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rvc_infer.py
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rvc_infer.py
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import os,sys,pdb,torch
now_dir = os.getcwd()
sys.path.append(now_dir)
import argparse
import glob
import sys
import torch
import numpy as np
import yaml
import pkg_resources
import logging
from multiprocessing import cpu_count
from vc_infer_pipeline import VC
from lib.infer_pack.models import SynthesizerTrnMs256NSFsid, SynthesizerTrnMs256NSFsid_nono, SynthesizerTrnMs768NSFsid, SynthesizerTrnMs768NSFsid_nono
from lib.audio import load_audio
from fairseq import checkpoint_utils
from scipy.io import wavfile
class Config:
def __init__(self,device,is_half):
self.device = device
self.is_half = is_half
self.n_cpu = 0
self.gpu_name = None
self.gpu_mem = None
self.x_pad, self.x_query, self.x_center, self.x_max = self.device_config()
def device_config(self) -> tuple:
if torch.cuda.is_available():
i_device = int(self.device.split(":")[-1])
self.gpu_name = torch.cuda.get_device_name(i_device)
if (
("16" in self.gpu_name and "V100" not in self.gpu_name.upper())
or "P40" in self.gpu_name.upper()
or "1060" in self.gpu_name
or "1070" in self.gpu_name
or "1080" in self.gpu_name
):
print("16系/10系显卡和P40强制单精度")
self.is_half = False
for config_file in ["32k.json", "40k.json", "48k.json"]:
with open(f"configs/{config_file}", "r") as f:
strr = f.read().replace("true", "false")
with open(f"configs/{config_file}", "w") as f:
f.write(strr)
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
else:
self.gpu_name = None
self.gpu_mem = int(
torch.cuda.get_device_properties(i_device).total_memory
/ 1024
/ 1024
/ 1024
+ 0.4
)
if self.gpu_mem <= 4:
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
elif torch.backends.mps.is_available():
print("没有发现支持的N卡, 使用MPS进行推理")
self.device = "mps"
else:
print("没有发现支持的N卡, 使用CPU进行推理")
self.device = "cpu"
self.is_half = True
if self.n_cpu == 0:
self.n_cpu = cpu_count()
if self.is_half:
# 6G显存配置
x_pad = 3
x_query = 10
x_center = 60
x_max = 65
else:
# 5G显存配置
x_pad = 1
x_query = 6
x_center = 38
x_max = 41
if self.gpu_mem != None and self.gpu_mem <= 4:
x_pad = 1
x_query = 5
x_center = 30
x_max = 32
return x_pad, x_query, x_center, x_max
def get_path(name):
'''
Built to get the path of a file based on where the initial script is being run from.
Args:
- name(str) : name of the file/folder
'''
return os.path.join(os.getcwd(), name)
def create_directory(name):
'''
Creates a directory based on the location from which the script is run. Relies on
get_path()
Args:
- name(str) : name of the file/folder
'''
dir_name = get_path(name)
if not os.path.exists(dir_name):
os.makedirs(dir_name)
def load_hubert(file_path="hubert_base.pt"):
'''
Args:
file_fath (str) : Direct path location to the hubert_base. If not specified, defaults to top level directory.
'''
global hubert_model
file_path = file_path
models, _, _ = checkpoint_utils.load_model_ensemble_and_task(
[file_path],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(config.device)
if config.is_half:
hubert_model = hubert_model.half()
else:
hubert_model = hubert_model.float()
hubert_model.eval()
def vc_single(
sid,
input_audio_path,
f0_up_key,
f0_file,
f0_method,
file_index,
file_index2,
# file_big_npy,
index_rate,
filter_radius,
resample_sr,
rms_mix_rate,
protect,
): # spk_item, input_audio0, vc_transform0,f0_file,f0method0
global tgt_sr, net_g, vc, hubert_model, version
f0_file = None
if input_audio_path is None:
return "You need to upload an audio", None
f0_up_key = int(f0_up_key)
audio = load_audio(input_audio_path, 16000)
audio_max = np.abs(audio).max() / 0.95
if audio_max > 1:
audio /= audio_max
times = [0, 0, 0]
if not hubert_model:
load_hubert()
if_f0 = cpt.get("f0", 1)
file_index = (
(
file_index.strip(" ")
.strip('"')
.strip("\n")
.strip('"')
.strip(" ")
.replace("trained", "added")
)
if file_index != ""
else file_index2
) # 防止小白写错,自动帮他替换掉
# file_big_npy = (
# file_big_npy.strip(" ").strip('"').strip("\n").strip('"').strip(" ")
# )
audio_opt = vc.pipeline(
hubert_model,
net_g,
sid,
audio,
input_audio_path,
times,
f0_up_key,
f0_method,
file_index,
# file_big_npy,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
f0_file=f0_file,
