, samples], dim=1)
if early_stop_num != -1 and (y.shape[1] - prefix_len) > early_stop_num:
stop = True
if torch.argmax(logits, dim=-1)[0] == self.EOS or samples[0, 0] == self.EOS:
stop = True
if stop:
if y.shape[1] == 0:
y = torch.concat([y, torch.zeros_like(samples)], dim=1)
break
y_emb = self.ar_audio_embedding(y[:, -1:])
xy_pos = y_emb * self.ar_audio_position.x_scale + self.ar_audio_position.alpha * self.ar_audio_position.pe[:, y_len + idx].to(dtype=y_emb.dtype,device=y_emb.device)
y[0,-1] = 0
return y[:, -idx:].unsqueeze(0)
bert_path = os.environ.get(
"bert_path", "GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large"
)
cnhubert_base_path = "GPT_SoVITS/pretrained_models/chinese-hubert-base"
cnhubert.cnhubert_base_path = cnhubert_base_path
@torch.jit.script
def build_phone_level_feature(res:Tensor, word2ph:IntTensor):
phone_level_feature = []
for i in range(word2ph.shape[0]):
repeat_feature = res[i].repeat(word2ph[i].item(), 1)
phone_level_feature.append(repeat_feature)
phone_level_feature = torch.cat(phone_level_feature, dim=0)
# [sum(word2ph), 1024]
return phone_level_feature
class MyBertModel(torch.nn.Module):
def __init__(self, bert_model):
super(MyBertModel, self).__init__()
self.bert = bert_model
def forward(self, input_ids:torch.Tensor, attention_mask:torch.Tensor, token_type_ids:torch.Tensor, word2ph:IntTensor):
outputs = self.bert(input_ids=input_ids, attention_mask=attention_mask, token_type_ids=token_type_ids)
# res = torch.cat(outputs["hidden_states"][-3:-2], -1)[0][1:-1]
res = torch.cat(outputs[1][-3:-2], -1)[0][1:-1]
return build_phone_level_feature(res, word2ph)
class SSLModel(torch.nn.Module):
def __init__(self):
super().__init__()
self.ssl = cnhubert.get_model().model
def forward(self, ref_audio_16k)-> torch.Tensor:
ssl_content = self.ssl(ref_audio_16k)["last_hidden_state"].transpose(1, 2)
return ssl_content
class ExportSSLModel(torch.nn.Module):
def __init__(self,ssl:SSLModel):
super().__init__()
self.ssl = ssl
def forward(self, ref_audio:torch.Tensor):
return self.ssl(ref_audio)
@torch.jit.export
def resample(self,ref_audio:torch.Tensor,src_sr:int,dst_sr:int)->torch.Tensor:
audio = resamplex(ref_audio,src_sr,dst_sr).float()
return audio
def export_bert(output_path):
tokenizer = AutoTokenizer.from_pretrained(bert_path)
text = "叹息声一声接着一声传出,木兰对着房门织布.听不见织布机织布的声音,只听见木兰在叹息.问木兰在想什么?问木兰在惦记什么?木兰答道,我也没有在想什么,也没有在惦记什么."
ref_bert_inputs = tokenizer(text, return_tensors="pt")
word2ph = []
for c in text:
if c in ['','','','',",",".","?"]:
word2ph.append(1)
else:
word2ph.append(2)
ref_bert_inputs['word2ph'] = torch.Tensor(word2ph).int()
bert_model = AutoModelForMaskedLM.from_pretrained(bert_path,output_hidden_states=True,torchscript=True)
my_bert_model = MyBertModel(bert_model)
ref_bert_inputs = {
'input_ids': ref_bert_inputs['input_ids'],
'attention_mask': ref_bert_inputs['attention_mask'],
'token_type_ids': ref_bert_inputs['token_type_ids'],
'word2ph': ref_bert_inputs['word2ph']
}
torch._dynamo.mark_dynamic(ref_bert_inputs['input_ids'], 1)
torch._dynamo.mark_dynamic(ref_bert_inputs['attention_mask'], 1)
torch._dynamo.mark_dynamic(ref_bert_inputs['token_type_ids'], 1)
torch._dynamo.mark_dynamic(ref_bert_inputs['word2ph'], 0)
my_bert_model = torch.jit.trace(my_bert_model,example_kwarg_inputs=ref_bert_inputs)
output_path = os.path.join(output_path, "bert_model.pt")
my_bert_model.save(output_path)
print('#### exported bert ####')
def export(gpt_path, vits_path, ref_audio_path, ref_text, output_path, export_bert_and_ssl=False, device='cpu'):
if not os.path.exists(output_path):
os.makedirs(output_path)
print(f"目录已创建: {output_path}")
else:
print(f"目录已存在: {output_path}")
ref_audio = torch.tensor([load_audio(ref_audio_path, 16000)]).float()
ssl = SSLModel()
if export_bert_and_ssl:
s = ExportSSLModel(torch.jit.trace(ssl,example_inputs=(ref_audio)))
ssl_path = os.path.join(output_path, "ssl_model.pt")
torch.jit.script(s).save(ssl_path)
print('#### exported ssl ####')
export_bert(output_path)
else:
s = ExportSSLModel(ssl)
print(f"device: {device}")
ref_seq_id,ref_bert_T,ref_norm_text = get_phones_and_bert(ref_text,"all_zh",'v2')
ref_seq = torch.LongTensor([ref_seq_id]).to(device)
ref_bert = ref_bert_T.T.to(ref_seq.device)
text_seq_id,text_bert_T,norm_text = get_phones_and_bert("这是一条测试语音,说什么无所谓,只是给它一个例子","all_zh",'v2')
text_seq = torch.LongTensor([text_seq_id]).to(device)
text_bert = text_bert_T.T.to(text_seq.device)
ssl_content = ssl(ref_audio).to(device)
# vits_path = "SoVITS_weights_v2/xw_e8_s216.pth"
vits = VitsModel(vits_path).to(device)
vits.eval()
# gpt_path = "GPT_weights_v2/xw-e15.ckpt"
# dict_s1 = torch.load(gpt_path, map_location=device)
dict_s1 = torch.load(gpt_path)
raw_t2s = get_raw_t2s_model(dict_s1).to(device)
print('#### get_raw_t2s_model ####')
print(raw_t2s.config)
t2s_m = T2SModel(raw_t2s)
t2s_m.eval()
t2s = torch.jit.script(t2s_m).to(device)
print('#### script t2s_m ####')
print("vits.hps.data.sampling_rate:",vits.hps.data.sampling_rate)
gpt_sovits = GPT_SoVITS(t2s,vits).to(device)
gpt_sovits.eval()
ref_audio_sr = s.resample(ref_audio,16000,32000).to(device)
torch._dynamo.mark_dynamic(ssl_content, 2)
torch._dynamo.mark_dynamic(ref_audio_sr, 1)
torch._dynamo.mark_dynamic(ref_seq, 1)
torch._dynamo.mark_dynamic(text_seq, 1)
torch._dynamo.mark_dynamic(ref_bert, 0)
torch._dynamo.mark_dynamic(text_bert, 0)
top_k = torch.LongTensor([5]).to(device)
with torch.no_grad():
gpt_sovits_export = torch.jit.trace(
gpt_sovits,
example_inputs=(
ssl_content,
ref_audio_sr,
ref_seq,
text_seq,
ref_bert,
text_bert,
top_k))
gpt_sovits_path = os.path.join(output_path, "gpt_sovits_model.pt")
gpt_sovits_export.save(gpt_sovits_path)
print('#### exported gpt_sovits ####')
@torch.jit.script
def parse_audio(ref_audio):
ref_audio_16k = torchaudio.functional.resample(ref_audio,48000,16000).float()#.to(ref_audio.device)
ref_audio_sr = torchaudio.functional.resample(ref_audio,48000,32000).float()#.to(ref_audio.device)
return ref_audio_16k,ref_audio_sr
@torch.jit.script
def resamplex(ref_audio:torch.Tensor,src_sr:int,dst_sr:int)->torch.Tensor:
return torchaudio.functional.resample(ref_audio,src_sr,dst_sr).float()
class GPT_SoVITS(nn.Module):
def __init__(self, t2s:T2SModel,vits:VitsModel):
super().__init__()
self.t2s = t2s
self.vits = vits
def forward(
self,
ssl_content: torch.Tensor,
ref_audio_sr: torch.Tensor,
ref_seq: Tensor,
text_seq: Tensor,
ref_bert: Tensor,
text_bert: Tensor,
top_k: LongTensor,
speed=1.0,
):
codes = self.vits.vq_model.extract_latent(ssl_content)
prompt_semantic = codes[0, 0]
prompts = prompt_semantic.unsqueeze(0)
pred_semantic = self.t2s(prompts, ref_seq, text_seq, ref_bert, text_bert, top_k)
audio = self.vits(text_seq, pred_semantic, ref_audio_sr, speed)
return audio
def test():
parser = argparse.ArgumentParser(description="GPT-SoVITS Command Line Tool")
parser.add_argument('--gpt_model', required=True, help="Path to the GPT model file")
parser.add_argument('--sovits_model', required=True, help="Path to the SoVITS model file")
parser.add_argument('--ref_audio', required=True, help="Path to the reference audio file")
parser.add_argument('--ref_text', required=True, help="Path to the reference text file")
parser.add_argument('--output_path', required=True, help="Path to the output directory")
args = parser.parse_args()
gpt_path = args.gpt_model
vits_path = args.sovits_model
ref_audio_path = args.ref_audio
ref_text = args.ref_text
tokenizer = AutoTokenizer.from_pretrained(bert_path)
# bert_model = AutoModelForMaskedLM.from_pretrained(bert_path,output_hidden_states=True,torchscript=True)
# bert = MyBertModel(bert_model)
my_bert = torch.jit.load("onnx/bert_model.pt",map_location='cuda')
# dict_s1 = torch.load(gpt_path, map_location="cuda")
# raw_t2s = get_raw_t2s_model(dict_s1)
# t2s = T2SModel(raw_t2s)
# t2s.eval()
# t2s = torch.jit.load("onnx/xw/t2s_model.pt",map_location='cuda')
# vits_path = "SoVITS_weights_v2/xw_e8_s216.pth"
# vits = VitsModel(vits_path)
# vits.eval()
# ssl = ExportSSLModel(SSLModel()).to('cuda')
# ssl.eval()
ssl = torch.jit.load("onnx/by/ssl_model.pt",map_location='cuda')
# gpt_sovits = GPT_SoVITS(t2s,vits)
gpt_sovits = torch.jit.load("onnx/by/gpt_sovits_model.pt",map_location='cuda')
ref_seq_id,ref_bert_T,ref_norm_text = get_phones_and_bert(ref_text,"all_zh",'v2')
ref_seq = torch.LongTensor([ref_seq_id])
ref_bert = ref_bert_T.T.to(ref_seq.device)
# text_seq_id,text_bert_T,norm_text = get_phones_and_bert("昨天晚上看见征兵文书,知道君主在大规模征兵,那么多卷征兵文册,每一卷上都有父亲的名字.","all_zh",'v2')
text = "昨天晚上看见征兵文书,知道君主在大规模征兵,那么多卷征兵文册,每一卷上都有父亲的名字."
text_seq_id,text_bert_T,norm_text = get_phones_and_bert(text,"all_zh",'v2')
test_bert = tokenizer(text, return_tensors="pt")
word2ph = []
for c in text:
if c in ['','','','',"?",",","."]:
word2ph.append(1)
else:
word2ph.append(2)
test_bert['word2ph'] = torch.Tensor(word2ph).int()
test_bert = my_bert(
test_bert['input_ids'].to('cuda'),
test_bert['attention_mask'].to('cuda'),
test_bert['token_type_ids'].to('cuda'),
test_bert['word2ph'].to('cuda')
)
text_seq = torch.LongTensor([text_seq_id])
text_bert = text_bert_T.T.to(text_seq.device)
print('text_bert:',text_bert.shape,text_bert)
print('test_bert:',test_bert.shape,test_bert)
print(torch.allclose(text_bert.to('cuda'),test_bert))
print('text_seq:',text_seq.shape)
print('text_bert:',text_bert.shape,text_bert.type())
#[1,N]
ref_audio = torch.tensor([load_audio(ref_audio_path, 16000)]).float().to('cuda')
print('ref_audio:',ref_audio.shape)
ref_audio_sr = ssl.resample(ref_audio,16000,32000)
print('start ssl')
ssl_content = ssl(ref_audio)
print('start gpt_sovits:')
print('ssl_content:',ssl_content.shape)
print('ref_audio_sr:',ref_audio_sr.shape)
print('ref_seq:',ref_seq.shape)
ref_seq=ref_seq.to('cuda')
print('text_seq:',text_seq.shape)
text_seq=text_seq.to('cuda')
print('ref_bert:',ref_bert.shape)
ref_bert=ref_bert.to('cuda')
print('text_bert:',text_bert.shape)
text_bert=text_bert.to('cuda')
top_k = torch.LongTensor([5]).to('cuda')
with torch.no_grad():
audio = gpt_sovits(ssl_content, ref_audio_sr, ref_seq, text_seq, ref_bert, test_bert, top_k)
print('start write wav')
soundfile.write("out.wav", audio.detach().cpu().numpy(), 32000)
import text
import json
def export_symbel(version='v2'):
if version=='v1':
symbols = text._symbol_to_id_v1
with open(f"onnx/symbols_v1.json", "w") as file:
json.dump(symbols, file, indent=4)
else:
symbols = text._symbol_to_id_v2
with open(f"onnx/symbols_v2.json", "w") as file:
json.dump(symbols, file, indent=4)
def main():
parser = argparse.ArgumentParser(description="GPT-SoVITS Command Line Tool")
parser.add_argument('--gpt_model', required=True, help="Path to the GPT model file")
parser.add_argument('--sovits_model', required=True, help="Path to the SoVITS model file")
parser.add_argument('--ref_audio', required=True, help="Path to the reference audio file")
parser.add_argument('--ref_text', required=True, help="Path to the reference text file")
parser.add_argument('--output_path', required=True, help="Path to the output directory")
parser.add_argument('--export_common_model', action='store_true', help="Export Bert and SSL model")
parser.add_argument('--device', help="Device to use")
args = parser.parse_args()
export(
gpt_path=args.gpt_model,
vits_path=args.sovits_model,
ref_audio_path=args.ref_audio,
ref_text=args.ref_text,
output_path=args.output_path,
device=args.device,
export_bert_and_ssl=args.export_common_model,
)
import inference_webui
if __name__ == "__main__":
inference_webui.is_half=False
inference_webui.dtype=torch.float32
main()
# test()

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# from f5_tts.model.cfm import CFM
#
# from f5_tts.model.backbones.unett import UNetT
from GPT_SoVITS.f5_tts.model.backbones.dit import DiT
# from f5_tts.model.backbones.dit import DiTNoCond
# from f5_tts.model.backbones.dit import DiTNoCondNoT
# from f5_tts.model.backbones.mmdit import MMDiT
# from f5_tts.model.trainer import Trainer
# __all__ = ["CFM", "UNetT", "DiT", "MMDiT", "Trainer"]
# __all__ = ["CFM", "UNetT", "DiTNoCond","DiT", "MMDiT"]

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## Backbones quick introduction
### unett.py
- flat unet transformer
- structure same as in e2-tts & voicebox paper except using rotary pos emb
- update: allow possible abs pos emb & convnextv2 blocks for embedded text before concat
### dit.py
- adaln-zero dit
- embedded timestep as condition
- concatted noised_input + masked_cond + embedded_text, linear proj in
- possible abs pos emb & convnextv2 blocks for embedded text before concat
- possible long skip connection (first layer to last layer)
### mmdit.py
- sd3 structure
- timestep as condition
- left stream: text embedded and applied a abs pos emb
- right stream: masked_cond & noised_input concatted and with same conv pos emb as unett

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"""
ein notation:
b - batch
n - sequence
nt - text sequence
nw - raw wave length
d - dimension
"""
from __future__ import annotations
import torch
from torch import nn
import torch.nn.functional as F
from torch.utils.checkpoint import checkpoint
from x_transformers.x_transformers import RotaryEmbedding
from GPT_SoVITS.f5_tts.model.modules import (
TimestepEmbedding,
ConvNeXtV2Block,
ConvPositionEmbedding,
DiTBlock,
AdaLayerNormZero_Final,
precompute_freqs_cis,
get_pos_embed_indices,
)
from module.commons import sequence_mask
class TextEmbedding(nn.Module):
def __init__(self, text_dim, conv_layers=0, conv_mult=2):
super().__init__()
if conv_layers > 0:
self.extra_modeling = True
self.precompute_max_pos = 4096 # ~44s of 24khz audio
self.register_buffer("freqs_cis", precompute_freqs_cis(text_dim, self.precompute_max_pos), persistent=False)
self.text_blocks = nn.Sequential(
*[ConvNeXtV2Block(text_dim, text_dim * conv_mult) for _ in range(conv_layers)]
)
else:
self.extra_modeling = False
def forward(self, text: int["b nt"], seq_len, drop_text=False): # noqa: F722
batch, text_len = text.shape[0], text.shape[1]
if drop_text: # cfg for text
text = torch.zeros_like(text)
# possible extra modeling
if self.extra_modeling:
# sinus pos emb
batch_start = torch.zeros((batch,), dtype=torch.long)
pos_idx = get_pos_embed_indices(batch_start, seq_len, max_pos=self.precompute_max_pos)
text_pos_embed = self.freqs_cis[pos_idx]
# print(23333333,text.shape,text_pos_embed.shape)#torch.Size([7, 465, 256]) torch.Size([7, 465, 256])
text = text + text_pos_embed
# convnextv2 blocks
text = self.text_blocks(text)
return text
# noised input audio and context mixing embedding
class InputEmbedding(nn.Module):
def __init__(self, mel_dim, text_dim, out_dim):
super().__init__()
self.proj = nn.Linear(mel_dim * 2 + text_dim, out_dim)
self.conv_pos_embed = ConvPositionEmbedding(dim=out_dim)
def forward(self, x: float["b n d"], cond: float["b n d"], text_embed: float["b n d"], drop_audio_cond=False): # noqa: F722
if drop_audio_cond: # cfg for cond audio
cond = torch.zeros_like(cond)
x = self.proj(torch.cat((x, cond, text_embed), dim=-1))
x = self.conv_pos_embed(x) + x
return x
# Transformer backbone using DiT blocks
class DiT(nn.Module):
def __init__(
self,
*,
dim,
depth=8,
heads=8,
dim_head=64,
dropout=0.1,
ff_mult=4,
mel_dim=100,
text_dim=None,
conv_layers=0,
long_skip_connection=False,
):
super().__init__()
self.time_embed = TimestepEmbedding(dim)
self.d_embed = TimestepEmbedding(dim)
if text_dim is None:
text_dim = mel_dim
self.text_embed = TextEmbedding(text_dim, conv_layers=conv_layers)
self.input_embed = InputEmbedding(mel_dim, text_dim, dim)
self.rotary_embed = RotaryEmbedding(dim_head)
self.dim = dim
self.depth = depth
self.transformer_blocks = nn.ModuleList(
[DiTBlock(dim=dim, heads=heads, dim_head=dim_head, ff_mult=ff_mult, dropout=dropout) for _ in range(depth)]
)
self.long_skip_connection = nn.Linear(dim * 2, dim, bias=False) if long_skip_connection else None
self.norm_out = AdaLayerNormZero_Final(dim) # final modulation
self.proj_out = nn.Linear(dim, mel_dim)
def ckpt_wrapper(self, module):
# https://github.com/chuanyangjin/fast-DiT/blob/main/models.py
def ckpt_forward(*inputs):
outputs = module(*inputs)
return outputs
return ckpt_forward
def forward(#x, prompt_x, x_lens, t, style,cond
self,#d is channel,n is T
x0: float["b n d"], # nosied input audio # noqa: F722
cond0: float["b n d"], # masked cond audio # noqa: F722
x_lens,
time: float["b"] | float[""], # time step # noqa: F821 F722
dt_base_bootstrap,
text0, # : int["b nt"] # noqa: F722#####condition feature
use_grad_ckpt=False, # bool
###no-use
drop_audio_cond=False, # cfg for cond audio
drop_text=False, # cfg for text
# mask: bool["b n"] | None = None, # noqa: F722
):
x=x0.transpose(2,1)
cond=cond0.transpose(2,1)
text=text0.transpose(2,1)
mask = sequence_mask(x_lens,max_length=x.size(1)).to(x.device)
batch, seq_len = x.shape[0], x.shape[1]
if time.ndim == 0:
time = time.repeat(batch)
# t: conditioning time, c: context (text + masked cond audio), x: noised input audio
t = self.time_embed(time)
dt = self.d_embed(dt_base_bootstrap)
t+=dt
text_embed = self.text_embed(text, seq_len, drop_text=drop_text)###need to change
x = self.input_embed(x, cond, text_embed, drop_audio_cond=drop_audio_cond)
rope = self.rotary_embed.forward_from_seq_len(seq_len)
if self.long_skip_connection is not None:
residual = x
for block in self.transformer_blocks:
if use_grad_ckpt:
x = checkpoint(self.ckpt_wrapper(block), x, t, mask, rope, use_reentrant=False)
else:
x = block(x, t, mask=mask, rope=rope)
if self.long_skip_connection is not None:
x = self.long_skip_connection(torch.cat((x, residual), dim=-1))
x = self.norm_out(x, t)
output = self.proj_out(x)
return output

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"""
ein notation:
b - batch
n - sequence
nt - text sequence
nw - raw wave length
d - dimension
"""
from __future__ import annotations
import torch
from torch import nn
from x_transformers.x_transformers import RotaryEmbedding
from f5_tts.model.modules import (
TimestepEmbedding,
ConvPositionEmbedding,
MMDiTBlock,
AdaLayerNormZero_Final,
precompute_freqs_cis,
get_pos_embed_indices,
)
# text embedding
class TextEmbedding(nn.Module):
def __init__(self, out_dim, text_num_embeds):
super().__init__()
self.text_embed = nn.Embedding(text_num_embeds + 1, out_dim) # will use 0 as filler token
self.precompute_max_pos = 1024
self.register_buffer("freqs_cis", precompute_freqs_cis(out_dim, self.precompute_max_pos), persistent=False)
def forward(self, text: int["b nt"], drop_text=False) -> int["b nt d"]: # noqa: F722
text = text + 1
if drop_text:
text = torch.zeros_like(text)
text = self.text_embed(text)
# sinus pos emb
batch_start = torch.zeros((text.shape[0],), dtype=torch.long)
batch_text_len = text.shape[1]
pos_idx = get_pos_embed_indices(batch_start, batch_text_len, max_pos=self.precompute_max_pos)
text_pos_embed = self.freqs_cis[pos_idx]
text = text + text_pos_embed
return text
# noised input & masked cond audio embedding
class AudioEmbedding(nn.Module):
def __init__(self, in_dim, out_dim):
super().__init__()
self.linear = nn.Linear(2 * in_dim, out_dim)
self.conv_pos_embed = ConvPositionEmbedding(out_dim)
def forward(self, x: float["b n d"], cond: float["b n d"], drop_audio_cond=False): # noqa: F722
if drop_audio_cond:
cond = torch.zeros_like(cond)
x = torch.cat((x, cond), dim=-1)
x = self.linear(x)
x = self.conv_pos_embed(x) + x
return x
# Transformer backbone using MM-DiT blocks
class MMDiT(nn.Module):
def __init__(
self,
*,
dim,
depth=8,
heads=8,
dim_head=64,
dropout=0.1,
ff_mult=4,
text_num_embeds=256,
mel_dim=100,
):
super().__init__()
self.time_embed = TimestepEmbedding(dim)
self.text_embed = TextEmbedding(dim, text_num_embeds)
self.audio_embed = AudioEmbedding(mel_dim, dim)
self.rotary_embed = RotaryEmbedding(dim_head)
self.dim = dim
self.depth = depth
self.transformer_blocks = nn.ModuleList(
[
MMDiTBlock(
dim=dim,
heads=heads,
dim_head=dim_head,
dropout=dropout,
ff_mult=ff_mult,
context_pre_only=i == depth - 1,
)
for i in range(depth)
]
)
self.norm_out = AdaLayerNormZero_Final(dim) # final modulation
self.proj_out = nn.Linear(dim, mel_dim)
def forward(
self,
x: float["b n d"], # nosied input audio # noqa: F722
cond: float["b n d"], # masked cond audio # noqa: F722
text: int["b nt"], # text # noqa: F722
time: float["b"] | float[""], # time step # noqa: F821 F722
drop_audio_cond, # cfg for cond audio
drop_text, # cfg for text
mask: bool["b n"] | None = None, # noqa: F722
):
batch = x.shape[0]
if time.ndim == 0:
time = time.repeat(batch)
# t: conditioning (time), c: context (text + masked cond audio), x: noised input audio
t = self.time_embed(time)
c = self.text_embed(text, drop_text=drop_text)
x = self.audio_embed(x, cond, drop_audio_cond=drop_audio_cond)
seq_len = x.shape[1]
text_len = text.shape[1]
rope_audio = self.rotary_embed.forward_from_seq_len(seq_len)
rope_text = self.rotary_embed.forward_from_seq_len(text_len)
for block in self.transformer_blocks:
c, x = block(x, c, t, mask=mask, rope=rope_audio, c_rope=rope_text)
x = self.norm_out(x, t)
output = self.proj_out(x)
return output

