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Dev no emo (#123)

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* Create emo_gen.py

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* update server.py, fix bugs in func get_text() and infer(). (#52)

* Extract get_text() and infer() from webui.py. (#53)

* Extract get_text() and infer() from webui.py.

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* add emo emb

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* init emo gen

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* init emo

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* init emo

* Delete bert/bert-base-japanese-v3 directory

* Create .gitkeep

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* Create add_punc.py

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* fix bug in bert_gen.py (#54)

* Update README.md

* fix bug in models.py (#56)

* 更新 models.py

* Fix japanese cleaner (#61)

* 初步，睡觉明天继续写（

* 好好好放错分支了，熬夜是大忌

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* Create tokenizer_config.json

* update preprocess_text.py：过滤一个音频匹配多个文本的情况 (#57)

* update preprocess_text.py：过滤音频不存在的情况 (#58)

* 修复日语cleaner和bert

* better

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* Apply Code Formatter Change

* Add config.yml for global configuration. (#62)

* Add config.yml for global configuration.

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* Fix bug in webui.py.

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* Rename config.yml to default_config.yml. Add ./config.yml to gitignore.

* Add config.py to parse config.yml

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* Update webui.py (#65)

* Update webui.py:
1. Add auto translation from Chinese to Japanese.
2. Start to use config.py in webui.py to set config instead of using the command line.

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* Fix (#68)

* 加上ー

* fix

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* Update infer.py and webui.py.  Supports loading and inference models of 1.1.1 version. (#66)

* Update infer.py and webui.py. Supports loading and inference models of 1.1.1 version.

* Update config.json

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* Fix bug in translate.py (#69)

* Supports loading and inference models of 1.1、1.0.1、1.0 version. (#70)

* Supports loading and inference models of 1.1、1.0.1、1.0 version.

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* Delete useless file in OldVersion

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* Update japanese.py (#71)

Handling JA long pronunciations

* 使用配置文件配置bert_gen.py, preprocess_text.py, resample.py (#72)

* Update bert_gen.py, preprocess_text.py, resample.py. Support using config.yml in these files.

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* Update bert_gen.py

* Update bert_gen.py, fix bug.

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* Delete bert/bert-base-japanese-v3 directory

* Create config.json

* Create tokenizer_config.json

* Create vocab.txt

* Update server.py. 支持多版本多模型 (#76)

* Update server.py. 支持多版本多模型

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* Dev webui (#77)

* 申请pr (#75)

* 2023/10/11 update

界面优化

* Update webui.py

翻译英文页面为中文

* Update train_ms.py

单卡训练

* 加入图片

* Update extern_subprocess.py

* Update asr_transcript.py

* Update asr_transcript.py

* Update asr_transcript.py

* Update extern_subprocess.py

* Update asr_transcript.py

* Update asr_transcript.py

* Update asr_transcript.py

* Update all_process.py

* Update extern_subprocess.py

* Update all_process.py

* Update all_process.py

* Update asr_transcript.py

* Update extern_subprocess.py

* Update webui.py

* Create re_matching.py

* Update webui.py

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* Update all_process.py

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* Update all_process.py

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* Update all_process.py

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* Update asr_transcript.py

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* Pack 'update' functions into a module

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* Update all_process.py

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* Update asr_transcript.py

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* Update extern_subprocess.py

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* Update README.md

* Update README.md

* Update README.md

* Update README.md

* Update README.md

* Update all_process.py

* Update asr_transcript.py

* Update webui.py

* Add files via upload

* Update extern_subprocess.py

* Update all_process.py

* Update asr_transcript.py

* Update bert_gen.py

* Update extern_subprocess.py

* Update preprocess_text.py

* Update re_matching.py

* Update resample.py

* Update update_status.py

* Update update_status.py

* Update webui.py

* Update all_process.py

* Update preprocess_text.py

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* Update train_ms.py

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* Delete all_process.py

* Delete asr_transcript.py

* Delete extern_subprocess.py

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* Create config.json

* Create  preprocessor_config.json

* Create vocab.json

* Delete emotional/wav2vec2-large-robust-12-ft-emotion-msp-dim/.gitkeep

* Update emo_gen.py

* Delete add_punc.py

* add emotion_clustering.i

* Apply Code Formatter Change

* Update models.py

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* Update preprocess_text.py (#78)

* Update preprocess_text.py. 检测重复以及不存在的音频 (#79)

* Handle Janpanese long pronunciations (#80)

* Handle Janpanese long pronunciations

* Update japanese.py

* Update japanese.py

* Use unified phonemes for Japanese long vowel (#82)

* Use an unified phoneme for Japanese long vowel

`symbol.py` has not been updated to ensure compatibility with older version models.

