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https://github.com/RVC-Boss/GPT-SoVITS.git
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clean branch
This commit is contained in:
林北1941783147
parent
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commit
8ccb4d8b7d
5
.gitignore
vendored
5
.gitignore
vendored
@ -4,8 +4,3 @@ __pycache__
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env
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runtime
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.idea
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TEMP
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ffmpeg.exe
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ffprobe.exe
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GPT_weights/
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SoVITS_weights/
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@ -1,12 +1,5 @@
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import sys,os
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#model type name
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MODEL_TYPE_GPT = "GPT"
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MODEL_TYPE_SOVITS = "SOVITS"
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#model folder path
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MODEL_FOLDER_PATH_GPT = "GPT_weights"
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MODEL_FOLDER_PATH_SOVITS = "SoVITS_weights"
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# 推理用的指定模型
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sovits_path = ""
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2
tools/uvr5/uvr5_weights/.gitignore
vendored
2
tools/uvr5/uvr5_weights/.gitignore
vendored
@ -1,2 +0,0 @@
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*
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!.gitignore
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371
voice_loader.py
371
voice_loader.py
@ -1,371 +0,0 @@
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import config
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import sys,os
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import gradio as gr
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import torch
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import numpy as np
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import librosa,torch
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from transformers import AutoModelForMaskedLM, AutoTokenizer
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from feature_extractor import cnhubert
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from module.models import SynthesizerTrn
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from AR.models.t2s_lightning_module import Text2SemanticLightningModule
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from text import cleaned_text_to_sequence
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from text.cleaner import clean_text
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from time import time as ttime
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from module.mel_processing import spectrogram_torch
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from my_utils import load_audio
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from tools.i18n.i18n import I18nAuto
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hps = None
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ssl_model = None
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vq_model = None
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t2s_model = None
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is_half = config.is_half
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hz = 50
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max_sec = None
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top_k = None
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#后期可能将这里个path分离成变量
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bert_path = config.bert_path
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cnhubert_base_path = config.cnhubert_path
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cnhubert.cnhubert_base_path = cnhubert_base_path
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device = "cuda" #不确定能否支持cpu,先预留
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tokenizer = None
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bert_model = None
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i18n = I18nAuto()
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cwd = os.getcwd()
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sys.path.append(cwd)
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SUPPORT_LANGUAGE = [i18n("中文"),i18n("英文"),i18n("日文")]
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dict_language={
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i18n("中文"):"zh",
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i18n("英文"):"en",
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i18n("日文"):"ja"
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}
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def read_model_path(model_type):
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model_list = []
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if model_type == config.MODEL_TYPE_GPT:
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folder_path = os.path.join(cwd,config.MODEL_FOLDER_PATH_GPT)
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file_type = ".ckpt"
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elif model_type == config.MODEL_TYPE_SOVITS:
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folder_path = os.path.join(cwd,config.MODEL_FOLDER_PATH_SOVITS)
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file_type = ".pth"
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for root, dirs, files in os.walk(folder_path):
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for file_name in files:
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if file_name.endswith(file_type):
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file_path = os.path.join(root, file_name)
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model_list.append((file_name,file_path))
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return model_list
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def refresh_model_list():
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gpt_choices = read_model_path(config.MODEL_TYPE_GPT)
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sovits_choices = read_model_path(config.MODEL_TYPE_SOVITS)
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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)
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def get_bert_feature(text, word2ph):
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global tokenizer,bert_model
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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) #####输入是long不用管精度问题，精度随bert_model
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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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# if(is_half==True):phone_level_feature=phone_level_feature.half()
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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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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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def load_models(sovits_path, gpt_path):
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global tokenizer,bert_model,hps,ssl_model,vq_model,t2s_model,is_half,hz,max_sec,top_k
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print(f"SoVITS model path: {sovits_path}")
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print(f"GPT model path: {gpt_path}")
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if sovits_path is None or gpt_path is None:
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print("Choose both of two models before loading")
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return "请正确选择两个模型",gr.Button(interactive=False)
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torch.cuda.empty_cache()
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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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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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dict_s1 = torch.load(gpt_path, map_location="cpu")
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dict_s1_config = dict_s1["config"]
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max_sec = dict_s1_config["data"]["max_sec"]
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top_k=dict_s1_config["inference"]["top_k"]
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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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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 is_half:
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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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t2s_model = Text2SemanticLightningModule(dict_s1_config, "ojbk", 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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#加载模型成功
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return "模型加载成功",gr.Button(interactive=True)
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def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language):
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global hps,ssl_model,vq_model,t2s_model,is_half,hz,max_sec,top_k
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t0 = ttime()
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prompt_text = prompt_text.strip("\n")
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prompt_language, text = prompt_language, text.strip("\n")
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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 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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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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) # .float()
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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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prompt_language = dict_language[prompt_language]
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text_language = dict_language[text_language]
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phones1, word2ph1, norm_text1 = clean_text(prompt_text, prompt_language)
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phones1 = cleaned_text_to_sequence(phones1)
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texts = text.split("\n")
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audio_opt = []
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for text in texts:
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# 解决输入目标文本的空行导致报错的问题