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"""
ein notation:
b - batch
n - sequence
nt - text sequence
nw - raw wave length
d - dimension
"""
from __future__ import annotations
from typing import Literal
import torch
from torch import nn
import torch.nn.functional as F
from x_transformers import RMSNorm
from x_transformers.x_transformers import RotaryEmbedding
from f5_tts.model.modules import (
TimestepEmbedding,
ConvNeXtV2Block,
ConvPositionEmbedding,
Attention,
AttnProcessor,
FeedForward,
precompute_freqs_cis,
get_pos_embed_indices,
)
# Text embedding
class TextEmbedding(nn.Module):
def __init__(self, text_num_embeds, text_dim, conv_layers=0, conv_mult=2):
super().__init__()
self.text_embed = nn.Embedding(text_num_embeds + 1, text_dim) # use 0 as filler token
if conv_layers > 0:
self.extra_modeling = True
self.precompute_max_pos = 4096 # ~44s of 24khz audio
self.register_buffer("freqs_cis", precompute_freqs_cis(text_dim, self.precompute_max_pos), persistent=False)
self.text_blocks = nn.Sequential(
*[ConvNeXtV2Block(text_dim, text_dim * conv_mult) for _ in range(conv_layers)]
)
else:
self.extra_modeling = False
def forward(self, text: int["b nt"], seq_len, drop_text=False): # noqa: F722
text = text + 1 # use 0 as filler token. preprocess of batch pad -1, see list_str_to_idx()
text = text[:, :seq_len] # curtail if character tokens are more than the mel spec tokens
batch, text_len = text.shape[0], text.shape[1]
text = F.pad(text, (0, seq_len - text_len), value=0)
if drop_text: # cfg for text
text = torch.zeros_like(text)
text = self.text_embed(text) # b n -> b n d
# possible extra modeling
if self.extra_modeling:
# sinus pos emb
batch_start = torch.zeros((batch,), dtype=torch.long)
pos_idx = get_pos_embed_indices(batch_start, seq_len, max_pos=self.precompute_max_pos)
text_pos_embed = self.freqs_cis[pos_idx]
text = text + text_pos_embed
# convnextv2 blocks
text = self.text_blocks(text)
return text
# noised input audio and context mixing embedding
class InputEmbedding(nn.Module):
def __init__(self, mel_dim, text_dim, out_dim):
super().__init__()
self.proj = nn.Linear(mel_dim * 2 + text_dim, out_dim)
self.conv_pos_embed = ConvPositionEmbedding(dim=out_dim)
def forward(self, x: float["b n d"], cond: float["b n d"], text_embed: float["b n d"], drop_audio_cond=False): # noqa: F722
if drop_audio_cond: # cfg for cond audio
cond = torch.zeros_like(cond)
x = self.proj(torch.cat((x, cond, text_embed), dim=-1))
x = self.conv_pos_embed(x) + x
return x
# Flat UNet Transformer backbone
class UNetT(nn.Module):
def __init__(
self,
*,
dim,
depth=8,
heads=8,
dim_head=64,
dropout=0.1,
ff_mult=4,
mel_dim=100,
text_num_embeds=256,
text_dim=None,
conv_layers=0,
skip_connect_type: Literal["add", "concat", "none"] = "concat",
):
super().__init__()
assert depth % 2 == 0, "UNet-Transformer's depth should be even."
self.time_embed = TimestepEmbedding(dim)
if text_dim is None:
text_dim = mel_dim
self.text_embed = TextEmbedding(text_num_embeds, text_dim, conv_layers=conv_layers)
self.input_embed = InputEmbedding(mel_dim, text_dim, dim)
self.rotary_embed = RotaryEmbedding(dim_head)
# transformer layers & skip connections
self.dim = dim
self.skip_connect_type = skip_connect_type
needs_skip_proj = skip_connect_type == "concat"
self.depth = depth
self.layers = nn.ModuleList([])
for idx in range(depth):
is_later_half = idx >= (depth // 2)
attn_norm = RMSNorm(dim)
attn = Attention(
processor=AttnProcessor(),
dim=dim,
heads=heads,
dim_head=dim_head,
dropout=dropout,
)
ff_norm = RMSNorm(dim)
ff = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
skip_proj = nn.Linear(dim * 2, dim, bias=False) if needs_skip_proj and is_later_half else None
self.layers.append(
nn.ModuleList(
[
skip_proj,
attn_norm,
attn,
ff_norm,
ff,
]
)
)
self.norm_out = RMSNorm(dim)
self.proj_out = nn.Linear(dim, mel_dim)
def forward(
self,
x: float["b n d"], # nosied input audio # noqa: F722
cond: float["b n d"], # masked cond audio # noqa: F722
text: int["b nt"], # text # noqa: F722
time: float["b"] | float[""], # time step # noqa: F821 F722
drop_audio_cond, # cfg for cond audio
drop_text, # cfg for text
mask: bool["b n"] | None = None, # noqa: F722
):
batch, seq_len = x.shape[0], x.shape[1]
if time.ndim == 0:
time = time.repeat(batch)
# t: conditioning time, c: context (text + masked cond audio), x: noised input audio
t = self.time_embed(time)
text_embed = self.text_embed(text, seq_len, drop_text=drop_text)
x = self.input_embed(x, cond, text_embed, drop_audio_cond=drop_audio_cond)
# postfix time t to input x, [b n d] -> [b n+1 d]
x = torch.cat([t.unsqueeze(1), x], dim=1) # pack t to x
if mask is not None:
mask = F.pad(mask, (1, 0), value=1)
rope = self.rotary_embed.forward_from_seq_len(seq_len + 1)
# flat unet transformer
skip_connect_type = self.skip_connect_type
skips = []
for idx, (maybe_skip_proj, attn_norm, attn, ff_norm, ff) in enumerate(self.layers):
layer = idx + 1
# skip connection logic
is_first_half = layer <= (self.depth // 2)
is_later_half = not is_first_half
if is_first_half:
skips.append(x)
if is_later_half:
skip = skips.pop()
if skip_connect_type == "concat":
x = torch.cat((x, skip), dim=-1)
x = maybe_skip_proj(x)
elif skip_connect_type == "add":
x = x + skip
# attention and feedforward blocks
x = attn(attn_norm(x), rope=rope, mask=mask) + x
x = ff(ff_norm(x)) + x
assert len(skips) == 0
x = self.norm_out(x)[:, 1:, :] # unpack t from x
return self.proj_out(x)

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@ -0,0 +1,664 @@
"""
ein notation:
b - batch
n - sequence
nt - text sequence
nw - raw wave length
d - dimension
"""
from __future__ import annotations
import math
from typing import Optional
import torch
import torch.nn.functional as F
import torchaudio
from librosa.filters import mel as librosa_mel_fn
from torch import nn
from x_transformers.x_transformers import apply_rotary_pos_emb
# raw wav to mel spec
mel_basis_cache = {}
hann_window_cache = {}
def get_bigvgan_mel_spectrogram(
waveform,
n_fft=1024,
n_mel_channels=100,
target_sample_rate=24000,
hop_length=256,
win_length=1024,
fmin=0,
fmax=None,
center=False,
): # Copy from https://github.com/NVIDIA/BigVGAN/tree/main
device = waveform.device
key = f"{n_fft}_{n_mel_channels}_{target_sample_rate}_{hop_length}_{win_length}_{fmin}_{fmax}_{device}"
if key not in mel_basis_cache:
mel = librosa_mel_fn(sr=target_sample_rate, n_fft=n_fft, n_mels=n_mel_channels, fmin=fmin, fmax=fmax)
mel_basis_cache[key] = torch.from_numpy(mel).float().to(device) # TODO: why they need .float()?
hann_window_cache[key] = torch.hann_window(win_length).to(device)
mel_basis = mel_basis_cache[key]
hann_window = hann_window_cache[key]
padding = (n_fft - hop_length) // 2
waveform = torch.nn.functional.pad(waveform.unsqueeze(1), (padding, padding), mode="reflect").squeeze(1)
spec = torch.stft(
waveform,
n_fft,
hop_length=hop_length,
win_length=win_length,
window=hann_window,
center=center,
pad_mode="reflect",
normalized=False,
onesided=True,
return_complex=True,
)
spec = torch.sqrt(torch.view_as_real(spec).pow(2).sum(-1) + 1e-9)
mel_spec = torch.matmul(mel_basis, spec)
mel_spec = torch.log(torch.clamp(mel_spec, min=1e-5))
return mel_spec
def get_vocos_mel_spectrogram(
waveform,
n_fft=1024,
n_mel_channels=100,
target_sample_rate=24000,
hop_length=256,
win_length=1024,
):
mel_stft = torchaudio.transforms.MelSpectrogram(
sample_rate=target_sample_rate,
n_fft=n_fft,
win_length=win_length,
hop_length=hop_length,
n_mels=n_mel_channels,
power=1,
center=True,
normalized=False,
norm=None,
).to(waveform.device)
if len(waveform.shape) == 3:
waveform = waveform.squeeze(1) # 'b 1 nw -> b nw'
assert len(waveform.shape) == 2
mel = mel_stft(waveform)
mel = mel.clamp(min=1e-5).log()
return mel
class MelSpec(nn.Module):
def __init__(
self,
n_fft=1024,
hop_length=256,
win_length=1024,
n_mel_channels=100,
target_sample_rate=24_000,
mel_spec_type="vocos",
):
super().__init__()
assert mel_spec_type in ["vocos", "bigvgan"], print("We only support two extract mel backend: vocos or bigvgan")
self.n_fft = n_fft
self.hop_length = hop_length
self.win_length = win_length
self.n_mel_channels = n_mel_channels
self.target_sample_rate = target_sample_rate
if mel_spec_type == "vocos":
self.extractor = get_vocos_mel_spectrogram
elif mel_spec_type == "bigvgan":
self.extractor = get_bigvgan_mel_spectrogram
self.register_buffer("dummy", torch.tensor(0), persistent=False)
def forward(self, wav):
if self.dummy.device != wav.device:
self.to(wav.device)
mel = self.extractor(
waveform=wav,
n_fft=self.n_fft,
n_mel_channels=self.n_mel_channels,
target_sample_rate=self.target_sample_rate,
hop_length=self.hop_length,
win_length=self.win_length,
)
return mel
# sinusoidal position embedding
class SinusPositionEmbedding(nn.Module):
def __init__(self, dim):
super().__init__()
self.dim = dim
def forward(self, x, scale=1000):
device = x.device
half_dim = self.dim // 2
emb = math.log(10000) / (half_dim - 1)
emb = torch.exp(torch.arange(half_dim, device=device).float() * -emb)
emb = scale * x.unsqueeze(1) * emb.unsqueeze(0)
emb = torch.cat((emb.sin(), emb.cos()), dim=-1)
return emb
# convolutional position embedding
class ConvPositionEmbedding(nn.Module):
def __init__(self, dim, kernel_size=31, groups=16):
super().__init__()
assert kernel_size % 2 != 0
self.conv1d = nn.Sequential(
nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=kernel_size // 2),
nn.Mish(),
nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=kernel_size // 2),
nn.Mish(),
)
def forward(self, x: float["b n d"], mask: bool["b n"] | None = None): # noqa: F722
if mask is not None:
mask = mask[..., None]
x = x.masked_fill(~mask, 0.0)
x = x.permute(0, 2, 1)
x = self.conv1d(x)
out = x.permute(0, 2, 1)
if mask is not None:
out = out.masked_fill(~mask, 0.0)
return out
# rotary positional embedding related
def precompute_freqs_cis(dim: int, end: int, theta: float = 10000.0, theta_rescale_factor=1.0):
# proposed by reddit user bloc97, to rescale rotary embeddings to longer sequence length without fine-tuning
# has some connection to NTK literature
# https://www.reddit.com/r/LocalLLaMA/comments/14lz7j5/ntkaware_scaled_rope_allows_llama_models_to_have/
# https://github.com/lucidrains/rotary-embedding-torch/blob/main/rotary_embedding_torch/rotary_embedding_torch.py
theta *= theta_rescale_factor ** (dim / (dim - 2))
freqs = 1.0 / (theta ** (torch.arange(0, dim, 2)[: (dim // 2)].float() / dim))
t = torch.arange(end, device=freqs.device) # type: ignore
freqs = torch.outer(t, freqs).float() # type: ignore
freqs_cos = torch.cos(freqs) # real part
freqs_sin = torch.sin(freqs) # imaginary part
return torch.cat([freqs_cos, freqs_sin], dim=-1)
def get_pos_embed_indices(start, length, max_pos, scale=1.0):
# length = length if isinstance(length, int) else length.max()
scale = scale * torch.ones_like(start, dtype=torch.float32) # in case scale is a scalar
pos = (
start.unsqueeze(1)
+ (torch.arange(length, device=start.device, dtype=torch.float32).unsqueeze(0) * scale.unsqueeze(1)).long()
)
# avoid extra long error.
pos = torch.where(pos < max_pos, pos, max_pos - 1)
return pos
# Global Response Normalization layer (Instance Normalization ?)
class GRN(nn.Module):
def __init__(self, dim):
super().__init__()
self.gamma = nn.Parameter(torch.zeros(1, 1, dim))
self.beta = nn.Parameter(torch.zeros(1, 1, dim))
def forward(self, x):
Gx = torch.norm(x, p=2, dim=1, keepdim=True)
Nx = Gx / (Gx.mean(dim=-1, keepdim=True) + 1e-6)
return self.gamma * (x * Nx) + self.beta + x
# ConvNeXt-V2 Block https://github.com/facebookresearch/ConvNeXt-V2/blob/main/models/convnextv2.py
# ref: https://github.com/bfs18/e2_tts/blob/main/rfwave/modules.py#L108
class ConvNeXtV2Block(nn.Module):
def __init__(
self,
dim: int,
intermediate_dim: int,
dilation: int = 1,
):
super().__init__()
padding = (dilation * (7 - 1)) // 2
self.dwconv = nn.Conv1d(
dim, dim, kernel_size=7, padding=padding, groups=dim, dilation=dilation
) # depthwise conv
self.norm = nn.LayerNorm(dim, eps=1e-6)
self.pwconv1 = nn.Linear(dim, intermediate_dim) # pointwise/1x1 convs, implemented with linear layers
self.act = nn.GELU()
self.grn = GRN(intermediate_dim)
self.pwconv2 = nn.Linear(intermediate_dim, dim)
def forward(self, x: torch.Tensor) -> torch.Tensor:
residual = x
x = x.transpose(1, 2) # b n d -> b d n
x = self.dwconv(x)
x = x.transpose(1, 2) # b d n -> b n d
x = self.norm(x)
x = self.pwconv1(x)
x = self.act(x)
x = self.grn(x)
x = self.pwconv2(x)
return residual + x
# AdaLayerNormZero
# return with modulated x for attn input, and params for later mlp modulation
class AdaLayerNormZero(nn.Module):
def __init__(self, dim):
super().__init__()
self.silu = nn.SiLU()
self.linear = nn.Linear(dim, dim * 6)
self.norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
def forward(self, x, emb=None):
emb = self.linear(self.silu(emb))
shift_msa, scale_msa, gate_msa, shift_mlp, scale_mlp, gate_mlp = torch.chunk(emb, 6, dim=1)
x = self.norm(x) * (1 + scale_msa[:, None]) + shift_msa[:, None]
return x, gate_msa, shift_mlp, scale_mlp, gate_mlp
# AdaLayerNormZero for final layer
# return only with modulated x for attn input, cuz no more mlp modulation
class AdaLayerNormZero_Final(nn.Module):
def __init__(self, dim):
super().__init__()
self.silu = nn.SiLU()
self.linear = nn.Linear(dim, dim * 2)
self.norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
def forward(self, x, emb):
emb = self.linear(self.silu(emb))
scale, shift = torch.chunk(emb, 2, dim=1)
x = self.norm(x) * (1 + scale)[:, None, :] + shift[:, None, :]
return x
# FeedForward
class FeedForward(nn.Module):
def __init__(self, dim, dim_out=None, mult=4, dropout=0.0, approximate: str = "none"):
super().__init__()
inner_dim = int(dim * mult)
dim_out = dim_out if dim_out is not None else dim
activation = nn.GELU(approximate=approximate)
project_in = nn.Sequential(nn.Linear(dim, inner_dim), activation)
self.ff = nn.Sequential(project_in, nn.Dropout(dropout), nn.Linear(inner_dim, dim_out))
def forward(self, x):
return self.ff(x)
# Attention with possible joint part
# modified from diffusers/src/diffusers/models/attention_processor.py
class Attention(nn.Module):
def __init__(
self,
processor: JointAttnProcessor | AttnProcessor,
dim: int,
heads: int = 8,
dim_head: int = 64,
dropout: float = 0.0,
context_dim: Optional[int] = None, # if not None -> joint attention
context_pre_only=None,
):
super().__init__()
if not hasattr(F, "scaled_dot_product_attention"):
raise ImportError("Attention equires PyTorch 2.0, to use it, please upgrade PyTorch to 2.0.")
self.processor = processor
self.dim = dim
self.heads = heads
self.inner_dim = dim_head * heads
self.dropout = dropout
self.context_dim = context_dim
self.context_pre_only = context_pre_only
self.to_q = nn.Linear(dim, self.inner_dim)
self.to_k = nn.Linear(dim, self.inner_dim)
self.to_v = nn.Linear(dim, self.inner_dim)
if self.context_dim is not None:
self.to_k_c = nn.Linear(context_dim, self.inner_dim)
self.to_v_c = nn.Linear(context_dim, self.inner_dim)
if self.context_pre_only is not None:
self.to_q_c = nn.Linear(context_dim, self.inner_dim)
self.to_out = nn.ModuleList([])
self.to_out.append(nn.Linear(self.inner_dim, dim))
self.to_out.append(nn.Dropout(dropout))
if self.context_pre_only is not None and not self.context_pre_only:
self.to_out_c = nn.Linear(self.inner_dim, dim)
def forward(
self,
x: float["b n d"], # noised input x # noqa: F722
c: float["b n d"] = None, # context c # noqa: F722
mask: bool["b n"] | None = None, # noqa: F722
rope=None, # rotary position embedding for x
c_rope=None, # rotary position embedding for c
) -> torch.Tensor:
if c is not None:
return self.processor(self, x, c=c, mask=mask, rope=rope, c_rope=c_rope)
else:
return self.processor(self, x, mask=mask, rope=rope)
# Attention processor
# from torch.nn.attention import SDPBackend
# torch.backends.cuda.enable_flash_sdp(True)
class AttnProcessor:
def __init__(self):
pass
def __call__(
self,
attn: Attention,
x: float["b n d"], # noised input x # noqa: F722
mask: bool["b n"] | None = None, # noqa: F722
rope=None, # rotary position embedding
) -> torch.FloatTensor:
batch_size = x.shape[0]
# `sample` projections.
query = attn.to_q(x)
key = attn.to_k(x)
value = attn.to_v(x)
# apply rotary position embedding
if rope is not None:
freqs, xpos_scale = rope
q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
query = apply_rotary_pos_emb(query, freqs, q_xpos_scale)
key = apply_rotary_pos_emb(key, freqs, k_xpos_scale)
# attention
inner_dim = key.shape[-1]
head_dim = inner_dim // attn.heads
query = query.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
key = key.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
value = value.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
# mask. e.g. inference got a batch with different target durations, mask out the padding
if mask is not None:
attn_mask = mask
attn_mask = attn_mask.unsqueeze(1).unsqueeze(1) # 'b n -> b 1 1 n'
# print(3433333333,attn_mask.shape)
attn_mask = attn_mask.expand(batch_size, attn.heads, query.shape[-2], key.shape[-2])
else:
attn_mask = None
# with torch.nn.attention.sdpa_kernel(backends=[SDPBackend.EFFICIENT_ATTENTION]):
# with torch.backends.cuda.sdp_kernel(enable_flash=True, enable_math=False, enable_mem_efficient=True):
# with torch.backends.cuda.sdp_kernel(enable_flash=True, enable_math=True, enable_mem_efficient=False):
# print(torch.backends.cuda.flash_sdp_enabled())
# print(torch.backends.cuda.mem_efficient_sdp_enabled())
# print(torch.backends.cuda.math_sdp_enabled())
x = F.scaled_dot_product_attention(query, key, value, attn_mask=attn_mask, dropout_p=0.0, is_causal=False)
x = x.transpose(1, 2).reshape(batch_size, -1, attn.heads * head_dim)
x = x.to(query.dtype)
# linear proj
x = attn.to_out[0](x)
# dropout
x = attn.to_out[1](x)
if mask is not None:
mask = mask.unsqueeze(-1)
x = x.masked_fill(~mask, 0.0)
return x
# Joint Attention processor for MM-DiT
# modified from diffusers/src/diffusers/models/attention_processor.py
class JointAttnProcessor:
def __init__(self):
pass
def __call__(
self,
attn: Attention,
x: float["b n d"], # noised input x # noqa: F722
c: float["b nt d"] = None, # context c, here text # noqa: F722
mask: bool["b n"] | None = None, # noqa: F722
rope=None, # rotary position embedding for x
c_rope=None, # rotary position embedding for c
) -> torch.FloatTensor:
residual = x
batch_size = c.shape[0]
# `sample` projections.
query = attn.to_q(x)
key = attn.to_k(x)
value = attn.to_v(x)
# `context` projections.
c_query = attn.to_q_c(c)
c_key = attn.to_k_c(c)
c_value = attn.to_v_c(c)
# apply rope for context and noised input independently
if rope is not None:
freqs, xpos_scale = rope
q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
query = apply_rotary_pos_emb(query, freqs, q_xpos_scale)
key = apply_rotary_pos_emb(key, freqs, k_xpos_scale)
if c_rope is not None:
freqs, xpos_scale = c_rope
q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
c_query = apply_rotary_pos_emb(c_query, freqs, q_xpos_scale)
c_key = apply_rotary_pos_emb(c_key, freqs, k_xpos_scale)
# attention
query = torch.cat([query, c_query], dim=1)
key = torch.cat([key, c_key], dim=1)
value = torch.cat([value, c_value], dim=1)
inner_dim = key.shape[-1]
head_dim = inner_dim // attn.heads
query = query.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
key = key.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
value = value.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
# mask. e.g. inference got a batch with different target durations, mask out the padding
if mask is not None:
attn_mask = F.pad(mask, (0, c.shape[1]), value=True) # no mask for c (text)
attn_mask = attn_mask.unsqueeze(1).unsqueeze(1) # 'b n -> b 1 1 n'
attn_mask = attn_mask.expand(batch_size, attn.heads, query.shape[-2], key.shape[-2])
else:
attn_mask = None
x = F.scaled_dot_product_attention(query, key, value, attn_mask=attn_mask, dropout_p=0.0, is_causal=False)
x = x.transpose(1, 2).reshape(batch_size, -1, attn.heads * head_dim)
x = x.to(query.dtype)
# Split the attention outputs.
x, c = (
x[:, : residual.shape[1]],
x[:, residual.shape[1] :],
)
# linear proj
x = attn.to_out[0](x)
# dropout
x = attn.to_out[1](x)
if not attn.context_pre_only:
c = attn.to_out_c(c)
if mask is not None:
mask = mask.unsqueeze(-1)
x = x.masked_fill(~mask, 0.0)
# c = c.masked_fill(~mask, 0.) # no mask for c (text)
return x, c
# DiT Block
class DiTBlock(nn.Module):
def __init__(self, dim, heads, dim_head, ff_mult=4, dropout=0.1):
super().__init__()
self.attn_norm = AdaLayerNormZero(dim)
self.attn = Attention(
processor=AttnProcessor(),
dim=dim,
heads=heads,
dim_head=dim_head,
dropout=dropout,
)
self.ff_norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
self.ff = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
def forward(self, x, t, mask=None, rope=None): # x: noised input, t: time embedding
# pre-norm & modulation for attention input
norm, gate_msa, shift_mlp, scale_mlp, gate_mlp = self.attn_norm(x, emb=t)
# attention
attn_output = self.attn(x=norm, mask=mask, rope=rope)
# process attention output for input x
x = x + gate_msa.unsqueeze(1) * attn_output
norm = self.ff_norm(x) * (1 + scale_mlp[:, None]) + shift_mlp[:, None]
ff_output = self.ff(norm)
x = x + gate_mlp.unsqueeze(1) * ff_output
return x
# MMDiT Block https://arxiv.org/abs/2403.03206
class MMDiTBlock(nn.Module):
r"""
modified from diffusers/src/diffusers/models/attention.py
notes.
_c: context related. text, cond, etc. (left part in sd3 fig2.b)
_x: noised input related. (right part)
context_pre_only: last layer only do prenorm + modulation cuz no more ffn
"""
def __init__(self, dim, heads, dim_head, ff_mult=4, dropout=0.1, context_pre_only=False):
super().__init__()
self.context_pre_only = context_pre_only
self.attn_norm_c = AdaLayerNormZero_Final(dim) if context_pre_only else AdaLayerNormZero(dim)
self.attn_norm_x = AdaLayerNormZero(dim)
self.attn = Attention(
processor=JointAttnProcessor(),
dim=dim,
heads=heads,
dim_head=dim_head,
dropout=dropout,
context_dim=dim,
context_pre_only=context_pre_only,
)
if not context_pre_only:
self.ff_norm_c = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
self.ff_c = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
else:
self.ff_norm_c = None
self.ff_c = None
self.ff_norm_x = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
self.ff_x = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
def forward(self, x, c, t, mask=None, rope=None, c_rope=None): # x: noised input, c: context, t: time embedding
# pre-norm & modulation for attention input
if self.context_pre_only:
norm_c = self.attn_norm_c(c, t)
else:
norm_c, c_gate_msa, c_shift_mlp, c_scale_mlp, c_gate_mlp = self.attn_norm_c(c, emb=t)
norm_x, x_gate_msa, x_shift_mlp, x_scale_mlp, x_gate_mlp = self.attn_norm_x(x, emb=t)
# attention
x_attn_output, c_attn_output = self.attn(x=norm_x, c=norm_c, mask=mask, rope=rope, c_rope=c_rope)
# process attention output for context c
if self.context_pre_only:
c = None
else: # if not last layer
c = c + c_gate_msa.unsqueeze(1) * c_attn_output
norm_c = self.ff_norm_c(c) * (1 + c_scale_mlp[:, None]) + c_shift_mlp[:, None]
c_ff_output = self.ff_c(norm_c)
c = c + c_gate_mlp.unsqueeze(1) * c_ff_output
# process attention output for input x
x = x + x_gate_msa.unsqueeze(1) * x_attn_output
norm_x = self.ff_norm_x(x) * (1 + x_scale_mlp[:, None]) + x_shift_mlp[:, None]
x_ff_output = self.ff_x(norm_x)
x = x + x_gate_mlp.unsqueeze(1) * x_ff_output
return c, x
# time step conditioning embedding
class TimestepEmbedding(nn.Module):
def __init__(self, dim, freq_embed_dim=256):
super().__init__()
self.time_embed = SinusPositionEmbedding(freq_embed_dim)
self.time_mlp = nn.Sequential(nn.Linear(freq_embed_dim, dim), nn.SiLU(), nn.Linear(dim, dim))
def forward(self, timestep: float["b"]): # noqa: F821
time_hidden = self.time_embed(timestep)
time_hidden = time_hidden.to(timestep.dtype)
time = self.time_mlp(time_hidden) # b d
return time

