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import gradio as gr |
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import numpy as np |
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import soundfile as sf |
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from datetime import datetime |
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from time import time as ttime |
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from my_utils import load_audio |
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from transformers import pipeline |
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from text.cleaner import clean_text |
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from feature_extractor import cnhubert |
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from timeit import default_timer as timer |
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from text import cleaned_text_to_sequence |
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from module.models import SynthesizerTrn |
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import os,re,sys,LangSegment,librosa,pdb,torch,pytz |
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from module.mel_processing import spectrogram_torch |
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from transformers.pipelines.audio_utils import ffmpeg_read |
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from transformers import AutoModelForMaskedLM, AutoTokenizer |
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from AR.models.t2s_lightning_module import Text2SemanticLightningModule |
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import logging |
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logging.getLogger("markdown_it").setLevel(logging.ERROR) |
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logging.getLogger("urllib3").setLevel(logging.ERROR) |
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logging.getLogger("httpcore").setLevel(logging.ERROR) |
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logging.getLogger("httpx").setLevel(logging.ERROR) |
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logging.getLogger("asyncio").setLevel(logging.ERROR) |
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logging.getLogger("charset_normalizer").setLevel(logging.ERROR) |
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logging.getLogger("torchaudio._extension").setLevel(logging.ERROR) |
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logging.getLogger("multipart").setLevel(logging.WARNING) |
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if "_CUDA_VISIBLE_DEVICES" in os.environ: |
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os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"] |
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tz = pytz.timezone('Asia/Singapore') |
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device = "cuda" if torch.cuda.is_available() else "cpu" |
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def abs_path(dir): |
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global_dir = os.path.dirname(os.path.abspath(sys.argv[0])) |
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return(os.path.join(global_dir, dir)) |
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gpt_path = abs_path("MODELS/22/22.ckpt") |
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sovits_path=abs_path("MODELS/22/22.pth") |
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cnhubert_base_path = os.environ.get("cnhubert_base_path", "pretrained_models/chinese-hubert-base") |
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bert_path = os.environ.get("bert_path", "pretrained_models/chinese-roberta-wwm-ext-large") |
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if not os.path.exists(cnhubert_base_path): |
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cnhubert_base_path = "TencentGameMate/chinese-hubert-base" |
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if not os.path.exists(bert_path): |
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bert_path = "hfl/chinese-roberta-wwm-ext-large" |
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cnhubert.cnhubert_base_path = cnhubert_base_path |
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whisper_path = os.environ.get("whisper_path", "pretrained_models/whisper-tiny") |
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if not os.path.exists(whisper_path): |
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whisper_path = "openai/whisper-tiny" |
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pipe = pipeline( |
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task="automatic-speech-recognition", |
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model=whisper_path, |
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chunk_length_s=30, |
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device=device,) |
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is_half = eval( |
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os.environ.get("is_half", "True" if torch.cuda.is_available() else "False") |
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) |
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device = "cuda" if torch.cuda.is_available() else "cpu" |
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tokenizer = AutoTokenizer.from_pretrained(bert_path) |
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bert_model = AutoModelForMaskedLM.from_pretrained(bert_path) |
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if is_half == True: |
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bert_model = bert_model.half().to(device) |
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else: |
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bert_model = bert_model.to(device) |