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* 增加一个按钮，点击后可以按句子切分，添加“|” (#81)

* Update re_matching.py

* Update webui.py

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* Fix phonemer bug (#83)

* Fix phonemer bug

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* Fix long vowel handler bug (#84)

* Fix long vowel handler bug

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* 加入整合包管理器的特性：长文本合成可以自定义句间段间停顿 (#85)

* Update webui.py

* Update re_matching.py

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* Update train_ms.py

* fix'

* Update cleaner.py

* add en

* add en

* Update english.py

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* add en

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* add en

* add en

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* add en

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* 更新 README.md

* 更新 README.md

* 更新 README.md

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* Change phonemer to pyopenjtalk (#86)

* Change phonemer to pyopenjtalk

* 修改为openjtalk便于安装

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* 更新 english.py

* Fix english_bert_mock.py. (#87)

* Add punctuation execptions (#88)

* Add punctuation execptions

* Ellipses exceptions

* remove get bert

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* Fix bug in oldVersion. (#89)

* Update requirements.txt

* change to large

* rollback requirements.txt

* Feat: Enable 1.1.1 models using fix-ver infer. (#91)

* Feat: Enable 1.1.1 models using fix-ver infer.

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* Add Japanese accent (high-low) (#90)

* Add punctuation execptions

* Ellipses exceptions

* Add Japanese accent

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* Do not replace iteration mark (#92)

* Add punctuation execptions

* Ellipses exceptions

* Add Japanese accent

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* Do not replace iteration mark

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* Fix: fix import error in oldVersion (#93)

* Refactor: reusing model loading in webui.py and server.py. (#94)

* Feat: Enable using config.yml in train_ms.py (#96)

* 更新 emo_gen.py

* Change emo_gen.py (#97)

* Fix emo_gen bugs

* Add multiprocess

* Fix queue (#98)

* Fix emo_gen bugs

* Add multiprocess

* Del var

* Fix queue

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* Fix training bugs (#99)

* Updatge cluster notebook

* Fix train

* Fix filename

* Update infer.py (#100)

* Update infer.py

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* Add reference audio (#101)

* Add reference audio

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* Update

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* Update

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* Fix: fix 1.1.1-fix (#102)

* Fix infer bug (#103)

* Feat: Add server_fastapi.py. (#104)

* Feat: Add server_fastapi.py.

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* Fix: Update requirements.txt.

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* Fix: requirements.txt. (#105)

* Swith to deberta-v3-large (#106)

* Swith to deberta-v3-large

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* Feat: Update config.py. (#107)

* Feat: Update config.py.

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* Dev fix (#108)

* fix bugs when deploying

* fix bugs when deploying

* fix bugs when deploying

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* Revert "Dev fix (#108)" (#109)

This reverts commit 685e18a10498d602b1a9a26079340d11925646f0.

* Dev fix (#110)

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* fix bugs when deploying

* fix bugs when deploying

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* fix fixed bugs

* fix fixed bugs

* fix fixed bug 3

* fix fixed bug 4

* fix fixed bug 5

* fix

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* Add emo vec quantizer (#111)

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* Clean req and gitignore (#112)

* Clean req and gitignore

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* Switch to deberta-v2-large-japanese (#113)

* Switch to deberta-v2-large-japanese

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* Fix emo bugs (#114)

* Fix english (#115)

* Remove emo (#117)

* Don't train codebook

* Remove emo

* Update

* Update

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* Merge dev into no-emo (#122)

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* Don't train codebook (#116)

* Update requirements.txt

* Update english_bert_mock.py

* Fix: server_fastapi.py (#118)

* Fix: server_fastapi.py

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* Fix: don't print debug logging. (#119)

* Fix: don't print debug logging.