View File

@ -4,8 +4,11 @@ import librosa
import torch
import torch.nn.functional as F
import soundfile as sf
import logging
import os
from transformers import logging as tf_logging
tf_logging.set_verbosity_error()
import logging
logging.getLogger("numba").setLevel(logging.WARNING)
from transformers import (
@ -20,11 +23,15 @@ cnhubert_base_path = None
class CNHubert(nn.Module):
def __init__(self):
def __init__(self, base_path:str=None):
super().__init__()
self.model = HubertModel.from_pretrained(cnhubert_base_path)
if base_path is None:
base_path = cnhubert_base_path
if os.path.exists(base_path):...
else:raise FileNotFoundError(base_path)
self.model = HubertModel.from_pretrained(base_path, local_files_only=True)
self.feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(
cnhubert_base_path
base_path, local_files_only=True
)
def forward(self, x):

View File

@ -0,0 +1,55 @@
import argparse
import os
import soundfile as sf
from tools.i18n.i18n import I18nAuto
from GPT_SoVITS.inference_webui import change_gpt_weights, change_sovits_weights, get_tts_wav
i18n = I18nAuto()
def synthesize(GPT_model_path, SoVITS_model_path, ref_audio_path, ref_text_path, ref_language, target_text_path, target_language, output_path):
# Read reference text
with open(ref_text_path, 'r', encoding='utf-8') as file:
ref_text = file.read()
# Read target text
with open(target_text_path, 'r', encoding='utf-8') as file:
target_text = file.read()
# Change model weights
change_gpt_weights(gpt_path=GPT_model_path)
change_sovits_weights(sovits_path=SoVITS_model_path)
# Synthesize audio
synthesis_result = get_tts_wav(ref_wav_path=ref_audio_path,
prompt_text=ref_text,
prompt_language=i18n(ref_language),
text=target_text,
text_language=i18n(target_language), top_p=1, temperature=1)
result_list = list(synthesis_result)
if result_list:
last_sampling_rate, last_audio_data = result_list[-1]
output_wav_path = os.path.join(output_path, "output.wav")
sf.write(output_wav_path, last_audio_data, last_sampling_rate)
print(f"Audio saved to {output_wav_path}")
def main():
parser = argparse.ArgumentParser(description="GPT-SoVITS Command Line Tool")
parser.add_argument('--gpt_model', required=True, help="Path to the GPT model file")
parser.add_argument('--sovits_model', required=True, help="Path to the SoVITS model file")
parser.add_argument('--ref_audio', required=True, help="Path to the reference audio file")
parser.add_argument('--ref_text', required=True, help="Path to the reference text file")
parser.add_argument('--ref_language', required=True, choices=["中文", "英文", "日文"], help="Language of the reference audio")
parser.add_argument('--target_text', required=True, help="Path to the target text file")
parser.add_argument('--target_language', required=True, choices=["中文", "英文", "日文", "中英混合", "日英混合", "多语种混合"], help="Language of the target text")
parser.add_argument('--output_path', required=True, help="Path to the output directory")
args = parser.parse_args()
synthesize(args.gpt_model, args.sovits_model, args.ref_audio, args.ref_text, args.ref_language, args.target_text, args.target_language, args.output_path)
if __name__ == '__main__':
main()

View File

@ -1,3 +1,4 @@
import os
import sys
from PyQt5.QtCore import QEvent
from PyQt5.QtWidgets import QApplication, QMainWindow, QLabel, QLineEdit, QPushButton, QTextEdit
@ -7,16 +8,16 @@ import soundfile as sf
from tools.i18n.i18n import I18nAuto
i18n = I18nAuto()
from GPT_SoVITS.inference_webui import change_gpt_weights, change_sovits_weights, get_tts_wav
from inference_webui import gpt_path, sovits_path, change_gpt_weights, change_sovits_weights, get_tts_wav
class GPTSoVITSGUI(QMainWindow):
GPT_Path = gpt_path
SoVITS_Path = sovits_path
def __init__(self):
super().__init__()
self.init_ui()
def init_ui(self):
self.setWindowTitle('GPT-SoVITS GUI')
self.setGeometry(800, 450, 950, 850)
@ -71,6 +72,7 @@ class GPTSoVITSGUI(QMainWindow):
self.GPT_model_label = QLabel("选择GPT模型:")
self.GPT_model_input = QLineEdit()
self.GPT_model_input.setPlaceholderText("拖拽或选择文件")
self.GPT_model_input.setText(self.GPT_Path)
self.GPT_model_input.setReadOnly(True)
self.GPT_model_button = QPushButton("选择GPT模型文件")
self.GPT_model_button.clicked.connect(self.select_GPT_model)
@ -78,6 +80,7 @@ class GPTSoVITSGUI(QMainWindow):
self.SoVITS_model_label = QLabel("选择SoVITS模型:")
self.SoVITS_model_input = QLineEdit()
self.SoVITS_model_input.setPlaceholderText("拖拽或选择文件")
self.SoVITS_model_input.setText(self.SoVITS_Path)
self.SoVITS_model_input.setReadOnly(True)
self.SoVITS_model_button = QPushButton("选择SoVITS模型文件")
self.SoVITS_model_button.clicked.connect(self.select_SoVITS_model)
@ -91,25 +94,25 @@ class GPTSoVITSGUI(QMainWindow):
self.ref_text_label = QLabel("参考音频文本:")
self.ref_text_input = QLineEdit()
self.ref_text_input.setPlaceholderText("拖拽或选择文件")
self.ref_text_input.setReadOnly(True)
self.ref_text_input.setPlaceholderText("直接输入文字或上传文本")
self.ref_text_button = QPushButton("上传文本")
self.ref_text_button.clicked.connect(self.upload_ref_text)
self.language_label = QLabel("参考音频语言:")
self.language_combobox = QComboBox()
self.language_combobox.addItems(["中文", "英文", "日文"])
self.ref_language_label = QLabel("参考音频语言:")
self.ref_language_combobox = QComboBox()
self.ref_language_combobox.addItems(["中文", "英文", "日文", "中英混合", "日英混合", "多语种混合"])
self.ref_language_combobox.setCurrentText("多语种混合")
self.target_text_label = QLabel("合成目标文本:")
self.target_text_input = QLineEdit()
self.target_text_input.setPlaceholderText("拖拽或选择文件")
self.target_text_input.setReadOnly(True)
self.target_text_input.setPlaceholderText("直接输入文字或上传文本")
self.target_text_button = QPushButton("上传文本")
self.target_text_button.clicked.connect(self.upload_target_text)
self.language_label_02 = QLabel("合成音频语言:")
self.language_combobox_02 = QComboBox()
self.language_combobox_02.addItems(["中文", "英文", "日文"])
self.target_language_label = QLabel("合成音频语言:")
self.target_language_combobox = QComboBox()
self.target_language_combobox.addItems(["中文", "英文", "日文", "中英混合", "日英混合", "多语种混合"])
self.target_language_combobox.setCurrentText("多语种混合")
self.output_label = QLabel("输出音频路径:")
self.output_input = QLineEdit()
@ -140,10 +143,8 @@ class GPTSoVITSGUI(QMainWindow):
main_layout = QVBoxLayout()
input_layout = QGridLayout()
input_layout.setSpacing(10)
self.setLayout(input_layout)
input_layout = QGridLayout(self)
input_layout.setSpacing(10)
input_layout.addWidget(license_label, 0, 0, 1, 3)
@ -159,22 +160,22 @@ class GPTSoVITSGUI(QMainWindow):
input_layout.addWidget(self.ref_audio_input, 6, 0, 1, 2)
input_layout.addWidget(self.ref_audio_button, 6, 2)
input_layout.addWidget(self.language_label, 7, 0)
input_layout.addWidget(self.language_combobox, 8, 0, 1, 1)
input_layout.addWidget(self.ref_language_label, 7, 0)
input_layout.addWidget(self.ref_language_combobox, 8, 0, 1, 1)
input_layout.addWidget(self.ref_text_label, 9, 0)
input_layout.addWidget(self.ref_text_input, 10, 0, 1, 2)
input_layout.addWidget(self.ref_text_button, 10, 2)
input_layout.addWidget(self.language_label_02, 11, 0)
input_layout.addWidget(self.language_combobox_02, 12, 0, 1, 1)
input_layout.addWidget(self.target_language_label, 11, 0)
input_layout.addWidget(self.target_language_combobox, 12, 0, 1, 1)
input_layout.addWidget(self.target_text_label, 13, 0)
input_layout.addWidget(self.target_text_input, 14, 0, 1, 2)
input_layout.addWidget(self.target_text_button, 14, 2)
input_layout.addWidget(self.output_label, 15, 0)
input_layout.addWidget(self.output_input, 16, 0, 1, 2)
input_layout.addWidget(self.output_button, 16, 2)
main_layout.addLayout(input_layout)
output_layout = QVBoxLayout()
@ -198,10 +199,8 @@ class GPTSoVITSGUI(QMainWindow):
def dropEvent(self, event):
if event.mimeData().hasUrls():
file_paths = [url.toLocalFile() for url in event.mimeData().urls()]
if len(file_paths) == 1:
self.update_ref_audio(file_paths[0])
self.update_input_paths(self.ref_audio_input, file_paths[0])
else:
self.update_ref_audio(", ".join(file_paths))
@ -211,23 +210,13 @@ class GPTSoVITSGUI(QMainWindow):
widget.installEventFilter(self)
def eventFilter(self, obj, event):
if event.type() == QEvent.DragEnter:
if event.type() in (QEvent.DragEnter, QEvent.Drop):
mime_data = event.mimeData()
if mime_data.hasUrls():
event.acceptProposedAction()
elif event.type() == QEvent.Drop:
mime_data = event.mimeData()
if mime_data.hasUrls():
file_paths = [url.toLocalFile() for url in mime_data.urls()]
if len(file_paths) == 1:
self.update_input_paths(obj, file_paths[0])
else:
self.update_input_paths(obj, ", ".join(file_paths))
event.acceptProposedAction()
return super().eventFilter(obj, event)
def select_GPT_model(self):
file_path, _ = QFileDialog.getOpenFileName(self, "选择GPT模型文件", "", "GPT Files (*.ckpt)")
if file_path:
@ -239,24 +228,9 @@ class GPTSoVITSGUI(QMainWindow):
self.SoVITS_model_input.setText(file_path)
def select_ref_audio(self):
options = QFileDialog.Options()
options |= QFileDialog.DontUseNativeDialog
options |= QFileDialog.ShowDirsOnly
file_dialog = QFileDialog()
file_dialog.setOptions(options)
file_dialog.setFileMode(QFileDialog.AnyFile)
file_dialog.setNameFilter("Audio Files (*.wav *.mp3)")
if file_dialog.exec_():
file_paths = file_dialog.selectedFiles()
if len(file_paths) == 1:
self.update_ref_audio(file_paths[0])
self.update_input_paths(self.ref_audio_input, file_paths[0])
else:
self.update_ref_audio(", ".join(file_paths))
file_path, _ = QFileDialog.getOpenFileName(self, "选择参考音频文件", "", "Audio Files (*.wav *.mp3)")
if file_path:
self.update_ref_audio(file_path)
def upload_ref_text(self):
file_path, _ = QFileDialog.getOpenFileName(self, "选择文本文件", "", "Text Files (*.txt)")
@ -264,7 +238,6 @@ class GPTSoVITSGUI(QMainWindow):
with open(file_path, 'r', encoding='utf-8') as file:
content = file.read()
self.ref_text_input.setText(content)
self.update_input_paths(self.ref_text_input, file_path)
def upload_target_text(self):
file_path, _ = QFileDialog.getOpenFileName(self, "选择文本文件", "", "Text Files (*.txt)")
@ -272,7 +245,6 @@ class GPTSoVITSGUI(QMainWindow):
with open(file_path, 'r', encoding='utf-8') as file:
content = file.read()
self.target_text_input.setText(content)
self.update_input_paths(self.target_text_input, file_path)
def select_output_path(self):
options = QFileDialog.Options()
@ -290,9 +262,6 @@ class GPTSoVITSGUI(QMainWindow):
def update_ref_audio(self, file_path):
self.ref_audio_input.setText(file_path)
def update_input_paths(self, input_box, file_path):
input_box.setText(file_path)
def clear_output(self):
self.output_text.clear()
@ -300,23 +269,27 @@ class GPTSoVITSGUI(QMainWindow):
GPT_model_path = self.GPT_model_input.text()
SoVITS_model_path = self.SoVITS_model_input.text()
ref_audio_path = self.ref_audio_input.text()
language_combobox = self.language_combobox.currentText()
language_combobox = self.ref_language_combobox.currentText()
language_combobox = i18n(language_combobox)
ref_text = self.ref_text_input.text()
language_combobox_02 = self.language_combobox_02.currentText()
language_combobox_02 = i18n(language_combobox_02)
target_language_combobox = self.target_language_combobox.currentText()
target_language_combobox = i18n(target_language_combobox)
target_text = self.target_text_input.text()
output_path = self.output_input.text()
change_gpt_weights(gpt_path=GPT_model_path)
change_sovits_weights(sovits_path=SoVITS_model_path)
if GPT_model_path != self.GPT_Path:
change_gpt_weights(gpt_path=GPT_model_path)
self.GPT_Path = GPT_model_path
if SoVITS_model_path != self.SoVITS_Path:
change_sovits_weights(sovits_path=SoVITS_model_path)
self.SoVITS_Path = SoVITS_model_path
synthesis_result = get_tts_wav(ref_wav_path=ref_audio_path,
prompt_text=ref_text,
prompt_language=language_combobox,
text=target_text,
text_language=language_combobox_02)
text_language=target_language_combobox)
result_list = list(synthesis_result)
if result_list:
@ -329,12 +302,9 @@ class GPTSoVITSGUI(QMainWindow):
self.status_bar.showMessage("合成完成!输出路径:" + output_wav_path, 5000)
self.output_text.append("处理结果:\n" + result)
def main():
if __name__ == '__main__':
app = QApplication(sys.argv)
mainWin = GPTSoVITSGUI()
mainWin.show()
sys.exit(app.exec_())
if __name__ == '__main__':
main()
sys.exit(app.exec_())

File diff suppressed because it is too large Load Diff

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@ -0,0 +1,392 @@
'''
按中英混合识别
按日英混合识别
多语种启动切分识别语种
全部按中文识别
全部按英文识别
全部按日文识别
'''
import random
import os, re, logging, json
import sys
now_dir = os.getcwd()
sys.path.append(now_dir)
sys.path.append("%s/GPT_SoVITS" % (now_dir))
logging.getLogger("markdown_it").setLevel(logging.ERROR)
logging.getLogger("urllib3").setLevel(logging.ERROR)
logging.getLogger("httpcore").setLevel(logging.ERROR)
logging.getLogger("httpx").setLevel(logging.ERROR)
logging.getLogger("asyncio").setLevel(logging.ERROR)
logging.getLogger("charset_normalizer").setLevel(logging.ERROR)
logging.getLogger("torchaudio._extension").setLevel(logging.ERROR)
import pdb
import torch
try:
import gradio.analytics as analytics
analytics.version_check = lambda:None
except:...
infer_ttswebui = os.environ.get("infer_ttswebui", 9872)
infer_ttswebui = int(infer_ttswebui)
is_share = os.environ.get("is_share", "False")
is_share = eval(is_share)
if "_CUDA_VISIBLE_DEVICES" in os.environ:
os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"]
is_half = eval(os.environ.get("is_half", "True")) and torch.cuda.is_available()
gpt_path = os.environ.get("gpt_path", None)
sovits_path = os.environ.get("sovits_path", None)
cnhubert_base_path = os.environ.get("cnhubert_base_path", None)
bert_path = os.environ.get("bert_path", None)
version=model_version=os.environ.get("version","v2")
import gradio as gr
from TTS_infer_pack.TTS import TTS, TTS_Config, NO_PROMPT_ERROR
from TTS_infer_pack.text_segmentation_method import get_method
from tools.i18n.i18n import I18nAuto, scan_language_list
from inference_webui import DictToAttrRecursive
language=os.environ.get("language","Auto")
language=sys.argv[-1] if sys.argv[-1] in scan_language_list() else language
i18n = I18nAuto(language=language)
# os.environ['PYTORCH_ENABLE_MPS_FALLBACK'] = '1' # 确保直接启动推理UI时也能够设置。
if torch.cuda.is_available():
device = "cuda"
# elif torch.backends.mps.is_available():
# device = "mps"
else:
device = "cpu"
# is_half = False
# device = "cpu"
dict_language_v1 = {
i18n("中文"): "all_zh",#全部按中文识别
i18n("英文"): "en",#全部按英文识别#######不变
i18n("日文"): "all_ja",#全部按日文识别
i18n("中英混合"): "zh",#按中英混合识别####不变
i18n("日英混合"): "ja",#按日英混合识别####不变
i18n("多语种混合"): "auto",#多语种启动切分识别语种
}
dict_language_v2 = {
i18n("中文"): "all_zh",#全部按中文识别
i18n("英文"): "en",#全部按英文识别#######不变
i18n("日文"): "all_ja",#全部按日文识别
i18n("粤语"): "all_yue",#全部按中文识别
i18n("韩文"): "all_ko",#全部按韩文识别
i18n("中英混合"): "zh",#按中英混合识别####不变
i18n("日英混合"): "ja",#按日英混合识别####不变
i18n("粤英混合"): "yue",#按粤英混合识别####不变
i18n("韩英混合"): "ko",#按韩英混合识别####不变
i18n("多语种混合"): "auto",#多语种启动切分识别语种
i18n("多语种混合(粤语)"): "auto_yue",#多语种启动切分识别语种
}
dict_language = dict_language_v1 if version =='v1' else dict_language_v2
cut_method = {
i18n("不切"):"cut0",
i18n("凑四句一切"): "cut1",
i18n("凑50字一切"): "cut2",
i18n("按中文句号。切"): "cut3",
i18n("按英文句号.切"): "cut4",
i18n("按标点符号切"): "cut5",
}
tts_config = TTS_Config("GPT_SoVITS/configs/tts_infer.yaml")
tts_config.device = device
tts_config.is_half = is_half
tts_config.version = version
if gpt_path is not None:
tts_config.t2s_weights_path = gpt_path
if sovits_path is not None:
tts_config.vits_weights_path = sovits_path
if cnhubert_base_path is not None:
tts_config.cnhuhbert_base_path = cnhubert_base_path
if bert_path is not None:
tts_config.bert_base_path = bert_path
print(tts_config)
tts_pipeline = TTS(tts_config)
gpt_path = tts_config.t2s_weights_path
sovits_path = tts_config.vits_weights_path
version = tts_config.version
def inference(text, text_lang,
ref_audio_path,
aux_ref_audio_paths,
prompt_text,
prompt_lang, top_k,
top_p, temperature,
text_split_method, batch_size,
speed_factor, ref_text_free,
split_bucket,fragment_interval,
seed, keep_random, parallel_infer,
repetition_penalty, sample_steps, super_sampling,
):
seed = -1 if keep_random else seed
actual_seed = seed if seed not in [-1, "", None] else random.randint(0, 2**32 - 1)
inputs={
"text": text,
"text_lang": dict_language[text_lang],
"ref_audio_path": ref_audio_path,
"aux_ref_audio_paths": [item.name for item in aux_ref_audio_paths] if aux_ref_audio_paths is not None else [],
"prompt_text": prompt_text if not ref_text_free else "",
"prompt_lang": dict_language[prompt_lang],
"top_k": top_k,
"top_p": top_p,
"temperature": temperature,
"text_split_method": cut_method[text_split_method],
"batch_size":int(batch_size),
"speed_factor":float(speed_factor),
"split_bucket":split_bucket,
"return_fragment":False,
"fragment_interval":fragment_interval,
"seed":actual_seed,
"parallel_infer": parallel_infer,
"repetition_penalty": repetition_penalty,
"sample_steps": int(sample_steps),
"super_sampling": super_sampling,
}
try:
for item in tts_pipeline.run(inputs):
yield item, actual_seed
except NO_PROMPT_ERROR:
gr.Warning(i18n('V3不支持无参考文本模式请填写参考文本'))
def custom_sort_key(s):
# 使用正则表达式提取字符串中的数字部分和非数字部分
parts = re.split('(\d+)', s)
# 将数字部分转换为整数,非数字部分保持不变
parts = [int(part) if part.isdigit() else part for part in parts]
return parts
def change_choices():
SoVITS_names, GPT_names = get_weights_names(GPT_weight_root, SoVITS_weight_root)
return {"choices": sorted(SoVITS_names, key=custom_sort_key), "__type__": "update"}, {"choices": sorted(GPT_names, key=custom_sort_key), "__type__": "update"}
path_sovits_v3="GPT_SoVITS/pretrained_models/s2Gv3.pth"
pretrained_sovits_name=["GPT_SoVITS/pretrained_models/s2G488k.pth", "GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s2G2333k.pth",path_sovits_v3]
pretrained_gpt_name=["GPT_SoVITS/pretrained_models/s1bert25hz-2kh-longer-epoch=68e-step=50232.ckpt","GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s1bert25hz-5kh-longer-epoch=12-step=369668.ckpt", "GPT_SoVITS/pretrained_models/s1v3.ckpt"]
_ =[[],[]]
for i in range(3):
if os.path.exists(pretrained_gpt_name[i]):_[0].append(pretrained_gpt_name[i])
if os.path.exists(pretrained_sovits_name[i]):_[-1].append(pretrained_sovits_name[i])
pretrained_gpt_name,pretrained_sovits_name = _
if os.path.exists(f"./weight.json"):
pass
else:
with open(f"./weight.json", 'w', encoding="utf-8") as file:json.dump({'GPT':{},'SoVITS':{}},file)
with open(f"./weight.json", 'r', encoding="utf-8") as file:
weight_data = file.read()
weight_data=json.loads(weight_data)
gpt_path = os.environ.get(
"gpt_path", weight_data.get('GPT',{}).get(version,pretrained_gpt_name))
sovits_path = os.environ.get(
"sovits_path", weight_data.get('SoVITS',{}).get(version,pretrained_sovits_name))
if isinstance(gpt_path,list):
gpt_path = gpt_path[0]
if isinstance(sovits_path,list):
sovits_path = sovits_path[0]
SoVITS_weight_root=["SoVITS_weights","SoVITS_weights_v2","SoVITS_weights_v3"]
GPT_weight_root=["GPT_weights","GPT_weights_v2","GPT_weights_v3"]
for path in SoVITS_weight_root+GPT_weight_root:
os.makedirs(path,exist_ok=True)
def get_weights_names(GPT_weight_root, SoVITS_weight_root):
SoVITS_names = [i for i in pretrained_sovits_name]
for path in SoVITS_weight_root:
for name in os.listdir(path):
if name.endswith(".pth"): SoVITS_names.append("%s/%s" % (path, name))
GPT_names = [i for i in pretrained_gpt_name]
for path in GPT_weight_root:
for name in os.listdir(path):
if name.endswith(".ckpt"): GPT_names.append("%s/%s" % (path, name))
return SoVITS_names, GPT_names
SoVITS_names, GPT_names = get_weights_names(GPT_weight_root, SoVITS_weight_root)
from process_ckpt import get_sovits_version_from_path_fast,load_sovits_new
def change_sovits_weights(sovits_path,prompt_language=None,text_language=None):
global version, model_version, dict_language,if_lora_v3
version, model_version, if_lora_v3=get_sovits_version_from_path_fast(sovits_path)
# print(sovits_path,version, model_version, if_lora_v3)
if if_lora_v3 and not os.path.exists(path_sovits_v3):
info= path_sovits_v3 + i18n("SoVITS V3 底模缺失,无法加载相应 LoRA 权重")
gr.Warning(info)
raise FileExistsError(info)
dict_language = dict_language_v1 if version =='v1' else dict_language_v2
if prompt_language is not None and text_language is not None:
if prompt_language in list(dict_language.keys()):
prompt_text_update, prompt_language_update = {'__type__':'update'}, {'__type__':'update', 'value':prompt_language}
else:
prompt_text_update = {'__type__':'update', 'value':''}
prompt_language_update = {'__type__':'update', 'value':i18n("中文")}
if text_language in list(dict_language.keys()):
text_update, text_language_update = {'__type__':'update'}, {'__type__':'update', 'value':text_language}
else:
text_update = {'__type__':'update', 'value':''}
text_language_update = {'__type__':'update', 'value':i18n("中文")}
if model_version=="v3":
visible_sample_steps=True
visible_inp_refs=False
else:
visible_sample_steps=False
visible_inp_refs=True
#prompt_language,text_language,prompt_text,prompt_language,text,text_language,inp_refs,ref_text_free,
yield {'__type__':'update', 'choices':list(dict_language.keys())}, {'__type__':'update', 'choices':list(dict_language.keys())}, prompt_text_update, prompt_language_update, text_update, text_language_update,{"__type__": "update", "interactive": visible_sample_steps,"value":32},{"__type__": "update", "visible": visible_inp_refs},{"__type__": "update", "interactive": True if model_version!="v3"else False},{"__type__": "update", "value":i18n("模型加载中,请等待"),"interactive":False}
tts_pipeline.init_vits_weights(sovits_path)
yield {'__type__':'update', 'choices':list(dict_language.keys())}, {'__type__':'update', 'choices':list(dict_language.keys())}, prompt_text_update, prompt_language_update, text_update, text_language_update,{"__type__": "update", "interactive": visible_sample_steps,"value":32},{"__type__": "update", "visible": visible_inp_refs},{"__type__": "update", "interactive": True if model_version!="v3"else False},{"__type__": "update", "value":i18n("合成语音"),"interactive":True}
with open("./weight.json")as f:
data=f.read()
data=json.loads(data)
data["SoVITS"][version]=sovits_path
with open("./weight.json","w")as f:f.write(json.dumps(data))
with gr.Blocks(title="GPT-SoVITS WebUI") as app:
gr.Markdown(
value=i18n("本软件以MIT协议开源, 作者不对软件具备任何控制力, 使用软件者、传播软件导出的声音者自负全责.") + "<br>" + i18n("如不认可该条款, 则不能使用或引用软件包内任何代码和文件. 详见根目录LICENSE.")
)
with gr.Column():
# with gr.Group():
gr.Markdown(value=i18n("模型切换"))
with gr.Row():
GPT_dropdown = gr.Dropdown(label=i18n("GPT模型列表"), choices=sorted(GPT_names, key=custom_sort_key), value=gpt_path, interactive=True)
SoVITS_dropdown = gr.Dropdown(label=i18n("SoVITS模型列表"), choices=sorted(SoVITS_names, key=custom_sort_key), value=sovits_path, interactive=True)
refresh_button = gr.Button(i18n("刷新模型路径"), variant="primary")
refresh_button.click(fn=change_choices, inputs=[], outputs=[SoVITS_dropdown, GPT_dropdown])
with gr.Row():
with gr.Column():
gr.Markdown(value=i18n("*请上传并填写参考信息"))
with gr.Row():
inp_ref = gr.Audio(label=i18n("主参考音频(请上传3~10秒内参考音频超过会报错)"), type="filepath")
inp_refs = gr.File(label=i18n("辅参考音频(可选多个,或不选)"),file_count="multiple", visible=True if model_version!="v3"else False)
prompt_text = gr.Textbox(label=i18n("主参考音频的文本"), value="", lines=2)
with gr.Row():
prompt_language = gr.Dropdown(
label=i18n("主参考音频的语种"), choices=list(dict_language.keys()), value=i18n("中文")
)
with gr.Column():
ref_text_free = gr.Checkbox(label=i18n("开启无参考文本模式。不填参考文本亦相当于开启。"), value=False, interactive=True if model_version!="v3"else False, show_label=True)
gr.Markdown(i18n("使用无参考文本模式时建议使用微调的GPT")+"<br>"+i18n("听不清参考音频说的啥(不晓得写啥)可以开。开启后无视填写的参考文本。"))
with gr.Column():
gr.Markdown(value=i18n("*请填写需要合成的目标文本和语种模式"))
text = gr.Textbox(label=i18n("需要合成的文本"), value="", lines=20, max_lines=20)
text_language = gr.Dropdown(
label=i18n("需要合成的文本的语种"), choices=list(dict_language.keys()), value=i18n("中文")
)
with gr.Group():
gr.Markdown(value=i18n("推理设置"))
with gr.Row():
with gr.Column():
with gr.Row():
batch_size = gr.Slider(minimum=1,maximum=200,step=1,label=i18n("batch_size"),value=20,interactive=True)
sample_steps = gr.Radio(label=i18n("采样步数(仅对V3生效)"),value=32,choices=[4,8,16,32],visible=True)
with gr.Row():
fragment_interval = gr.Slider(minimum=0.01,maximum=1,step=0.01,label=i18n("分段间隔(秒)"),value=0.3,interactive=True)
speed_factor = gr.Slider(minimum=0.6,maximum=1.65,step=0.05,label="语速",value=1.0,interactive=True)
with gr.Row():
top_k = gr.Slider(minimum=1,maximum=100,step=1,label=i18n("top_k"),value=5,interactive=True)
top_p = gr.Slider(minimum=0,maximum=1,step=0.05,label=i18n("top_p"),value=1,interactive=True)
with gr.Row():
temperature = gr.Slider(minimum=0,maximum=1,step=0.05,label=i18n("temperature"),value=1,interactive=True)
repetition_penalty = gr.Slider(minimum=0,maximum=2,step=0.05,label=i18n("重复惩罚"),value=1.35,interactive=True)
with gr.Column():
with gr.Row():
how_to_cut = gr.Dropdown(
label=i18n("怎么切"),
choices=[i18n("不切"), i18n("凑四句一切"), i18n("凑50字一切"), i18n("按中文句号。切"), i18n("按英文句号.切"), i18n("按标点符号切"), ],
value=i18n("凑四句一切"),
interactive=True, scale=1
)
super_sampling = gr.Checkbox(label=i18n("音频超采样(仅对V3生效))"), value=False, interactive=True, show_label=True)
with gr.Row():
parallel_infer = gr.Checkbox(label=i18n("并行推理"), value=True, interactive=True, show_label=True)
split_bucket = gr.Checkbox(label=i18n("数据分桶(并行推理时会降低一点计算量)"), value=True, interactive=True, show_label=True)
with gr.Row():
seed = gr.Number(label=i18n("随机种子"),value=-1)
keep_random = gr.Checkbox(label=i18n("保持随机"), value=True, interactive=True, show_label=True)
output = gr.Audio(label=i18n("输出的语音"))
with gr.Row():
inference_button = gr.Button(i18n("合成语音"), variant="primary")
stop_infer = gr.Button(i18n("终止合成"), variant="primary")
inference_button.click(
inference,
[
text,text_language, inp_ref, inp_refs,
prompt_text, prompt_language,
top_k, top_p, temperature,
how_to_cut, batch_size,
speed_factor, ref_text_free,
split_bucket,fragment_interval,
seed, keep_random, parallel_infer,
repetition_penalty, sample_steps, super_sampling,
],
[output, seed],
)
stop_infer.click(tts_pipeline.stop, [], [])
SoVITS_dropdown.change(change_sovits_weights, [SoVITS_dropdown,prompt_language,text_language], [prompt_language,text_language,prompt_text,prompt_language,text,text_language,sample_steps,inp_refs,ref_text_free,inference_button])#
GPT_dropdown.change(tts_pipeline.init_t2s_weights, [GPT_dropdown], [])
with gr.Group():
gr.Markdown(value=i18n("文本切分工具。太长的文本合成出来效果不一定好,所以太长建议先切。合成会根据文本的换行分开合成再拼起来。"))
with gr.Row():
text_inp = gr.Textbox(label=i18n("需要合成的切分前文本"), value="", lines=4)
with gr.Column():
_how_to_cut = gr.Radio(
label=i18n("怎么切"),
choices=[i18n("不切"), i18n("凑四句一切"), i18n("凑50字一切"), i18n("按中文句号。切"), i18n("按英文句号.切"), i18n("按标点符号切"), ],
value=i18n("凑四句一切"),
interactive=True,
)
cut_text= gr.Button(i18n("切分"), variant="primary")
def to_cut(text_inp, how_to_cut):
if len(text_inp.strip()) == 0 or text_inp==[]:
return ""
method = get_method(cut_method[how_to_cut])
return method(text_inp)
text_opt = gr.Textbox(label=i18n("切分后文本"), value="", lines=4)
cut_text.click(to_cut, [text_inp, _how_to_cut], [text_opt])
gr.Markdown(value=i18n("后续将支持转音素、手工修改音素、语音合成分步执行。"))
if __name__ == '__main__':
app.queue().launch(#concurrency_count=511, max_size=1022
server_name="0.0.0.0",
inbrowser=True,
share=is_share,
server_port=infer_ttswebui,
quiet=True,
)