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def get_bert_feature(text, word2ph): |
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with torch.no_grad(): |
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inputs = tokenizer(text, return_tensors="pt") |
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for i in inputs: |
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inputs[i] = inputs[i].to(device) |
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res = bert_model(**inputs, output_hidden_states=True) |
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res = torch.cat(res["hidden_states"][-3:-2], -1)[0].cpu()[1:-1] |
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assert len(word2ph) == len(text) |
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phone_level_feature = [] |
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for i in range(len(word2ph)): |
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repeat_feature = res[i].repeat(word2ph[i], 1) |
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phone_level_feature.append(repeat_feature) |
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phone_level_feature = torch.cat(phone_level_feature, dim=0) |
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return phone_level_feature.T |
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class DictToAttrRecursive(dict): |
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def __init__(self, input_dict): |
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super().__init__(input_dict) |
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for key, value in input_dict.items(): |
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if isinstance(value, dict): |
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value = DictToAttrRecursive(value) |
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self[key] = value |
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setattr(self, key, value) |
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def __getattr__(self, item): |
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try: |
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return self[item] |
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except KeyError: |
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raise AttributeError(f"Attribute {item} not found") |
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def __setattr__(self, key, value): |
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if isinstance(value, dict): |
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value = DictToAttrRecursive(value) |
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super(DictToAttrRecursive, self).__setitem__(key, value) |
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super().__setattr__(key, value) |
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def __delattr__(self, item): |
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try: |
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del self[item] |
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except KeyError: |
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raise AttributeError(f"Attribute {item} not found") |
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ssl_model = cnhubert.get_model() |
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if is_half == True: |
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ssl_model = ssl_model.half().to(device) |
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else: |
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ssl_model = ssl_model.to(device) |
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def change_sovits_weights(sovits_path): |
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global vq_model, hps |
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dict_s2 = torch.load(sovits_path, map_location="cpu") |
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hps = dict_s2["config"] |
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hps = DictToAttrRecursive(hps) |
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hps.model.semantic_frame_rate = "25hz" |
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vq_model = SynthesizerTrn( |
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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 |
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) |
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if ("pretrained" not in sovits_path): |
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del vq_model.enc_q |
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if is_half == True: |
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vq_model = vq_model.half().to(device) |
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else: |
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vq_model = vq_model.to(device) |
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vq_model.eval() |
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print(vq_model.load_state_dict(dict_s2["weight"], strict=False)) |
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with open("./sweight.txt", "w", encoding="utf-8") as f: |
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f.write(sovits_path) |
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change_sovits_weights(sovits_path) |
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def change_gpt_weights(gpt_path): |
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global hz, max_sec, t2s_model, config |
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hz = 50 |
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dict_s1 = torch.load(gpt_path, map_location="cpu") |
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config = dict_s1["config"] |
