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import argparse
import os
from pathlib import Path
import logging
import re_matching
logging.getLogger("numba").setLevel(logging.WARNING)
logging.getLogger("markdown_it").setLevel(logging.WARNING)
logging.getLogger("urllib3").setLevel(logging.WARNING)
logging.getLogger("matplotlib").setLevel(logging.WARNING)
logging.basicConfig(
level=logging.INFO, format="| %(name)s | %(levelname)s | %(message)s"
)
logger = logging.getLogger(__name__)
import librosa
import numpy as np
import torch
import torch.nn as nn
from torch.utils.data import Dataset
from torch.utils.data import DataLoader, Dataset
from tqdm import tqdm
from clap_wrapper import get_clap_audio_feature, get_clap_text_feature
import uuid
from flask import Flask, request, jsonify, render_template_string
from flask_cors import CORS
import gradio as gr
import utils
from config import config
import torch
import commons
from text import cleaned_text_to_sequence, get_bert
from text.cleaner import clean_text
import utils
from models import SynthesizerTrn
from text.symbols import symbols
import sys
from scipy.io.wavfile import write
from threading import Thread
net_g = None
device = (
"cuda:0"
if torch.cuda.is_available()
else (
"mps"
if sys.platform == "darwin" and torch.backends.mps.is_available()
else "cpu"
)
)
#device = "cpu"
BandList = {
"PoppinParty":["香澄","有咲","たえ","りみ","沙綾"],
"Afterglow":["蘭","モカ","ひまり","巴","つぐみ"],
"HelloHappyWorld":["こころ","美咲","薫","花音","はぐみ"],
"PastelPalettes":["彩","日菜","千聖","イヴ","麻弥"],
"Roselia":["友希那","紗夜","リサ","燐子","あこ"],
"RaiseASuilen":["レイヤ","ロック","ますき","チュチュ","パレオ"],
"Morfonica":["ましろ","瑠唯","つくし","七深","透子"],
"MyGo":["燈","愛音","そよ","立希","楽奈"],
"AveMujica":["祥子","睦","海鈴","にゃむ","初華"],
"圣翔音乐学园":["華戀","光","香子","雙葉","真晝","純那","克洛迪娜","真矢","奈奈"],
"凛明馆女子学校":["珠緒","壘","文","悠悠子","一愛"],
"弗隆提亚艺术学校":["艾露","艾露露","菈樂菲","司","靜羽"],
"西克菲尔特音乐学院":["晶","未知留","八千代","栞","美帆"]
}
def get_net_g(model_path: str, device: str, 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(model_path, net_g, None, skip_optimizer=True)
return net_g
def get_text(text, language_str, hps, device, style_text=None, style_weight=0.7):
style_text = None if style_text == "" else style_text
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_ori = get_bert(
norm_text, word2ph, language_str, device, style_text, style_weight
)
del word2ph
assert bert_ori.shape[-1] == len(phone), phone
if language_str == "ZH":
bert = bert_ori
ja_bert = torch.randn(1024, len(phone))
en_bert = torch.randn(1024, len(phone))
elif language_str == "JP":
bert = torch.randn(1024, len(phone))
ja_bert = bert_ori
en_bert = torch.randn(1024, len(phone))
elif language_str == "EN":
bert = torch.randn(1024, len(phone))
ja_bert = torch.randn(1024, len(phone))
en_bert = bert_ori
else:
raise ValueError("language_str should be ZH, JP or EN")
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, en_bert, phone, tone, language
def infer(
text,
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
sid,
style_text=None,
style_weight=0.7,
):
language= 'JP' if is_japanese(text) else 'ZH'
bert, ja_bert, en_bert, phones, tones, lang_ids = get_text(
text,
language,
hps,
device,
style_text=style_text,
style_weight=style_weight,
)
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)
en_bert = en_bert.to(device).unsqueeze(0)
x_tst_lengths = torch.LongTensor([phones.size(0)]).to(device)
# emo = emo.to(device).unsqueeze(0)
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,
en_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,
ja_bert,
en_bert,
) # , emo
if torch.cuda.is_available():
torch.cuda.empty_cache()
return (hps.data.sampling_rate,gr.processing_utils.convert_to_16_bit_wav(audio))
def inferAPI(
text,
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
sid,
style_text=None,
style_weight=0.7,
):
language= 'JP' if is_japanese(text) else 'ZH'
bert, ja_bert, en_bert, phones, tones, lang_ids = get_text(
text,
language,
hps,
device,
style_text=style_text,
style_weight=style_weight,
)
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)
en_bert = en_bert.to(device).unsqueeze(0)
x_tst_lengths = torch.LongTensor([phones.size(0)]).to(device)
# emo = emo.to(device).unsqueeze(0)
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,
en_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,
ja_bert,
en_bert,
) # , emo
if torch.cuda.is_available():
torch.cuda.empty_cache()