View File

@ -4,8 +4,8 @@ from torch import nn
from torch.nn import functional as F
from module import commons
from module.modules import LayerNorm
from typing import Optional
class LayerNorm(nn.Module):
def __init__(self, channels, eps=1e-5):
@ -59,6 +59,7 @@ class Encoder(nn.Module):
# self.cond_layer = weight_norm(cond_layer, name='weight')
# self.gin_channels = 256
self.cond_layer_idx = self.n_layers
self.spk_emb_linear = nn.Linear(256, self.hidden_channels)
if "gin_channels" in kwargs:
self.gin_channels = kwargs["gin_channels"]
if self.gin_channels != 0:
@ -98,22 +99,36 @@ class Encoder(nn.Module):
)
self.norm_layers_2.append(LayerNorm(hidden_channels))
def forward(self, x, x_mask, g=None):
# def forward(self, x, x_mask, g=None):
# attn_mask = x_mask.unsqueeze(2) * x_mask.unsqueeze(-1)
# x = x * x_mask
# for i in range(self.n_layers):
# if i == self.cond_layer_idx and g is not None:
# g = self.spk_emb_linear(g.transpose(1, 2))
# g = g.transpose(1, 2)
# x = x + g
# x = x * x_mask
# y = self.attn_layers[i](x, x, attn_mask)
# y = self.drop(y)
# x = self.norm_layers_1[i](x + y)
# y = self.ffn_layers[i](x, x_mask)
# y = self.drop(y)
# x = self.norm_layers_2[i](x + y)
# x = x * x_mask
# return x
def forward(self, x, x_mask):
attn_mask = x_mask.unsqueeze(2) * x_mask.unsqueeze(-1)
x = x * x_mask
for i in range(self.n_layers):
if i == self.cond_layer_idx and g is not None:
g = self.spk_emb_linear(g.transpose(1, 2))
g = g.transpose(1, 2)
x = x + g
x = x * x_mask
y = self.attn_layers[i](x, x, attn_mask)
for attn_layers,norm_layers_1,ffn_layers,norm_layers_2 in zip(self.attn_layers,self.norm_layers_1,self.ffn_layers,self.norm_layers_2):
y = attn_layers(x, x, attn_mask)
y = self.drop(y)
x = self.norm_layers_1[i](x + y)
x = norm_layers_1(x + y)
y = self.ffn_layers[i](x, x_mask)
y = ffn_layers(x, x_mask)
y = self.drop(y)
x = self.norm_layers_2[i](x + y)
x = norm_layers_2(x + y)
x = x * x_mask
return x
@ -172,17 +187,18 @@ class MultiHeadAttention(nn.Module):
self.conv_k.weight.copy_(self.conv_q.weight)
self.conv_k.bias.copy_(self.conv_q.bias)
def forward(self, x, c, attn_mask=None):
def forward(self, x, c, attn_mask:Optional[torch.Tensor]=None):
q = self.conv_q(x)
k = self.conv_k(c)
v = self.conv_v(c)
x, self.attn = self.attention(q, k, v, mask=attn_mask)
# x, self.attn = self.attention(q, k, v, mask=attn_mask)
x, _ = self.attention(q, k, v, mask=attn_mask)
x = self.conv_o(x)
return x
def attention(self, query, key, value, mask=None):
def attention(self, query, key, value, mask:Optional[torch.Tensor]=None):
# reshape [b, d, t] -> [b, n_h, t, d_k]
b, d, t_s, _ = (*key.size(), query.size(2))
query = query.view(b, self.n_heads, self.k_channels, -1).transpose(2, 3)
@ -304,7 +320,7 @@ class FFN(nn.Module):
filter_channels,
kernel_size,
p_dropout=0.0,
activation=None,
activation="",
causal=False,
):
super().__init__()
@ -316,10 +332,11 @@ class FFN(nn.Module):
self.activation = activation
self.causal = causal
if causal:
self.padding = self._causal_padding
else:
self.padding = self._same_padding
# 从上下文看这里一定是 False
# if causal:
# self.padding = self._causal_padding
# else:
# self.padding = self._same_padding
self.conv_1 = nn.Conv1d(in_channels, filter_channels, kernel_size)
self.conv_2 = nn.Conv1d(filter_channels, out_channels, kernel_size)
@ -334,6 +351,9 @@ class FFN(nn.Module):
x = self.drop(x)
x = self.conv_2(self.padding(x * x_mask))
return x * x_mask
def padding(self, x):
return self._same_padding(x)
def _causal_padding(self, x):
if self.kernel_size == 1:
@ -352,3 +372,35 @@ class FFN(nn.Module):
padding = [[0, 0], [0, 0], [pad_l, pad_r]]
x = F.pad(x, commons.convert_pad_shape(padding))
return x
class MRTE(nn.Module):
def __init__(
self,
content_enc_channels=192,
hidden_size=512,
out_channels=192,
kernel_size=5,
n_heads=4,
ge_layer=2,
):
super(MRTE, self).__init__()
self.cross_attention = MultiHeadAttention(hidden_size, hidden_size, n_heads)
self.c_pre = nn.Conv1d(content_enc_channels, hidden_size, 1)
self.text_pre = nn.Conv1d(content_enc_channels, hidden_size, 1)
self.c_post = nn.Conv1d(hidden_size, out_channels, 1)
def forward(self, ssl_enc, ssl_mask, text, text_mask, ge):
attn_mask = text_mask.unsqueeze(2) * ssl_mask.unsqueeze(-1)
ssl_enc = self.c_pre(ssl_enc * ssl_mask)
text_enc = self.text_pre(text * text_mask)
x = (
self.cross_attention(
ssl_enc * ssl_mask, text_enc * text_mask, attn_mask
)
+ ssl_enc
+ ge
)
x = self.c_post(x * ssl_mask)
return x

View File

@ -13,10 +13,10 @@ def get_padding(kernel_size, dilation=1):
return int((kernel_size * dilation - dilation) / 2)
def convert_pad_shape(pad_shape):
l = pad_shape[::-1]
pad_shape = [item for sublist in l for item in sublist]
return pad_shape
# def convert_pad_shape(pad_shape):
# l = pad_shape[::-1]
# pad_shape = [item for sublist in l for item in sublist]
# return pad_shape
def intersperse(lst, item):

View File

@ -9,7 +9,7 @@ import torch.utils.data
from tqdm import tqdm
from module import commons
from module.mel_processing import spectrogram_torch
from module.mel_processing import spectrogram_torch,spec_to_mel_torch
from text import cleaned_text_to_sequence
from utils import load_wav_to_torch, load_filepaths_and_text
import torch.nn.functional as F
@ -17,8 +17,8 @@ from functools import lru_cache
import requests
from scipy.io import wavfile
from io import BytesIO
from my_utils import load_audio
from tools.my_utils import load_audio
version = os.environ.get('version',None)
# ZeroDivisionError fixed by Tybost (https://github.com/RVC-Boss/GPT-SoVITS/issues/79)
class TextAudioSpeakerLoader(torch.utils.data.Dataset):
"""
@ -77,7 +77,7 @@ class TextAudioSpeakerLoader(torch.utils.data.Dataset):
try:
phoneme = self.phoneme_data[audiopath][0]
phoneme = phoneme.split(' ')
phoneme_ids = cleaned_text_to_sequence(phoneme)
phoneme_ids = cleaned_text_to_sequence(phoneme, version)
except Exception:
print(f"{audiopath} not in self.phoneme_data !")
skipped_phone += 1
@ -170,8 +170,6 @@ class TextAudioSpeakerLoader(torch.utils.data.Dataset):
assert abs(ssl.shape[-1] - wav2.shape[-1] // self.hop_length) < 3, (
ssl.shape, wav.shape, wav2.shape, mel.shape, sep_point, self.hop_length, sep_point * self.hop_length, dir)
return reference_mel, ssl, wav2, mel
class TextAudioSpeakerCollate():
""" Zero-pads model inputs and targets
"""
@ -232,7 +230,457 @@ class TextAudioSpeakerCollate():
text_lengths[i] = text.size(0)
return ssl_padded, ssl_lengths, spec_padded, spec_lengths, wav_padded, wav_lengths, text_padded, text_lengths
class TextAudioSpeakerLoaderV3(torch.utils.data.Dataset):
"""
1) loads audio, speaker_id, text pairs
2) normalizes text and converts them to sequences of integers
3) computes spectrograms from audio files.
"""
def __init__(self, hparams, val=False):
exp_dir = hparams.exp_dir
self.path2 = "%s/2-name2text.txt" % exp_dir
self.path4 = "%s/4-cnhubert" % exp_dir
self.path5 = "%s/5-wav32k" % exp_dir
assert os.path.exists(self.path2)
assert os.path.exists(self.path4)
assert os.path.exists(self.path5)
names4 = set([name[:-3] for name in list(os.listdir(self.path4))]) # 去除.pt后缀
names5 = set(os.listdir(self.path5))
self.phoneme_data = {}
with open(self.path2, "r", encoding="utf8") as f:
lines = f.read().strip("\n").split("\n")
for line in lines:
tmp = line.split("\t")
if (len(tmp) != 4):
continue
self.phoneme_data[tmp[0]] = [tmp[1]]
self.audiopaths_sid_text = list(set(self.phoneme_data) & names4 & names5)
tmp = self.audiopaths_sid_text
leng = len(tmp)
min_num = 100
if (leng < min_num):
self.audiopaths_sid_text = []
for _ in range(max(2, int(min_num / leng))):
self.audiopaths_sid_text += tmp
self.max_wav_value = hparams.max_wav_value
self.sampling_rate = hparams.sampling_rate
self.filter_length = hparams.filter_length
self.hop_length = hparams.hop_length
self.win_length = hparams.win_length
self.sampling_rate = hparams.sampling_rate
self.val = val
random.seed(1234)
random.shuffle(self.audiopaths_sid_text)
print("phoneme_data_len:", len(self.phoneme_data.keys()))
print("wav_data_len:", len(self.audiopaths_sid_text))
audiopaths_sid_text_new = []
lengths = []
skipped_phone = 0
skipped_dur = 0
for audiopath in tqdm(self.audiopaths_sid_text):
try:
phoneme = self.phoneme_data[audiopath][0]
phoneme = phoneme.split(' ')
phoneme_ids = cleaned_text_to_sequence(phoneme, version)
except Exception:
print(f"{audiopath} not in self.phoneme_data !")
skipped_phone += 1
continue
size = os.path.getsize("%s/%s" % (self.path5, audiopath))
duration = size / self.sampling_rate / 2
if duration == 0:
print(f"Zero duration for {audiopath}, skipping...")
skipped_dur += 1
continue
if 54 > duration > 0.6 or self.val:
audiopaths_sid_text_new.append([audiopath, phoneme_ids])
lengths.append(size // (2 * self.hop_length))
else:
skipped_dur += 1
continue
print("skipped_phone: ", skipped_phone, ", skipped_dur: ", skipped_dur)
print("total left: ", len(audiopaths_sid_text_new))
assert len(audiopaths_sid_text_new) > 1 # 至少能凑够batch size这里todo
self.audiopaths_sid_text = audiopaths_sid_text_new
self.lengths = lengths
self.spec_min=-12
self.spec_max=2
self.filter_length_mel=self.win_length_mel=1024
self.hop_length_mel=256
self.n_mel_channels=100
self.sampling_rate_mel=24000
self.mel_fmin=0
self.mel_fmax=None
def norm_spec(self, x):
return (x - self.spec_min) / (self.spec_max - self.spec_min) * 2 - 1
def get_audio_text_speaker_pair(self, audiopath_sid_text):
audiopath, phoneme_ids = audiopath_sid_text
text = torch.FloatTensor(phoneme_ids)
try:
spec, mel = self.get_audio("%s/%s" % (self.path5, audiopath))
with torch.no_grad():
ssl = torch.load("%s/%s.pt" % (self.path4, audiopath), map_location="cpu")
if (ssl.shape[-1] != spec.shape[-1]):
typee = ssl.dtype
ssl = F.pad(ssl.float(), (0, 1), mode="replicate").to(typee)
ssl.requires_grad = False
except:
traceback.print_exc()
mel = torch.zeros(100, 180)
# wav = torch.zeros(1, 96 * self.hop_length)
spec = torch.zeros(1025, 96)
ssl = torch.zeros(1, 768, 96)
text = text[-1:]
print("load audio or ssl error!!!!!!", audiopath)
return (ssl, spec, mel, text)
def get_audio(self, filename):
audio_array = load_audio(filename,self.sampling_rate)#load_audio的方法是已经归一化到-1~1之间的不用再/32768
audio=torch.FloatTensor(audio_array)#/32768
audio_norm = audio
audio_norm = audio_norm.unsqueeze(0)
audio_array24 = load_audio(filename,24000)#load_audio的方法是已经归一化到-1~1之间的不用再/32768######这里可以用GPU重采样加速
audio24=torch.FloatTensor(audio_array24)#/32768
audio_norm24 = audio24
audio_norm24 = audio_norm24.unsqueeze(0)
spec = spectrogram_torch(audio_norm, self.filter_length,
self.sampling_rate, self.hop_length, self.win_length,
center=False)
spec = torch.squeeze(spec, 0)
spec1 = spectrogram_torch(audio_norm24, self.filter_length_mel,self.sampling_rate_mel, self.hop_length_mel, self.win_length_mel,center=False)
mel = spec_to_mel_torch(spec1, self.filter_length_mel, self.n_mel_channels, self.sampling_rate_mel, self.mel_fmin, self.mel_fmax)
mel = torch.squeeze(mel, 0)
mel=self.norm_spec(mel)
# print(1111111,spec.shape,mel.shape)
return spec, mel
def get_sid(self, sid):
sid = torch.LongTensor([int(sid)])
return sid
def __getitem__(self, index):
# with torch.no_grad():
return self.get_audio_text_speaker_pair(self.audiopaths_sid_text[index])
def __len__(self):
return len(self.audiopaths_sid_text)
class TextAudioSpeakerCollateV3():
""" Zero-pads model inputs and targets
"""
def __init__(self, return_ids=False):
self.return_ids = return_ids
def __call__(self, batch):
"""Collate's training batch from normalized text, audio and speaker identities
PARAMS
------
batch: [text_normalized, spec_normalized, wav_normalized, sid]
"""
#ssl, spec, wav,mel, text
# Right zero-pad all one-hot text sequences to max input length
_, ids_sorted_decreasing = torch.sort(
torch.LongTensor([x[1].size(1) for x in batch]),
dim=0, descending=True)
#(ssl, spec,mel, text)
max_ssl_len = max([x[0].size(2) for x in batch])
max_ssl_len1 = int(8 * ((max_ssl_len // 8) + 1))
max_ssl_len = int(2 * ((max_ssl_len // 2) + 1))
# max_ssl_len = int(8 * ((max_ssl_len // 8) + 1))
# max_ssl_len1=max_ssl_len
max_spec_len = max([x[1].size(1) for x in batch])
max_spec_len = int(2 * ((max_spec_len // 2) + 1))
# max_wav_len = max([x[2].size(1) for x in batch])
max_text_len = max([x[3].size(0) for x in batch])
max_mel_len=int(max_ssl_len1*1.25*1.5)###24000/256,32000/640=16000/320
ssl_lengths = torch.LongTensor(len(batch))
spec_lengths = torch.LongTensor(len(batch))
text_lengths = torch.LongTensor(len(batch))
# wav_lengths = torch.LongTensor(len(batch))
mel_lengths = torch.LongTensor(len(batch))
spec_padded = torch.FloatTensor(len(batch), batch[0][1].size(0), max_spec_len)
mel_padded = torch.FloatTensor(len(batch), batch[0][2].size(0), max_mel_len)
ssl_padded = torch.FloatTensor(len(batch), batch[0][0].size(1), max_ssl_len)
text_padded = torch.LongTensor(len(batch), max_text_len)
# wav_padded = torch.FloatTensor(len(batch), 1, max_wav_len)
spec_padded.zero_()
mel_padded.zero_()
ssl_padded.zero_()
text_padded.zero_()
# wav_padded.zero_()
for i in range(len(ids_sorted_decreasing)):
row = batch[ids_sorted_decreasing[i]]
# ssl, spec, wav,mel, text
ssl = row[0]
ssl_padded[i, :, :ssl.size(2)] = ssl[0, :, :]
ssl_lengths[i] = ssl.size(2)
spec = row[1]
spec_padded[i, :, :spec.size(1)] = spec
spec_lengths[i] = spec.size(1)
# wav = row[2]
# wav_padded[i, :, :wav.size(1)] = wav
# wav_lengths[i] = wav.size(1)
mel = row[2]
mel_padded[i, :, :mel.size(1)] = mel
mel_lengths[i] = mel.size(1)
text = row[3]
text_padded[i, :text.size(0)] = text
text_lengths[i] = text.size(0)
# return ssl_padded, spec_padded,mel_padded, ssl_lengths, spec_lengths, text_padded, text_lengths, wav_padded, wav_lengths,mel_lengths
return ssl_padded, spec_padded,mel_padded, ssl_lengths, spec_lengths, text_padded, text_lengths,mel_lengths
class TextAudioSpeakerLoaderV3b(torch.utils.data.Dataset):
"""
1) loads audio, speaker_id, text pairs
2) normalizes text and converts them to sequences of integers
3) computes spectrograms from audio files.
"""
def __init__(self, hparams, val=False):
exp_dir = hparams.exp_dir
self.path2 = "%s/2-name2text.txt" % exp_dir
self.path4 = "%s/4-cnhubert" % exp_dir
self.path5 = "%s/5-wav32k" % exp_dir
assert os.path.exists(self.path2)
assert os.path.exists(self.path4)
assert os.path.exists(self.path5)
names4 = set([name[:-3] for name in list(os.listdir(self.path4))]) # 去除.pt后缀
names5 = set(os.listdir(self.path5))
self.phoneme_data = {}
with open(self.path2, "r", encoding="utf8") as f:
lines = f.read().strip("\n").split("\n")
for line in lines:
tmp = line.split("\t")
if (len(tmp) != 4):
continue
self.phoneme_data[tmp[0]] = [tmp[1]]
self.audiopaths_sid_text = list(set(self.phoneme_data) & names4 & names5)
tmp = self.audiopaths_sid_text
leng = len(tmp)
min_num = 100
if (leng < min_num):
self.audiopaths_sid_text = []
for _ in range(max(2, int(min_num / leng))):
self.audiopaths_sid_text += tmp
self.max_wav_value = hparams.max_wav_value
self.sampling_rate = hparams.sampling_rate
self.filter_length = hparams.filter_length
self.hop_length = hparams.hop_length
self.win_length = hparams.win_length
self.sampling_rate = hparams.sampling_rate
self.val = val
random.seed(1234)
random.shuffle(self.audiopaths_sid_text)
print("phoneme_data_len:", len(self.phoneme_data.keys()))
print("wav_data_len:", len(self.audiopaths_sid_text))
audiopaths_sid_text_new = []
lengths = []
skipped_phone = 0
skipped_dur = 0
for audiopath in tqdm(self.audiopaths_sid_text):
try:
phoneme = self.phoneme_data[audiopath][0]
phoneme = phoneme.split(' ')
phoneme_ids = cleaned_text_to_sequence(phoneme, version)
except Exception:
print(f"{audiopath} not in self.phoneme_data !")
skipped_phone += 1
continue
size = os.path.getsize("%s/%s" % (self.path5, audiopath))
duration = size / self.sampling_rate / 2
if duration == 0:
print(f"Zero duration for {audiopath}, skipping...")
skipped_dur += 1
continue
if 54 > duration > 0.6 or self.val:
audiopaths_sid_text_new.append([audiopath, phoneme_ids])
lengths.append(size // (2 * self.hop_length))
else:
skipped_dur += 1
continue
print("skipped_phone: ", skipped_phone, ", skipped_dur: ", skipped_dur)
print("total left: ", len(audiopaths_sid_text_new))
assert len(audiopaths_sid_text_new) > 1 # 至少能凑够batch size这里todo
self.audiopaths_sid_text = audiopaths_sid_text_new
self.lengths = lengths
self.spec_min=-12
self.spec_max=2
self.filter_length_mel=self.win_length_mel=1024
self.hop_length_mel=256
self.n_mel_channels=100
self.sampling_rate_mel=24000
self.mel_fmin=0
self.mel_fmax=None
def norm_spec(self, x):
return (x - self.spec_min) / (self.spec_max - self.spec_min) * 2 - 1
def get_audio_text_speaker_pair(self, audiopath_sid_text):
audiopath, phoneme_ids = audiopath_sid_text
text = torch.FloatTensor(phoneme_ids)
try:
spec, mel,wav = self.get_audio("%s/%s" % (self.path5, audiopath))
with torch.no_grad():
ssl = torch.load("%s/%s.pt" % (self.path4, audiopath), map_location="cpu")
if (ssl.shape[-1] != spec.shape[-1]):
typee = ssl.dtype
ssl = F.pad(ssl.float(), (0, 1), mode="replicate").to(typee)
ssl.requires_grad = False
except:
traceback.print_exc()
mel = torch.zeros(100, 180)
wav = torch.zeros(1, 96 * self.hop_length)
spec = torch.zeros(1025, 96)
ssl = torch.zeros(1, 768, 96)
text = text[-1:]
print("load audio or ssl error!!!!!!", audiopath)
return (ssl, spec, wav, mel, text)
def get_audio(self, filename):
audio_array = load_audio(filename,self.sampling_rate)#load_audio的方法是已经归一化到-1~1之间的不用再/32768
audio=torch.FloatTensor(audio_array)#/32768
audio_norm = audio
audio_norm = audio_norm.unsqueeze(0)
audio_array24 = load_audio(filename,24000)#load_audio的方法是已经归一化到-1~1之间的不用再/32768######这里可以用GPU重采样加速
audio24=torch.FloatTensor(audio_array24)#/32768
audio_norm24 = audio24
audio_norm24 = audio_norm24.unsqueeze(0)
spec = spectrogram_torch(audio_norm, self.filter_length,
self.sampling_rate, self.hop_length, self.win_length,
center=False)
spec = torch.squeeze(spec, 0)
spec1 = spectrogram_torch(audio_norm24, self.filter_length_mel,self.sampling_rate_mel, self.hop_length_mel, self.win_length_mel,center=False)
mel = spec_to_mel_torch(spec1, self.filter_length_mel, self.n_mel_channels, self.sampling_rate_mel, self.mel_fmin, self.mel_fmax)
mel = torch.squeeze(mel, 0)
mel=self.norm_spec(mel)
# print(1111111,spec.shape,mel.shape)
return spec, mel,audio_norm
def get_sid(self, sid):
sid = torch.LongTensor([int(sid)])
return sid
def __getitem__(self, index):
# with torch.no_grad():
return self.get_audio_text_speaker_pair(self.audiopaths_sid_text[index])
def __len__(self):
return len(self.audiopaths_sid_text)
class TextAudioSpeakerCollateV3b():
""" Zero-pads model inputs and targets
"""
def __init__(self, return_ids=False):
self.return_ids = return_ids
def __call__(self, batch):
"""Collate's training batch from normalized text, audio and speaker identities
PARAMS
------
batch: [text_normalized, spec_normalized, wav_normalized, sid]
"""
#ssl, spec, wav,mel, text
# Right zero-pad all one-hot text sequences to max input length
_, ids_sorted_decreasing = torch.sort(
torch.LongTensor([x[1].size(1) for x in batch]),
dim=0, descending=True)
#(ssl, spec,mel, text)
max_ssl_len = max([x[0].size(2) for x in batch])
max_ssl_len1 = int(8 * ((max_ssl_len // 8) + 1))
max_ssl_len = int(2 * ((max_ssl_len // 2) + 1))
# max_ssl_len = int(8 * ((max_ssl_len // 8) + 1))
# max_ssl_len1=max_ssl_len
max_spec_len = max([x[1].size(1) for x in batch])
max_spec_len = int(2 * ((max_spec_len // 2) + 1))
max_wav_len = max([x[2].size(1) for x in batch])
max_text_len = max([x[4].size(0) for x in batch])
max_mel_len=int(max_ssl_len1*1.25*1.5)###24000/256,32000/640=16000/320
ssl_lengths = torch.LongTensor(len(batch))
spec_lengths = torch.LongTensor(len(batch))
text_lengths = torch.LongTensor(len(batch))
wav_lengths = torch.LongTensor(len(batch))
mel_lengths = torch.LongTensor(len(batch))
spec_padded = torch.FloatTensor(len(batch), batch[0][1].size(0), max_spec_len)
mel_padded = torch.FloatTensor(len(batch), batch[0][3].size(0), max_mel_len)
ssl_padded = torch.FloatTensor(len(batch), batch[0][0].size(1), max_ssl_len)
text_padded = torch.LongTensor(len(batch), max_text_len)
wav_padded = torch.FloatTensor(len(batch), 1, max_wav_len)
spec_padded.zero_()
mel_padded.zero_()
ssl_padded.zero_()
text_padded.zero_()
wav_padded.zero_()
for i in range(len(ids_sorted_decreasing)):
row = batch[ids_sorted_decreasing[i]]
# ssl, spec, wav,mel, text
ssl = row[0]
ssl_padded[i, :, :ssl.size(2)] = ssl[0, :, :]
ssl_lengths[i] = ssl.size(2)
spec = row[1]
spec_padded[i, :, :spec.size(1)] = spec
spec_lengths[i] = spec.size(1)
wav = row[2]
wav_padded[i, :, :wav.size(1)] = wav
wav_lengths[i] = wav.size(1)
mel = row[3]
mel_padded[i, :, :mel.size(1)] = mel
mel_lengths[i] = mel.size(1)
text = row[4]
text_padded[i, :text.size(0)] = text
text_lengths[i] = text.size(0)
return ssl_padded, spec_padded,mel_padded, ssl_lengths, spec_lengths, text_padded, text_lengths, wav_padded, wav_lengths,mel_lengths
# return ssl_padded, spec_padded,mel_padded, ssl_lengths, spec_lengths, text_padded, text_lengths,mel_lengths
class DistributedBucketSampler(torch.utils.data.distributed.DistributedSampler):
"""

