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import os |
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import numpy as np |
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import torch |
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from torch import no_grad, LongTensor |
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import argparse |
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import commons |
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from mel_processing import spectrogram_torch |
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import utils |
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from models import SynthesizerTrn |
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import gradio as gr |
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import librosa |
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import webbrowser |
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import time |
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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 cleaned_text_to_sequence,_symbol_to_id, get_bert |
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from text.cleaner import clean_text |
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from scipy.io import wavfile |
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device = "cuda:0" if torch.cuda.is_available() else "cpu" |
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import logging |
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logging.getLogger("PIL").setLevel(logging.WARNING) |
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logging.getLogger("urllib3").setLevel(logging.WARNING) |
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logging.getLogger("markdown_it").setLevel(logging.WARNING) |
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logging.getLogger("httpx").setLevel(logging.WARNING) |
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logging.getLogger("asyncio").setLevel(logging.WARNING) |
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language_marks = { |
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"简体中文": "[ZH]", |
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} |
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lang = ['简体中文'] |
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def get_text(text, language_str, hps): |
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norm_text, phone, tone, word2ph = clean_text(text, language_str) |
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print([f"{p}{t}" for p, t in zip(phone, tone)]) |
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phone, tone, language = cleaned_text_to_sequence(phone, tone, language_str) |
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if hps.data.add_blank: |
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phone = commons.intersperse(phone, 0) |
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tone = commons.intersperse(tone, 0) |
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language = commons.intersperse(language, 0) |
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for i in range(len(word2ph)): |
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word2ph[i] = word2ph[i] * 2 |
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word2ph[0] += 1 |
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bert = get_bert(norm_text, word2ph, language_str) |
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assert bert.shape[-1] == len(phone) |
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phone = torch.LongTensor(phone) |
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tone = torch.LongTensor(tone) |
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language = torch.LongTensor(language) |
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return bert, phone, tone, language |
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''' |
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def create_tts_fn(model, hps, speaker_ids): |
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def tts_fn(text, speaker, language, speed): |
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if language is not None: |
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text = language_marks[language] + text + language_marks[language] |
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speaker_id = speaker_ids[speaker] |
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stn_tst = get_text(text, hps, False) |
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with no_grad(): |
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x_tst = stn_tst.unsqueeze(0).to(device) |
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x_tst_lengths = LongTensor([stn_tst.size(0)]).to(device) |
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sid = LongTensor([speaker_id]).to(device) |
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audio = model.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=.667, noise_scale_w=0.8, |
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length_scale=1.0 / speed)[0][0, 0].data.cpu().float().numpy() |
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del stn_tst, x_tst, x_tst_lengths, sid |
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return "Success", (hps.data.sampling_rate, audio) |
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return tts_fn |
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''' |
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dev='cuda' |
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def infer(text, sdp_ratio, noise_scale, noise_scale_w,length_scale,sid): |
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bert, phones, tones, lang_ids = get_text(text,"ZH", hps,) |
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print(sid) |
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with torch.no_grad(): |
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x_tst=phones.to(dev).unsqueeze(0) |
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tones=tones.to(dev).unsqueeze(0) |
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lang_ids=lang_ids.to(dev).unsqueeze(0) |
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bert = bert.to(dev).unsqueeze(0) |
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x_tst_lengths = torch.LongTensor([phones.size(0)]).to(dev) |
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del phones |
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speakers = torch.LongTensor([hps.data.spk2id[sid]]).to(dev) |
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audio = net_g.infer(x_tst, x_tst_lengths, speakers, tones, lang_ids,bert, sdp_ratio=sdp_ratio |
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, noise_scale=noise_scale, noise_scale_w=noise_scale_w, length_scale=length_scale)[0][0,0].data.cpu().float().numpy() |
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del x_tst, tones, lang_ids, bert, x_tst_lengths, speakers |
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return "Success",(hps.data.sampling_rate, audio) |
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if __name__ == "__main__": |
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parser = argparse.ArgumentParser() |
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parser.add_argument("--model_dir", default="./G_latest.pth", help="directory to your fine-tuned model") |
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parser.add_argument("--config_dir", default="./configs\config.json", help="directory to your model config file") |
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parser.add_argument("--share", default=False, help="make link public (used in colab)") |
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args = parser.parse_args() |
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hps = utils.get_hparams_from_file(args.config_dir) |
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net_g = SynthesizerTrn( |
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len(symbols), |
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hps.data.filter_length // 2 + 1, |
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hps.train.segment_size // hps.data.hop_length, |
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n_speakers=hps.data.n_speakers, |
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**hps.model).to(dev) |
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_ = net_g.eval() |
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_ = utils.load_checkpoint(args.model_dir, net_g, None,skip_optimizer=True) |
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speaker_ids = hps.data.spk2id |
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speakers = list(hps.data.spk2id.keys()) |
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#inf = infer(net_g, hps, speaker_ids) |
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app = gr.Blocks() |
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with app: |
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with gr.Tab("Text-to-Speech"): |
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with gr.Row(): |
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with gr.Column(): |
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textbox = gr.TextArea(label="Text", |
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placeholder="Type your sentence here", |
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value="生活就像海洋,只有意志坚强的人,才能到达彼岸。", elem_id=f"tts-input") |
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# select character |
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char_dropdown = gr.Dropdown(choices=speakers, value=speakers[0], label='character') |
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language_dropdown = gr.Dropdown(choices=lang, value=lang[0], label='language') |
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sdp_ratio = gr.Slider(minimum=0.1, maximum=0.9, value=0.2, step=0.1, |
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label='SDP/DP混合比-语调方差') |
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noise_scale = gr.Slider(minimum=0.1, maximum=1.5, value=0.5, step=0.1, |
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label='noise/感情变化') |
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noise_scale_w = gr.Slider(minimum=0.1, maximum=1.4, value=0.9, step=0.1, |
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label='noisew/音节发音长度变化') |
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length_scale = gr.Slider(minimum=0.1, maximum=2, value=1.0, step=0.1, |
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label='length/语速') |
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with gr.Column(): |
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text_output = gr.Textbox(label="Message") |
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audio_output = gr.Audio(label="Output Audio", elem_id="tts-audio") |
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btn = gr.Button("Generate!") |
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btn.click(infer, |
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inputs=[textbox,sdp_ratio,noise_scale,noise_scale_w,length_scale,char_dropdown], |
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outputs=[text_output, audio_output]) |
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webbrowser.open("http://127.0.0.1:7860") |
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app.launch(share=args.share) |
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