import config import sys,os import gradio as gr import torch import numpy as np import librosa,torch from transformers import AutoModelForMaskedLM, AutoTokenizer from feature_extractor import cnhubert from module.models import SynthesizerTrn from AR.models.t2s_lightning_module import Text2SemanticLightningModule from text import cleaned_text_to_sequence from text.cleaner import clean_text from time import time as ttime from module.mel_processing import spectrogram_torch from my_utils import load_audio from tools.i18n.i18n import I18nAuto hps = None ssl_model = None vq_model = None t2s_model = None is_half = config.is_half hz = 50 max_sec = None top_k = None #后期可能将这里个path分离成变量 bert_path = config.bert_path cnhubert_base_path = config.cnhubert_path cnhubert.cnhubert_base_path = cnhubert_base_path device = "cuda" #不确定能否支持cpu,先预留 tokenizer = None bert_model = None i18n = I18nAuto() cwd = os.getcwd() sys.path.append(cwd) SUPPORT_LANGUAGE = [i18n("中文"),i18n("英文"),i18n("日文")] dict_language={ i18n("中文"):"zh", i18n("英文"):"en", i18n("日文"):"ja" } def read_model_path(model_type): model_list = [] if model_type == config.MODEL_TYPE_GPT: folder_path = os.path.join(cwd,config.MODEL_FOLDER_PATH_GPT) file_type = ".ckpt" elif model_type == config.MODEL_TYPE_SOVITS: folder_path = os.path.join(cwd,config.MODEL_FOLDER_PATH_SOVITS) file_type = ".pth" for root, dirs, files in os.walk(folder_path): for file_name in files: if file_name.endswith(file_type): file_path = os.path.join(root, file_name) model_list.append((file_name,file_path)) return model_list def refresh_model_list(): gpt_choices = read_model_path(config.MODEL_TYPE_GPT) sovits_choices = read_model_path(config.MODEL_TYPE_SOVITS) return gr.Dropdown(choices=sorted(gpt_choices),value=gpt_choices[0]if len(gpt_choices)>0 else "",interactive=True),gr.Dropdown(choices=sorted(sovits_choices),value=sovits_choices[0]if len(sovits_choices)>0 else None,interactive=True) def get_bert_feature(text, word2ph): global tokenizer,bert_model 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 class DictToAttrRecursive(dict): def __init__(self, input_dict): super().__init__(input_dict) for key, value in input_dict.items(): if isinstance(value, dict): value = DictToAttrRecursive(value) self[key] = value setattr(self, key, value) def __getattr__(self, item): try: return self[item] except KeyError: raise AttributeError(f"Attribute {item} not found") def __setattr__(self, key, value): if isinstance(value, dict): value = DictToAttrRecursive(value) super(DictToAttrRecursive, self).__setitem__(key, value) super().__setattr__(key, value) def __delattr__(self, item): try: del self[item] except KeyError: raise AttributeError(f"Attribute {item} not found") 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 def load_models(sovits_path, gpt_path): global tokenizer,bert_model,hps,ssl_model,vq_model,t2s_model,is_half,hz,max_sec,top_k print(f"SoVITS model path: {sovits_path}") print(f"GPT model path: {gpt_path}") if sovits_path is None or gpt_path is None: print("Choose both of two models before loading") return "请正确选择两个模型",gr.Button(interactive=False) torch.cuda.empty_cache() 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) dict_s2=torch.load(sovits_path,map_location="cpu") hps=dict_s2["config"] hps = DictToAttrRecursive(hps) hps.model.semantic_frame_rate = "25hz" dict_s1 = torch.load(gpt_path, map_location="cpu") dict_s1_config = dict_s1["config"] max_sec = dict_s1_config["data"]["max_sec"] top_k=dict_s1_config["inference"]["top_k"] 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: 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)) t2s_model = Text2SemanticLightningModule(dict_s1_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)) #加载模型成功 return "模型加载成功",gr.Button(interactive=True) def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language): global hps,ssl_model,vq_model,t2s_model,is_half,hz,max_sec,top_k t0 = ttime() prompt_text = prompt_text.strip("\n") prompt_language, text = prompt_language, text.strip("\n") zero_wav = np.zeros( int(hps.data.sampling_rate * 0.3), dtype=np.float16 if is_half else np.float32, ) with torch.no_grad(): wav16k, sr = librosa.load(ref_wav_path, sr=16000) wav16k = torch.from_numpy(wav16k) zero_wav_torch = torch.from_numpy(zero_wav) if is_half == True: wav16k = wav16k.half().to(device) zero_wav_torch = zero_wav_torch.half().to(device) else: wav16k = wav16k.to(device) zero_wav_torch = zero_wav_torch.to(device) wav16k=torch.cat([wav16k,zero_wav_torch]) 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 = [] for text in texts: # 解决输入目标文本的空行导致报错的问题 if (len(text.strip()) == 0): continue 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=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() print("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3)) yield hps.data.sampling_rate, (np.concatenate(audio_opt, 0) * 32768).astype( np.int16 ) splits = { ",", "。", "?", "!", ",", ".", "?", "!", "~", ":", ":", "—", "…", } # 不考虑省略号 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 # 结尾一定有标点,所以直接跳出即可,最后一段在上次已加入 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 start_webui(): ngpu = torch.cuda.device_count() gpu_list = [] for i in range(ngpu): gpu_list.append((torch.cuda.get_device_name(i),i)) print(gpu_list) gpt_choices = read_model_path(config.MODEL_TYPE_GPT) sovits_choices = read_model_path(config.MODEL_TYPE_SOVITS) with gr.Blocks() as demo: with gr.Row(): message_text = gr.Textbox("信息",interactive=False) with gr.Accordion(label="设备"): with gr.Row(): cuda_device_index = gr.Dropdown(choices=gpu_list,value=0 if len(gpu_list)>0 else None,label="CUDA设备",interactive=True) with gr.Accordion(label="模型"): with gr.Row(): gpt_dropdown = gr.Dropdown(choices=sorted(gpt_choices),value=gpt_choices[0][1]if len(gpt_choices)>0 else None,label="选择GPT模型",interactive=True) sovits_dropdown = gr.Dropdown(choices=sorted(sovits_choices),value=sovits_choices[0][1]if len(sovits_choices)>0 else None,label="选择SoVITS模型",interactive=True) with gr.Row(): model_load_button = gr.Button("加载模型",variant="primary") model_refresh_button = gr.Button("刷新模型", variant="secondary") with gr.Accordion(label="参考"): with gr.Group(): with gr.Row(): with gr.Row(): ref_wav_path = gr.Audio(label="参考音频", type="filepath", scale=3) ref_language = gr.Dropdown(choices=SUPPORT_LANGUAGE,value=i18n("中文"),label="参考语种",interactive=True,min_width=50, scale=1) ref_text = gr.TextArea(label="参考文本",scale=1) with gr.Row(): output_language = gr.Dropdown(choices=SUPPORT_LANGUAGE,value=i18n("中文"),label="合成语种",interactive=True, scale=2) preprocess_output_text_button = gr.Button("合成文本预处理",variant="primary",scale=3) inference_button = gr.Button(i18n("合成语音"), interactive=False,variant="primary") output_text = gr.TextArea(label="合成文本",interactive=True) output_audio = gr.Audio(label="输出结果") model_load_button.click(load_models,[gpt_dropdown,sovits_dropdown],[message_text,inference_button]) model_refresh_button.click(refresh_model_list,[],[gpt_dropdown,sovits_dropdown]) inference_button.click( get_tts_wav, [ref_wav_path, ref_text, ref_language, output_text, output_language], [output_audio], ) demo.queue(max_size=1022).launch(server_port=2777) if __name__ == "__main__": start_webui()