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max_sec = config["data"]["max_sec"] |
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t2s_model = Text2SemanticLightningModule(config, "****", is_train=False) |
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t2s_model.load_state_dict(dict_s1["weight"]) |
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if is_half == True: |
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t2s_model = t2s_model.half() |
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t2s_model = t2s_model.to(device) |
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t2s_model.eval() |
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total = sum([param.nelement() for param in t2s_model.parameters()]) |
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print("Number of parameter: %.2fM" % (total / 1e6)) |
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with open("./gweight.txt", "w", encoding="utf-8") as f: f.write(gpt_path) |
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change_gpt_weights(gpt_path) |
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def get_spepc(hps, filename): |
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audio = load_audio(filename, int(hps.data.sampling_rate)) |
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audio = torch.FloatTensor(audio) |
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audio_norm = audio |
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audio_norm = audio_norm.unsqueeze(0) |
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spec = spectrogram_torch( |
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audio_norm, |
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hps.data.filter_length, |
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hps.data.sampling_rate, |
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hps.data.hop_length, |
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hps.data.win_length, |
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center=False, |
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) |
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return spec |
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dict_language = { |
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("中文1"): "all_zh", |
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("English"): "en", |
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("日文1"): "all_ja", |
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("中文"): "zh", |
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("日本語"): "ja", |
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("混合"): "auto", |
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} |
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def splite_en_inf(sentence, language): |
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pattern = re.compile(r'[a-zA-Z ]+') |
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textlist = [] |
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langlist = [] |
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pos = 0 |
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for match in pattern.finditer(sentence): |
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start, end = match.span() |
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if start > pos: |
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textlist.append(sentence[pos:start]) |
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langlist.append(language) |
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textlist.append(sentence[start:end]) |
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langlist.append("en") |
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pos = end |
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if pos < len(sentence): |
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textlist.append(sentence[pos:]) |
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langlist.append(language) |
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for i in range(len(textlist)-1, 0, -1): |
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if re.match(r'^[\W_]+$', textlist[i]): |
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textlist[i-1] += textlist[i] |
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del textlist[i] |
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del langlist[i] |
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i = 0 |
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while i < len(langlist) - 1: |
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if langlist[i] == langlist[i+1]: |
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textlist[i] += textlist[i+1] |
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del textlist[i+1] |
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del langlist[i+1] |
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else: |
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i += 1 |
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return textlist, langlist |
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def clean_text_inf(text, language): |
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formattext = "" |
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language = language.replace("all_","") |
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for tmp in LangSegment.getTexts(text): |
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if language == "ja": |
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if tmp["lang"] == language or tmp["lang"] == "zh": |
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formattext += tmp["text"] + " " |
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continue |
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if tmp["lang"] == language: |