* Feat: support emo_gen config

* Fix config

* Apply Code Formatter Change

* 更新，修正bug (#121)

* Feat: Update infer.py preprocess_text.py server_fastapi.py.

* Fix resample.py. Maintain same directory structure in out_dir as in_dir.

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* Update server_fastapi.py to no-emo ver

* Update config.py, no emo config

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* Update train_ms.py

* Update latest version info (#124)

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This commit is contained in:
Stardust·减
2023-10-28 10:36:21 +08:00
committed by GitHub
parent c1ba4c7b3a
commit 82ae8f6928
102 changed files with 175574 additions and 1062 deletions
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356
webui.py
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@@ -1,7 +1,7 @@
# flake8: noqa: E402
import sys, os
import os
import logging
import re_matching
logging.getLogger("numba").setLevel(logging.WARNING)
logging.getLogger("markdown_it").setLevel(logging.WARNING)
@@ -15,197 +15,242 @@ logging.basicConfig(
logger = logging.getLogger(__name__)
import torch
import argparse
import commons
import utils
from models import SynthesizerTrn
from text.symbols import symbols
from text import cleaned_text_to_sequence, get_bert
from text.cleaner import clean_text
from infer import infer, latest_version, get_net_g
import gradio as gr
import webbrowser
import numpy as np
from config import config
from tools.translate import translate
net_g = None
if sys.platform == "darwin" and torch.backends.mps.is_available():
device = "mps"
device = config.webui_config.device
if device == "mps":
os.environ["PYTORCH_ENABLE_MPS_FALLBACK"] = "1"
else:
device = "cuda"
def get_text(text, language_str, hps):
norm_text, phone, tone, word2ph = clean_text(text, language_str)
phone, tone, language = cleaned_text_to_sequence(phone, tone, language_str)
if hps.data.add_blank:
phone = commons.intersperse(phone, 0)
tone = commons.intersperse(tone, 0)
language = commons.intersperse(language, 0)
for i in range(len(word2ph)):
word2ph[i] = word2ph[i] * 2
word2ph[0] += 1
bert = get_bert(norm_text, word2ph, language_str, device)
del word2ph
assert bert.shape[-1] == len(phone), phone
if language_str == "ZH":
bert = bert
ja_bert = torch.zeros(768, len(phone))
elif language_str == "JP":
ja_bert = bert
bert = torch.zeros(1024, len(phone))
else:
bert = torch.zeros(1024, len(phone))
ja_bert = torch.zeros(768, len(phone))
assert bert.shape[-1] == len(
phone
), f"Bert seq len {bert.shape[-1]} != {len(phone)}"
phone = torch.LongTensor(phone)
tone = torch.LongTensor(tone)
language = torch.LongTensor(language)
return bert, ja_bert, phone, tone, language
def infer(text, sdp_ratio, noise_scale, noise_scale_w, length_scale, sid, language):
global net_g
bert, ja_bert, phones, tones, lang_ids = get_text(text, language, hps)
with torch.no_grad():
x_tst = phones.to(device).unsqueeze(0)
tones = tones.to(device).unsqueeze(0)
lang_ids = lang_ids.to(device).unsqueeze(0)
bert = bert.to(device).unsqueeze(0)
ja_bert = ja_bert.to(device).unsqueeze(0)
x_tst_lengths = torch.LongTensor([phones.size(0)]).to(device)
del phones
speakers = torch.LongTensor([hps.data.spk2id[sid]]).to(device)
audio = (
net_g.infer(
x_tst,
x_tst_lengths,
speakers,
tones,
lang_ids,
bert,
ja_bert,
sdp_ratio=sdp_ratio,
noise_scale=noise_scale,
noise_scale_w=noise_scale_w,
length_scale=length_scale,
)[0][0, 0]
.data.cpu()
.float()
.numpy()
)