View File

@ -145,7 +145,7 @@ def mel_spectrogram_torch(
return_complex=False,
)
spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-6)
spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-9)
spec = torch.matmul(mel_basis[fmax_dtype_device], spec)
spec = spectral_normalize_torch(spec)

View File

@ -1,5 +1,10 @@
import warnings
warnings.filterwarnings("ignore")
import copy
import math
import os
import pdb
import torch
from torch import nn
from torch.nn import functional as F
@ -7,14 +12,17 @@ from torch.nn import functional as F
from module import commons
from module import modules
from module import attentions
from f5_tts.model import DiT
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
from module.commons import init_weights, get_padding
from module.mrte_model import MRTE
from module.quantize import ResidualVectorQuantizer
from text import symbols
# from text import symbols
from text import symbols as symbols_v1
from text import symbols2 as symbols_v2
from torch.cuda.amp import autocast
import contextlib,random
class StochasticDurationPredictor(nn.Module):
@ -182,6 +190,7 @@ class TextEncoder(nn.Module):
kernel_size,
p_dropout,
latent_channels=192,
version = "v2",
):
super().__init__()
self.out_channels = out_channels
@ -192,6 +201,7 @@ class TextEncoder(nn.Module):
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.latent_channels = latent_channels
self.version = version
self.ssl_proj = nn.Conv1d(768, hidden_channels, 1)
@ -207,6 +217,11 @@ class TextEncoder(nn.Module):
self.encoder_text = attentions.Encoder(
hidden_channels, filter_channels, n_heads, n_layers, kernel_size, p_dropout
)
if self.version == "v1":
symbols = symbols_v1.symbols
else:
symbols = symbols_v2.symbols
self.text_embedding = nn.Embedding(len(symbols), hidden_channels)
self.mrte = MRTE()
@ -222,13 +237,13 @@ class TextEncoder(nn.Module):
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, y, y_lengths, text, text_lengths, ge, test=None):
def forward(self, y, y_lengths, text, text_lengths, ge, speed=1,test=None):
y_mask = torch.unsqueeze(commons.sequence_mask(y_lengths, y.size(2)), 1).to(
y.dtype
)
y = self.ssl_proj(y * y_mask) * y_mask
y = self.encoder_ssl(y * y_mask, y_mask)
text_mask = torch.unsqueeze(
@ -239,9 +254,10 @@ class TextEncoder(nn.Module):
text = self.text_embedding(text).transpose(1, 2)
text = self.encoder_text(text * text_mask, text_mask)
y = self.mrte(y, y_mask, text, text_mask, ge)
y = self.encoder2(y * y_mask, y_mask)
if(speed!=1):
y = F.interpolate(y, size=int(y.shape[-1] / speed)+1, mode="linear")
y_mask = F.interpolate(y_mask, size=y.shape[-1], mode="nearest")
stats = self.proj(y) * y_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
return y, m, logs, y_mask
@ -355,6 +371,37 @@ class PosteriorEncoder(nn.Module):
return z, m, logs, x_mask
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, 1)
def forward(self, x, x_lengths, g=None):
if(g!=None):
g = g.detach()
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
return stats, x_mask
class WNEncoder(nn.Module):
def __init__(
self,
@ -823,6 +870,7 @@ class SynthesizerTrn(nn.Module):
use_sdp=True,
semantic_frame_rate=None,
freeze_quantizer=None,
version = "v2",
**kwargs
):
super().__init__()
@ -843,6 +891,7 @@ class SynthesizerTrn(nn.Module):
self.segment_size = segment_size
self.n_speakers = n_speakers
self.gin_channels = gin_channels
self.version = version
self.use_sdp = use_sdp
self.enc_p = TextEncoder(
@ -853,6 +902,7 @@ class SynthesizerTrn(nn.Module):
n_layers,
kernel_size,
p_dropout,
version = version,
)
self.dec = Generator(
inter_channels,
@ -877,9 +927,11 @@ class SynthesizerTrn(nn.Module):
inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels
)
self.ref_enc = modules.MelStyleEncoder(
spec_channels, style_vector_dim=gin_channels
)
# self.version=os.environ.get("version","v1")
if(self.version=="v1"):
self.ref_enc = modules.MelStyleEncoder(spec_channels, style_vector_dim=gin_channels)
else:
self.ref_enc = modules.MelStyleEncoder(704, style_vector_dim=gin_channels)
ssl_dim = 768
assert semantic_frame_rate in ["25hz", "50hz"]
@ -890,21 +942,22 @@ class SynthesizerTrn(nn.Module):
self.ssl_proj = nn.Conv1d(ssl_dim, ssl_dim, 1, stride=1)
self.quantizer = ResidualVectorQuantizer(dimension=ssl_dim, n_q=1, bins=1024)
if freeze_quantizer:
self.ssl_proj.requires_grad_(False)
self.quantizer.requires_grad_(False)
#self.quantizer.eval()
# self.enc_p.text_embedding.requires_grad_(False)
# self.enc_p.encoder_text.requires_grad_(False)
# self.enc_p.mrte.requires_grad_(False)
self.freeze_quantizer = freeze_quantizer
def forward(self, ssl, y, y_lengths, text, text_lengths):
y_mask = torch.unsqueeze(commons.sequence_mask(y_lengths, y.size(2)), 1).to(
y.dtype
)
ge = self.ref_enc(y * y_mask, y_mask)
if(self.version=="v1"):
ge = self.ref_enc(y * y_mask, y_mask)
else:
ge = self.ref_enc(y[:,:704] * y_mask, y_mask)
with autocast(enabled=False):
maybe_no_grad = torch.no_grad() if self.freeze_quantizer else contextlib.nullcontext()
with maybe_no_grad:
if self.freeze_quantizer:
self.ssl_proj.eval()
self.quantizer.eval()
ssl = self.ssl_proj(ssl)
quantized, codes, commit_loss, quantized_list = self.quantizer(
ssl, layers=[0]
@ -939,7 +992,10 @@ class SynthesizerTrn(nn.Module):
y_mask = torch.unsqueeze(commons.sequence_mask(y_lengths, y.size(2)), 1).to(
y.dtype
)
ge = self.ref_enc(y * y_mask, y_mask)
if(self.version=="v1"):
ge = self.ref_enc(y * y_mask, y_mask)
else:
ge = self.ref_enc(y[:,:704] * y_mask, y_mask)
ssl = self.ssl_proj(ssl)
quantized, codes, commit_loss, _ = self.quantizer(ssl, layers=[0])
@ -959,14 +1015,27 @@ class SynthesizerTrn(nn.Module):
return o, y_mask, (z, z_p, m_p, logs_p)
@torch.no_grad()
def decode(self, codes, text, refer, noise_scale=0.5):
ge = None
if refer is not None:
refer_lengths = torch.LongTensor([refer.size(2)]).to(refer.device)
refer_mask = torch.unsqueeze(
commons.sequence_mask(refer_lengths, refer.size(2)), 1
).to(refer.dtype)
ge = self.ref_enc(refer * refer_mask, refer_mask)
def decode(self, codes, text, refer, noise_scale=0.5,speed=1):
def get_ge(refer):
ge = None
if refer is not None:
refer_lengths = torch.LongTensor([refer.size(2)]).to(refer.device)
refer_mask = torch.unsqueeze(
commons.sequence_mask(refer_lengths, refer.size(2)), 1
).to(refer.dtype)
if (self.version == "v1"):
ge = self.ref_enc(refer * refer_mask, refer_mask)
else:
ge = self.ref_enc(refer[:, :704] * refer_mask, refer_mask)
return ge
if(type(refer)==list):
ges=[]
for _refer in refer:
ge=get_ge(_refer)
ges.append(ge)
ge=torch.stack(ges,0).mean(0)
else:
ge=get_ge(refer)
y_lengths = torch.LongTensor([codes.size(2) * 2]).to(codes.device)
text_lengths = torch.LongTensor([text.size(-1)]).to(text.device)
@ -976,9 +1045,8 @@ class SynthesizerTrn(nn.Module):
quantized = F.interpolate(
quantized, size=int(quantized.shape[-1] * 2), mode="nearest"
)
x, m_p, logs_p, y_mask = self.enc_p(
quantized, y_lengths, text, text_lengths, ge
quantized, y_lengths, text, text_lengths, ge,speed
)
z_p = m_p + torch.randn_like(m_p) * torch.exp(logs_p) * noise_scale
@ -991,3 +1059,371 @@ class SynthesizerTrn(nn.Module):
ssl = self.ssl_proj(x)
quantized, codes, commit_loss, quantized_list = self.quantizer(ssl)
return codes.transpose(0, 1)
class CFM(torch.nn.Module):
def __init__(
self,
in_channels,dit
):
super().__init__()
self.sigma_min = 1e-6
self.estimator = dit
self.in_channels = in_channels
self.criterion = torch.nn.MSELoss()
@torch.inference_mode()
def inference(self, mu, x_lens, prompt, n_timesteps, temperature=1.0, inference_cfg_rate=0):
"""Forward diffusion"""
B, T = mu.size(0), mu.size(1)
x = torch.randn([B, self.in_channels, T], device=mu.device,dtype=mu.dtype) * temperature
prompt_len = prompt.size(-1)
prompt_x = torch.zeros_like(x,dtype=mu.dtype)
prompt_x[..., :prompt_len] = prompt[..., :prompt_len]
x[..., :prompt_len] = 0
mu=mu.transpose(2,1)
t = 0
d = 1 / n_timesteps
for j in range(n_timesteps):
t_tensor = torch.ones(x.shape[0], device=x.device,dtype=mu.dtype) * t
d_tensor = torch.ones(x.shape[0], device=x.device,dtype=mu.dtype) * d
# v_pred = model(x, t_tensor, d_tensor, **extra_args)
v_pred = self.estimator(x, prompt_x, x_lens, t_tensor,d_tensor, mu, use_grad_ckpt=False,drop_audio_cond=False,drop_text=False).transpose(2, 1)
if inference_cfg_rate>1e-5:
neg = self.estimator(x, prompt_x, x_lens, t_tensor, d_tensor, mu, use_grad_ckpt=False, drop_audio_cond=True, drop_text=True).transpose(2, 1)
v_pred=v_pred+(v_pred-neg)*inference_cfg_rate
x = x + d * v_pred
t = t + d
x[:, :, :prompt_len] = 0
return x
def forward(self, x1, x_lens, prompt_lens, mu, use_grad_ckpt):
b, _, t = x1.shape
t = torch.rand([b], device=mu.device, dtype=x1.dtype)
x0 = torch.randn_like(x1,device=mu.device)
vt = x1 - x0
xt = x0 + t[:, None, None] * vt
dt = torch.zeros_like(t,device=mu.device)
prompt = torch.zeros_like(x1)
for i in range(b):
prompt[i, :, :prompt_lens[i]] = x1[i, :, :prompt_lens[i]]
xt[i, :, :prompt_lens[i]] = 0
gailv=0.3# if ttime()>1736250488 else 0.1
if random.random() < gailv:
base = torch.randint(2, 8, (t.shape[0],), device=mu.device)
d = 1/torch.pow(2, base)
d_input = d.clone()
d_input[d_input < 1e-2] = 0
# with torch.no_grad():
v_pred_1 = self.estimator(xt, prompt, x_lens, t, d_input, mu, use_grad_ckpt).transpose(2, 1).detach()
# v_pred_1 = self.diffusion(xt, t, d_input, cond=conditioning).detach()
x_mid = xt + d[:, None, None] * v_pred_1
# v_pred_2 = self.diffusion(x_mid, t+d, d_input, cond=conditioning).detach()
v_pred_2 = self.estimator(x_mid, prompt, x_lens, t+d, d_input, mu, use_grad_ckpt).transpose(2, 1).detach()
vt = (v_pred_1 + v_pred_2) / 2
vt = vt.detach()
dt = 2*d
vt_pred = self.estimator(xt, prompt, x_lens, t,dt, mu, use_grad_ckpt).transpose(2,1)
loss = 0
for i in range(b):
loss += self.criterion(vt_pred[i, :, prompt_lens[i]:x_lens[i]], vt[i, :, prompt_lens[i]:x_lens[i]])
loss /= b
return loss
def set_no_grad(net_g):
for name, param in net_g.named_parameters():
param.requires_grad=False
class SynthesizerTrnV3(nn.Module):
"""
Synthesizer for Training
"""
def __init__(self,
spec_channels,
segment_size,
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
n_speakers=0,
gin_channels=0,
use_sdp=True,
semantic_frame_rate=None,
freeze_quantizer=None,
version="v3",
**kwargs):
super().__init__()
self.spec_channels = spec_channels
self.inter_channels = inter_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.resblock = resblock
self.resblock_kernel_sizes = resblock_kernel_sizes
self.resblock_dilation_sizes = resblock_dilation_sizes
self.upsample_rates = upsample_rates
self.upsample_initial_channel = upsample_initial_channel
self.upsample_kernel_sizes = upsample_kernel_sizes
self.segment_size = segment_size
self.n_speakers = n_speakers
self.gin_channels = gin_channels
self.version = version
self.model_dim=512
self.use_sdp = use_sdp
self.enc_p = TextEncoder(inter_channels,hidden_channels,filter_channels,n_heads,n_layers,kernel_size,p_dropout)
# self.ref_enc = modules.MelStyleEncoder(spec_channels, style_vector_dim=gin_channels)###Rollback
self.ref_enc = modules.MelStyleEncoder(704, style_vector_dim=gin_channels)###Rollback
# self.dec = Generator(inter_channels, resblock, resblock_kernel_sizes, resblock_dilation_sizes, upsample_rates,
# upsample_initial_channel, upsample_kernel_sizes, gin_channels=gin_channels)
# self.enc_q = PosteriorEncoder(spec_channels, inter_channels, hidden_channels, 5, 1, 16,
# gin_channels=gin_channels)
# self.flow = ResidualCouplingBlock(inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels)
ssl_dim = 768
assert semantic_frame_rate in ['25hz', "50hz"]
self.semantic_frame_rate = semantic_frame_rate
if semantic_frame_rate == '25hz':
self.ssl_proj = nn.Conv1d(ssl_dim, ssl_dim, 2, stride=2)
else:
self.ssl_proj = nn.Conv1d(ssl_dim, ssl_dim, 1, stride=1)
self.quantizer = ResidualVectorQuantizer(
dimension=ssl_dim,
n_q=1,
bins=1024
)
self.freeze_quantizer=freeze_quantizer
inter_channels2=512
self.bridge=nn.Sequential(
nn.Conv1d(inter_channels, inter_channels2, 1, stride=1),
nn.LeakyReLU()
)
self.wns1=Encoder(inter_channels2, inter_channels2, inter_channels2, 5, 1, 8,gin_channels=gin_channels)
self.linear_mel=nn.Conv1d(inter_channels2,100,1,stride=1)
self.cfm = CFM(100,DiT(**dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=inter_channels2, conv_layers=4)),)#text_dim is condition feature dim
if self.freeze_quantizer==True:
set_no_grad(self.ssl_proj)
set_no_grad(self.quantizer)
set_no_grad(self.enc_p)
def forward(self, ssl, y, mel,ssl_lengths,y_lengths, text, text_lengths,mel_lengths, use_grad_ckpt):#ssl_lengths no need now
with autocast(enabled=False):
y_mask = torch.unsqueeze(commons.sequence_mask(y_lengths, y.size(2)), 1).to(y.dtype)
ge = self.ref_enc(y[:,:704] * y_mask, y_mask)
maybe_no_grad = torch.no_grad() if self.freeze_quantizer else contextlib.nullcontext()
with maybe_no_grad:
if self.freeze_quantizer:
self.ssl_proj.eval()#
self.quantizer.eval()
self.enc_p.eval()
ssl = self.ssl_proj(ssl)
quantized, codes, commit_loss, quantized_list = self.quantizer(
ssl, layers=[0]
)
quantized = F.interpolate(quantized, scale_factor=2, mode="nearest")##BCT
x, m_p, logs_p, y_mask = self.enc_p(quantized, y_lengths, text, text_lengths, ge)
fea=self.bridge(x)
fea = F.interpolate(fea, scale_factor=1.875, mode="nearest")##BCT
fea, y_mask_ = self.wns1(fea, mel_lengths, ge)##If the 1-minute fine-tuning works fine, no need to manually adjust the learning rate.
B=ssl.shape[0]
prompt_len_max = mel_lengths*2/3
prompt_len = (torch.rand([B], device=fea.device) * prompt_len_max).floor().to(dtype=torch.long)
minn=min(mel.shape[-1],fea.shape[-1])
mel=mel[:,:,:minn]
fea=fea[:,:,:minn]
cfm_loss= self.cfm(mel, mel_lengths, prompt_len, fea, use_grad_ckpt)
return cfm_loss
@torch.no_grad()
def decode_encp(self, codes,text, refer,ge=None,speed=1):
# print(2333333,refer.shape)
# ge=None
if(ge==None):
refer_lengths = torch.LongTensor([refer.size(2)]).to(refer.device)
refer_mask = torch.unsqueeze(commons.sequence_mask(refer_lengths, refer.size(2)), 1).to(refer.dtype)
ge = self.ref_enc(refer[:,:704] * refer_mask, refer_mask)
y_lengths = torch.LongTensor([int(codes.size(2)*2)]).to(codes.device)
if speed==1:
sizee=int(codes.size(2)*2.5*1.5)
else:
sizee=int(codes.size(2)*2.5*1.5/speed)+1
y_lengths1 = torch.LongTensor([sizee]).to(codes.device)
text_lengths = torch.LongTensor([text.size(-1)]).to(text.device)
quantized = self.quantizer.decode(codes)
if self.semantic_frame_rate == '25hz':
quantized = F.interpolate(quantized, scale_factor=2, mode="nearest")##BCT
x, m_p, logs_p, y_mask = self.enc_p(quantized, y_lengths, text, text_lengths, ge,speed)
fea=self.bridge(x)
fea = F.interpolate(fea, scale_factor=1.875, mode="nearest")##BCT
####more wn paramter to learn mel
fea, y_mask_ = self.wns1(fea, y_lengths1, ge)
return fea,ge
def extract_latent(self, x):
ssl = self.ssl_proj(x)
quantized, codes, commit_loss, quantized_list = self.quantizer(ssl)
return codes.transpose(0,1)
class SynthesizerTrnV3b(nn.Module):
"""
Synthesizer for Training
"""
def __init__(self,
spec_channels,
segment_size,
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
n_speakers=0,
gin_channels=0,
use_sdp=True,
semantic_frame_rate=None,
freeze_quantizer=None,
**kwargs):
super().__init__()
self.spec_channels = spec_channels
self.inter_channels = inter_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.resblock = resblock
self.resblock_kernel_sizes = resblock_kernel_sizes
self.resblock_dilation_sizes = resblock_dilation_sizes
self.upsample_rates = upsample_rates
self.upsample_initial_channel = upsample_initial_channel
self.upsample_kernel_sizes = upsample_kernel_sizes
self.segment_size = segment_size
self.n_speakers = n_speakers
self.gin_channels = gin_channels
self.model_dim=512
self.use_sdp = use_sdp
self.enc_p = TextEncoder(inter_channels,hidden_channels,filter_channels,n_heads,n_layers,kernel_size,p_dropout)
# self.ref_enc = modules.MelStyleEncoder(spec_channels, style_vector_dim=gin_channels)###Rollback
self.ref_enc = modules.MelStyleEncoder(704, style_vector_dim=gin_channels)###Rollback
self.dec = Generator(inter_channels, resblock, resblock_kernel_sizes, resblock_dilation_sizes, upsample_rates,
upsample_initial_channel, upsample_kernel_sizes, gin_channels=gin_channels)
self.enc_q = PosteriorEncoder(spec_channels, inter_channels, hidden_channels, 5, 1, 16,
gin_channels=gin_channels)
self.flow = ResidualCouplingBlock(inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels)
ssl_dim = 768
assert semantic_frame_rate in ['25hz', "50hz"]
self.semantic_frame_rate = semantic_frame_rate
if semantic_frame_rate == '25hz':
self.ssl_proj = nn.Conv1d(ssl_dim, ssl_dim, 2, stride=2)
else:
self.ssl_proj = nn.Conv1d(ssl_dim, ssl_dim, 1, stride=1)
self.quantizer = ResidualVectorQuantizer(
dimension=ssl_dim,
n_q=1,
bins=1024
)
self.freeze_quantizer=freeze_quantizer
inter_channels2=512
self.bridge=nn.Sequential(
nn.Conv1d(inter_channels, inter_channels2, 1, stride=1),
nn.LeakyReLU()
)
self.wns1=Encoder(inter_channels2, inter_channels2, inter_channels2, 5, 1, 8,gin_channels=gin_channels)
self.linear_mel=nn.Conv1d(inter_channels2,100,1,stride=1)
self.cfm = CFM(100,DiT(**dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=inter_channels2, conv_layers=4)),)#text_dim is condition feature dim
def forward(self, ssl, y, mel,ssl_lengths,y_lengths, text, text_lengths,mel_lengths):#ssl_lengths no need now
with autocast(enabled=False):
y_mask = torch.unsqueeze(commons.sequence_mask(y_lengths, y.size(2)), 1).to(y.dtype)
ge = self.ref_enc(y[:,:704] * y_mask, y_mask)
# ge = self.ref_enc(y * y_mask, y_mask)#change back, new spec setting is whole 24k
# ge=None
maybe_no_grad = torch.no_grad() if self.freeze_quantizer else contextlib.nullcontext()
with maybe_no_grad:
if self.freeze_quantizer:
self.ssl_proj.eval()
self.quantizer.eval()
ssl = self.ssl_proj(ssl)
quantized, codes, commit_loss, quantized_list = self.quantizer(
ssl, layers=[0]
)
quantized = F.interpolate(quantized, scale_factor=2, mode="nearest")##BCT
x, m_p, logs_p, y_mask = self.enc_p(quantized, y_lengths, text, text_lengths, ge)
z, m_q, logs_q, y_mask = self.enc_q(y, y_lengths, g=ge)
z_p = self.flow(z, y_mask, g=ge)
z_slice, ids_slice = commons.rand_slice_segments(z, y_lengths, self.segment_size)
o = self.dec(z_slice, g=ge)
fea=self.bridge(x)
fea = F.interpolate(fea, scale_factor=1.875, mode="nearest")##BCT
fea, y_mask_ = self.wns1(fea, mel_lengths, ge)
learned_mel = self.linear_mel(fea)
B=ssl.shape[0]
prompt_len_max = mel_lengths*2/3
prompt_len = (torch.rand([B], device=fea.device) * prompt_len_max).floor().to(dtype=torch.long)#
minn=min(mel.shape[-1],fea.shape[-1])
mel=mel[:,:,:minn]
fea=fea[:,:,:minn]
cfm_loss= self.cfm(mel, mel_lengths, prompt_len, fea)#fea==cond,y_lengths==target_mel_lengths#ge not need
return commit_loss,cfm_loss,F.mse_loss(learned_mel, mel),o, ids_slice, y_mask, y_mask, (z, z_p, m_p, logs_p, m_q, logs_q), quantized
@torch.no_grad()
def decode_encp(self, codes,text, refer,ge=None):
# print(2333333,refer.shape)
# ge=None
if(ge==None):
refer_lengths = torch.LongTensor([refer.size(2)]).to(refer.device)
refer_mask = torch.unsqueeze(commons.sequence_mask(refer_lengths, refer.size(2)), 1).to(refer.dtype)
ge = self.ref_enc(refer[:,:704] * refer_mask, refer_mask)
y_lengths = torch.LongTensor([int(codes.size(2)*2)]).to(codes.device)
y_lengths1 = torch.LongTensor([int(codes.size(2)*2.5*1.5)]).to(codes.device)
text_lengths = torch.LongTensor([text.size(-1)]).to(text.device)
quantized = self.quantizer.decode(codes)
if self.semantic_frame_rate == '25hz':
quantized = F.interpolate(quantized, scale_factor=2, mode="nearest")##BCT
x, m_p, logs_p, y_mask = self.enc_p(quantized, y_lengths, text, text_lengths, ge)
fea=self.bridge(x)
fea = F.interpolate(fea, scale_factor=1.875, mode="nearest")##BCT
####more wn paramter to learn mel
fea, y_mask_ = self.wns1(fea, y_lengths1, ge)
return fea,ge
def extract_latent(self, x):
ssl = self.ssl_proj(x)
quantized, codes, commit_loss, quantized_list = self.quantizer(ssl)
return codes.transpose(0,1)