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if (len(text.strip()) == 0):
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continue
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phones2, word2ph2, norm_text2 = clean_text(text, text_language)
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phones2 = cleaned_text_to_sequence(phones2)
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if prompt_language == "zh":
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bert1 = get_bert_feature(norm_text1, word2ph1).to(device)
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else:
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bert1 = torch.zeros(
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(1024, len(phones1)),
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dtype=torch.float16 if is_half == True else torch.float32,
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).to(device)
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if text_language == "zh":
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bert2 = get_bert_feature(norm_text2, word2ph2).to(device)
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else:
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bert2 = torch.zeros((1024, len(phones2))).to(bert1)
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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)
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bert = bert.to(device).unsqueeze(0)
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all_phoneme_len = torch.tensor([all_phoneme_ids.shape[-1]]).to(device)
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prompt = prompt_semantic.unsqueeze(0).to(device)
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t2 = ttime()
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with torch.no_grad():
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# pred_semantic = t2s_model.model.infer(
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pred_semantic, idx = t2s_model.model.infer_panel(
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all_phoneme_ids,
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all_phoneme_len,
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prompt,
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bert,
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# prompt_phone_len=ph_offset,
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top_k=top_k,
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early_stop_num=hz * max_sec,
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)
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t3 = ttime()
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# print(pred_semantic.shape,idx)
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pred_semantic = pred_semantic[:, -idx:].unsqueeze(
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0
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) # .unsqueeze(0)#mq要多unsqueeze一次
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refer = get_spepc(hps, ref_wav_path) # .to(device)
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if is_half == True:
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refer = refer.half().to(device)
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else:
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refer = refer.to(device)
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# audio = vq_model.decode(pred_semantic, all_phoneme_ids, refer).detach().cpu().numpy()[0, 0]
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audio = (
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vq_model.decode(
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pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refer
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)
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.detach()
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.cpu()
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.numpy()[0, 0]
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) ###试试重建不带上prompt部分
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audio_opt.append(audio)
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audio_opt.append(zero_wav)
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t4 = ttime()
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print("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3))
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yield hps.data.sampling_rate, (np.concatenate(audio_opt, 0) * 32768).astype(
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np.int16
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)
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splits = {
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"，",
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"。",
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"？",
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"！",
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",",
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".",
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"?",
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"!",
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"~",
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":",
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"：",
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"—",
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"…",
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} # 不考虑省略号
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def split(todo_text):
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todo_text = todo_text.replace("……", "。").replace("——", "，")
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if todo_text[-1] not in splits:
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todo_text += "。"
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i_split_head = i_split_tail = 0
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len_text = len(todo_text)
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todo_texts = []
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while 1:
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if i_split_head >= len_text:
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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
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if todo_text[i_split_head] in splits:
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i_split_head += 1
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todo_texts.append(todo_text[i_split_tail:i_split_head])
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i_split_tail = i_split_head
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else:
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i_split_head += 1
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return todo_texts
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def start_webui():
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ngpu = torch.cuda.device_count()
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gpu_list = []
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for i in range(ngpu):
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gpu_list.append((torch.cuda.get_device_name(i),i))
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print(gpu_list)
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gpt_choices = read_model_path(config.MODEL_TYPE_GPT)
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sovits_choices = read_model_path(config.MODEL_TYPE_SOVITS)
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with gr.Blocks() as demo:
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with gr.Row():
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message_text = gr.Textbox("信息",interactive=False)
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with gr.Accordion(label="设备"):
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with gr.Row():
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cuda_device_index = gr.Dropdown(choices=gpu_list,value=0 if len(gpu_list)>0 else None,label="CUDA设备",interactive=True)
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with gr.Accordion(label="模型"):
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with gr.Row():
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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)
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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)
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with gr.Row():
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model_load_button = gr.Button("加载模型",variant="primary")
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model_refresh_button = gr.Button("刷新模型", variant="secondary")
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with gr.Accordion(label="参考"):
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with gr.Group():
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with gr.Row():
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with gr.Row():
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ref_wav_path = gr.Audio(label="参考音频", type="filepath", scale=3)
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ref_language = gr.Dropdown(choices=SUPPORT_LANGUAGE,value=i18n("中文"),label="参考语种",interactive=True,min_width=50, scale=1)
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ref_text = gr.TextArea(label="参考文本",scale=1)
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with gr.Row():
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output_language = gr.Dropdown(choices=SUPPORT_LANGUAGE,value=i18n("中文"),label="合成语种",interactive=True, scale=2)
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preprocess_output_text_button = gr.Button("合成文本预处理",variant="primary",scale=3)
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inference_button = gr.Button(i18n("合成语音"), interactive=False,variant="primary")
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output_text = gr.TextArea(label="合成文本",interactive=True)
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output_audio = gr.Audio(label="输出结果")
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model_load_button.click(load_models,[gpt_dropdown,sovits_dropdown],[message_text,inference_button])
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model_refresh_button.click(refresh_model_list,[],[gpt_dropdown,sovits_dropdown])
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inference_button.click(
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get_tts_wav,
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[ref_wav_path, ref_text, ref_language, output_text, output_language],
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[output_audio],
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)
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demo.queue(max_size=1022).launch(server_port=2777)
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if __name__ == "__main__":
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start_webui()
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