diff --git a/webui.py b/webui.py new file mode 100644 index 0000000..6d5c3e2 --- /dev/null +++ b/webui.py @@ -0,0 +1,95 @@ +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 webbrowser + +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) + + 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): + 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) + 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() + return audio + +def tts_fn(text, speaker, sdp_ratio, noise_scale, noise_scale_w, length_scale): + 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) + return "Success", (hps.data.sampling_rate, audio) + + +if __name__ == "__main__": + parser = argparse.ArgumentParser() + parser.add_argument("-m", "--model", default="./logs/as/G_8000.pth", help="path of your model") + parser.add_argument("-c", "--config", default="./configs/config.json", help="path of your config file") + parser.add_argument("--share", default=False, help="make link public") + + args = parser.parse_args() + hps = utils.get_hparams_from_file(args.config) + + 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() + + _ = utils.load_checkpoint(args.model, net_g, None,skip_optimizer=True) + + speaker_ids = hps.data.spk2id + speakers = list(speaker_ids.keys()) + app = gr.Blocks() + with app: + with gr.Row(): + with gr.Column(): + text = gr.TextArea(label="Text", placeholder="Input Text Here", + value="吃葡萄不吐葡萄皮,不吃葡萄倒吐葡萄皮。") + speaker = gr.Dropdown(choices=speakers, value=speakers[0], label='Speaker') + sdp_ratio = gr.Slider(minimum=0.1, maximum=2, value=0.2, step=0.1, label='SDP Ratio') + noise_scale = gr.Slider(minimum=0.1, maximum=2, value=0.5, step=0.1, label='Noise Scale') + noise_scale_w = gr.Slider(minimum=0.1, maximum=2, value=0.6, step=0.1, label='Noise Scale W') + length_scale = gr.Slider(minimum=0.1, maximum=2, value=1.2, step=0.1, label='Length Scale') + btn = gr.Button("Generate!", variant="primary") + with gr.Column(): + text_output = gr.Textbox(label="Message") + audio_output = gr.Audio(label="Output Audio") + + btn.click(tts_fn, + inputs=[text, speaker, sdp_ratio, noise_scale, noise_scale_w, length_scale], + outputs=[text_output, audio_output]) + + webbrowser.open("http://127.0.0.1:7860") + app.launch(share=args.share)