310 lines
16 KiB
Python
310 lines
16 KiB
Python
import io
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import os
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# os.system("wget -P cvec/ https://huggingface.co/spaces/innnky/nanami/resolve/main/checkpoint_best_legacy_500.pt")
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import gradio as gr
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import gradio.processing_utils as gr_pu
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import librosa
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import numpy as np
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import soundfile
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from inference.infer_tool import Svc
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import logging
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import re
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import json
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import subprocess
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import edge_tts
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import asyncio
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from scipy.io import wavfile
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import librosa
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import torch
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import time
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import traceback
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from itertools import chain
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from utils import mix_model
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logging.getLogger('numba').setLevel(logging.WARNING)
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logging.getLogger('markdown_it').setLevel(logging.WARNING)
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logging.getLogger('urllib3').setLevel(logging.WARNING)
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logging.getLogger('matplotlib').setLevel(logging.WARNING)
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logging.getLogger('multipart').setLevel(logging.WARNING)
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model = None
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spk = None
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debug = False
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cuda = {}
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if torch.cuda.is_available():
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for i in range(torch.cuda.device_count()):
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device_name = torch.cuda.get_device_properties(i).name
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cuda[f"CUDA:{i} {device_name}"] = f"cuda:{i}"
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def upload_mix_append_file(files,sfiles):
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try:
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if(sfiles == None):
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file_paths = [file.name for file in files]
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else:
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file_paths = [file.name for file in chain(files,sfiles)]
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p = {file:100 for file in file_paths}
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return file_paths,mix_model_output1.update(value=json.dumps(p,indent=2))
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except Exception as e:
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if debug: traceback.print_exc()
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raise gr.Error(e)
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def mix_submit_click(js,mode):
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try:
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assert js.lstrip()!=""
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modes = {"凸组合":0, "线性组合":1}
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mode = modes[mode]
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data = json.loads(js)
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data = list(data.items())
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model_path,mix_rate = zip(*data)
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path = mix_model(model_path,mix_rate,mode)
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return f"成功，文件被保存在了{path}"
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except Exception as e:
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if debug: traceback.print_exc()
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raise gr.Error(e)
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def updata_mix_info(files):
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try:
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if files == None : return mix_model_output1.update(value="")
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p = {file.name:100 for file in files}
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return mix_model_output1.update(value=json.dumps(p,indent=2))
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except Exception as e:
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if debug: traceback.print_exc()
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raise gr.Error(e)
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def modelAnalysis(model_path,config_path,cluster_model_path,device,enhance):
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global model
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try:
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device = cuda[device] if "CUDA" in device else device
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model = Svc(model_path.name, config_path.name, device=device if device!="Auto" else None, cluster_model_path = cluster_model_path.name if cluster_model_path != None else "",nsf_hifigan_enhance=enhance)
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spks = list(model.spk2id.keys())
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device_name = torch.cuda.get_device_properties(model.dev).name if "cuda" in str(model.dev) else str(model.dev)
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msg = f"成功加载模型到设备{device_name}上\n"
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if cluster_model_path is None:
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msg += "未加载聚类模型\n"
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else:
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msg += f"聚类模型{cluster_model_path.name}加载成功\n"
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msg += "当前模型的可用音色：\n"
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for i in spks:
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msg += i + " "
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return sid.update(choices = spks,value=spks[0]), msg
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except Exception as e:
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if debug: traceback.print_exc()
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raise gr.Error(e)
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def modelUnload():
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global model
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if model is None:
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return sid.update(choices = [],value=""),"没有模型需要卸载!"
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else:
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model.unload_model()
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model = None
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torch.cuda.empty_cache()
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return sid.update(choices = [],value=""),"模型卸载完毕!"