del x_tst, tones, lang_ids, bert, x_tst_lengths, speakers
torch.cuda.empty_cache()
return audio
def tts_fn(
text, speaker, sdp_ratio, noise_scale, noise_scale_w, length_scale, language
def generate_audio(
slices,
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
speaker,
language,
):
slices = text.split("|")
audio_list = []
silence = np.zeros(hps.data.sampling_rate // 2)
with torch.no_grad():
for slice in slices:
for piece in slices:
audio = infer(
slice,
piece,
sdp_ratio=sdp_ratio,
noise_scale=noise_scale,
noise_scale_w=noise_scale_w,
length_scale=length_scale,
sid=speaker,
language=language,
hps=hps,
net_g=net_g,
device=device,
)
audio_list.append(audio)
silence = np.zeros(hps.data.sampling_rate) # 生成1秒的静音
audio_list.append(silence) # 将静音添加到列表中
return audio_list
def tts_split(
text: str,
speaker,
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
language,
cut_by_sent,
interval_between_para,
interval_between_sent,
):
if language == "mix":
return ("invalid", None)
while text.find("\n\n") != -1:
text = text.replace("\n\n", "\n")
para_list = re_matching.cut_para(text)
audio_list = []
if not cut_by_sent:
for p in para_list:
audio = infer(
p,
sdp_ratio=sdp_ratio,
noise_scale=noise_scale,
noise_scale_w=noise_scale_w,
length_scale=length_scale,
sid=speaker,
language=language,
hps=hps,
net_g=net_g,
device=device,
)
audio_list.append(audio)
silence = np.zeros((int)(44100 * interval_between_para))
audio_list.append(silence)
else:
for p in para_list:
sent_list = re_matching.cut_sent(p)
for s in sent_list:
audio = infer(
s,
sdp_ratio=sdp_ratio,
noise_scale=noise_scale,
noise_scale_w=noise_scale_w,
length_scale=length_scale,
sid=speaker,
language=language,
hps=hps,
net_g=net_g,
device=device,
)
audio_list.append(audio)
silence = np.zeros((int)(44100 * interval_between_sent))
audio_list.append(silence)
if (interval_between_para - interval_between_sent) > 0:
silence = np.zeros(
(int)(44100 * (interval_between_para - interval_between_sent))
)
audio_list.append(silence)
audio_concat = np.concatenate(audio_list)
return ("Success", (44100, audio_concat))
def tts_fn(
text: str,
speaker,
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
language,
):
audio_list = []
if language == "mix":
bool_valid, str_valid = re_matching.validate_text(text)
if not bool_valid:
return str_valid, (
hps.data.sampling_rate,
np.concatenate([np.zeros(hps.data.sampling_rate // 2)]),
)
result = re_matching.text_matching(text)
for one in result:
_speaker = one.pop()
for lang, content in one:
audio_list.extend(
generate_audio(
content.split("|"),
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
_speaker + "_" + lang.lower(),
lang,
)
)
else:
audio_list.extend(
generate_audio(
text.split("|"),
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
speaker,
language,
)
)
audio_concat = np.concatenate(audio_list)
return "Success", (hps.data.sampling_rate, audio_concat)
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument(
"-m", "--model", default="./logs/as/G_8000.pth", help="path of your model"
)
parser.add_argument(
"-c",
"--config",
default="./configs/config.json",
help="path of your config file",
)
parser.add_argument(
"--share", default=False, help="make link public", action="store_true"
)
parser.add_argument(
"-d", "--debug", action="store_true", help="enable DEBUG-LEVEL log"
)
args = parser.parse_args()
if args.debug:
if config.webui_config.debug:
logger.info("Enable DEBUG-LEVEL log")
logging.basicConfig(level=logging.DEBUG)
hps = utils.get_hparams_from_file(args.config)