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formattext += tmp["text"] + " " |
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while " " in formattext: |
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formattext = formattext.replace(" ", " ") |
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phones, word2ph, norm_text = clean_text(formattext, language) |
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phones = cleaned_text_to_sequence(phones) |
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return phones, word2ph, norm_text |
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dtype=torch.float16 if is_half == True else torch.float32 |
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def get_bert_inf(phones, word2ph, norm_text, language): |
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language=language.replace("all_","") |
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if language == "zh": |
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bert = get_bert_feature(norm_text, word2ph).to(device) |
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else: |
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bert = torch.zeros( |
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(1024, len(phones)), |
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dtype=torch.float16 if is_half == True else torch.float32, |
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).to(device) |
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return bert |
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def nonen_clean_text_inf(text, language): |
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if(language!="auto"): |
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textlist, langlist = splite_en_inf(text, language) |
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else: |
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textlist=[] |
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langlist=[] |
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for tmp in LangSegment.getTexts(text): |
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langlist.append(tmp["lang"]) |
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textlist.append(tmp["text"]) |
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print(textlist) |
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print(langlist) |
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phones_list = [] |
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word2ph_list = [] |
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norm_text_list = [] |
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for i in range(len(textlist)): |
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lang = langlist[i] |
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phones, word2ph, norm_text = clean_text_inf(textlist[i], lang) |
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phones_list.append(phones) |
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if lang == "zh": |
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word2ph_list.append(word2ph) |
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norm_text_list.append(norm_text) |
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print(word2ph_list) |
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phones = sum(phones_list, []) |
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word2ph = sum(word2ph_list, []) |
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norm_text = ' '.join(norm_text_list) |
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return phones, word2ph, norm_text |
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def nonen_get_bert_inf(text, language): |
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if(language!="auto"): |
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textlist, langlist = splite_en_inf(text, language) |
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else: |
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textlist=[] |
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langlist=[] |
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for tmp in LangSegment.getTexts(text): |
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langlist.append(tmp["lang"]) |
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textlist.append(tmp["text"]) |
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print(textlist) |
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print(langlist) |
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bert_list = [] |
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for i in range(len(textlist)): |
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lang = langlist[i] |
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phones, word2ph, norm_text = clean_text_inf(textlist[i], lang) |
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bert = get_bert_inf(phones, word2ph, norm_text, lang) |
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bert_list.append(bert) |
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bert = torch.cat(bert_list, dim=1) |
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return bert |
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splits = {",", "。", "?", "!", ",", ".", "?", "!", "~", ":", ":", "—", "…", } |
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def get_first(text): |
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pattern = "[" + "".join(re.escape(sep) for sep in splits) + "]" |
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text = re.split(pattern, text)[0].strip() |
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return text |
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def get_cleaned_text_final(text,language): |
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if language in {"en","all_zh","all_ja"}: |