View File

@ -1,5 +1,6 @@
import copy
import math
from typing import Optional
import torch
from torch import nn
from torch.nn import functional as F
@ -8,12 +9,15 @@ from module import commons
from module import modules
from module import attentions_onnx as attentions
from f5_tts.model import DiT
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
from module.commons import init_weights, get_padding
from module.mrte_model import MRTE
from module.quantize import ResidualVectorQuantizer
from text import symbols
# from text import symbols
from text import symbols as symbols_v1
from text import symbols2 as symbols_v2
from torch.cuda.amp import autocast
@ -182,6 +186,7 @@ class TextEncoder(nn.Module):
kernel_size,
p_dropout,
latent_channels=192,
version="v2",
):
super().__init__()
self.out_channels = out_channels
@ -192,6 +197,7 @@ class TextEncoder(nn.Module):
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.latent_channels = latent_channels
self.version = version
self.ssl_proj = nn.Conv1d(768, hidden_channels, 1)
@ -207,9 +213,14 @@ class TextEncoder(nn.Module):
self.encoder_text = attentions.Encoder(
hidden_channels, filter_channels, n_heads, n_layers, kernel_size, p_dropout
)
if self.version == "v1":
symbols = symbols_v1.symbols
else:
symbols = symbols_v2.symbols
self.text_embedding = nn.Embedding(len(symbols), hidden_channels)
self.mrte = MRTE()
self.mrte = attentions.MRTE()
self.encoder2 = attentions.Encoder(
hidden_channels,
@ -222,7 +233,7 @@ class TextEncoder(nn.Module):
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, y, text, ge):
def forward(self, y, text, ge, speed=1):
y_mask = torch.ones_like(y[:1,:1,:])
y = self.ssl_proj(y * y_mask) * y_mask
@ -235,30 +246,14 @@ class TextEncoder(nn.Module):
y = self.mrte(y, y_mask, text, text_mask, ge)
y = self.encoder2(y * y_mask, y_mask)
if(speed!=1):
y = F.interpolate(y, size=int(y.shape[-1] / speed)+1, mode="linear")
y_mask = F.interpolate(y_mask, size=y.shape[-1], mode="nearest")
stats = self.proj(y) * y_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
return y, m, logs, y_mask
def extract_latent(self, x):
x = self.ssl_proj(x)
quantized, codes, commit_loss, quantized_list = self.quantizer(x)
return codes.transpose(0, 1)
def decode_latent(self, codes, y_mask, refer, refer_mask, ge):
quantized = self.quantizer.decode(codes)
y = self.vq_proj(quantized) * y_mask
y = self.encoder_ssl(y * y_mask, y_mask)
y = self.mrte(y, y_mask, refer, refer_mask, ge)
y = self.encoder2(y * y_mask, y_mask)
stats = self.proj(y) * y_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
return y, m, logs, y_mask, quantized
class ResidualCouplingBlock(nn.Module):
def __init__(
@ -349,6 +344,37 @@ class PosteriorEncoder(nn.Module):
return z, m, logs, x_mask
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, 1)
def forward(self, x, x_lengths, g=None):
if(g!=None):
g = g.detach()
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
return stats, x_mask
class WNEncoder(nn.Module):
def __init__(
self,
@ -439,7 +465,7 @@ class Generator(torch.nn.Module):
if gin_channels != 0:
self.cond = nn.Conv1d(gin_channels, upsample_initial_channel, 1)
def forward(self, x, g=None):
def forward(self, x, g:Optional[torch.Tensor]=None):
x = self.conv_pre(x)
if g is not None:
x = x + self.cond(g)
@ -817,6 +843,7 @@ class SynthesizerTrn(nn.Module):
use_sdp=True,
semantic_frame_rate=None,
freeze_quantizer=None,
version="v2",
**kwargs
):
super().__init__()
@ -837,6 +864,7 @@ class SynthesizerTrn(nn.Module):
self.segment_size = segment_size
self.n_speakers = n_speakers
self.gin_channels = gin_channels
self.version = version
self.use_sdp = use_sdp
self.enc_p = TextEncoder(
@ -847,6 +875,7 @@ class SynthesizerTrn(nn.Module):
n_layers,
kernel_size,
p_dropout,
version=version,
)
self.dec = Generator(
inter_channels,
@ -858,22 +887,24 @@ class SynthesizerTrn(nn.Module):
upsample_kernel_sizes,
gin_channels=gin_channels,
)
self.enc_q = PosteriorEncoder(
spec_channels,
inter_channels,
hidden_channels,
5,
1,
16,
gin_channels=gin_channels,
)
# self.enc_q = PosteriorEncoder(
# spec_channels,
# inter_channels,
# hidden_channels,
# 5,
# 1,
# 16,
# gin_channels=gin_channels,
# )
self.flow = ResidualCouplingBlock(
inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels
)
self.ref_enc = modules.MelStyleEncoder(
spec_channels, style_vector_dim=gin_channels
)
# self.version=os.environ.get("version","v1")
if self.version == "v1":
self.ref_enc = modules.MelStyleEncoder(spec_channels, style_vector_dim=gin_channels)
else:
self.ref_enc = modules.MelStyleEncoder(704, style_vector_dim=gin_channels)
ssl_dim = 768
self.ssl_dim = ssl_dim
@ -892,9 +923,12 @@ class SynthesizerTrn(nn.Module):
# self.enc_p.encoder_text.requires_grad_(False)
# self.enc_p.mrte.requires_grad_(False)
def forward(self, codes, text, refer):
def forward(self, codes, text, refer,noise_scale=0.5, speed=1):
refer_mask = torch.ones_like(refer[:1,:1,:])
ge = self.ref_enc(refer * refer_mask, refer_mask)
if (self.version == "v1"):
ge = self.ref_enc(refer * refer_mask, refer_mask)
else:
ge = self.ref_enc(refer[:, :704] * refer_mask, refer_mask)
quantized = self.quantizer.decode(codes)
if self.semantic_frame_rate == "25hz":
@ -902,10 +936,10 @@ class SynthesizerTrn(nn.Module):
quantized = dquantized.contiguous().view(1, self.ssl_dim, -1)
x, m_p, logs_p, y_mask = self.enc_p(
quantized, text, ge
quantized, text, ge, speed
)
z_p = m_p + torch.randn_like(m_p) * torch.exp(logs_p)
z_p = m_p + torch.randn_like(m_p) * torch.exp(logs_p) * noise_scale
z = self.flow(z_p, y_mask, g=ge, reverse=True)
@ -915,4 +949,175 @@ class SynthesizerTrn(nn.Module):
def extract_latent(self, x):
ssl = self.ssl_proj(x)
quantized, codes, commit_loss, quantized_list = self.quantizer(ssl)
return codes.transpose(0, 1)
return codes.transpose(0, 1)
class CFM(torch.nn.Module):
def __init__(
self,
in_channels,dit
):
super().__init__()
# self.sigma_min = 1e-6
self.estimator = dit
self.in_channels = in_channels
# self.criterion = torch.nn.MSELoss()
def forward(self, mu:torch.Tensor, x_lens:torch.LongTensor, prompt:torch.Tensor, n_timesteps:torch.LongTensor, temperature:float=1.0):
"""Forward diffusion"""
B, T = mu.size(0), mu.size(1)
x = torch.randn([B, self.in_channels, T], device=mu.device,dtype=mu.dtype)
ntimesteps = int(n_timesteps)
prompt_len = prompt.size(-1)
prompt_x = torch.zeros_like(x,dtype=mu.dtype)
prompt_x[..., :prompt_len] = prompt[..., :prompt_len]
x[..., :prompt_len] = 0.0
mu=mu.transpose(2,1)
t = torch.tensor(0.0,dtype=x.dtype,device=x.device)
d = torch.tensor(1.0/ntimesteps,dtype=x.dtype,device=x.device)
d_tensor = torch.ones(x.shape[0], device=x.device,dtype=mu.dtype) * d
for j in range(ntimesteps):
t_tensor = torch.ones(x.shape[0], device=x.device,dtype=mu.dtype) * t
# d_tensor = torch.ones(x.shape[0], device=x.device,dtype=mu.dtype) * d
# v_pred = model(x, t_tensor, d_tensor, **extra_args)
v_pred = self.estimator(x, prompt_x, x_lens, t_tensor,d_tensor, mu).transpose(2, 1)
# if inference_cfg_rate>1e-5:
# neg = self.estimator(x, prompt_x, x_lens, t_tensor, d_tensor, mu, use_grad_ckpt=False, drop_audio_cond=True, drop_text=True).transpose(2, 1)
# v_pred=v_pred+(v_pred-neg)*inference_cfg_rate
x = x + d * v_pred
t = t + d
x[:, :, :prompt_len] = 0.0
return x
def set_no_grad(net_g):
for name, param in net_g.named_parameters():
param.requires_grad=False
@torch.jit.script_if_tracing
def compile_codes_length(codes):
y_lengths1 = torch.LongTensor([codes.size(2)]).to(codes.device)
return y_lengths1 * 2.5 * 1.5
@torch.jit.script_if_tracing
def compile_ref_length(refer):
refer_lengths = torch.LongTensor([refer.size(2)]).to(refer.device)
return refer_lengths
class SynthesizerTrnV3(nn.Module):
"""
Synthesizer for Training
"""
def __init__(self,
spec_channels,
segment_size,
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
n_speakers=0,
gin_channels=0,
use_sdp=True,
semantic_frame_rate=None,
freeze_quantizer=None,
version="v3",
**kwargs):
super().__init__()
self.spec_channels = spec_channels
self.inter_channels = inter_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.resblock = resblock
self.resblock_kernel_sizes = resblock_kernel_sizes
self.resblock_dilation_sizes = resblock_dilation_sizes
self.upsample_rates = upsample_rates
self.upsample_initial_channel = upsample_initial_channel
self.upsample_kernel_sizes = upsample_kernel_sizes
self.segment_size = segment_size
self.n_speakers = n_speakers
self.gin_channels = gin_channels
self.version = version
self.model_dim=512
self.use_sdp = use_sdp
self.enc_p = TextEncoder(inter_channels,hidden_channels,filter_channels,n_heads,n_layers,kernel_size,p_dropout)
# self.ref_enc = modules.MelStyleEncoder(spec_channels, style_vector_dim=gin_channels)###Rollback
self.ref_enc = modules.MelStyleEncoder(704, style_vector_dim=gin_channels)###Rollback
# self.dec = Generator(inter_channels, resblock, resblock_kernel_sizes, resblock_dilation_sizes, upsample_rates,
# upsample_initial_channel, upsample_kernel_sizes, gin_channels=gin_channels)
# self.enc_q = PosteriorEncoder(spec_channels, inter_channels, hidden_channels, 5, 1, 16,
# gin_channels=gin_channels)
# self.flow = ResidualCouplingBlock(inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels)
ssl_dim = 768
assert semantic_frame_rate in ['25hz', "50hz"]
self.semantic_frame_rate = semantic_frame_rate
if semantic_frame_rate == '25hz':
self.ssl_proj = nn.Conv1d(ssl_dim, ssl_dim, 2, stride=2)
else:
self.ssl_proj = nn.Conv1d(ssl_dim, ssl_dim, 1, stride=1)
self.quantizer = ResidualVectorQuantizer(
dimension=ssl_dim,
n_q=1,
bins=1024
)
freeze_quantizer
inter_channels2=512
self.bridge=nn.Sequential(
nn.Conv1d(inter_channels, inter_channels2, 1, stride=1),
nn.LeakyReLU()
)
self.wns1=Encoder(inter_channels2, inter_channels2, inter_channels2, 5, 1, 8,gin_channels=gin_channels)
self.linear_mel=nn.Conv1d(inter_channels2,100,1,stride=1)
self.cfm = CFM(100,DiT(**dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=inter_channels2, conv_layers=4)),)#text_dim is condition feature dim
if freeze_quantizer==True:
set_no_grad(self.ssl_proj)
set_no_grad(self.quantizer)
set_no_grad(self.enc_p)
def create_ge(self, refer):
refer_lengths = compile_ref_length(refer)
refer_mask = torch.unsqueeze(commons.sequence_mask(refer_lengths, refer.size(2)), 1).to(refer.dtype)
ge = self.ref_enc(refer[:,:704] * refer_mask, refer_mask)
return ge
def forward(self, codes, text,ge,speed=1):
y_lengths1=compile_codes_length(codes)
quantized = self.quantizer.decode(codes)
if self.semantic_frame_rate == '25hz':
quantized = F.interpolate(quantized, scale_factor=2, mode="nearest")##BCT
x, m_p, logs_p, y_mask = self.enc_p(quantized, text, ge,speed)
fea=self.bridge(x)
fea = F.interpolate(fea, scale_factor=1.875, mode="nearest")##BCT
####more wn paramter to learn mel
fea, y_mask_ = self.wns1(fea, y_lengths1, ge)
return fea
def extract_latent(self, x):
ssl = self.ssl_proj(x)
quantized, codes, commit_loss, quantized_list = self.quantizer(ssl)
return codes.transpose(0,1)

View File

@ -1,21 +0,0 @@
import ffmpeg
import numpy as np
def load_audio(file, sr):
try:
# https://github.com/openai/whisper/blob/main/whisper/audio.py#L26
# This launches a subprocess to decode audio while down-mixing and resampling as necessary.
# Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.
file = (
file.strip(" ").strip('"').strip("\n").strip('"').strip(" ")
) # 防止小白拷路径头尾带了空格和"和回车
out, _ = (
ffmpeg.input(file, threads=0)
.output("-", format="f32le", acodec="pcm_f32le", ac=1, ar=sr)
.run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
)
except Exception as e:
raise RuntimeError(f"Failed to load audio: {e}")
return np.frombuffer(out, np.float32).flatten()

View File

@ -1,15 +1,16 @@
from module.models_onnx import SynthesizerTrn, symbols
from module.models_onnx import SynthesizerTrn, symbols_v1, symbols_v2
from AR.models.t2s_lightning_module_onnx import Text2SemanticLightningModule
import torch
import torchaudio
from torch import nn
from feature_extractor import cnhubert
cnhubert_base_path = "pretrained_models/chinese-hubert-base"
cnhubert.cnhubert_base_path=cnhubert_base_path
cnhubert_base_path = "GPT_SoVITS/pretrained_models/chinese-hubert-base"
cnhubert.cnhubert_base_path = cnhubert_base_path
ssl_model = cnhubert.get_model()
from text import cleaned_text_to_sequence
import soundfile
from my_utils import load_audio
from tools.my_utils import load_audio
import os
import json
@ -196,6 +197,11 @@ class VitsModel(nn.Module):
super().__init__()
dict_s2 = torch.load(vits_path,map_location="cpu")
self.hps = dict_s2["config"]
if dict_s2['weight']['enc_p.text_embedding.weight'].shape[0] == 322:
self.hps["model"]["version"] = "v1"
else:
self.hps["model"]["version"] = "v2"
self.hps = DictToAttrRecursive(self.hps)
self.hps.model.semantic_frame_rate = "25hz"
self.vq_model = SynthesizerTrn(
@ -267,13 +273,13 @@ class SSLModel(nn.Module):
return self.ssl.model(ref_audio_16k)["last_hidden_state"].transpose(1, 2)
def export(vits_path, gpt_path, project_name):
def export(vits_path, gpt_path, project_name, vits_model="v2"):
vits = VitsModel(vits_path)
gpt = T2SModel(gpt_path, vits)
gpt_sovits = GptSoVits(vits, gpt)
ssl = SSLModel()
ref_seq = torch.LongTensor([cleaned_text_to_sequence(["n", "i2", "h", "ao3", ",", "w", "o3", "sh", "i4", "b", "ai2", "y", "e4"])])
text_seq = torch.LongTensor([cleaned_text_to_sequence(["w", "o3", "sh", "i4", "b", "ai2", "y", "e4", "w", "o3", "sh", "i4", "b", "ai2", "y", "e4", "w", "o3", "sh", "i4", "b", "ai2", "y", "e4"])])
ref_seq = torch.LongTensor([cleaned_text_to_sequence(["n", "i2", "h", "ao3", ",", "w", "o3", "sh", "i4", "b", "ai2", "y", "e4"],version=vits_model)])
text_seq = torch.LongTensor([cleaned_text_to_sequence(["w", "o3", "sh", "i4", "b", "ai2", "y", "e4", "w", "o3", "sh", "i4", "b", "ai2", "y", "e4", "w", "o3", "sh", "i4", "b", "ai2", "y", "e4"],version=vits_model)])
ref_bert = torch.randn((ref_seq.shape[1], 1024)).float()
text_bert = torch.randn((text_seq.shape[1], 1024)).float()
ref_audio = torch.randn((1, 48000 * 5)).float()
@ -287,34 +293,38 @@ def export(vits_path, gpt_path, project_name):
pass
ssl_content = ssl(ref_audio_16k).float()
debug = False
# debug = False
debug = True
# gpt_sovits.export(ref_seq, text_seq, ref_bert, text_bert, ref_audio_sr, ssl_content, project_name)
if debug:
a, b = gpt_sovits(ref_seq, text_seq, ref_bert, text_bert, ref_audio_sr, ssl_content, debug=debug)
soundfile.write("out1.wav", a.cpu().detach().numpy(), vits.hps.data.sampling_rate)
soundfile.write("out2.wav", b[0], vits.hps.data.sampling_rate)
return
a = gpt_sovits(ref_seq, text_seq, ref_bert, text_bert, ref_audio_sr, ssl_content).detach().cpu().numpy()
else:
a = gpt_sovits(ref_seq, text_seq, ref_bert, text_bert, ref_audio_sr, ssl_content).detach().cpu().numpy()
soundfile.write("out.wav", a, vits.hps.data.sampling_rate)
soundfile.write("out.wav", a, vits.hps.data.sampling_rate)
gpt_sovits.export(ref_seq, text_seq, ref_bert, text_bert, ref_audio_sr, ssl_content, project_name)
if vits_model == "v1":
symbols = symbols_v1
else:
symbols = symbols_v2
MoeVSConf = {
"Folder" : f"{project_name}",
"Name" : f"{project_name}",
"Type" : "GPT-SoVits",
"Rate" : vits.hps.data.sampling_rate,
"NumLayers": gpt.t2s_model.num_layers,
"EmbeddingDim": gpt.t2s_model.embedding_dim,
"Dict": "BasicDict",
"BertPath": "chinese-roberta-wwm-ext-large",
"Symbol": symbols,
"AddBlank": False
}
"Folder": f"{project_name}",
"Name": f"{project_name}",
"Type": "GPT-SoVits",
"Rate": vits.hps.data.sampling_rate,
"NumLayers": gpt.t2s_model.num_layers,
"EmbeddingDim": gpt.t2s_model.embedding_dim,
"Dict": "BasicDict",
"BertPath": "chinese-roberta-wwm-ext-large",
# "Symbol": symbols,
"AddBlank": False,
}
MoeVSConfJson = json.dumps(MoeVSConf)
with open(f"onnx/{project_name}.json", 'w') as MoeVsConfFile:
json.dump(MoeVSConf, MoeVsConfFile, indent = 4)

View File

@ -7,18 +7,21 @@ inp_wav_dir = os.environ.get("inp_wav_dir")
exp_name = os.environ.get("exp_name")
i_part = os.environ.get("i_part")
all_parts = os.environ.get("all_parts")
os.environ["CUDA_VISIBLE_DEVICES"] = os.environ.get("_CUDA_VISIBLE_DEVICES")
if "_CUDA_VISIBLE_DEVICES" in os.environ:
os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"]
opt_dir = os.environ.get("opt_dir")
bert_pretrained_dir = os.environ.get("bert_pretrained_dir")
is_half = eval(os.environ.get("is_half", "True"))
import torch
is_half = eval(os.environ.get("is_half", "True")) and torch.cuda.is_available()
version = os.environ.get('version', None)
import sys, numpy as np, traceback, pdb
import os.path
from glob import glob
from tqdm import tqdm
from text.cleaner import clean_text
import torch
from transformers import AutoModelForMaskedLM, AutoTokenizer
import numpy as np
from tools.my_utils import clean_path
# inp_text=sys.argv[1]
# inp_wav_dir=sys.argv[2]
@ -53,6 +56,8 @@ if os.path.exists(txt_path) == False:
# device = "mps"
else:
device = "cpu"
if os.path.exists(bert_pretrained_dir):...
else:raise FileNotFoundError(bert_pretrained_dir)
tokenizer = AutoTokenizer.from_pretrained(bert_pretrained_dir)
bert_model = AutoModelForMaskedLM.from_pretrained(bert_pretrained_dir)
if is_half == True:
@ -81,9 +86,11 @@ if os.path.exists(txt_path) == False:
def process(data, res):
for name, text, lan in data:
try:
name=clean_path(name)
name = os.path.basename(name)
print(name)
phones, word2ph, norm_text = clean_text(
text.replace("%", "-").replace("", ","), lan
text.replace("%", "-").replace("", ","), lan, version
)
path_bert = "%s/%s.pt" % (bert_dir, name)
if os.path.exists(path_bert) == False and lan == "zh":
@ -112,14 +119,23 @@ if os.path.exists(txt_path) == False:
"EN": "en",
"en": "en",
"En": "en",
"KO": "ko",
"Ko": "ko",
"ko": "ko",
"yue": "yue",
"YUE": "yue",
"Yue": "yue",
}
for line in lines[int(i_part) :: int(all_parts)]:
try:
wav_name, spk_name, language, text = line.split("|")
# todo.append([name,text,"zh"])
todo.append(
[wav_name, text, language_v1_to_language_v2.get(language, language)]
)
if language in language_v1_to_language_v2.keys():
todo.append(
[wav_name, text, language_v1_to_language_v2.get(language, language)]
)
else:
print(f"\033[33m[Waring] The {language = } of {wav_name} is not supported for training.\033[0m")
except:
print(line, traceback.format_exc())

View File

@ -6,18 +6,20 @@ inp_wav_dir= os.environ.get("inp_wav_dir")
exp_name= os.environ.get("exp_name")
i_part= os.environ.get("i_part")
all_parts= os.environ.get("all_parts")
os.environ["CUDA_VISIBLE_DEVICES"]= os.environ.get("_CUDA_VISIBLE_DEVICES")
if "_CUDA_VISIBLE_DEVICES" in os.environ:
os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"]
from feature_extractor import cnhubert
opt_dir= os.environ.get("opt_dir")
cnhubert.cnhubert_base_path= os.environ.get("cnhubert_base_dir")
is_half=eval(os.environ.get("is_half","True"))
import torch
is_half = eval(os.environ.get("is_half", "True")) and torch.cuda.is_available()
import pdb,traceback,numpy as np,logging
from scipy.io import wavfile
import librosa,torch
import librosa
now_dir = os.getcwd()
sys.path.append(now_dir)
from my_utils import load_audio
from tools.my_utils import load_audio,clean_path
# from config import cnhubert_base_path
# cnhubert.cnhubert_base_path=cnhubert_base_path
@ -82,7 +84,7 @@ def name2go(wav_name,wav_path):
tensor_wav16 = tensor_wav16.to(device)
ssl=model.model(tensor_wav16.unsqueeze(0))["last_hidden_state"].transpose(1,2).cpu()#torch.Size([1, 768, 215])
if np.isnan(ssl.detach().numpy()).sum()!= 0:
nan_fails.append(wav_name)
nan_fails.append((wav_name,wav_path))
print("nan filtered:%s"%wav_name)
return
wavfile.write(
@ -90,7 +92,7 @@ def name2go(wav_name,wav_path):
32000,
tmp_audio32.astype("int16"),
)
my_save(ssl,hubert_path )
my_save(ssl,hubert_path)
with open(inp_text,"r",encoding="utf8")as f:
lines=f.read().strip("\n").split("\n")
@ -99,6 +101,7 @@ for line in lines[int(i_part)::int(all_parts)]:
try:
# wav_name,text=line.split("\t")
wav_name, spk_name, language, text = line.split("|")
wav_name=clean_path(wav_name)
if (inp_wav_dir != "" and inp_wav_dir != None):
wav_name = os.path.basename(wav_name)
wav_path = "%s/%s"%(inp_wav_dir, wav_name)
@ -113,8 +116,8 @@ for line in lines[int(i_part)::int(all_parts)]:
if(len(nan_fails)>0 and is_half==True):
is_half=False
model=model.float()
for wav_name in nan_fails:
for wav in nan_fails:
try:
name2go(wav_name)
name2go(wav[0],wav[1])
except:
print(wav_name,traceback.format_exc())