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def vc_fn(sid, input_audio, vc_transform, auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,F0_mean_pooling,enhancer_adaptive_key):
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global model
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try:
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if input_audio is None:
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raise gr.Error("你需要上传音频")
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if model is None:
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raise gr.Error("你需要指定模型")
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sampling_rate, audio = input_audio
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# print(audio.shape,sampling_rate)
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audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
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if len(audio.shape) > 1:
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audio = librosa.to_mono(audio.transpose(1, 0))
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temp_path = "temp.wav"
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soundfile.write(temp_path, audio, sampling_rate, format="wav")
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_audio = model.slice_inference(temp_path, sid, vc_transform, slice_db, cluster_ratio, auto_f0, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,F0_mean_pooling,enhancer_adaptive_key)
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model.clear_empty()
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os.remove(temp_path)
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#构建保存文件的路径，并保存到results文件夹内
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try:
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timestamp = str(int(time.time()))
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filename = sid + "_" + timestamp + ".wav"
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output_file = os.path.join("./results", filename)
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soundfile.write(output_file, _audio, model.target_sample, format="wav")
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return f"推理成功，音频文件保存为results/{filename}", (model.target_sample, _audio)
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except Exception as e:
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if debug: traceback.print_exc()
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raise gr.Error(e)
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except Exception as e:
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if debug: traceback.print_exc()
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raise gr.Error(e)
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def tts_func(_text,_rate,_voice):
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#使用edge-tts把文字转成音频
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# voice = "zh-CN-XiaoyiNeural"#女性，较高音
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# voice = "zh-CN-YunxiNeural"#男性
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voice = "zh-CN-YunxiNeural"#男性
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if ( _voice == "女" ) : voice = "zh-CN-XiaoyiNeural"
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output_file = _text[0:10]+".wav"
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# communicate = edge_tts.Communicate(_text, voice)
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# await communicate.save(output_file)
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if _rate>=0:
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ratestr="+{:.0%}".format(_rate)
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elif _rate<0:
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ratestr="{:.0%}".format(_rate)#减号自带
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p=subprocess.Popen("edge-tts "+
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" --text "+_text+
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" --write-media "+output_file+
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" --voice "+voice+
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" --rate="+ratestr
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,shell=True,
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stdout=subprocess.PIPE,
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stdin=subprocess.PIPE)
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p.wait()
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return output_file
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def text_clear(text):
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return re.sub(r"[\n\,\(\) ]", "", text)
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def vc_fn2(sid, input_audio, vc_transform, auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,text2tts,tts_rate,tts_voice,F0_mean_pooling,enhancer_adaptive_key):
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#使用edge-tts把文字转成音频
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text2tts=text_clear(text2tts)
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output_file=tts_func(text2tts,tts_rate,tts_voice)
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#调整采样率
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sr2=44100
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wav, sr = librosa.load(output_file)
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wav2 = librosa.resample(wav, orig_sr=sr, target_sr=sr2)
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save_path2= text2tts[0:10]+"_44k"+".wav"
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wavfile.write(save_path2,sr2,
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(wav2 * np.iinfo(np.int16).max).astype(np.int16)
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)
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#读取音频
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sample_rate, data=gr_pu.audio_from_file(save_path2)
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vc_input=(sample_rate, data)
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a,b=vc_fn(sid, vc_input, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,F0_mean_pooling,enhancer_adaptive_key)
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os.remove(output_file)
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os.remove(save_path2)
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return a,b
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def debug_change():
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global debug
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debug = debug_button.value
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with gr.Blocks(
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theme=gr.themes.Base(
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primary_hue = gr.themes.colors.green,
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font=["Source Sans Pro", "Arial", "sans-serif"],
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font_mono=['JetBrains mono', "Consolas", 'Courier New']
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),
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) as app:
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with gr.Tabs():
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with gr.TabItem("推理"):
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gr.Markdown(value="""
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So-vits-svc 4.0 推理 webui
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""")
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with gr.Row(variant="panel"):
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with gr.Column():
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gr.Markdown(value="""
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<font size=2> 模型设置</font>
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""")
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model_path = gr.File(label="选择模型文件")
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config_path = gr.File(label="选择配置文件")
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cluster_model_path = gr.File(label="选择聚类模型文件（没有可以不选）")
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device = gr.Dropdown(label="推理设备，默认为自动选择CPU和GPU", choices=["Auto",*cuda.keys(),"CPU"], value="Auto")