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phones, word2ph, norm_text = clean_text_inf(text, language) |
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elif language in {"zh", "ja","auto"}: |
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phones, word2ph, norm_text = nonen_clean_text_inf(text, language) |
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return phones, word2ph, norm_text |
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def get_bert_final(phones, word2ph, text,language,device): |
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if language == "en": |
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bert = get_bert_inf(phones, word2ph, text, language) |
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elif language in {"zh", "ja","auto"}: |
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bert = nonen_get_bert_inf(text, language) |
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elif language == "all_zh": |
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bert = get_bert_feature(text, word2ph).to(device) |
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else: |
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bert = torch.zeros((1024, len(phones))).to(device) |
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return bert |
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|
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def merge_short_text_in_array(texts, threshold): |
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if (len(texts)) < 2: |
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return texts |
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result = [] |
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text = "" |
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for ele in texts: |
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text += ele |
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if len(text) >= threshold: |
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result.append(text) |
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text = "" |
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if (len(text) > 0): |
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if len(result) == 0: |
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result.append(text) |
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else: |
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result[len(result) - 1] += text |
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return result |
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def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language, how_to_cut=("Do not split"), volume_scale=1.0): |
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if not duration(ref_wav_path): |
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return None |
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if text == '': |
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wprint("Please enter text to generate/请输入生成文字") |
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return None |
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t0 = ttime() |
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startTime=timer() |
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text=trim_text(text,text_language) |
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change_sovits_weights(sovits_path) |
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tprint(f'🏕️LOADED SoVITS Model: {sovits_path}') |
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change_gpt_weights(gpt_path) |
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tprint(f'🏕️LOADED GPT Model: {gpt_path}') |
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prompt_language = dict_language[prompt_language] |
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text_language = dict_language[text_language] |
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prompt_text = prompt_text.strip("\n") |
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if (prompt_text[-1] not in splits): prompt_text += "。" if prompt_language != "en" else "." |
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text = text.strip("\n") |
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if (text[0] not in splits and len(get_first(text)) < 4): text = "。" + text if text_language != "en" else "." + text |
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zero_wav = np.zeros( |
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int(hps.data.sampling_rate * 0.3), |
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dtype=np.float16 if is_half == True else np.float32, |
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) |
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with torch.no_grad(): |
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wav16k, sr = librosa.load(ref_wav_path, sr=16000) |
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if (wav16k.shape[0] > 160000 or wav16k.shape[0] < 48000): |
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errinfo='参考音频在3~10秒范围外,请更换!' |
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raise OSError((errinfo)) |
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wav16k = torch.from_numpy(wav16k) |
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zero_wav_torch = torch.from_numpy(zero_wav) |
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if is_half == True: |
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wav16k = wav16k.half().to(device) |
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zero_wav_torch = zero_wav_torch.half().to(device) |
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else: |
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wav16k = wav16k.to(device) |
