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kevinwang676
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Parent(s):
7a44c08
Update app.py
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app.py
CHANGED
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import
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import
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import os
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import json
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import math
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import requests
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import torch
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from torch import nn
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from torch.nn import functional as F
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from torch.utils.data import DataLoader
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import commons
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import utils
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from data_utils import TextAudioLoader, TextAudioCollate, TextAudioSpeakerLoader, TextAudioSpeakerCollate
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from models import SynthesizerTrn
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from text.symbols import symbols
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from text import text_to_sequence
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import langdetect
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from scipy.io.wavfile import write
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import re
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from scipy import signal
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import gradio as gr
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'''
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from phonemizer.backend.espeak.wrapper import EspeakWrapper
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_ESPEAK_LIBRARY = 'C:\Program Files\eSpeak NG\libespeak-ng.dll'
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EspeakWrapper.set_library(_ESPEAK_LIBRARY)
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'''
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# check device
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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def get_text(text, hps):
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text_norm = text_to_sequence(text, hps.data.text_cleaners)
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text_norm = torch.LongTensor(text_norm)
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return text_norm
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def langdetector(text): # from PolyLangVITS
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try:
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lang = langdetect.detect(text).lower()
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if lang == 'ko':
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return f'[KO]{text}[KO]'
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elif lang == 'ja':
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return f'[JA]{text}[JA]'
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elif lang == 'en':
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return f'[EN]{text}[EN]'
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elif lang == 'zh-cn':
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return f'[ZH]{text}[ZH]'
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else:
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return text
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except Exception as e:
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return text
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def vcss(inputstr): # single
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fltstr = re.sub(r"[\[\]\(\)\{\}]", "", inputstr)
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#fltstr = langdetector(fltstr) #- optional for cjke/cjks type cleaners
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stn_tst = get_text(fltstr, hps)
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speed = 1
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output_dir = 'output'
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sid = 0
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with torch.no_grad():
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x_tst =
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x_tst_lengths = torch.LongTensor([stn_tst.size(0)]).to(device)
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audio = net_g.infer(x_tst, x_tst_lengths, noise_scale=.667, noise_scale_w=0.8, length_scale=1 / speed)[0][
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0, 0].data.cpu().float().numpy()
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write("output.wav", hps.data.sampling_rate, audio)
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return "output.wav"
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"""
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def vcms(inputstr, sid):
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fltstr = re.sub(r"[\[\]\(\)\{\}]", "", inputstr)
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fltstr = langdetector(fltstr)
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stn_tst = get_text(fltstr, hps)
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speed = 1
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output_dir = 'output'
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with torch.no_grad():
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x_tst = stn_tst.to(device).unsqueeze(0)
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x_tst_lengths = torch.LongTensor([stn_tst.size(0)]).to(device)
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sid = torch.LongTensor([sid]).to(device)
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audio = net_g.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=.667, noise_scale_w=0.8, length_scale=1 / speed)[0][
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0, 0].data.cpu().float().numpy()
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write(f'./{output_dir}/output_{sid}.wav', hps.data.sampling_rate, audio)
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print(f'./{output_dir}/output_{sid}.wav Generated!')
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"""
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hps = utils.get_hparams_from_file("./configs/config.json")
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if "use_mel_posterior_encoder" in hps.model.keys() and hps.model.use_mel_posterior_encoder == True:
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print("Using mel posterior encoder for VITS2")
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posterior_channels = 80 # vits2
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hps.data.use_mel_posterior_encoder = True
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else:
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print("Using lin posterior encoder for VITS1")
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posterior_channels = hps.data.filter_length // 2 + 1
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hps.data.use_mel_posterior_encoder = False
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net_g = SynthesizerTrn(
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len(symbols),
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posterior_channels,
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hps.train.segment_size // hps.data.hop_length,
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# n_speakers=hps.data.n_speakers, #- for multi speaker
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**hps.model).to(device)
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_ = net_g.eval()
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_ = utils.load_checkpoint("./logs/G_6100.pth", net_g, None)
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return vcss(text)
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gr.Markdown("### <center>🌊 - 更多精彩应用,敬请关注[滔滔AI](http://www.talktalkai.com);滔滔AI,为爱滔滔!💕</center>")
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inp1 = gr.Textbox(label="请在这里填写您想合成的文本", placeholder="想说却还没说的 还很多...", lines=3)
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btn1 = gr.Button("3.一键推理", variant="primary")
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with gr.Column():
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out1 = gr.Audio(type="filepath", label="为您合成的神里绫华语音")
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</p>
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</div>
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''')
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app.launch(show_error=True)
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import argparse
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import gradio as gr
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from gradio import components
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import os
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import torch
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import commons
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import utils
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from models import SynthesizerTrn
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from text.symbols import symbols
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from text import text_to_sequence
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from scipy.io.wavfile import write
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def get_text(text, hps):
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text_norm = text_to_sequence(text, hps.data.text_cleaners)
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text_norm = torch.LongTensor(text_norm)
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return text_norm
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def tts(model_path, config_path, text):
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model_path = "./logs/G_23300.pth"
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config_path = "./configs/config.json"
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hps = utils.get_hparams_from_file(config_path)
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if "use_mel_posterior_encoder" in hps.model.keys() and hps.model.use_mel_posterior_encoder == True:
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posterior_channels = 80
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hps.data.use_mel_posterior_encoder = True
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else:
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posterior_channels = hps.data.filter_length // 2 + 1
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hps.data.use_mel_posterior_encoder = False
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net_g = SynthesizerTrn(
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len(symbols),
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posterior_channels,
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hps.train.segment_size // hps.data.hop_length,
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**hps.model).cuda()
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_ = net_g.eval()
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_ = utils.load_checkpoint(model_path, net_g, None)
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stn_tst = get_text(text, hps)
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x_tst = stn_tst.cuda().unsqueeze(0)
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x_tst_lengths = torch.LongTensor([stn_tst.size(0)]).cuda()
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with torch.no_grad():
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audio = net_g.infer(x_tst, x_tst_lengths, noise_scale=.667, noise_scale_w=0.8, length_scale=1)[0][0,0].data.cpu().float().numpy()
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output_wav_path = "output.wav"
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write(output_wav_path, hps.data.sampling_rate, audio)
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return output_wav_path
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument('--model_path', type=str, default="./logs/G_23300.pth", help='Path to the model file.')
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parser.add_argument('--config_path', type=str, default="./configs/config.json", help='Path to the config file.')
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args = parser.parse_args()
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model_files = [f for f in os.listdir('./logs/') if f.endswith('.pth')]
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model_files.sort(key=lambda x: int(x.split('_')[-1].split('.')[0]), reverse=True)
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config_files = [f for f in os.listdir('./configs/') if f.endswith('.json')]
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default_model_file = args.model_path if args.model_path else (model_files[0] if model_files else None)
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default_config_file = args.config_path if args.config_path else 'config.json'
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gr.Interface(
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fn=tts,
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inputs=components.Textbox(label="Text Input"),
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outputs=components.Audio(type='filepath', label="Generated Speech"),
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live=False
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).launch(show_error=True)
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