V-Voice / app.py
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import sys, os
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
import gradio as gr
import soundfile as sf
from datetime import datetime
import pytz
net_g = None
models = {
"V1": "./MODELS/v1100.pth",
"V2": "./MODELS/180_3000.pth",
"V3":"./MODELS/v3_8000.pth"
}
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)
del word2ph
assert bert.shape[-1] == len(phone)
phone = torch.LongTensor(phone)
tone = torch.LongTensor(tone)
language = torch.LongTensor(language)
return bert, phone, tone, language
def infer(text, sdp_ratio, noise_scale, noise_scale_w, length_scale, sid, model_dir):
global net_g
bert, phones, tones, lang_ids = get_text(text, "ZH", 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)
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, 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
sf.write("tmp.wav", audio, 44100)
return audio
tz = pytz.timezone('Asia/Shanghai')
def convert_wav_to_mp3(wav_file):
global tz
now = datetime.now(tz).strftime('%m%d%H%M%S')
os.makedirs('out', exist_ok=True)
output_path_mp3 = os.path.join('out', f"{now}.mp3")
renamed_input_path = os.path.join('in', f"in.wav")
os.makedirs('in', exist_ok=True)
os.rename(wav_file.name, renamed_input_path)
command = ["ffmpeg", "-i", renamed_input_path, "-acodec", "libmp3lame", "-y", output_path_mp3]
os.system(" ".join(command))
return output_path_mp3
def tts_generator(text, sdp_ratio, noise_scale, noise_scale_w, length_scale, model):
global net_g,speakers
model_path = models[model]
net_g, _, _, _ = utils.load_checkpoint(model_path, net_g, None, skip_optimizer=True)
text = text[:500]
try:
with torch.no_grad():
audio = infer(text, sdp_ratio=sdp_ratio, noise_scale=noise_scale, noise_scale_w=noise_scale_w, length_scale=length_scale, sid=speaker,model_dir=model)
with open('tmp.wav', 'rb') as wav_file:
mp3 = convert_wav_to_mp3(wav_file)
return "生成语音成功", (hps.data.sampling_rate, audio), mp3
except Exception as e:
return "生成语音失败:" + str(e), None, None
if __name__ == "__main__":
hps = utils.get_hparams_from_file("./configs/config.json")
device = "cuda:0" if torch.cuda.is_available() else "cpu"
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()
speaker_ids = hps.data.spk2id
speaker = list(speaker_ids.keys())[0]
theme='remilia/Ghostly'
with gr.Blocks(theme=theme) as app:
with gr.Row():
with gr.Column():
gr.Markdown("""**测试用**""")
text = gr.TextArea(label="✨输入需要生成语音的文字", placeholder="输入文字",
value="漩涡帮可不是吃素的,我是碰巧路过听人说,他们要整一个全金属和尚",
info="使用huggingface的免费CPU进行推理,因此速度不快,最多生成500字,多余的会被忽略。字数越多越耗时,请耐心等待,只会说中文",
)
model = gr.Radio(choices=list(models.keys()), value=list(models.keys())[0], label='📢音声模型')
with gr.Accordion(label="💡展开设置生成参数", open=False):
sdp_ratio = gr.Slider(minimum=0, maximum=1, value=0.2, step=0.01, label='SDP/DP混合比',info='可控制一定程度的语调变化')
noise_scale = gr.Slider(minimum=0.1, maximum=1.5, value=0.5, step=0.01, label='感情变化')
noise_scale_w = gr.Slider(minimum=0.1, maximum=1.4, value=0.9, step=0.01, label='音节长度')
length_scale = gr.Slider(minimum=0.1, maximum=2, value=1, step=0.01, label='生成语音总长度',info='数值越大,语速越慢')
btn = gr.Button("🪄生成", variant="primary")
with gr.Column():
audio_output = gr.Audio(label="🔊试听")
MP3_output = gr.File(label="💾下载")
text_output = gr.Textbox(label="❗调试信息")
gr.Markdown("""
""")
btn.click(
tts_generator,
inputs=[text, sdp_ratio, noise_scale, noise_scale_w, length_scale, model],
outputs=[text_output, audio_output,MP3_output]
)
gr.Examples(
fn=tts_generator,
examples=[
[
"我?当警察,上次我说这话的时候才六岁"
],
[
"但对我来说,回忆中的夜之城反而笼罩在一种暖暖淡淡的,有奶油质感的颜色中。"
],
[
"与我打卡过的北京其他几家社区图书馆一样,环境那叫一个整洁优雅,工作日那叫一个人烟稀少。书虽不多,但好书不少,而且崭新得烫手。"
],
[
"《神笔狗良》冒险解谜涂色游戏,对小朋友来说或许有点幼稚但对我来说刚刚好!"
],
[
"不知道有没有使用过不同读取速度内存卡的姐妹,游戏加载和运行速度会差很多吗?"
]
,
],
inputs=[text],
outputs=[audio_output]
)
#gr.HTML('''<div align=center><img id="visitor-badge" alt="visitor badge" src="https://visitor-badge.laobi.icu/badge?page_id=Ailyth/DLM" /></div>''')
app.launch(show_error=True)