View File

@ -4,11 +4,28 @@ inp_text = os.environ.get("inp_text")
exp_name = os.environ.get("exp_name")
i_part = os.environ.get("i_part")
all_parts = os.environ.get("all_parts")
os.environ["CUDA_VISIBLE_DEVICES"] = os.environ.get("_CUDA_VISIBLE_DEVICES")
if "_CUDA_VISIBLE_DEVICES" in os.environ:
os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"]
opt_dir = os.environ.get("opt_dir")
pretrained_s2G = os.environ.get("pretrained_s2G")
s2config_path = os.environ.get("s2config_path")
is_half = eval(os.environ.get("is_half", "True"))
if os.path.exists(pretrained_s2G):...
else:raise FileNotFoundError(pretrained_s2G)
# version=os.environ.get("version","v2")
size = os.path.getsize(pretrained_s2G)
if size < 82978 * 1024:
version = "v1"
elif size < 100 * 1024 * 1024:
version = "v2"
elif size < 103520 * 1024:
version = "v1"
elif size < 700 * 1024 * 1024:
version = "v2"
else:
version = "v3"
import torch
is_half = eval(os.environ.get("is_half", "True")) and torch.cuda.is_available()
import math, traceback
import multiprocessing
import sys, pdb
@ -19,9 +36,12 @@ from random import shuffle
import torch.multiprocessing as mp
from glob import glob
from tqdm import tqdm
import logging, librosa, utils, torch
from module.models import SynthesizerTrn
import logging, librosa, utils
if version!="v3":
from module.models import SynthesizerTrn
else:
from module.models import SynthesizerTrnV3 as SynthesizerTrn
from tools.my_utils import clean_path
logging.getLogger("numba").setLevel(logging.WARNING)
# from config import pretrained_s2G
@ -49,6 +69,7 @@ if os.path.exists(semantic_path) == False:
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
version=version,
**hps.model
)
if is_half == True:
@ -60,7 +81,7 @@ if os.path.exists(semantic_path) == False:
# utils.load_checkpoint(pretrained_s2G, vq_model, None, True)
print(
vq_model.load_state_dict(
torch.load(pretrained_s2G, map_location="cpu")["weight"], strict=False
torch.load(pretrained_s2G, map_location="cpu", weights_only=False)["weight"], strict=False
)
)
@ -86,6 +107,7 @@ if os.path.exists(semantic_path) == False:
try:
# wav_name,text=line.split("\t")
wav_name, spk_name, language, text = line.split("|")
wav_name=clean_path(wav_name)
wav_name = os.path.basename(wav_name)
# name2go(name,lines1)
name2go(wav_name, lines1)

View File

@ -14,7 +14,24 @@ def my_save(fea,path):#####fix issue: torch.save doesn't support chinese path
torch.save(fea,tmp_path)
shutil.move(tmp_path,"%s/%s"%(dir,name))
def savee(ckpt, name, epoch, steps, hps):
'''
00:v1
01:v2
02:v3
03:v3lora
'''
from io import BytesIO
def my_save2(fea,path):
bio = BytesIO()
torch.save(fea, bio)
bio.seek(0)
data = bio.getvalue()
data = b'03' + data[2:]###temp for v3lora only, todo
with open(path, "wb") as f: f.write(data)
def savee(ckpt, name, epoch, steps, hps,lora_rank=None):
try:
opt = OrderedDict()
opt["weight"] = {}
@ -24,8 +41,66 @@ def savee(ckpt, name, epoch, steps, hps):
opt["weight"][key] = ckpt[key].half()
opt["config"] = hps
opt["info"] = "%sepoch_%siteration" % (epoch, steps)
# torch.save(opt, "%s/%s.pth" % (hps.save_weight_dir, name))
my_save(opt, "%s/%s.pth" % (hps.save_weight_dir, name))
if lora_rank:
opt["lora_rank"]=lora_rank
my_save2(opt, "%s/%s.pth" % (hps.save_weight_dir, name))
else:
my_save(opt, "%s/%s.pth" % (hps.save_weight_dir, name))
return "Success."
except:
return traceback.format_exc()
head2version={
b'00':["v1","v1",False],
b'01':["v2","v2",False],
b'02':["v2","v3",False],
b'03':["v2","v3",True],
}
hash_pretrained_dict={
"dc3c97e17592963677a4a1681f30c653":["v2","v2",False],#s2G488k.pth#sovits_v1_pretrained
"43797be674a37c1c83ee81081941ed0f":["v2","v3",False],#s2Gv3.pth#sovits_v3_pretrained
"6642b37f3dbb1f76882b69937c95a5f3":["v2","v2",False],#s2G2333K.pth#sovits_v2_pretrained
}
import hashlib
def get_hash_from_file(sovits_path):
with open(sovits_path,"rb")as f:data=f.read(8192)
hash_md5 = hashlib.md5()
hash_md5.update(data)
return hash_md5.hexdigest()
def get_sovits_version_from_path_fast(sovits_path):
###1-if it is pretrained sovits models, by hash
hash=get_hash_from_file(sovits_path)
if hash in hash_pretrained_dict:
return hash_pretrained_dict[hash]
###2-new weights or old weights, by head
with open(sovits_path,"rb")as f:version=f.read(2)
if version!=b"PK":
return head2version[version]
###3-old weights, by file size
if_lora_v3=False
size=os.path.getsize(sovits_path)
'''
v1weights:about 82942KB
half thr:82978KB
v2weights:about 83014KB
v3weights:about 750MB
'''
if size < 82978 * 1024:
model_version = version = "v1"
elif size < 700 * 1024 * 1024:
model_version = version = "v2"
else:
version = "v2"
model_version = "v3"
return version,model_version,if_lora_v3
def load_sovits_new(sovits_path):
f=open(sovits_path,"rb")
meta=f.read(2)
if meta!="PK":
data = b'PK' + f.read()
bio = BytesIO()
bio.write(data)
bio.seek(0)
return torch.load(bio, map_location="cpu", weights_only=False)
return torch.load(sovits_path,map_location="cpu", weights_only=False)

View File

@ -26,12 +26,7 @@ from AR.utils import get_newest_ckpt
from collections import OrderedDict
from time import time as ttime
import shutil
def my_save(fea,path):#####fix issue: torch.save doesn't support chinese path
dir=os.path.dirname(path)
name=os.path.basename(path)
tmp_path="%s.pth"%(ttime())
torch.save(fea,tmp_path)
shutil.move(tmp_path,"%s/%s"%(dir,name))
from process_ckpt import my_save
class my_model_ckpt(ModelCheckpoint):
@ -79,15 +74,17 @@ class my_model_ckpt(ModelCheckpoint):
to_save_od["config"] = self.config
to_save_od["info"] = "GPT-e%s" % (trainer.current_epoch + 1)
# torch.save(
my_save(
to_save_od,
"%s/%s-e%s.ckpt"
% (
self.half_weights_save_dir,
self.exp_name,
trainer.current_epoch + 1,
),
)
# print(os.environ)
if(os.environ.get("LOCAL_RANK","0")=="0"):
my_save(
to_save_od,
"%s/%s-e%s.ckpt"
% (
self.half_weights_save_dir,
self.exp_name,
trainer.current_epoch + 1,
),
)
self._save_last_checkpoint(trainer, monitor_candidates)
@ -116,6 +113,7 @@ def main(args):
)
logger = TensorBoardLogger(name=output_dir.stem, save_dir=output_dir)
os.environ["MASTER_ADDR"]="localhost"
os.environ["USE_LIBUV"] = "0"
trainer: Trainer = Trainer(
max_epochs=config["train"]["epochs"],
accelerator="gpu" if torch.cuda.is_available() else "cpu",
@ -132,6 +130,7 @@ def main(args):
logger=logger,
num_sanity_val_steps=0,
callbacks=[ckpt_callback],
use_distributed_sampler=False, # 非常简单的修改,但解决了采用自定义的 bucket_sampler 下训练步数不一致的问题!
)
model: Text2SemanticLightningModule = Text2SemanticLightningModule(

View File

@ -1,5 +1,6 @@
import warnings
warnings.filterwarnings("ignore")
import utils, os
hps = utils.get_hparams(stage=2)
os.environ["CUDA_VISIBLE_DEVICES"] = hps.train.gpu_numbers.replace("-", ",")
import torch
@ -74,7 +75,7 @@ def run(rank, n_gpus, hps):
dist.init_process_group(
backend = "gloo" if os.name == "nt" or not torch.cuda.is_available() else "nccl",
init_method="env://",
init_method="env://?use_libuv=False",
world_size=n_gpus,
rank=rank,
)
@ -119,7 +120,7 @@ def run(rank, n_gpus, hps):
collate_fn=collate_fn,
batch_sampler=train_sampler,
persistent_workers=True,
prefetch_factor=16,
prefetch_factor=4,
)
# if rank == 0:
# eval_dataset = TextAudioSpeakerLoader(hps.data.validation_files, hps.data, val=True)
@ -192,7 +193,7 @@ def run(rank, n_gpus, hps):
try: # 如果能加载自动resume
_, _, _, epoch_str = utils.load_checkpoint(
utils.latest_checkpoint_path("%s/logs_s2" % hps.data.exp_dir, "D_*.pth"),
utils.latest_checkpoint_path("%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "D_*.pth"),
net_d,
optim_d,
) # D多半加载没事
@ -200,10 +201,11 @@ def run(rank, n_gpus, hps):
logger.info("loaded D")
# _, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "G_*.pth"), net_g, optim_g,load_opt=0)
_, _, _, epoch_str = utils.load_checkpoint(
utils.latest_checkpoint_path("%s/logs_s2" % hps.data.exp_dir, "G_*.pth"),
utils.latest_checkpoint_path("%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "G_*.pth"),
net_g,
optim_g,
)
epoch_str+=1
global_step = (epoch_str - 1) * len(train_loader)
# epoch_str = 1
# global_step = 0
@ -211,10 +213,10 @@ def run(rank, n_gpus, hps):
# traceback.print_exc()
epoch_str = 1
global_step = 0
if hps.train.pretrained_s2G != "":
if hps.train.pretrained_s2G != ""and hps.train.pretrained_s2G != None and os.path.exists(hps.train.pretrained_s2G):
if rank == 0:
logger.info("loaded pretrained %s" % hps.train.pretrained_s2G)
print(
print("loaded pretrained %s" % hps.train.pretrained_s2G,
net_g.module.load_state_dict(
torch.load(hps.train.pretrained_s2G, map_location="cpu")["weight"],
strict=False,
@ -223,10 +225,10 @@ def run(rank, n_gpus, hps):
strict=False,
)
) ##测试不加载优化器
if hps.train.pretrained_s2D != "":
if hps.train.pretrained_s2D != ""and hps.train.pretrained_s2D != None and os.path.exists(hps.train.pretrained_s2D):
if rank == 0:
logger.info("loaded pretrained %s" % hps.train.pretrained_s2D)
print(
print("loaded pretrained %s" % hps.train.pretrained_s2D,
net_d.module.load_state_dict(
torch.load(hps.train.pretrained_s2D, map_location="cpu")["weight"]
) if torch.cuda.is_available() else net_d.load_state_dict(
@ -249,6 +251,7 @@ def run(rank, n_gpus, hps):
scaler = GradScaler(enabled=hps.train.fp16_run)
print("start training from epoch %s" % epoch_str)
for epoch in range(epoch_str, hps.train.epochs + 1):
if rank == 0:
train_and_evaluate(
@ -279,6 +282,7 @@ def run(rank, n_gpus, hps):
)
scheduler_g.step()
scheduler_d.step()
print("training done")
def train_and_evaluate(
@ -305,7 +309,7 @@ def train_and_evaluate(
y_lengths,
text,
text_lengths,
) in tqdm(enumerate(train_loader)):
) in enumerate(tqdm(train_loader)):
if torch.cuda.is_available():
spec, spec_lengths = spec.cuda(rank, non_blocking=True), spec_lengths.cuda(
rank, non_blocking=True
@ -425,26 +429,25 @@ def train_and_evaluate(
# scalar_dict.update({"loss/g/{}".format(i): v for i, v in enumerate(losses_gen)})
# scalar_dict.update({"loss/d_r/{}".format(i): v for i, v in enumerate(losses_disc_r)})
# scalar_dict.update({"loss/d_g/{}".format(i): v for i, v in enumerate(losses_disc_g)})
image_dict = {
"slice/mel_org": utils.plot_spectrogram_to_numpy(
y_mel[0].data.cpu().numpy()
),
"slice/mel_gen": utils.plot_spectrogram_to_numpy(
y_hat_mel[0].data.cpu().numpy()
),
"all/mel": utils.plot_spectrogram_to_numpy(
mel[0].data.cpu().numpy()
),
"all/stats_ssl": utils.plot_spectrogram_to_numpy(
stats_ssl[0].data.cpu().numpy()
),
}
utils.summarize(
writer=writer,
global_step=global_step,
images=image_dict,
scalars=scalar_dict,
)
image_dict=None
try:###Some people installed the wrong version of matplotlib.
image_dict = {
"slice/mel_org": utils.plot_spectrogram_to_numpy(
y_mel[0].data.cpu().numpy()
),
"slice/mel_gen": utils.plot_spectrogram_to_numpy(
y_hat_mel[0].data.cpu().numpy()
),
"all/mel": utils.plot_spectrogram_to_numpy(
mel[0].data.cpu().numpy()
),
"all/stats_ssl": utils.plot_spectrogram_to_numpy(
stats_ssl[0].data.cpu().numpy()
),
}
except:pass
if image_dict:utils.summarize(writer=writer,global_step=global_step,images=image_dict,scalars=scalar_dict,)
else:utils.summarize(writer=writer,global_step=global_step,scalars=scalar_dict,)
global_step += 1
if epoch % hps.train.save_every_epoch == 0 and rank == 0:
if hps.train.if_save_latest == 0:
@ -454,7 +457,7 @@ def train_and_evaluate(
hps.train.learning_rate,
epoch,
os.path.join(
"%s/logs_s2" % hps.data.exp_dir, "G_{}.pth".format(global_step)
"%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "G_{}.pth".format(global_step)
),
)
utils.save_checkpoint(
@ -463,7 +466,7 @@ def train_and_evaluate(
hps.train.learning_rate,
epoch,
os.path.join(
"%s/logs_s2" % hps.data.exp_dir, "D_{}.pth".format(global_step)
"%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "D_{}.pth".format(global_step)
),
)
else:
@ -473,7 +476,7 @@ def train_and_evaluate(
hps.train.learning_rate,
epoch,
os.path.join(
"%s/logs_s2" % hps.data.exp_dir, "G_{}.pth".format(233333333333)
"%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "G_{}.pth".format(233333333333)
),
)
utils.save_checkpoint(
@ -482,7 +485,7 @@ def train_and_evaluate(
hps.train.learning_rate,
epoch,
os.path.join(
"%s/logs_s2" % hps.data.exp_dir, "D_{}.pth".format(233333333333)
"%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "D_{}.pth".format(233333333333)
),
)
if rank == 0 and hps.train.if_save_every_weights == True:

416
GPT_SoVITS/s2_train_v3.py Normal file
View File

@ -0,0 +1,416 @@
import warnings
warnings.filterwarnings("ignore")
import utils, os
hps = utils.get_hparams(stage=2)
os.environ["CUDA_VISIBLE_DEVICES"] = hps.train.gpu_numbers.replace("-", ",")
import torch
from torch.nn import functional as F
from torch.utils.data import DataLoader
from torch.utils.tensorboard import SummaryWriter
import torch.multiprocessing as mp
import torch.distributed as dist, traceback
from torch.nn.parallel import DistributedDataParallel as DDP
from torch.cuda.amp import autocast, GradScaler
from tqdm import tqdm
import logging, traceback
logging.getLogger("matplotlib").setLevel(logging.INFO)
logging.getLogger("h5py").setLevel(logging.INFO)
logging.getLogger("numba").setLevel(logging.INFO)
from random import randint
from module import commons
from module.data_utils import (
TextAudioSpeakerLoaderV3 as TextAudioSpeakerLoader,
TextAudioSpeakerCollateV3 as TextAudioSpeakerCollate,
DistributedBucketSampler,
)
from module.models import (
SynthesizerTrnV3 as SynthesizerTrn,
MultiPeriodDiscriminator,
)
from module.losses import generator_loss, discriminator_loss, feature_loss, kl_loss
from module.mel_processing import mel_spectrogram_torch, spec_to_mel_torch
from process_ckpt import savee
torch.backends.cudnn.benchmark = False
torch.backends.cudnn.deterministic = False
###反正A100fp32更快那试试tf32吧
torch.backends.cuda.matmul.allow_tf32 = True
torch.backends.cudnn.allow_tf32 = True
torch.set_float32_matmul_precision("medium") # 最低精度但最快(也就快一丁点),对于结果造成不了影响
# from config import pretrained_s2G,pretrained_s2D
global_step = 0
device = "cpu" # cuda以外的设备等mps优化后加入
def main():
if torch.cuda.is_available():
n_gpus = torch.cuda.device_count()
else:
n_gpus = 1
os.environ["MASTER_ADDR"] = "localhost"
os.environ["MASTER_PORT"] = str(randint(20000, 55555))
mp.spawn(
run,
nprocs=n_gpus,
args=(
n_gpus,
hps,
),
)
def run(rank, n_gpus, hps):
global global_step
if rank == 0:
logger = utils.get_logger(hps.data.exp_dir)
logger.info(hps)
# utils.check_git_hash(hps.s2_ckpt_dir)
writer = SummaryWriter(log_dir=hps.s2_ckpt_dir)
writer_eval = SummaryWriter(log_dir=os.path.join(hps.s2_ckpt_dir, "eval"))
dist.init_process_group(
backend = "gloo" if os.name == "nt" or not torch.cuda.is_available() else "nccl",
init_method="env://?use_libuv=False",
world_size=n_gpus,
rank=rank,
)
torch.manual_seed(hps.train.seed)
if torch.cuda.is_available():
torch.cuda.set_device(rank)
train_dataset = TextAudioSpeakerLoader(hps.data) ########
train_sampler = DistributedBucketSampler(
train_dataset,
hps.train.batch_size,
[
32,
300,
400,
500,
600,
700,
800,
900,
1000,
# 1100,
# 1200,
# 1300,
# 1400,
# 1500,
# 1600,
# 1700,
# 1800,
# 1900,
],
num_replicas=n_gpus,
rank=rank,
shuffle=True,
)
collate_fn = TextAudioSpeakerCollate()
train_loader = DataLoader(
train_dataset,
num_workers=6,
shuffle=False,
pin_memory=True,
collate_fn=collate_fn,
batch_sampler=train_sampler,
persistent_workers=True,
prefetch_factor=4,
)
# if rank == 0:
# eval_dataset = TextAudioSpeakerLoader(hps.data.validation_files, hps.data, val=True)
# eval_loader = DataLoader(eval_dataset, num_workers=0, shuffle=False,
# batch_size=1, pin_memory=True,
# drop_last=False, collate_fn=collate_fn)
net_g = SynthesizerTrn(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
**hps.model,
).cuda(rank) if torch.cuda.is_available() else SynthesizerTrn(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
**hps.model,
).to(device)
# net_d = MultiPeriodDiscriminator(hps.model.use_spectral_norm).cuda(rank) if torch.cuda.is_available() else MultiPeriodDiscriminator(hps.model.use_spectral_norm).to(device)
# for name, param in net_g.named_parameters():
# if not param.requires_grad:
# print(name, "not requires_grad")
optim_g = torch.optim.AdamW(
filter(lambda p: p.requires_grad, net_g.parameters()),###默认所有层lr一致
hps.train.learning_rate,
betas=hps.train.betas,
eps=hps.train.eps,
)
# optim_d = torch.optim.AdamW(
# net_d.parameters(),
# hps.train.learning_rate,
# betas=hps.train.betas,
# eps=hps.train.eps,
# )
if torch.cuda.is_available():
net_g = DDP(net_g, device_ids=[rank], find_unused_parameters=True)
# net_d = DDP(net_d, device_ids=[rank], find_unused_parameters=True)
else:
net_g = net_g.to(device)
# net_d = net_d.to(device)
try: # 如果能加载自动resume
# _, _, _, epoch_str = utils.load_checkpoint(
# utils.latest_checkpoint_path("%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "D_*.pth"),
# net_d,
# optim_d,
# ) # D多半加载没事
# if rank == 0:
# logger.info("loaded D")
# _, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "G_*.pth"), net_g, optim_g,load_opt=0)
_, _, _, epoch_str = utils.load_checkpoint(
utils.latest_checkpoint_path("%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "G_*.pth"),
net_g,
optim_g,
)
epoch_str+=1
global_step = (epoch_str - 1) * len(train_loader)
# epoch_str = 1
# global_step = 0
except: # 如果首次不能加载加载pretrain
# traceback.print_exc()
epoch_str = 1
global_step = 0
if hps.train.pretrained_s2G != ""and hps.train.pretrained_s2G != None and os.path.exists(hps.train.pretrained_s2G):
if rank == 0:
logger.info("loaded pretrained %s" % hps.train.pretrained_s2G)
print("loaded pretrained %s" % hps.train.pretrained_s2G,
net_g.module.load_state_dict(
torch.load(hps.train.pretrained_s2G, map_location="cpu")["weight"],
strict=False,
) if torch.cuda.is_available() else net_g.load_state_dict(
torch.load(hps.train.pretrained_s2G, map_location="cpu")["weight"],
strict=False,
)
) ##测试不加载优化器
# if hps.train.pretrained_s2D != ""and hps.train.pretrained_s2D != None and os.path.exists(hps.train.pretrained_s2D):
# if rank == 0:
# logger.info("loaded pretrained %s" % hps.train.pretrained_s2D)
# print(
# net_d.module.load_state_dict(
# torch.load(hps.train.pretrained_s2D, map_location="cpu")["weight"]
# ) if torch.cuda.is_available() else net_d.load_state_dict(
# torch.load(hps.train.pretrained_s2D, map_location="cpu")["weight"]
# )
# )
# scheduler_g = torch.optim.lr_scheduler.ExponentialLR(optim_g, gamma=hps.train.lr_decay, last_epoch=epoch_str - 2)
# scheduler_d = torch.optim.lr_scheduler.ExponentialLR(optim_d, gamma=hps.train.lr_decay, last_epoch=epoch_str - 2)
scheduler_g = torch.optim.lr_scheduler.ExponentialLR(
optim_g, gamma=hps.train.lr_decay, last_epoch=-1
)
# scheduler_d = torch.optim.lr_scheduler.ExponentialLR(
# optim_d, gamma=hps.train.lr_decay, last_epoch=-1
# )
for _ in range(epoch_str):
scheduler_g.step()
# scheduler_d.step()
scaler = GradScaler(enabled=hps.train.fp16_run)
net_d=optim_d=scheduler_d=None
print("start training from epoch %s" % epoch_str)
for epoch in range(epoch_str, hps.train.epochs + 1):
if rank == 0:
train_and_evaluate(
rank,
epoch,
hps,
[net_g, net_d],
[optim_g, optim_d],
[scheduler_g, scheduler_d],
scaler,
# [train_loader, eval_loader], logger, [writer, writer_eval])
[train_loader, None],
logger,
[writer, writer_eval],
)
else:
train_and_evaluate(
rank,
epoch,
hps,
[net_g, net_d],
[optim_g, optim_d],
[scheduler_g, scheduler_d],
scaler,
[train_loader, None],
None,
None,
)
scheduler_g.step()
# scheduler_d.step()
print("training done")
def train_and_evaluate(
rank, epoch, hps, nets, optims, schedulers, scaler, loaders, logger, writers
):
net_g, net_d = nets
optim_g, optim_d = optims
# scheduler_g, scheduler_d = schedulers
train_loader, eval_loader = loaders
if writers is not None:
writer, writer_eval = writers
train_loader.batch_sampler.set_epoch(epoch)
global global_step
net_g.train()
# net_d.train()
# for batch_idx, (
# ssl,
# ssl_lengths,
# spec,
# spec_lengths,
# y,
# y_lengths,
# text,
# text_lengths,
# ) in enumerate(tqdm(train_loader)):
for batch_idx, (ssl, spec, mel, ssl_lengths, spec_lengths, text, text_lengths, mel_lengths) in enumerate(tqdm(train_loader)):
if torch.cuda.is_available():
spec, spec_lengths = spec.cuda(rank, non_blocking=True), spec_lengths.cuda(
rank, non_blocking=True
)
mel, mel_lengths = mel.cuda(rank, non_blocking=True), mel_lengths.cuda(
rank, non_blocking=True
)
ssl = ssl.cuda(rank, non_blocking=True)
ssl.requires_grad = False
# ssl_lengths = ssl_lengths.cuda(rank, non_blocking=True)
text, text_lengths = text.cuda(rank, non_blocking=True), text_lengths.cuda(
rank, non_blocking=True
)
else:
spec, spec_lengths = spec.to(device), spec_lengths.to(device)
mel, mel_lengths = mel.to(device), mel_lengths.to(device)
ssl = ssl.to(device)
ssl.requires_grad = False
# ssl_lengths = ssl_lengths.cuda(rank, non_blocking=True)
text, text_lengths = text.to(device), text_lengths.to(device)
with autocast(enabled=hps.train.fp16_run):
cfm_loss = net_g(ssl, spec, mel,ssl_lengths,spec_lengths, text, text_lengths,mel_lengths, use_grad_ckpt=hps.train.grad_ckpt)
loss_gen_all=cfm_loss
optim_g.zero_grad()
scaler.scale(loss_gen_all).backward()
scaler.unscale_(optim_g)
grad_norm_g = commons.clip_grad_value_(net_g.parameters(), None)
scaler.step(optim_g)
scaler.update()
if rank == 0:
if global_step % hps.train.log_interval == 0:
lr = optim_g.param_groups[0]['lr']
# losses = [commit_loss,cfm_loss,mel_loss,loss_disc, loss_gen, loss_fm, loss_mel, loss_kl]
losses = [cfm_loss]
logger.info('Train Epoch: {} [{:.0f}%]'.format(
epoch,
100. * batch_idx / len(train_loader)))
logger.info([x.item() for x in losses] + [global_step, lr])
scalar_dict = {"loss/g/total": loss_gen_all, "learning_rate": lr, "grad_norm_g": grad_norm_g}
# image_dict = {
# "slice/mel_org": utils.plot_spectrogram_to_numpy(y_mel[0].data.cpu().numpy()),
# "slice/mel_gen": utils.plot_spectrogram_to_numpy(y_hat_mel[0].data.cpu().numpy()),
# "all/mel": utils.plot_spectrogram_to_numpy(mel[0].data.cpu().numpy()),
# "all/stats_ssl": utils.plot_spectrogram_to_numpy(stats_ssl[0].data.cpu().numpy()),
# }
utils.summarize(
writer=writer,
global_step=global_step,
# images=image_dict,
scalars=scalar_dict)
# if global_step % hps.train.eval_interval == 0:
# # evaluate(hps, net_g, eval_loader, writer_eval)
# utils.save_checkpoint(net_g, optim_g, hps.train.learning_rate, epoch,os.path.join(hps.s2_ckpt_dir, "G_{}.pth".format(global_step)),scaler)
# # utils.save_checkpoint(net_d, optim_d, hps.train.learning_rate, epoch,os.path.join(hps.s2_ckpt_dir, "D_{}.pth".format(global_step)),scaler)
# # keep_ckpts = getattr(hps.train, 'keep_ckpts', 3)
# # if keep_ckpts > 0:
# # utils.clean_checkpoints(path_to_models=hps.s2_ckpt_dir, n_ckpts_to_keep=keep_ckpts, sort_by_time=True)
global_step += 1
if epoch % hps.train.save_every_epoch == 0 and rank == 0:
if hps.train.if_save_latest == 0:
utils.save_checkpoint(
net_g,
optim_g,
hps.train.learning_rate,
epoch,
os.path.join(
"%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "G_{}.pth".format(global_step)
),
)
# utils.save_checkpoint(
# net_d,
# optim_d,
# hps.train.learning_rate,
# epoch,
# os.path.join(
# "%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "D_{}.pth".format(global_step)
# ),
# )
else:
utils.save_checkpoint(
net_g,
optim_g,
hps.train.learning_rate,
epoch,
os.path.join(
"%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "G_{}.pth".format(233333333333)
),
)
# utils.save_checkpoint(
# net_d,
# optim_d,
# hps.train.learning_rate,
# epoch,
# os.path.join(
# "%s/logs_s2_%s" % (hps.data.exp_dir,hps.model.version), "D_{}.pth".format(233333333333)
# ),
# )
if rank == 0 and hps.train.if_save_every_weights == True:
if hasattr(net_g, "module"):
ckpt = net_g.module.state_dict()
else:
ckpt = net_g.state_dict()
logger.info(
"saving ckpt %s_e%s:%s"
% (
hps.name,
epoch,
savee(
ckpt,
hps.name + "_e%s_s%s" % (epoch, global_step),
epoch,
global_step,
hps,
),
)
)
if rank == 0:
logger.info("====> Epoch: {}".format(epoch))
if __name__ == "__main__":
main()