unique_filename = f"temp{uuid.uuid4()}.wav"
write(unique_filename, 44100, audio)
return unique_filename
def is_japanese(string):
for ch in string:
if ord(ch) > 0x3040 and ord(ch) < 0x30FF:
return True
return False
def loadmodel(model):
try:
_ = net_g.eval()
_ = utils.load_checkpoint(model, net_g, None, skip_optimizer=True)
return "success"
except:
return "error"
Flaskapp = Flask(__name__)
CORS(Flaskapp)
@Flaskapp.route('/')
@Flaskapp.route('/')
def tts():
global last_text, last_model
speaker = request.args.get('speaker')
sdp_ratio = float(request.args.get('sdp_ratio', 0.2))
noise_scale = float(request.args.get('noise_scale', 0.6))
noise_scale_w = float(request.args.get('noise_scale_w', 0.8))
length_scale = float(request.args.get('length_scale', 1))
style_weight = float(request.args.get('style_weight', 0.7))
style_text = request.args.get('style_text', 'happy')
text = request.args.get('text')
is_chat = request.args.get('is_chat', 'false').lower() == 'true'
model = request.args.get('model',modelPaths[-1])
if not speaker or not text:
return render_template_string("""
<!DOCTYPE html>
<html>
<head>
<title>TTS API Documentation</title>
</head>
<body>
<iframe src="http://127.0.0.1:7860" style="width:100%; height:100vh; border:none;"></iframe>
</body>
</html>
""")
if model != last_model:
unique_filename = loadmodel(model)
last_model = model
if is_chat and text == last_text:
# Generate 1 second of silence and return
unique_filename = 'blank.wav'
silence = np.zeros(44100, dtype=np.int16)
write(unique_filename , 44100, silence)
else:
last_text = text
unique_filename = inferAPI(text, sdp_ratio=sdp_ratio, noise_scale=noise_scale, noise_scale_w=noise_scale_w, length_scale=length_scale,sid = speaker, style_text=style_text, style_weight=style_weight)
with open(unique_filename ,'rb') as bit:
wav_bytes = bit.read()
os.remove(unique_filename)
headers = {
'Content-Type': 'audio/wav',
'Text': unique_filename .encode('utf-8')}
return wav_bytes, 200, headers
def gradio_interface():
return app.launch(share=True)
if __name__ == "__main__":
languages = [ "Auto", "ZH", "JP"]
modelPaths = []
for dirpath, dirnames, filenames in os.walk('Data/Data/V23/models/'):
for filename in filenames:
modelPaths.append(os.path.join(dirpath, filename))
hps = utils.get_hparams_from_file('Data/Data/V23/configs/config.json')
net_g = get_net_g(
model_path=modelPaths[-1], device=device, hps=hps
)
speaker_ids = hps.data.spk2id
speakers = list(speaker_ids.keys())
last_text = ""
last_model = modelPaths[-1]
with gr.Blocks() as app:
for band in BandList:
with gr.TabItem(band):
for name in BandList[band]:
with gr.TabItem(name):
with gr.Row():
with gr.Column():
with gr.Row():
gr.Markdown(
'<div align="center">'
f'<img style="width:auto;height:400px;" src="https://mahiruoshi-bangdream-bert-vits2.hf.space/file/image/{name}.png">'
'</div>'
)
length_scale = gr.Slider(
minimum=0.1, maximum=2, value=1, step=0.01, label="语速调节"
)
with gr.Accordion(label="参数设定", open=False):
sdp_ratio = gr.Slider(
minimum=0, maximum=1, value=0.5, step=0.01, label="SDP/DP混合比"
)
noise_scale = gr.Slider(
minimum=0.1, maximum=2, value=0.6, step=0.01, label="感情调节"
)
noise_scale_w = gr.Slider(
minimum=0.1, maximum=2, value=0.667, step=0.01, label="音素长度"
)
speaker = gr.Dropdown(
choices=speakers, value=name, label="说话人"
)
with gr.Accordion(label="切换模型", open=False):
modelstrs = gr.Dropdown(label = "模型", choices = modelPaths, value = modelPaths[0], type = "value")
btnMod = gr.Button("载入模型")
statusa = gr.TextArea()
btnMod.click(loadmodel, inputs=[modelstrs], outputs = [statusa])
with gr.Column():
text = gr.TextArea(
label="输入纯日语或者中文",
placeholder="输入纯日语或者中文",
value="为什么要演奏春日影!",
)
style_text = gr.Textbox(label="辅助文本")
style_weight = gr.Slider(
minimum=0,
maximum=1,
value=0.7,
step=0.1,
label="Weight",
info="主文本和辅助文本的bert混合比率,0表示仅主文本,1表示仅辅助文本",
)
btn = gr.Button("点击生成", variant="primary")
audio_output = gr.Audio(label="Output Audio")
'''
btntran = gr.Button("快速中翻日")
translateResult = gr.TextArea("从这复制翻译后的文本")
btntran.click(translate, inputs=[text], outputs = [translateResult])
'''
btn.click(
infer,
inputs=[
text,
sdp_ratio,
noise_scale,
noise_scale_w,
length_scale,
speaker,
style_text,
style_weight,
],
outputs=[audio_output],
)
api_thread = Thread(target=Flaskapp.run, args=("0.0.0.0", 5000))
gradio_thread = Thread(target=gradio_interface)
gradio_thread.start()
print("推理页面已开启!")
api_thread.start()
print("api页面已开启!运行在5000端口")