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zero_wav_torch = zero_wav_torch.to(device) |
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wav16k = torch.cat([wav16k, zero_wav_torch]) |
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ssl_content = ssl_model.model(wav16k.unsqueeze(0))[ |
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"last_hidden_state" |
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].transpose( |
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1, 2 |
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) |
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codes = vq_model.extract_latent(ssl_content) |
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prompt_semantic = codes[0, 0] |
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t1 = ttime() |
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|
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phones1, word2ph1, norm_text1=get_cleaned_text_final(prompt_text, prompt_language) |
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|
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if (how_to_cut == ("Split into groups of 4 sentences")): |
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text = cut1(text) |
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elif (how_to_cut == ("Split every 50 characters")): |
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text = cut2(text) |
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elif (how_to_cut == ("Split at CN/JP periods (。)")): |
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text = cut3(text) |
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elif (how_to_cut == ("Split at English periods (.)")): |
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text = cut4(text) |
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elif (how_to_cut == ("Split at punctuation marks")): |
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text = cut5(text) |
|
while "\n\n" in text: |
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text = text.replace("\n\n", "\n") |
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print(f"🧨实际输入的目标文本(切句后):{text}\n") |
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texts = text.split("\n") |
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texts = merge_short_text_in_array(texts, 5) |
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audio_opt = [] |
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bert1=get_bert_final(phones1, word2ph1, norm_text1,prompt_language,device).to(dtype) |
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|
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for text in texts: |
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if (len(text.strip()) == 0): |
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continue |
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if (text[-1] not in splits): text += "。" if text_language != "en" else "." |
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print(("\n🎈实际输入的目标文本(每句):"), text) |
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phones2, word2ph2, norm_text2 = get_cleaned_text_final(text, text_language) |
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bert2 = get_bert_final(phones2, word2ph2, norm_text2, text_language, device).to(dtype) |
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bert = torch.cat([bert1, bert2], 1) |
|
|
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all_phoneme_ids = torch.LongTensor(phones1 + phones2).to(device).unsqueeze(0) |
|
bert = bert.to(device).unsqueeze(0) |
|
all_phoneme_len = torch.tensor([all_phoneme_ids.shape[-1]]).to(device) |
|
prompt = prompt_semantic.unsqueeze(0).to(device) |
|
t2 = ttime() |
|
with torch.no_grad(): |
|
|
|
pred_semantic, idx = t2s_model.model.infer_panel( |
|
all_phoneme_ids, |
|
all_phoneme_len, |
|
prompt, |
|
bert, |
|
|
|
top_k=config["inference"]["top_k"], |
|
early_stop_num=hz * max_sec, |
|
) |
|
t3 = ttime() |
|
|
|
pred_semantic = pred_semantic[:, -idx:].unsqueeze( |
|
0 |
|
) |
|
refer = get_spepc(hps, ref_wav_path) |
|
if is_half == True: |
|
refer = refer.half().to(device) |
|
else: |
|
refer = refer.to(device) |
|
|
|
try: |
|
audio = ( |
|
vq_model.decode( |
|
pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refer |
|
) |
|
.detach() |
|
.cpu() |
|
.numpy()[0, 0] |
|
) |
|
except RuntimeError as e: |
|
wprint(f"The input text does not match the language/输入文本与语言不匹配: {e}") |
|
return None |
|
|
|
max_audio=np.abs(audio).max() |
|
if max_audio>1:audio/=max_audio |
|
audio_opt.append(audio) |
|
audio_opt.append(zero_wav) |
|
t4 = ttime() |
|
print("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3)) |
|
|
|
audio_data = (np.concatenate(audio_opt, 0) * 32768).astype(np.int16) |
|
|
|
audio_data = (audio_data.astype(np.float32) * volume_scale).astype(np.int16) |
|
output_wav = "output_audio.wav" |
|
sf.write(output_wav, audio_data, hps.data.sampling_rate) |
|
endTime=timer() |
|
tprint(f'🆗TTS COMPLETE,{round(endTime-startTime,4)}s') |
|
return output_wav |
|
|
|
def split(todo_text): |
|
todo_text = todo_text.replace("……", "。").replace("——", ",") |
|
if todo_text[-1] not in splits: |
|
todo_text += "。" |
|
i_split_head = i_split_tail = 0 |
|
len_text = len(todo_text) |
|
todo_texts = [] |
|
while 1: |
|
if i_split_head >= len_text: |
|
break |
|
if todo_text[i_split_head] in splits: |
|
i_split_head += 1 |
|
todo_texts.append(todo_text[i_split_tail:i_split_head]) |
|
i_split_tail = i_split_head |
|
else: |
|
i_split_head += 1 |
|
return todo_texts |
|
|
|
|
|
def cut1(inp): |
|
inp = inp.strip("\n") |
|
inps = split(inp) |
|
split_idx = list(range(0, len(inps), 4)) |
|
split_idx[-1] = None |
|
if len(split_idx) > 1: |
|
opts = [] |
|
for idx in range(len(split_idx) - 1): |
|
opts.append("".join(inps[split_idx[idx]: split_idx[idx + 1]])) |
|
else: |
|
opts = [inp] |
|