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enhance = gr.Checkbox(label="是否使用NSF_HIFIGAN增强,该选项对部分训练集少的模型有一定的音质增强效果，但是对训练好的模型有反面效果，默认关闭", value=False)
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with gr.Column():
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gr.Markdown(value="""
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<font size=3>左侧文件全部选择完毕后(全部文件模块显示download)，点击“加载模型”进行解析：</font>
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""")
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model_load_button = gr.Button(value="加载模型", variant="primary")
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model_unload_button = gr.Button(value="卸载模型", variant="primary")
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sid = gr.Dropdown(label="音色（说话人）")
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sid_output = gr.Textbox(label="Output Message")
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with gr.Row(variant="panel"):
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with gr.Column():
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gr.Markdown(value="""
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<font size=2> 推理设置</font>
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""")
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auto_f0 = gr.Checkbox(label="自动f0预测，配合聚类模型f0预测效果更好,会导致变调功能失效（仅限转换语音，歌声勾选此项会究极跑调）", value=False)
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F0_mean_pooling = gr.Checkbox(label="是否对F0使用均值滤波器(池化)，对部分哑音有改善。注意，启动该选项会导致推理速度下降，默认关闭", value=False)
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vc_transform = gr.Number(label="变调（整数，可以正负，半音数量，升高八度就是12）", value=0)
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cluster_ratio = gr.Number(label="聚类模型混合比例，0-1之间，0即不启用聚类。使用聚类模型能提升音色相似度，但会导致咬字下降（如果使用建议0.5左右）", value=0)
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slice_db = gr.Number(label="切片阈值", value=-40)
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noise_scale = gr.Number(label="noise_scale 建议不要动，会影响音质，玄学参数", value=0.4)
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with gr.Column():
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pad_seconds = gr.Number(label="推理音频pad秒数，由于未知原因开头结尾会有异响，pad一小段静音段后就不会出现", value=0.5)
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cl_num = gr.Number(label="音频自动切片，0为不切片，单位为秒(s)", value=0)
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lg_num = gr.Number(label="两端音频切片的交叉淡入长度，如果自动切片后出现人声不连贯可调整该数值，如果连贯建议采用默认值0，注意，该设置会影响推理速度，单位为秒/s", value=0)
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lgr_num = gr.Number(label="自动音频切片后，需要舍弃每段切片的头尾。该参数设置交叉长度保留的比例，范围0-1,左开右闭", value=0.75)
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enhancer_adaptive_key = gr.Number(label="使增强器适应更高的音域(单位为半音数)|默认为0", value=0)
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with gr.Tabs():
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with gr.TabItem("音频转音频"):
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vc_input3 = gr.Audio(label="选择音频")
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vc_submit = gr.Button("音频转换", variant="primary")
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with gr.TabItem("文字转音频"):
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text2tts=gr.Textbox(label="在此输入要转译的文字。注意，使用该功能建议打开F0预测，不然会很怪")
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tts_rate = gr.Number(label="tts语速", value=0)
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tts_voice = gr.Radio(label="性别",choices=["男","女"], value="男")
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vc_submit2 = gr.Button("文字转换", variant="primary")
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with gr.Row():
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with gr.Column():
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vc_output1 = gr.Textbox(label="Output Message")
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with gr.Column():
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vc_output2 = gr.Audio(label="Output Audio", interactive=False)
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with gr.TabItem("小工具/实验室特性"):
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gr.Markdown(value="""
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<font size=2> So-vits-svc 4.0 小工具/实验室特性</font>
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""")
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with gr.Tabs():
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with gr.TabItem("静态声线融合"):
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gr.Markdown(value="""
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<font size=2> 介绍:该功能可以将多个声音模型合成为一个声音模型(多个模型参数的凸组合或线性组合)，从而制造出现实中不存在的声线
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注意：
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1.该功能仅支持单说话人的模型
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2.如果强行使用多说话人模型，需要保证多个模型的说话人数量相同，这样可以混合同一个SpaekerID下的声音
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3.保证所有待混合模型的config.json中的model字段是相同的
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4.输出的混合模型可以使用待合成模型的任意一个config.json，但聚类模型将不能使用
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5.批量上传模型的时候最好把模型放到一个文件夹选中后一起上传
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6.混合比例调整建议大小在0-100之间，也可以调为其他数字，但在线性组合模式下会出现未知的效果
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7.混合完毕后，文件将会保存在项目根目录中，文件名为output.pth
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8.凸组合模式会将混合比例执行Softmax使混合比例相加为1，而线性组合模式不会
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</font>
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""")
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mix_model_path = gr.Files(label="选择需要混合模型文件")
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mix_model_upload_button = gr.UploadButton("选择/追加需要混合模型文件", file_count="multiple", variant="primary")
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mix_model_output1 = gr.Textbox(
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label="混合比例调整，单位/%",
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interactive = True
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)
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mix_mode = gr.Radio(choices=["凸组合", "线性组合"], label="融合模式",value="凸组合",interactive = True)
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mix_submit = gr.Button("声线融合启动", variant="primary")
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mix_model_output2 = gr.Textbox(
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label="Output Message"
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)
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mix_model_path.change(updata_mix_info,[mix_model_path],[mix_model_output1])
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mix_model_upload_button.upload(upload_mix_append_file, [mix_model_upload_button,mix_model_path], [mix_model_path,mix_model_output1])
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mix_submit.click(mix_submit_click, [mix_model_output1,mix_mode], [mix_model_output2])
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with gr.Tabs():
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with gr.Row(variant="panel"):
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with gr.Column():
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gr.Markdown(value="""
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<font size=2> WebUI设置</font>
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""")
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debug_button = gr.Checkbox(label="Debug模式，如果向社区反馈BUG需要打开，打开后控制台可以显示具体错误提示", value=debug)
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vc_submit.click(vc_fn, [sid, vc_input3, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,F0_mean_pooling,enhancer_adaptive_key], [vc_output1, vc_output2])
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vc_submit2.click(vc_fn2, [sid, vc_input3, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,text2tts,tts_rate,tts_voice,F0_mean_pooling,enhancer_adaptive_key], [vc_output1, vc_output2])
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debug_button.change(debug_change,[],[])
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model_load_button.click(modelAnalysis,[model_path,config_path,cluster_model_path,device,enhance],[sid,sid_output])
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model_unload_button.click(modelUnload,[],[sid,sid_output])
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app.launch()
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