)
return audio_opt
def get_vc(model_path):
global n_spk,tgt_sr,net_g,vc,cpt,device,is_half, version
print("loading pth %s"%model_path)
cpt = torch.load(model_path, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3]=cpt["weight"]["emb_g.weight"].shape[0]#n_spk
if_f0=cpt.get("f0",1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(
*cpt["config"], is_half=config.is_half
)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1:
net_g = SynthesizerTrnMs768NSFsid(
*cpt["config"], is_half=config.is_half
)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del net_g.enc_q
print(net_g.load_state_dict(cpt["weight"], strict=False))
net_g.eval().to(device)
if (is_half):net_g = net_g.half()
else:net_g = net_g.float()
vc = VC(tgt_sr, config)
n_spk=cpt["config"][-3]
# return {"visible": True,"maximum": n_spk, "__type__": "update"}
def load_config():
current_dir = os.path.dirname(os.path.abspath(__file__))
yaml_file = os.path.join(current_dir, "rvc.yaml")
with open(yaml_file, "r") as file:
rvc_conf = yaml.safe_load(file)
return rvc_conf
def rvc_convert(model_path,
f0_up_key=0,
input_path=None,
output_dir_path=None,
_is_half="False",
f0method="rmvpe",
file_index="",
file_index2="",
index_rate=1,
filter_radius=3,
resample_sr=0,
rms_mix_rate=0.5,
protect=0.33,
verbose=False
):
'''
Function to call for the rvc voice conversion. All parameters are the same present in that of the webui
Args:
model_path (str) : path to the rvc voice model you're using
f0_up_key (int) : transpose of the audio file, changes pitch (positive makes voice higher pitch)
input_path (str) : path to audio file (use wav file)
output_dir_path (str) : path to output directory, defaults to parent directory in output folder
_is_half (str) : Determines half-precision
f0method (str) : picks which f0 method to use: dio, harvest, crepe, rmvpe (requires rmvpe.pt)
file_index (str) : path to file_index, defaults to None
file_index2 (str) : path to file_index2, defaults to None. #honestly don't know what this is for
index_rate (int) : strength of the index file if provided
filter_radius (int) : if >=3: apply median filtering to the harvested pitch results. The value represents the filter radius and can reduce breathiness.
resample_sr (int) : quality at which to resample audio to, defaults to no resample
rmx_mix_rate (int) : adjust the volume envelope scaling. Closer to 0, the more it mimicks the volume of the original vocals. Can help mask noise and make volume sound more natural when set relatively low. Closer to 1 will be more of a consistently loud volume
protect (int) : protect voiceless consonants and breath sounds to prevent artifacts such as tearing in electronic music. Set to 0.5 to disable. Decrease the value to increase protection, but it may reduce indexing accuracy
Returns:
output_file_path (str) : file path and name of tshe output wav file
'''
global config, now_dir, hubert_model, tgt_sr, net_g, vc, cpt, device, is_half, version
if torch.cuda.is_available():
device = "cuda:0"
elif torch.backends.mps.is_available():
device = "mps:0"
else:
print("Cuda or MPS not detected")
if not verbose:
logging.getLogger('fairseq').setLevel(logging.ERROR)
logging.getLogger('rvc').setLevel(logging.ERROR)
is_half = _is_half
if output_dir_path == None:
output_dir_path = "output"
output_file_name = "out.wav"
output_dir = os.getcwd()
output_file_path = os.path.join(output_dir,output_dir_path, output_file_name)
else:
# Mainly for Jarod's Vivy project, specify entire path + wav name
output_file_path = output_dir_path
pass
create_directory(output_dir_path)
output_dir = get_path(output_dir_path)
if(is_half.lower() == 'true'):
is_half = True
else:
is_half = False
config=Config(device,is_half)
now_dir=os.getcwd()
sys.path.append(now_dir)
hubert_model=None
get_vc(model_path)
wav_opt=vc_single(0,input_path,f0_up_key,None,f0method,file_index,file_index2,index_rate,filter_radius,resample_sr,rms_mix_rate,protect)
wavfile.write(output_file_path, tgt_sr, wav_opt)
print(f"\nFile finished writing to: {output_file_path}")
return output_file_path
def main():
# Need to comment out yaml setting for input audio
rvc_convert(model_path="models\\ado.pth", input_path="delilah.wav")
if __name__ == "__main__":
main()