GPT-SoVITS/GPT_SoVITS/inference_webui.py

import os
gpt_path=os.environ.get("gpt_path","pretrained_models/s1bert25hz-2kh-longer-epoch=68e-step=50232.ckpt")
sovits_path=os.environ.get("sovits_path","pretrained_models/s2G488k.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")
infer_ttswebui=os.environ.get("infer_ttswebui",9872)
infer_ttswebui=int(infer_ttswebui)
if("_CUDA_VISIBLE_DEVICES"in os.environ):
    os.environ["CUDA_VISIBLE_DEVICES"]=os.environ["_CUDA_VISIBLE_DEVICES"]
is_half=eval(os.environ.get("is_half","True"))
import gradio as gr
from transformers import AutoModelForMaskedLM, AutoTokenizer
import sys,torch,numpy as np
from pathlib import Path
import os,pdb,utils,librosa,math,traceback,requests,argparse,torch,multiprocessing,pandas as pd,torch.multiprocessing as mp,soundfile
# torch.backends.cuda.sdp_kernel("flash")
# torch.backends.cuda.enable_flash_sdp(True)
# torch.backends.cuda.enable_mem_efficient_sdp(True)  # Not avaliable if torch version is lower than 2.0
# torch.backends.cuda.enable_math_sdp(True)
from random import shuffle
from AR.utils import get_newest_ckpt
from glob import glob
from tqdm import tqdm
from feature_extractor import cnhubert
cnhubert.cnhubert_base_path=cnhubert_base_path
from io import BytesIO
from module.models import SynthesizerTrn
from AR.models.t2s_lightning_module import Text2SemanticLightningModule
from AR.utils.io import load_yaml_config
from text import cleaned_text_to_sequence
from text.cleaner import text_to_sequence, clean_text
from time import time as ttime
from module.mel_processing import spectrogram_torch
from my_utils import load_audio

device="cuda"
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={
    "中文":"zh",
    "英文":"en",
    "日文":"ja"
}
def get_tts_wav(ref_wav_path,prompt_text,prompt_language,text,text_language):
    t0 = ttime()
    prompt_text=prompt_text.strip("\n")
    prompt_language,text=prompt_language,text.strip("\n")
    with torch.no_grad():
        wav16k, sr = librosa.load(ref_wav_path, sr=16000)  # 派蒙
        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()
    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 cut1(inp):
    inp=inp.strip("\n")
    inps=split(inp)
    split_idx=list(range(0,len(inps),5))
    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])<50):##如果最后一个太短了，和前一个合一起
        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("。")])

with gr.Blocks(title="GPT-SoVITS WebUI") as app:
    gr.Markdown(
        value=
            "本软件以MIT协议开源, 作者不对软件具备任何控制力, 使用软件者、传播软件导出的声音者自负全责. <br>如不认可该条款, 则不能使用或引用软件包内任何代码和文件. 详见根目录<b>LICENSE</b>."
    )
    # with gr.Tabs():
    #     with gr.TabItem(i18n("伴奏人声分离&去混响&去回声")):
    with gr.Group():
        gr.Markdown(
            value=
                "*请上传并填写参考信息"
        )
        with gr.Row():
            inp_ref = gr.Audio(label="请上传参考音频", type="filepath")
            prompt_text= gr.Textbox(label="参考音频的文本",value="")
            prompt_language= gr.Dropdown(label="参考音频的语种",choices=["中文","英文","日文"],value="中文")
        gr.Markdown(
            value=
                "*请填写需要合成的目标文本"
        )
        with gr.Row():
            text=gr.Textbox(label="需要合成的文本",value="")
            text_language = gr.Dropdown(label="需要合成的语种", choices=["中文", "英文", "日文"],value="中文")
            inference_button=gr.Button("合成语音", variant="primary")
            output = gr.Audio(label="输出的语音")
        inference_button.click(get_tts_wav, [inp_ref, prompt_text,prompt_language, text,text_language], [output])

        gr.Markdown(
            value=
                "文本切分工具。太长的文本合成出来效果不一定好，所以太长建议先切。合成会根据文本的换行分开合成再拼起来。"
        )
        with gr.Row():
            text_inp=gr.Textbox(label="需要合成的切分前文本",value="")
            button1 = gr.Button("凑五句一切", variant="primary")
            button2 = gr.Button("凑50字一切", variant="primary")
            button3 = gr.Button("按中文句号。切", variant="primary")
            text_opt = gr.Textbox(label="切分后文本", value="")
            button1.click(cut1,[text_inp],[text_opt])
            button2.click(cut2,[text_inp],[text_opt])
            button3.click(cut3,[text_inp],[text_opt])
        gr.Markdown(
            value=
                "后续将支持混合语种编码文本输入。"
        )

app.queue(concurrency_count=511, max_size=1022).launch(
    server_name="0.0.0.0",
    inbrowser=True,
    server_port=infer_ttswebui,
    quiet=True,
)