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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 | |
tz = pytz.timezone('Asia/Shanghai') | |
net_g = None | |
models = { | |
"Mellowdear": "./MODELS/adorabledarling.pth", | |
"MistyNikki": "./MODELS/nikki9400.pth", | |
"Silverleg": "./MODELS/J8900.pth", | |
"Umemura": "./MODELS/take2.pth", | |
"LucidMoon": "./MODELS/lucid.pth", | |
"Rrabbitt": "./MODELS/rabbit4900.pth", | |
"VivaciousViolet": "./MODELS/vv.pth", | |
"AlluWin": "./MODELS/AW.pth", | |
"ImmenseStar": "./MODELS/endlessstar.pth", | |
"LightHammer": "./MODELS/hammer.pth", | |
"ArasakaAI": "Arasaka.pth", | |
"DLM": "./MODELS/DLM.pth", | |
"BadGirlDLM": "./MODELS/BG1300.pth", | |
"BadBoyDLM": "./MODELS/BAD1100.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 | |
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,tz | |
now = datetime.now(tz).strftime('%m-%d %H:%M:%S') | |
model_path = models[model] | |
net_g, _, _, _ = utils.load_checkpoint(model_path, net_g, None, skip_optimizer=True) | |
print(now+text) | |
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.Column(): | |
with gr.Column(): | |
gr.Markdown(''' | |
**仅供测试用** | |
''') | |
text = gr.TextArea(label="输入需要生成语音的文字", placeholder="输入文字", | |
value="在不在?能不能借给我三百块钱买可乐", | |
info="使用huggingface的免费CPU进行推理,因此速度不快,一次性不要输入超过500汉字。字数越多,生成速度越慢,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.HTML('''<div align=center><img id="visitor-badge" alt="visitor badge" src="https://visitor-badge.laobi.icu/badge?page_id=Ailyth/DLMP9" /></div>''') | |
app.launch(show_error=True) | |