return "\n".join(opts) |
|
|
|
|
|
def cut2(inp): |
|
inp = inp.strip("\n") |
|
inps = split(inp) |
|
if len(inps) < 2: |
|
return inp |
|
opts = [] |
|
summ = 0 |
|
tmp_str = "" |
|
for i in range(len(inps)): |
|
summ += len(inps[i]) |
|
tmp_str += inps[i] |
|
if summ > 50: |
|
summ = 0 |
|
opts.append(tmp_str) |
|
tmp_str = "" |
|
if tmp_str != "": |
|
opts.append(tmp_str) |
|
|
|
if len(opts) > 1 and len(opts[-1]) < 50: |
|
opts[-2] = opts[-2] + opts[-1] |
|
opts = opts[:-1] |
|
return "\n".join(opts) |
|
|
|
|
|
def cut3(inp): |
|
inp = inp.strip("\n") |
|
return "\n".join(["%s" % item for item in inp.strip("。").split("。")]) |
|
|
|
|
|
def cut4(inp): |
|
inp = inp.strip("\n") |
|
return "\n".join(["%s" % item for item in inp.strip(".").split(".")]) |
|
|
|
|
|
|
|
def cut5(inp): |
|
|
|
|
|
inp = inp.strip("\n") |
|
punds = r'[,.;?!、,。?!;:…]' |
|
items = re.split(f'({punds})', inp) |
|
mergeitems = ["".join(group) for group in zip(items[::2], items[1::2])] |
|
if len(items)%2 == 1: |
|
mergeitems.append(items[-1]) |
|
opt = "\n".join(mergeitems) |
|
return opt |
|
|
|
|
|
|
|
def custom_sort_key(s): |
|
|
|
parts = re.split('(\d+)', s) |
|
|
|
parts = [int(part) if part.isdigit() else part for part in parts] |
|
return parts |
|
|
|
def tprint(text): |
|
now=datetime.now(tz).strftime('%H:%M:%S') |
|
print(f'UTC+8 - {now} - {text}') |
|
|
|
def wprint(text): |
|
tprint(text) |
|
gr.Warning(text) |
|
|
|
|
|
def trim_text(text,language): |
|
limit_cj = 120 |
|
limit_en = 60 |
|
search_limit_cj = limit_cj+30 |
|
search_limit_en = limit_en +30 |
|
text = text.replace('\n', '').strip() |
|
|
|
if language =='English': |
|
words = text.split() |
|
if len(words) <= limit_en: |
|
return text |
|
|
|
for i in range(limit_en, -1, -1): |
|
if any(punct in words[i] for punct in splits): |
|
return ' '.join(words[:i+1]) |
|
for i in range(limit_en, min(len(words), search_limit_en)): |
|
if any(punct in words[i] for punct in splits): |
|
return ' '.join(words[:i+1]) |
|
return ' '.join(words[:limit_en]) |
|
|
|
else: |
|
if len(text) <= limit_cj: |
|
return text |
|
for i in range(limit_cj, -1, -1): |
|
if text[i] in splits: |
|
return text[:i+1] |
|
for i in range(limit_cj, min(len(text), search_limit_cj)): |
|
if text[i] in splits: |
|
return text[:i+1] |
|
return text[:limit_cj] |
|
|
|
def duration(audio_file_path): |
|
try: |
|
audio_duration = librosa.get_duration(filename=audio_file_path) |
|
if not 3 < audio_duration < 10: |
|
wprint("The audio length must be between 3~10 seconds/音频时长须在3~10秒之间") |
|
return False |
|
return True |
|
except FileNotFoundError: |
|
wprint("Failed to obtain uploaded audio/未找到音频文件") |
|
return False |
|
|
|
def update_model(choice): |
|
global gpt_path, sovits_path |
|
model_info = models[choice] |
|
gpt_path = abs_path(model_info["gpt_weight"]) |
|
sovits_path = abs_path(model_info["sovits_weight"]) |
|
model_name = choice |
|
tone_info = model_info["tones"]["tone1"] |
|
tone_sample_path = abs_path(tone_info["sample"]) |
|
tprint(f'✅SELECT MODEL:{choice}') |
|
|
|
return ( |
|
tone_info["example_voice_wav"], |
|
tone_info["example_voice_wav_words"], |
|
model_info["default_language"], |
|
model_info["default_language"], |
|
model_name, |
|
"tone1" , |
|
tone_sample_path |
|
) |
|
|
|
def update_tone(model_choice, tone_choice): |
|
model_info = models[model_choice] |
|
tone_info = model_info["tones"][tone_choice] |
|
example_voice_wav = abs_path(tone_info["example_voice_wav"]) |
|
example_voice_wav_words = tone_info["example_voice_wav_words"] |
|
tone_sample_path = abs_path(tone_info["sample"]) |
|
return example_voice_wav, example_voice_wav_words,tone_sample_path |
|
|
|
def transcribe(voice): |
|
time1=timer() |
|
tprint('⚡Start Clone - transcribe') |
|
task="transcribe" |
|
if voice is None: |
|
wprint("No audio file submitted! Please upload or record an audio file before submitting your request.") |
|
R = pipe(voice, batch_size=8, generate_kwargs={"task": task}, return_timestamps=True,return_language=True) |
|
text=R['text'] |
|
lang=R['chunks'][0]['language'] |
|
if lang=='english': |
|
language='English' |
|
elif lang =='chinese': |
|
language='中文' |
|
elif lang=='japanese': |
|
language = '日本語' |
|
|
|
time2=timer() |
|
tprint(f'transcribe COMPLETE,{round(time2-time1,4)}s') |
|
tprint(f'\n🔣转录结果:\n 🔣Language:{language} \n 🔣Text:{text}' ) |
|
return text,language |
|
|
|
def clone_voice(user_voice,user_text,user_lang): |
|
if not duration(user_voice): |
|
return None |
|
if user_text == '': |
|
wprint("Please enter text to generate/请输入生成文字") |
|
return None |
|
|
|
user_text=trim_text(user_text,user_lang) |
|
time1=timer() |
|
global gpt_path, sovits_path |
|
gpt_path = abs_path("pretrained_models/s1bert25hz-2kh-longer-epoch=68e-step=50232.ckpt") |
|
|
|
sovits_path = abs_path("pretrained_models/s2G488k.pth") |
|
|
|
prompt_text, prompt_language = transcribe(user_voice) |
|
output_wav = get_tts_wav( |
|
user_voice, |
|
prompt_text, |
|
prompt_language, |
|
user_text, |
|
user_lang, |
|
how_to_cut="Do not split", |
|
volume_scale=1.0) |
|
time2=timer() |
|
tprint(f'🆗CLONE COMPLETE,{round(time2-time1,4)}s') |
|
return output_wav |
|
|
|
|
|
from info import models |
|
models_by_language = { |
|
"English": [], |
|
"中文": [], |
|
"日本語": [] |
|
} |
|
for model_name, model_info in models.items(): |
|
language = model_info["default_language"] |
|
models_by_language[language].append((model_name, model_info)) |
|
|
|
|
|
|
|
with gr.Blocks(theme='Kasien/ali_theme_custom') as app: |
|
gr.HTML(''' |
|
<h1 style="font-size: 25px;">A TTS GENERATOR</h1> |
|
<p style="margin-bottom: 10px; font-size: 100%"> |
|
If you like this space, please click the ❤️ at the top of the page..如喜欢,请点一下页面顶部的❤️<br> |
|
💡This space is based on the innovative text-to-speech generation solution |
|
<a href="https://github.com/RVC-Boss/GPT-SoVITS" target="_blank">GPT-SoVITS</a> . |
