# Modified from https://github.com/RVC-Boss/GPT-SoVITS/blob/main/GPT_SoVITS/inference_webui.py import os gpt_path = os.environ.get( "gpt_path", "pretrained_models/jay-e15.ckpt" ) sovits_path = os.environ.get("sovits_path", "pretrained_models/jay_e10_s60.pth") cnhubert_base_path = os.environ.get( "cnhubert_base_path", "pretrained_models/chinese-hubert-base" ) bert_path = os.environ.get( "bert_path", "pretrained_models/chinese-roberta-wwm-ext-large" ) if "_CUDA_VISIBLE_DEVICES" in os.environ: os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"] import gradio as gr import librosa import numpy as np import torch from transformers import AutoModelForMaskedLM, AutoTokenizer from feature_extractor import cnhubert cnhubert.cnhubert_base_path = cnhubert_base_path from time import time as ttime import datetime from AR.models.t2s_lightning_module import Text2SemanticLightningModule from module.mel_processing import spectrogram_torch from module.models import SynthesizerTrn from my_utils import load_audio from text import cleaned_text_to_sequence from text.cleaner import clean_text device = "cuda" if torch.cuda.is_available() else "cpu" is_half = eval( os.environ.get("is_half", "True" if torch.cuda.is_available() else "False") ) tokenizer = AutoTokenizer.from_pretrained(bert_path) bert_model = AutoModelForMaskedLM.from_pretrained(bert_path) if is_half == True: bert_model = bert_model.half().to(device) else: bert_model = bert_model.to(device) # bert_model=bert_model.to(device) def get_bert_feature(text, word2ph): with torch.no_grad(): inputs = tokenizer(text, return_tensors="pt") for i in inputs: inputs[i] = inputs[i].to(device) #####输入是long不用管精度问题,精度随bert_model res = bert_model(**inputs, output_hidden_states=True) res = torch.cat(res["hidden_states"][-3:-2], -1)[0].cpu()[1:-1] assert len(word2ph) == len(text) phone_level_feature = [] for i in range(len(word2ph)): repeat_feature = res[i].repeat(word2ph[i], 1) phone_level_feature.append(repeat_feature) phone_level_feature = torch.cat(phone_level_feature, dim=0) # if(is_half==True):phone_level_feature=phone_level_feature.half() return phone_level_feature.T n_semantic = 1024 dict_s2 = torch.load(sovits_path, map_location="cpu") hps = dict_s2["config"] class DictToAttrRecursive: def __init__(self, input_dict): for key, value in input_dict.items(): if isinstance(value, dict): # 如果值是字典,递归调用构造函数 setattr(self, key, DictToAttrRecursive(value)) else: setattr(self, key, value) hps = DictToAttrRecursive(hps) hps.model.semantic_frame_rate = "25hz" dict_s1 = torch.load(gpt_path, map_location="cpu") config = dict_s1["config"] ssl_model = cnhubert.get_model() if is_half == True: ssl_model = ssl_model.half().to(device) else: ssl_model = ssl_model.to(device) vq_model = SynthesizerTrn( hps.data.filter_length // 2 + 1, hps.train.segment_size // hps.data.hop_length, n_speakers=hps.data.n_speakers, **hps.model, ) if is_half == True: vq_model = vq_model.half().to(device) else: vq_model = vq_model.to(device) vq_model.eval() print(vq_model.load_state_dict(dict_s2["weight"], strict=False)) hz = 50 max_sec = config["data"]["max_sec"] # t2s_model = Text2SemanticLightningModule.load_from_checkpoint(checkpoint_path=gpt_path, config=config, map_location="cpu")#########todo t2s_model = Text2SemanticLightningModule(config, "ojbk", is_train=False) t2s_model.load_state_dict(dict_s1["weight"]) if is_half == True: t2s_model = t2s_model.half() t2s_model = t2s_model.to(device) t2s_model.eval() total = sum([param.nelement() for param in t2s_model.parameters()]) print("Number of parameter: %.2fM" % (total / 1e6)) def get_spepc(hps, filename): audio = load_audio(filename, int(hps.data.sampling_rate)) audio = torch.FloatTensor(audio) audio_norm = audio audio_norm = audio_norm.unsqueeze(0) spec = spectrogram_torch( audio_norm, hps.data.filter_length, hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length, center=False, ) return spec dict_language = {"Chinese": "zh", "English": "en", "Japanese": "ja"} def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language): start_time = datetime.datetime.now() print(f"---START---{start_time}---") print(f"ref_wav_path: {ref_wav_path}") print(f"prompt_text: {prompt_text}") print(f"prompt_language: {prompt_language}") print(f"text: {text}") print(f"text_language: {text_language}") if len(prompt_text) > 100 or len(text) > 100: print("Input text is limited to 100 characters.") return "Input text is limited to 100 characters.", None t0 = ttime() prompt_text = prompt_text.strip("\n") prompt_language, text = prompt_language, text.strip("\n") with torch.no_grad(): wav16k, _ = librosa.load(ref_wav_path, sr=16000) # 派蒙 # length of wav16k in sec should be in 60s if len(wav16k) > 16000 * 60: print("Input audio is limited to 60 seconds.") return "Input audio is limited to 60 seconds.", None wav16k = wav16k[: int(hps.data.sampling_rate * max_sec)] wav16k = torch.from_numpy(wav16k) if is_half == True: wav16k = wav16k.half().to(device) else: wav16k = wav16k.to(device) ssl_content = ssl_model.model(wav16k.unsqueeze(0))[ "last_hidden_state" ].transpose( 1, 2 ) # .float() codes = vq_model.extract_latent(ssl_content) prompt_semantic = codes[0, 0] t1 = ttime() prompt_language = dict_language[prompt_language] text_language = dict_language[text_language] phones1, word2ph1, norm_text1 = clean_text(prompt_text, prompt_language) phones1 = cleaned_text_to_sequence(phones1) texts = text.split("\n") audio_opt = [] zero_wav = np.zeros( int(hps.data.sampling_rate * 0.3), dtype=np.float16 if is_half == True else np.float32, ) for text in texts: phones2, word2ph2, norm_text2 = clean_text(text, text_language) phones2 = cleaned_text_to_sequence(phones2) if prompt_language == "zh": bert1 = get_bert_feature(norm_text1, word2ph1).to(device) else: bert1 = torch.zeros( (1024, len(phones1)), dtype=torch.float16 if is_half == True else torch.float32, ).to(device) if text_language == "zh": bert2 = get_bert_feature(norm_text2, word2ph2).to(device) else: bert2 = torch.zeros((1024, len(phones2))).to(bert1) bert = torch.cat([bert1, bert2], 1) 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 = t2s_model.model.infer( pred_semantic, idx = t2s_model.model.infer_panel( all_phoneme_ids, all_phoneme_len, prompt, bert, # prompt_phone_len=ph_offset, top_k=config["inference"]["top_k"], early_stop_num=hz * max_sec, ) t3 = ttime() # print(pred_semantic.shape,idx) pred_semantic = pred_semantic[:, -idx:].unsqueeze( 0 ) # .unsqueeze(0)#mq要多unsqueeze一次 refer = get_spepc(hps, ref_wav_path) # .to(device) if is_half == True: refer = refer.half().to(device) else: refer = refer.to(device) # audio = vq_model.decode(pred_semantic, all_phoneme_ids, refer).detach().cpu().numpy()[0, 0] audio = ( vq_model.decode( pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refer ) .detach() .cpu() .numpy()[0, 0] ) ###试试重建不带上prompt部分 audio_opt.append(audio) audio_opt.append(zero_wav) t4 = ttime() end_time = datetime.datetime.now() dur = end_time - start_time print( f"Success! total time: {dur.seconds:.3f} sec,\ndetail time: {t1 - t0:.3f}, {t2 - t1:.3f}, {t3 - t2:.3f}, {t4 - t3:.3f}" ) print(f"---END---{end_time}---") return ( f"Success! total time: {dur.seconds:.3f} sec,\ndetail time: {t1 - t0:.3f}, {t2 - t1:.3f}, {t3 - t2:.3f}, {t4 - t3:.3f}", ( hps.data.sampling_rate, (np.concatenate(audio_opt, 0) * 32768).astype(np.int16), ), ) with gr.Blocks(title="GPT-SoVITS Zero-shot TTS Demo") as app: gr.Markdown("#
🥳💕🎶 GPT-SoVITS 1分钟完美声音克隆,版本大升级!
") gr.Markdown("##
🌟 支持中日英三语自动标注及训练 + 中文方言训练!Powered by [GPT-SoVITS](https://github.com/RVC-Boss/GPT-SoVITS)
") gr.Markdown("###
🌊 更多精彩应用,尽在[滔滔AI](http://www.talktalkai.com);滔滔AI,为爱滔滔!💕
") gr.Markdown("## 请上传参考音频") with gr.Row(): inp_ref = gr.Audio(label="请上传数据集中的参考音频", type="filepath", value="jay_speech.wav_1361600_1516800.wav") prompt_text = gr.Textbox(label="参考音频对应的文字内容", value="你喜欢这样的东西,所以你我觉得人要有一技之长呢。") prompt_language = gr.Dropdown( label="参考音频的语言", choices=["Chinese", "English", "Japanese"], value="Chinese", ) gr.Markdown("## 开始真实拟声之旅吧!") with gr.Row(): text = gr.Textbox(label="需要合成的内容", lines=5) text_language = gr.Dropdown( label="合成内容的语言", choices=["Chinese", "English", "Japanese"], value="Chinese", ) inference_button = gr.Button("开始真实拟声吧!", variant="primary") with gr.Column(): info = gr.Textbox(label="Info", visible=False) output = gr.Audio(label="为您合成的专属音频") inference_button.click( get_tts_wav, [inp_ref, prompt_text, prompt_language, text, text_language], [info, output], ) gr.Markdown("###
注意❗:请不要生成会对个人以及组织造成侵害的内容,此程序仅供科研、学习及个人娱乐使用。
") gr.HTML(''' ''') app.queue(max_size=10) app.launch(inbrowser=True)