device = (
"cuda:0"
if torch.cuda.is_available()
else (
"mps"
if sys.platform == "darwin" and torch.backends.mps.is_available()
else "cpu"
)
hps = utils.get_hparams_from_file(config.webui_config.config_path)
# 若config.json中未指定版本则默认为最新版本
version = hps.version if hasattr(hps, "version") else latest_version
net_g = get_net_g(
model_path=config.webui_config.model, version=version, device=device, hps=hps
)
net_g = SynthesizerTrn(
len(symbols),
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_g.eval()
_ = utils.load_checkpoint(args.model, net_g, None, skip_optimizer=True)
speaker_ids = hps.data.spk2id
speakers = list(speaker_ids.keys())
languages = ["ZH", "JP"]
languages = ["ZH", "JP", "EN", "mix"]
with gr.Blocks() as app:
with gr.Row():
with gr.Column():
text = gr.TextArea(
label="Text",
placeholder="Input Text Here",
value="吃葡萄不吐葡萄皮，不吃葡萄倒吐葡萄皮。",
label="输入文本内容",
placeholder="""
如果你选择语言为\'mix\'，必须按照格式输入，否则报错:
格式举例(zh是中文，jp是日语，不区分大小写；说话人举例:gongzi):
[说话人1]<zh>你好，こんにちは！ <jp>こんにちは，世界。
[说话人2]<zh>你好吗？<jp>元気ですか？
[说话人3]<zh>谢谢。<jp>どういたしまして。
...
另外，所有的语言选项都可以用'|'分割长段实现分句生成。
""",
)
trans = gr.Button("中翻日", variant="primary")
slicer = gr.Button("快速切分", variant="primary")
speaker = gr.Dropdown(
choices=speakers, value=speakers[0], label="Speaker"
choices=speakers, value=speakers[0], label="选择说话人"
)
sdp_ratio = gr.Slider(
minimum=0, maximum=1, value=0.2, step=0.1, label="SDP Ratio"
minimum=0, maximum=1, value=0.2, step=0.1, label="SDP/DP混合比"
)
noise_scale = gr.Slider(
minimum=0.1, maximum=2, value=0.6, step=0.1, label="Noise Scale"
minimum=0.1, maximum=2, value=0.6, step=0.1, label="感情"
)
noise_scale_w = gr.Slider(
minimum=0.1, maximum=2, value=0.8, step=0.1, label="Noise Scale W"
minimum=0.1, maximum=2, value=0.8, step=0.1, label="音素长度"
)
length_scale = gr.Slider(
minimum=0.1, maximum=2, value=1, step=0.1, label="Length Scale"
minimum=0.1, maximum=2, value=1.0, step=0.1, label="语速"
)
language = gr.Dropdown(
choices=languages, value=languages[0], label="Language"
choices=languages, value=languages[0], label="选择语言(新增mix混合选项)"
)
btn = gr.Button("Generate!", variant="primary")
btn = gr.Button("生成音频！", variant="primary")
with gr.Column():
text_output = gr.Textbox(label="Message")
audio_output = gr.Audio(label="Output Audio")
with gr.Row():
with gr.Column():
interval_between_sent = gr.Slider(
minimum=0,
maximum=5,
value=0.2,
step=0.1,
label="句间停顿(秒)，勾选按句切分才生效",
)
interval_between_para = gr.Slider(
minimum=0,
maximum=10,
value=1,
step=0.1,
label="段间停顿(秒)，需要大于句间停顿才有效",
)
opt_cut_by_sent = gr.Checkbox(
label="按句切分 在按段落切分的基础上再按句子切分文本"
)
slicer = gr.Button("切分生成", variant="primary")
text_output = gr.Textbox(label="状态信息")
audio_output = gr.Audio(label="输出音频")
# explain_image = gr.Image(
# label="参数解释信息",
# show_label=True,
# show_share_button=False,
# show_download_button=False,
# value=os.path.abspath("./img/参数说明.png"),
# )
btn.click(
tts_fn,
inputs=[
@@ -220,5 +265,28 @@ if __name__ == "__main__":
outputs=[text_output, audio_output],
)
webbrowser.open("http://127.0.0.1:7860")
app.launch(share=args.share)
trans.click(
translate,
inputs=[text],
outputs=[text],
)
slicer.click(
tts_split,
inputs=[
text,
speaker,
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
language,
opt_cut_by_sent,
interval_between_para,
interval_between_sent,
],
outputs=[text_output, audio_output],
)
print("推理页面已开启!")
webbrowser.open(f"http://127.0.0.1:{config.webui_config.port}")
app.launch(share=config.webui_config.share, server_port=config.webui_config.port)