import re import os import numpy as np import torch from torch import no_grad, LongTensor import argparse import commons from mel_processing import spectrogram_torch import utils from models import SynthesizerTrn import gradio as gr import librosa import webbrowser from text import text_to_sequence, _clean_text device = "cuda:0" if torch.cuda.is_available() else "cpu" language_marks = { "Japanese": "", "日本語": "[JA]", "简体中文": "[ZH]", "English": "[EN]", "Mix": "", } def get_text(text, hps, is_symbol): text_norm = text_to_sequence(text, hps.symbols, [] if is_symbol else hps.data.text_cleaners) if hps.data.add_blank: text_norm = commons.intersperse(text_norm, 0) text_norm = LongTensor(text_norm) return text_norm def create_tts_fn(model, hps, speaker_ids): def tts_fn(text, speaker, language, ns, nsw, speed, is_symbol): if language is not None: text = language_marks[language] + text + language_marks[language] speaker_id = speaker_ids[speaker] stn_tst = get_text(text, hps, is_symbol) with no_grad(): x_tst = stn_tst.unsqueeze(0).to(device) x_tst_lengths = LongTensor([stn_tst.size(0)]).to(device) sid = LongTensor([speaker_id]).to(device) audio = model.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=ns, noise_scale_w=nsw, length_scale=1.0 / speed)[0][0, 0].data.cpu().float().numpy() del stn_tst, x_tst, x_tst_lengths, sid return "Success", (hps.data.sampling_rate, audio) return tts_fn def create_vc_fn(model, hps, speaker_ids): def vc_fn(original_speaker, target_speaker, record_audio, upload_audio): input_audio = record_audio if record_audio is not None else upload_audio if input_audio is None: return "You need to record or upload an audio", None sampling_rate, audio = input_audio original_speaker_id = speaker_ids[original_speaker] target_speaker_id = speaker_ids[target_speaker] audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32) if len(audio.shape) > 1: audio = librosa.to_mono(audio.transpose(1, 0)) if sampling_rate != hps.data.sampling_rate: audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=hps.data.sampling_rate) with no_grad(): y = torch.FloatTensor(audio) y = y / max(-y.min(), y.max()) / 0.99 y = y.to(device) y = y.unsqueeze(0) spec = spectrogram_torch(y, hps.data.filter_length, hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length, center=False).to(device) spec_lengths = LongTensor([spec.size(-1)]).to(device) sid_src = LongTensor([original_speaker_id]).to(device) sid_tgt = LongTensor([target_speaker_id]).to(device) audio = model.voice_conversion(spec, spec_lengths, sid_src=sid_src, sid_tgt=sid_tgt)[0][ 0, 0].data.cpu().float().numpy() del y, spec, spec_lengths, sid_src, sid_tgt return "Success", (hps.data.sampling_rate, audio) return vc_fn def get_text(text, hps, is_symbol): text_norm = text_to_sequence(text, hps.symbols, [] if is_symbol else hps.data.text_cleaners) if hps.data.add_blank: text_norm = commons.intersperse(text_norm, 0) text_norm = LongTensor(text_norm) return text_norm def create_to_symbol_fn(hps): def to_symbol_fn(is_symbol_input, input_text, temp_text): return (_clean_text(input_text, hps.data.text_cleaners), input_text) if is_symbol_input \ else (temp_text, temp_text) return to_symbol_fn models_info = [ { "languages": ['日本語', '简体中文', 'English', 'Mix'], "description": """ 这个模型包含公主连结Re:Dive的161名角色,能合成中日英三语。\n\n 若需要在同一个句子中混合多种语言,使用相应的语言标记包裹句子。 (日语用[JA], 中文用[ZH], 英文用[EN]),参考Examples中的示例。 """, "model_path": "./models/G_9700.pth", "config_path": "./configs/modified_finetune_speaker.json", "examples": [['大切な人の誕生日を祝えるって、すごく幸せなことなんですよね。', '佩可莉姆', '日本語', 1, False], ['その…この制服、どうですか?', '栞', '日本語', 1, False], ['你们全都给我让开!敢挡路的家伙,我统统斩了!', '矛依未', '简体中文', 1, False], ['Can you tell me how much the shirt is?', '美咲', 'English', 1, False], ['[EN]Excuse me?[EN][JA]お帰りなさい,お兄様![JA]', '咲恋(夏日)', 'Mix', 1, False]], } ] models_tts = [] models_vc = [] if __name__ == "__main__": parser = argparse.ArgumentParser() parser.add_argument("--share", action="store_true", default=False, help="share gradio app") args = parser.parse_args() categories = ["Princess Connect! Re:Dive"] others = { "Umamusume": "https://huggingface.co/spaces/FrankZxShen/vits-fast-finetuning-umamusume", "Blue Archive": "https://huggingface.co/spaces/FrankZxShen/vits-fast-fineturning-models-ba", "PCR so-vits-svc(歌声语音转换)": "https://huggingface.co/spaces/FrankZxShen/so-vits-svc-models-pcr", } for info in models_info: lang = info['languages'] examples = info['examples'] config_path = info['config_path'] model_path = info['model_path'] description = info['description'] hps = utils.get_hparams_from_file(config_path) net_g = SynthesizerTrn( len(hps.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(model_path, net_g, None) speaker_ids = hps.speakers speakers = list(hps.speakers.keys()) models_tts.append((description, speakers, lang, examples, hps.symbols, create_tts_fn(net_g, hps, speaker_ids), create_to_symbol_fn(hps))) models_vc.append((description, speakers, create_vc_fn(net_g, hps, speaker_ids))) app = gr.Blocks() with app: gr.Markdown( "#