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@ -0,0 +1,345 @@
import warnings
warnings.filterwarnings("ignore")
import utils, os
hps = utils.get_hparams(stage=2)
os.environ["CUDA_VISIBLE_DEVICES"] = hps.train.gpu_numbers.replace("-", ",")
import torch
from torch.nn import functional as F
from torch.utils.data import DataLoader
from torch.utils.tensorboard import SummaryWriter
import torch.multiprocessing as mp
import torch.distributed as dist, traceback
from torch.nn.parallel import DistributedDataParallel as DDP
from torch.cuda.amp import autocast, GradScaler
from tqdm import tqdm
import logging, traceback
logging.getLogger("matplotlib").setLevel(logging.INFO)
logging.getLogger("h5py").setLevel(logging.INFO)
logging.getLogger("numba").setLevel(logging.INFO)
from random import randint
from module import commons
from peft import LoraConfig, PeftModel, get_peft_model
from module.data_utils import (
TextAudioSpeakerLoaderV3 as TextAudioSpeakerLoader,
TextAudioSpeakerCollateV3 as TextAudioSpeakerCollate,
DistributedBucketSampler,
)
from module.models import (
SynthesizerTrnV3 as SynthesizerTrn,
MultiPeriodDiscriminator,
)
from module.losses import generator_loss, discriminator_loss, feature_loss, kl_loss
from module.mel_processing import mel_spectrogram_torch, spec_to_mel_torch
from process_ckpt import savee
from collections import OrderedDict as od
torch.backends.cudnn.benchmark = False
torch.backends.cudnn.deterministic = False
###反正A100fp32更快那试试tf32吧
torch.backends.cuda.matmul.allow_tf32 = True
torch.backends.cudnn.allow_tf32 = True
torch.set_float32_matmul_precision("medium") # 最低精度但最快(也就快一丁点),对于结果造成不了影响
# from config import pretrained_s2G,pretrained_s2D
global_step = 0
device = "cpu" # cuda以外的设备等mps优化后加入
def main():
if torch.cuda.is_available():
n_gpus = torch.cuda.device_count()
else:
n_gpus = 1
os.environ["MASTER_ADDR"] = "localhost"
os.environ["MASTER_PORT"] = str(randint(20000, 55555))
mp.spawn(
run,
nprocs=n_gpus,
args=(
n_gpus,
hps,
),
)
def run(rank, n_gpus, hps):
global global_step,no_grad_names,save_root,lora_rank
if rank == 0:
logger = utils.get_logger(hps.data.exp_dir)
logger.info(hps)
# utils.check_git_hash(hps.s2_ckpt_dir)
writer = SummaryWriter(log_dir=hps.s2_ckpt_dir)
writer_eval = SummaryWriter(log_dir=os.path.join(hps.s2_ckpt_dir, "eval"))
dist.init_process_group(
backend = "gloo" if os.name == "nt" or not torch.cuda.is_available() else "nccl",
init_method="env://?use_libuv=False",
world_size=n_gpus,
rank=rank,
)
torch.manual_seed(hps.train.seed)
if torch.cuda.is_available():
torch.cuda.set_device(rank)
train_dataset = TextAudioSpeakerLoader(hps.data) ########
train_sampler = DistributedBucketSampler(
train_dataset,
hps.train.batch_size,
[
32,
300,
400,
500,
600,
700,
800,
900,
1000,
# 1100,
# 1200,
# 1300,
# 1400,
# 1500,
# 1600,
# 1700,
# 1800,
# 1900,
],
num_replicas=n_gpus,
rank=rank,
shuffle=True,
)
collate_fn = TextAudioSpeakerCollate()
train_loader = DataLoader(
train_dataset,
num_workers=6,
shuffle=False,
pin_memory=True,
collate_fn=collate_fn,
batch_sampler=train_sampler,
persistent_workers=True,
prefetch_factor=4,
)
save_root="%s/logs_s2_%s_lora_%s" % (hps.data.exp_dir,hps.model.version,hps.train.lora_rank)
os.makedirs(save_root,exist_ok=True)
lora_rank=int(hps.train.lora_rank)
lora_config = LoraConfig(
target_modules=["to_k", "to_q", "to_v", "to_out.0"],
r=lora_rank,
lora_alpha=lora_rank,
init_lora_weights=True,
)
def get_model(hps):return SynthesizerTrn(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
**hps.model,
)
def get_optim(net_g):
return torch.optim.AdamW(
filter(lambda p: p.requires_grad, net_g.parameters()), ###默认所有层lr一致
hps.train.learning_rate,
betas=hps.train.betas,
eps=hps.train.eps,
)
def model2cuda(net_g,rank):
if torch.cuda.is_available():
net_g = DDP(net_g.cuda(rank), device_ids=[rank], find_unused_parameters=True)
else:
net_g = net_g.to(device)
return net_g
try:# 如果能加载自动resume
net_g = get_model(hps)
net_g.cfm = get_peft_model(net_g.cfm, lora_config)
net_g=model2cuda(net_g,rank)
optim_g=get_optim(net_g)
# _, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "G_*.pth"), net_g, optim_g,load_opt=0)
_, _, _, epoch_str = utils.load_checkpoint(
utils.latest_checkpoint_path(save_root, "G_*.pth"),
net_g,
optim_g,
)
epoch_str+=1
global_step = (epoch_str - 1) * len(train_loader)
except: # 如果首次不能加载加载pretrain
# traceback.print_exc()
epoch_str = 1
global_step = 0
net_g = get_model(hps)
if hps.train.pretrained_s2G != ""and hps.train.pretrained_s2G != None and os.path.exists(hps.train.pretrained_s2G):
if rank == 0:
logger.info("loaded pretrained %s" % hps.train.pretrained_s2G)
print("loaded pretrained %s" % hps.train.pretrained_s2G,
net_g.load_state_dict(
torch.load(hps.train.pretrained_s2G, map_location="cpu")["weight"],
strict=False,
)
)
net_g.cfm = get_peft_model(net_g.cfm, lora_config)
net_g=model2cuda(net_g,rank)
optim_g = get_optim(net_g)
no_grad_names=set()
for name, param in net_g.named_parameters():
if not param.requires_grad:
no_grad_names.add(name.replace("module.",""))
# print(name, "not requires_grad")
# print(no_grad_names)
# os._exit(233333)
scheduler_g = torch.optim.lr_scheduler.ExponentialLR(
optim_g, gamma=hps.train.lr_decay, last_epoch=-1
)
for _ in range(epoch_str):
scheduler_g.step()
scaler = GradScaler(enabled=hps.train.fp16_run)
net_d=optim_d=scheduler_d=None
print("start training from epoch %s"%epoch_str)
for epoch in range(epoch_str, hps.train.epochs + 1):
if rank == 0:
train_and_evaluate(
rank,
epoch,
hps,
[net_g, net_d],
[optim_g, optim_d],
[scheduler_g, scheduler_d],
scaler,
# [train_loader, eval_loader], logger, [writer, writer_eval])
[train_loader, None],
logger,
[writer, writer_eval],
)
else:
train_and_evaluate(
rank,
epoch,
hps,
[net_g, net_d],
[optim_g, optim_d],
[scheduler_g, scheduler_d],
scaler,
[train_loader, None],
None,
None,
)
scheduler_g.step()
print("training done")
def train_and_evaluate(
rank, epoch, hps, nets, optims, schedulers, scaler, loaders, logger, writers
):
net_g, net_d = nets
optim_g, optim_d = optims
# scheduler_g, scheduler_d = schedulers
train_loader, eval_loader = loaders
if writers is not None:
writer, writer_eval = writers
train_loader.batch_sampler.set_epoch(epoch)
global global_step
net_g.train()
for batch_idx, (ssl, spec, mel, ssl_lengths, spec_lengths, text, text_lengths, mel_lengths) in enumerate(tqdm(train_loader)):
if torch.cuda.is_available():
spec, spec_lengths = spec.cuda(rank, non_blocking=True), spec_lengths.cuda(
rank, non_blocking=True
)
mel, mel_lengths = mel.cuda(rank, non_blocking=True), mel_lengths.cuda(
rank, non_blocking=True
)
ssl = ssl.cuda(rank, non_blocking=True)
ssl.requires_grad = False
text, text_lengths = text.cuda(rank, non_blocking=True), text_lengths.cuda(
rank, non_blocking=True
)
else:
spec, spec_lengths = spec.to(device), spec_lengths.to(device)
mel, mel_lengths = mel.to(device), mel_lengths.to(device)
ssl = ssl.to(device)
ssl.requires_grad = False
text, text_lengths = text.to(device), text_lengths.to(device)
with autocast(enabled=hps.train.fp16_run):
cfm_loss = net_g(ssl, spec, mel,ssl_lengths,spec_lengths, text, text_lengths,mel_lengths, use_grad_ckpt=hps.train.grad_ckpt)
loss_gen_all=cfm_loss
optim_g.zero_grad()
scaler.scale(loss_gen_all).backward()
scaler.unscale_(optim_g)
grad_norm_g = commons.clip_grad_value_(net_g.parameters(), None)
scaler.step(optim_g)
scaler.update()
if rank == 0:
if global_step % hps.train.log_interval == 0:
lr = optim_g.param_groups[0]['lr']
losses = [cfm_loss]
logger.info('Train Epoch: {} [{:.0f}%]'.format(
epoch,
100. * batch_idx / len(train_loader)))
logger.info([x.item() for x in losses] + [global_step, lr])
scalar_dict = {"loss/g/total": loss_gen_all, "learning_rate": lr, "grad_norm_g": grad_norm_g}
utils.summarize(
writer=writer,
global_step=global_step,
scalars=scalar_dict)
global_step += 1
if epoch % hps.train.save_every_epoch == 0 and rank == 0:
if hps.train.if_save_latest == 0:
utils.save_checkpoint(
net_g,
optim_g,
hps.train.learning_rate,
epoch,
os.path.join(
save_root, "G_{}.pth".format(global_step)
),
)
else:
utils.save_checkpoint(
net_g,
optim_g,
hps.train.learning_rate,
epoch,
os.path.join(
save_root, "G_{}.pth".format(233333333333)
),
)
if rank == 0 and hps.train.if_save_every_weights == True:
if hasattr(net_g, "module"):
ckpt = net_g.module.state_dict()
else:
ckpt = net_g.state_dict()
sim_ckpt=od()
for key in ckpt:
# if "cfm"not in key:
# print(key)
if key not in no_grad_names:
sim_ckpt[key]=ckpt[key].half().cpu()
logger.info(
"saving ckpt %s_e%s:%s"
% (
hps.name,
epoch,
savee(
sim_ckpt,
hps.name + "_e%s_s%s_l%s" % (epoch, global_step,lora_rank),
epoch,
global_step,
hps,lora_rank=lora_rank
),
)
)
if rank == 0:
logger.info("====> Epoch: {}".format(epoch))
if __name__ == "__main__":
main()

3
GPT_SoVITS/text/.gitignore vendored Normal file
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@ -0,0 +1,3 @@
G2PWModel
__pycache__
*.zip

View File

@ -0,0 +1 @@
from .langsegmenter import LangSegmenter

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@ -0,0 +1,158 @@
import logging
import re
# jieba静音
import jieba
jieba.setLogLevel(logging.CRITICAL)
# 更改fast_langdetect大模型位置
from pathlib import Path
import fast_langdetect
fast_langdetect.infer._default_detector = fast_langdetect.infer.LangDetector(fast_langdetect.infer.LangDetectConfig(cache_dir=Path(__file__).parent.parent.parent / "pretrained_models" / "fast_langdetect"))
from split_lang import LangSplitter
def full_en(text):
pattern = r'^[A-Za-z0-9\s\u0020-\u007E\u2000-\u206F\u3000-\u303F\uFF00-\uFFEF]+$'
return bool(re.match(pattern, text))
def full_cjk(text):
# 来自wiki
cjk_ranges = [
(0x4E00, 0x9FFF), # CJK Unified Ideographs
(0x3400, 0x4DB5), # CJK Extension A
(0x20000, 0x2A6DD), # CJK Extension B
(0x2A700, 0x2B73F), # CJK Extension C
(0x2B740, 0x2B81F), # CJK Extension D
(0x2B820, 0x2CEAF), # CJK Extension E
(0x2CEB0, 0x2EBEF), # CJK Extension F
(0x30000, 0x3134A), # CJK Extension G
(0x31350, 0x323AF), # CJK Extension H
(0x2EBF0, 0x2EE5D), # CJK Extension H
]
pattern = r'[0-9、-〜。!?.!?… ]+$'
cjk_text = ""
for char in text:
code_point = ord(char)
in_cjk = any(start <= code_point <= end for start, end in cjk_ranges)
if in_cjk or re.match(pattern, char):
cjk_text += char
return cjk_text
def split_jako(tag_lang,item):
if tag_lang == "ja":
pattern = r"([\u3041-\u3096\u3099\u309A\u30A1-\u30FA\u30FC]+(?:[0-9、-〜。!?.!?… ]+[\u3041-\u3096\u3099\u309A\u30A1-\u30FA\u30FC]*)*)"
else:
pattern = r"([\u1100-\u11FF\u3130-\u318F\uAC00-\uD7AF]+(?:[0-9、-〜。!?.!?… ]+[\u1100-\u11FF\u3130-\u318F\uAC00-\uD7AF]*)*)"
lang_list: list[dict] = []
tag = 0
for match in re.finditer(pattern, item['text']):
if match.start() > tag:
lang_list.append({'lang':item['lang'],'text':item['text'][tag:match.start()]})
tag = match.end()
lang_list.append({'lang':tag_lang,'text':item['text'][match.start():match.end()]})
if tag < len(item['text']):
lang_list.append({'lang':item['lang'],'text':item['text'][tag:len(item['text'])]})
return lang_list
def merge_lang(lang_list, item):
if lang_list and item['lang'] == lang_list[-1]['lang']:
lang_list[-1]['text'] += item['text']
else:
lang_list.append(item)
return lang_list
class LangSegmenter():
# 默认过滤器, 基于gsv目前四种语言
DEFAULT_LANG_MAP = {
"zh": "zh",
"yue": "zh", # 粤语
"wuu": "zh", # 吴语
"zh-cn": "zh",
"zh-tw": "x", # 繁体设置为x
"ko": "ko",
"ja": "ja",
"en": "en",
}
def getTexts(text):
lang_splitter = LangSplitter(lang_map=LangSegmenter.DEFAULT_LANG_MAP)
substr = lang_splitter.split_by_lang(text=text)
lang_list: list[dict] = []
for _, item in enumerate(substr):
dict_item = {'lang':item.lang,'text':item.text}
# 处理短英文被识别为其他语言的问题
if full_en(dict_item['text']):
dict_item['lang'] = 'en'
lang_list = merge_lang(lang_list,dict_item)
continue
# 处理非日语夹日文的问题(不包含CJK)
ja_list: list[dict] = []
if dict_item['lang'] != 'ja':
ja_list = split_jako('ja',dict_item)
if not ja_list:
ja_list.append(dict_item)
# 处理非韩语夹韩语的问题(不包含CJK)
ko_list: list[dict] = []
temp_list: list[dict] = []
for _, ko_item in enumerate(ja_list):
if ko_item["lang"] != 'ko':
ko_list = split_jako('ko',ko_item)
if ko_list:
temp_list.extend(ko_list)
else:
temp_list.append(ko_item)
# 未存在非日韩文夹日韩文
if len(temp_list) == 1:
# 未知语言检查是否为CJK
if dict_item['lang'] == 'x':
cjk_text = full_cjk(dict_item['text'])
if cjk_text:
dict_item = {'lang':'zh','text':cjk_text}
lang_list = merge_lang(lang_list,dict_item)
continue
else:
lang_list = merge_lang(lang_list,dict_item)
continue
# 存在非日韩文夹日韩文
for _, temp_item in enumerate(temp_list):
# 未知语言检查是否为CJK
if temp_item['lang'] == 'x':
cjk_text = full_cjk(dict_item['text'])
if cjk_text:
dict_item = {'lang':'zh','text':cjk_text}
lang_list = merge_lang(lang_list,dict_item)
else:
lang_list = merge_lang(lang_list,temp_item)
return lang_list
if __name__ == "__main__":
text = "MyGO?,你也喜欢まいご吗?"
print(LangSegmenter.getTexts(text))
text = "ねえ、知ってる?最近、僕は天文学を勉強してるんだ。君の瞳が星空みたいにキラキラしてるからさ。"
print(LangSegmenter.getTexts(text))

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@ -1,15 +1,27 @@
from text.symbols import *
import os
# if os.environ.get("version","v1")=="v1":
# from text.symbols import symbols
# else:
# from text.symbols2 import symbols
from text import symbols as symbols_v1
from text import symbols2 as symbols_v2
_symbol_to_id = {s: i for i, s in enumerate(symbols)}
_symbol_to_id_v1 = {s: i for i, s in enumerate(symbols_v1.symbols)}
_symbol_to_id_v2 = {s: i for i, s in enumerate(symbols_v2.symbols)}
def cleaned_text_to_sequence(cleaned_text):
def cleaned_text_to_sequence(cleaned_text, version=None):
'''Converts a string of text to a sequence of IDs corresponding to the symbols in the text.
Args:
text: string to convert to a sequence
Returns:
List of integers corresponding to the symbols in the text
'''
phones = [_symbol_to_id[symbol] for symbol in cleaned_text]
if version is None:version=os.environ.get('version', 'v2')
if version == "v1":
phones = [_symbol_to_id_v1[symbol] for symbol in cleaned_text]
else:
phones = [_symbol_to_id_v2[symbol] for symbol in cleaned_text]
return phones

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# reference: https://huggingface.co/spaces/Naozumi0512/Bert-VITS2-Cantonese-Yue/blob/main/text/chinese.py
import sys
import re
import cn2an
import ToJyutping
from text.symbols import punctuation
from text.zh_normalization.text_normlization import TextNormalizer
normalizer = lambda x: cn2an.transform(x, "an2cn")
INITIALS = [
"aa",
"aai",
"aak",
"aap",
"aat",
"aau",
"ai",
"au",
"ap",
"at",
"ak",
"a",
"p",
"b",
"e",
"ts",
"t",
"dz",
"d",
"kw",
"k",
"gw",
"g",
"f",
"h",
"l",
"m",
"ng",
"n",
"s",
"y",
"w",
"c",
"z",
"j",
"ong",
"on",
"ou",
"oi",
"ok",
"o",
"uk",
"ung",
]
INITIALS += ["sp", "spl", "spn", "sil"]
rep_map = {
"": ",",
"": ",",
"": ",",
"": ".",
"": "!",
"": "?",
"\n": ".",
"·": ",",
"": ",",
"...": "",
"$": ".",
"": "'",
"": "'",
'"': "'",
"": "'",
"": "'",
"": "'",
"": "'",
"(": "'",
")": "'",
"": "'",
"": "'",
"": "'",
"": "'",
"[": "'",
"]": "'",
"": "-",
"": "-",
"~": "-",
"": "'",
"": "'",
}
def replace_punctuation(text):
# text = text.replace("嗯", "恩").replace("呣", "母")
pattern = re.compile("|".join(re.escape(p) for p in rep_map.keys()))
replaced_text = pattern.sub(lambda x: rep_map[x.group()], text)
replaced_text = re.sub(
r"[^\u4e00-\u9fa5" + "".join(punctuation) + r"]+", "", replaced_text
)
return replaced_text
def text_normalize(text):
tx = TextNormalizer()
sentences = tx.normalize(text)
dest_text = ""
for sentence in sentences:
dest_text += replace_punctuation(sentence)
return dest_text
punctuation_set=set(punctuation)
def jyuping_to_initials_finals_tones(jyuping_syllables):
initials_finals = []
tones = []
word2ph = []
for syllable in jyuping_syllables:
if syllable in punctuation:
initials_finals.append(syllable)
tones.append(0)
word2ph.append(1) # Add 1 for punctuation
elif syllable == "_":
initials_finals.append(syllable)
tones.append(0)
word2ph.append(1) # Add 1 for underscore
else:
try:
tone = int(syllable[-1])
syllable_without_tone = syllable[:-1]
except ValueError:
tone = 0
syllable_without_tone = syllable
for initial in INITIALS:
if syllable_without_tone.startswith(initial):
if syllable_without_tone.startswith("nga"):
initials_finals.extend(
[
syllable_without_tone[:2],
syllable_without_tone[2:] or syllable_without_tone[-1],
]
)
# tones.extend([tone, tone])
tones.extend([-1, tone])
word2ph.append(2)
else:
final = syllable_without_tone[len(initial) :] or initial[-1]
initials_finals.extend([initial, final])
# tones.extend([tone, tone])
tones.extend([-1, tone])
word2ph.append(2)
break
assert len(initials_finals) == len(tones)
###魔改为辅音+带音调的元音
phones=[]
for a,b in zip(initials_finals,tones):
if(b not in [-1,0]):###防止粤语和普通话重合开头加Y如果是标点不加。
todo="%s%s"%(a,b)
else:todo=a
if(todo not in punctuation_set):todo="Y%s"%todo
phones.append(todo)
# return initials_finals, tones, word2ph
return phones, word2ph
def get_jyutping(text):
jyutping_array = []
punct_pattern = re.compile(r"^[{}]+$".format(re.escape("".join(punctuation))))
syllables = ToJyutping.get_jyutping_list(text)
for word, syllable in syllables:
if punct_pattern.match(word):
puncts = re.split(r"([{}])".format(re.escape("".join(punctuation))), word)
for punct in puncts:
if len(punct) > 0:
jyutping_array.append(punct)
else:
# match multple jyutping eg: liu4 ge3, or single jyutping eg: liu4
if not re.search(r"^([a-z]+[1-6]+[ ]?)+$", syllable):
raise ValueError(f"Failed to convert {word} to jyutping: {syllable}")
jyutping_array.append(syllable)
return jyutping_array
def get_bert_feature(text, word2ph):
from text import chinese_bert
return chinese_bert.get_bert_feature(text, word2ph)
def g2p(text):
# word2ph = []
jyuping = get_jyutping(text)
# print(jyuping)
# phones, tones, word2ph = jyuping_to_initials_finals_tones(jyuping)
phones, word2ph = jyuping_to_initials_finals_tones(jyuping)
# phones = ["_"] + phones + ["_"]
# tones = [0] + tones + [0]
# word2ph = [1] + word2ph + [1]
return phones, word2ph
if __name__ == "__main__":
# text = "啊!但是《原神》是由,米哈\游自主, [研发]的一款全.新开放世界.冒险游戏"
text = "佢個鋤頭太短啦。"
text = text_normalize(text)
# phones, tones, word2ph = g2p(text)
phones, word2ph = g2p(text)
# print(phones, tones, word2ph)
print(phones, word2ph)

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