|
You can visit the repo's github homepage to learn training and inference.<br> |
|
本空间基于新式的文字转语音生成方案 <a href="https://github.com/RVC-Boss/GPT-SoVITS" target="_blank">GPT-SoVITS</a> . |
|
你可以前往项目的github主页学习如何推理和训练。<br> |
|
✏️Generating voice is very slow due to using HuggingFace's free CPU in this space. For faster generation, |
|
click the Colab icon below to use this space in Colab, which will significantly improve the speed.<br> |
|
由于本空间使用huggingface的免费CPU进行推理,因此速度很慢,如想快速生成, |
|
请点击下方的Colab图标,前往Colab使用已获得更快的生成速度。 |
|
</p> |
|
<a href="https://colab.research.google.com/drive/1fTuPZ4tZsAjS-TrhQWMCb7KRdnU8aF6j#scrollTo=MDtJIbLdLHe9" target="_blank"><img src="https://camo.githubusercontent.com/dd83d4a334eab7ada034c13747d9e2237182826d32e3fda6629740b6e02f18d8/68747470733a2f2f696d672e736869656c64732e696f2f62616467652f436f6c61622d4639414230303f7374796c653d666f722d7468652d6261646765266c6f676f3d676f6f676c65636f6c616226636f6c6f723d353235323532" alt="colab"></a> |
|
''') |
|
|
|
default_voice_wav, default_voice_wav_words, default_language, _, default_model_name, _, default_tone_sample_path = update_model("Trump") |
|
english_models = [name for name, _ in models_by_language["English"]] |
|
chinese_models = [name for name, _ in models_by_language["中文"]] |
|
japanese_models = [name for name, _ in models_by_language["日本語"]] |
|
with gr.Row(): |
|
english_choice = gr.Radio(english_models, label="EN|English Model",value="Trump",scale=3) |
|
chinese_choice = gr.Radio(chinese_models, label="CN|中文模型",scale=2) |
|
japanese_choice = gr.Radio(japanese_models, label="JP|日本語モデル",scale=4) |
|
|
|
plsh='Text must match the selected language option to prevent errors, for example, if English is input but Chinese is selected for generation.\n文字一定要和语言选项匹配,不然要报错,比如输入的是英文,生成语言选中文' |
|
limit='Max 70 words. Excess will be ignored./单次最多处理120字左右,多余的会被忽略' |
|
|
|
with gr.Row(): |
|
model_name = gr.Textbox(label="Seleted Model/已选模型", value=default_model_name, scale=1) |
|
text = gr.Textbox(label="Input some text for voice generation/输入想要生成语音的文字", lines=5,scale=8, |
|
placeholder=plsh,info=limit) |
|
|
|
|
|
with gr.Row(): |
|
with gr.Column(scale=2): |
|
tone_select = gr.Radio( |
|
label="Select Tone/选择语气", |
|
choices=["tone1","tone2","tone3"], |
|
value="tone1", |
|
info='Tone influences the emotional expression ',scale=1) |
|
|
|
text_language = gr.Radio( |
|
label="Select language for input text/输入的文字对应语言", |
|
choices=["中文","English","日本語"], |
|
value=default_language, |
|
info='Input text and language must match.',scale=1, |
|
) |
|
|
|
tone_sample=gr.Audio(label="🔊Preview tone/试听语气 ", scale=5) |
|
|
|
|
|
with gr.Accordion(label="prpt voice", open=False,visible=False): |
|
with gr.Row(visible=True): |
|
inp_ref = gr.Audio(label="Reference audio", type="filepath", value=default_voice_wav, scale=3) |
|
prompt_text = gr.Textbox(label="Reference text", value=default_voice_wav_words, scale=3) |
|
prompt_language = gr.Dropdown(label="Language of the reference audio", choices=["中文", "English", "日本語"], value=default_language, scale=1,interactive=False) |
|
|
|
|
|
|
|
with gr.Accordion(label="Additional generation options/附加生成选项", open=False): |
|
how_to_cut = gr.Dropdown( |
|
label=("How to split?"), |
|
choices=[("Do not split"), ("Split into groups of 4 sentences"), ("Split every 50 characters"), |
|
("Split at CN/JP periods (。)"), ("Split at English periods (.)"), ("Split at punctuation marks"), ], |
|
value=("Split into groups of 4 sentences"), |
|
interactive=True, |
|
info='A suitable splitting method can achieve better generation results' |
|
) |
|
volume = gr.Slider(minimum=0.5, maximum=2, value=1, step=0.01, label='Volume/音量') |
|
|
|
|
|
|
|
with gr.Row(): |
|
main_button = gr.Button("✨Generate Voice", variant="primary", scale=1) |
|
output = gr.Audio(label="💾Download it by clicking ⬇️", scale=3) |
|
|
|
|
|
gr.HTML(''' |
|
Generation is slower, please be patient and wait/合成比较慢,请耐心等待<br> |
|
If it generated silence, please try again./如果生成了空白声音,请重试 |
|
<br><br><br><br> |
|
<h1 style="font-size: 25px;">Clone custom Voice/克隆自定义声音</h1> |
|
<p style="margin-bottom: 10px; font-size: 100%">Need 3~10s audio.This involves voice-to-text conversion followed by text-to-voice conversion, so it takes longer time<br> |
|
需要3~10秒语音,这个会涉及语音转文字,之后再转语音,所以耗时比较久 |
|
</p>''') |
|
|
|
with gr.Row(): |
|
user_voice = gr.Audio(type="filepath", label="(3~10s)Upload or Record audio/上传或录制声音",scale=3) |
|
user_lang = gr.Dropdown(label="Language/生成语言", choices=["中文", "English", "日本語"],scale=1,value='English') |
|
user_text= gr.Textbox(label="Text for generation/输入想要生成语音的文字", lines=5,scale=5, |
|
placeholder=plsh,info=limit) |
|
|
|
user_button = gr.Button("✨Clone Voice", variant="primary") |
|
user_output = gr.Audio(label="💾Download it by clicking ⬇️") |
|
|
|
gr.HTML('''<div align=center><img id="visitor-badge" alt="visitor badge" src="https://visitor-badge.laobi.icu/badge?page_id=Ailyth/DLMP9" /></div>''') |
|
|
|
english_choice.change(update_model, inputs=[english_choice], outputs=[inp_ref, prompt_text, prompt_language, text_language, model_name, tone_select, tone_sample]) |
|
chinese_choice.change(update_model, inputs=[chinese_choice], outputs=[inp_ref, prompt_text, prompt_language, text_language, model_name, tone_select, tone_sample]) |
|
japanese_choice.change(update_model, inputs=[japanese_choice], outputs=[inp_ref, prompt_text, prompt_language, text_language, model_name, tone_select, tone_sample]) |
|
tone_select.change(update_tone, inputs=[model_name, tone_select], outputs=[inp_ref, prompt_text, tone_sample]) |
|
|
|
main_button.click( |
|
get_tts_wav, |
|
inputs=[inp_ref, prompt_text, prompt_language, text, text_language, how_to_cut,volume], |
|
outputs=[output]) |
|
|
|
user_button.click( |
|
clone_voice, |
|
inputs=[user_voice,user_text,user_lang], |
|
outputs=[user_output]) |
|
|
|
app.launch(share=True) |