mirror of
https://github.com/serp-ai/bark-with-voice-clone.git
synced 2025-12-14 18:57:56 +01:00
170 lines
6.3 KiB
Python
170 lines
6.3 KiB
Python
import os,sys,pdb,torch
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now_dir = os.getcwd()
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sys.path.append(now_dir)
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import argparse
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import glob
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import sys
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import torch
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from multiprocessing import cpu_count
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import ffmpeg
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import numpy as np
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def load_audio(file, sr):
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try:
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# https://github.com/openai/whisper/blob/main/whisper/audio.py#L26
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# This launches a subprocess to decode audio while down-mixing and resampling as necessary.
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# Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.
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file = (
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file.strip(" ").strip('"').strip("\n").strip('"').strip(" ")
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) # 防止小白拷路径头尾带了空格和"和回车
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out, _ = (
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ffmpeg.input(file, threads=0)
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.output("-", format="f32le", acodec="pcm_f32le", ac=1, ar=sr)
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.run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
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)
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except Exception as e:
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raise RuntimeError(f"Failed to load audio: {e}")
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return np.frombuffer(out, np.float32).flatten()
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class Config:
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def __init__(self,device,is_half):
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self.device = device
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self.is_half = is_half
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self.n_cpu = 0
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self.gpu_name = None
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self.gpu_mem = None
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self.x_pad, self.x_query, self.x_center, self.x_max = self.device_config()
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def device_config(self) -> tuple:
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if torch.cuda.is_available():
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i_device = int(self.device.split(":")[-1])
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self.gpu_name = torch.cuda.get_device_name(i_device)
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if (
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("16" in self.gpu_name and "V100" not in self.gpu_name.upper())
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or "P40" in self.gpu_name.upper()
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or "1060" in self.gpu_name
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or "1070" in self.gpu_name
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or "1080" in self.gpu_name
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):
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print("16系/10系显卡和P40强制单精度")
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self.is_half = False
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for config_file in ["32k.json", "40k.json", "48k.json"]:
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with open(f"configs/{config_file}", "r") as f:
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strr = f.read().replace("true", "false")
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with open(f"configs/{config_file}", "w") as f:
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f.write(strr)
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with open("trainset_preprocess_pipeline_print.py", "r") as f:
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strr = f.read().replace("3.7", "3.0")
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with open("trainset_preprocess_pipeline_print.py", "w") as f:
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f.write(strr)
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else:
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self.gpu_name = None
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self.gpu_mem = int(
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torch.cuda.get_device_properties(i_device).total_memory
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/ 1024
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/ 1024
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/ 1024
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+ 0.4
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)
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if self.gpu_mem <= 4:
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with open("trainset_preprocess_pipeline_print.py", "r") as f:
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strr = f.read().replace("3.7", "3.0")
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with open("trainset_preprocess_pipeline_print.py", "w") as f:
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f.write(strr)
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elif torch.backends.mps.is_available():
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print("没有发现支持的N卡, 使用MPS进行推理")
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self.device = "mps"
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else:
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print("没有发现支持的N卡, 使用CPU进行推理")
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self.device = "cpu"
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self.is_half = True
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if self.n_cpu == 0:
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self.n_cpu = cpu_count()
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if self.is_half:
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# 6G显存配置
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x_pad = 3
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x_query = 10
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x_center = 60
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x_max = 65
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else:
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# 5G显存配置
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x_pad = 1
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x_query = 6
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x_center = 38
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x_max = 41
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if self.gpu_mem != None and self.gpu_mem <= 4:
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x_pad = 1
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x_query = 5
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x_center = 30
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x_max = 32
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return x_pad, x_query, x_center, x_max
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now_dir=os.getcwd()
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sys.path.append(now_dir)
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sys.path.append(os.path.join(now_dir,"Retrieval-based-Voice-Conversion-WebUI"))
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from vc_infer_pipeline import VC
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from lib.infer_pack.models import SynthesizerTrnMs256NSFsid, SynthesizerTrnMs256NSFsid_nono, SynthesizerTrnMs768NSFsid, SynthesizerTrnMs768NSFsid_nono
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from fairseq import checkpoint_utils
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from scipy.io import wavfile
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hubert_model=None
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def load_hubert():
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global hubert_model
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models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task(["hubert_base.pt"],suffix="",)
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hubert_model = models[0]
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hubert_model = hubert_model.to(device)
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if(is_half):hubert_model = hubert_model.half()
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else:hubert_model = hubert_model.float()
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hubert_model.eval()
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def vc_single(sid,input_audio,f0_up_key,f0_file,f0_method,file_index,index_rate,filter_radius=3,resample_sr=48000,rms_mix_rate=0.25, protect=0.33):
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global tgt_sr,net_g,vc,hubert_model
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if input_audio is None:return "You need to upload an audio", None
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f0_up_key = int(f0_up_key)
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audio=load_audio(input_audio,16000)
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times = [0, 0, 0]
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if(hubert_model==None):load_hubert()
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if_f0 = cpt.get("f0", 1)
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version = cpt.get("version")
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audio_opt=vc.pipeline(hubert_model,net_g,sid,audio,input_audio,times,f0_up_key,f0_method,file_index,index_rate,if_f0,filter_radius=filter_radius,tgt_sr=tgt_sr,resample_sr=resample_sr,rms_mix_rate=rms_mix_rate,version=version,protect=protect,f0_file=f0_file)
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# print(times)
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return audio_opt
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def get_vc(model_path, device_, is_half_):
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global n_spk,tgt_sr,net_g,vc,cpt,device,is_half
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device = device_
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is_half = is_half_
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config = Config(device, is_half)
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print("loading pth %s"%model_path)
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cpt = torch.load(model_path, map_location="cpu")
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tgt_sr = cpt["config"][-1]
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cpt["config"][-3]=cpt["weight"]["emb_g.weight"].shape[0]#n_spk
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if_f0=cpt.get("f0",1)
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version=cpt.get("version", "v2")
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if(if_f0==1):
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if version == "v1":
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net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=is_half)
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else:
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net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=is_half)
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else:
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if version == "v1":
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net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
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else:
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net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
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del net_g.enc_q
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print(net_g.load_state_dict(cpt["weight"], strict=False)) # 不加这一行清不干净,真奇葩
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net_g.eval().to(device)
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if (is_half):net_g = net_g.half()
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else:net_g = net_g.float()
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vc = VC(tgt_sr, config)
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n_spk=cpt["config"][-3]
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