Update rvc-v2 to f63783c34888a80db37a28c68f4b4765040b6257

https://github.com/RVC-Project/Retrieval-based-Voice-Conversion-WebUI
This commit is contained in:
Brandon Thomas
2023-07-17 14:31:22 -04:00
parent 7eb9920534
commit ca4ea0fbcd
3 changed files with 29 additions and 231 deletions
@@ -1,222 +0,0 @@
# From https://github.com/RVC-Project/Retrieval-based-Voice-Conversion-WebUI/issues/299#issuecomment-1562141634
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
from multiprocessing import cpu_count
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
f0_up_key=int(sys.argv[1]) #transpose value
input_path=sys.argv[2]
opt_path=sys.argv[3]
model_path=sys.argv[4]
device=sys.argv[5]
is_half=sys.argv[6]
f0method=sys.argv[7] #pm or harvest
file_index=sys.argv[8] #.index file
file_index2=sys.argv[9]
index_rate=float(sys.argv[10]) #search feature ratio
filter_radius=float(sys.argv[11]) #median filter
resample_sr=float(sys.argv[12]) #resample audio in post processing
rms_mix_rate=float(sys.argv[13]) #search feature
print(sys.argv)
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)
from vc_infer_pipeline import VC
from infer_pack.models import SynthesizerTrnMs256NSFsid, SynthesizerTrnMs256NSFsid_nono
from my_utils import load_audio
from fairseq import checkpoint_utils
from scipy.io import wavfile
hubert_model=None
def load_hubert():
global hubert_model
models, _, _ = checkpoint_utils.load_model_ensemble_and_task(
["hubert_base.pt"],
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=0,
input_audio_path=None,
f0_up_key=0,
f0_file=None,
f0_method="pm",
file_index="", #.index file
file_index2="",
# file_big_npy,
index_rate=1.0,
filter_radius=3,
resample_sr=0,
rms_mix_rate=1.0,
):
global tgt_sr, net_g, vc, hubert_model, version
if input_audio_path is None:
return "You need to upload an audio file", 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 hubert_model == None:
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
)
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,
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(if_f0==1):
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=is_half)
else:
net_g = SynthesizerTrnMs256NSFsid_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"}
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)
wavfile.write(opt_path, tgt_sr, wav_opt)
+29 -8
View File
@@ -1,5 +1,8 @@
import os, sys
if sys.platform == "darwin":
os.environ["PYTORCH_ENABLE_MPS_FALLBACK"] = "1"
now_dir = os.getcwd()
sys.path.append(now_dir)
import multiprocessing
@@ -45,7 +48,11 @@ if __name__ == "__main__":
from i18n import I18nAuto
i18n = I18nAuto()
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
device = torch.device(
"cuda"
if torch.cuda.is_available()
else ("mps" if torch.backends.mps.is_available() else "cpu")
)
current_dir = os.getcwd()
inp_q = Queue()
opt_q = Queue()
@@ -441,8 +448,9 @@ if __name__ == "__main__":
"""
接受音频输入
"""
channels = 1 if sys.platform == "darwin" else 2
with sd.Stream(
channels=2,
channels=channels,
callback=self.audio_callback,
blocksize=self.block_frame,
samplerate=self.config.samplerate,
@@ -524,6 +532,9 @@ if __name__ == "__main__":
)
+ 1e-8
)
if sys.platform == "darwin":
cor_nom = cor_nom.cpu()
cor_den = cor_den.cpu()
sola_offset = torch.argmax(cor_nom[0, 0] / cor_den[0, 0])
print("sola offset: " + str(int(sola_offset)))
self.output_wav[:] = infer_wav[sola_offset : sola_offset + self.block_frame]
@@ -545,14 +556,24 @@ if __name__ == "__main__":
infer_wav[-self.crossfade_frame :] * self.fade_out_window
)
if self.config.O_noise_reduce:
outdata[:] = np.tile(
nr.reduce_noise(
if sys.platform == "darwin":
noise_reduced_signal = nr.reduce_noise(
y=self.output_wav[:].cpu().numpy(), sr=self.config.samplerate
),
(2, 1),
).T
)
outdata[:] = noise_reduced_signal[:, np.newaxis]
else:
outdata[:] = np.tile(
nr.reduce_noise(
y=self.output_wav[:].cpu().numpy(),
sr=self.config.samplerate,
),
(2, 1),
).T
else:
outdata[:] = self.output_wav[:].repeat(2, 1).t().cpu().numpy()
if sys.platform == "darwin":
outdata[:] = self.output_wav[:].cpu().numpy()[:, np.newaxis]
else:
outdata[:] = self.output_wav[:].repeat(2, 1).t().cpu().numpy()
total_time = time.perf_counter() - start_time
self.window["infer_time"].update(int(total_time * 1000))
print("infer time:" + str(total_time))
@@ -21,7 +21,6 @@ praat-parselmouth>=0.4.2
Pillow>=9.1.1
resampy>=0.4.2
scikit-learn
#starlette>=0.25.0
tensorboard
tqdm>=4.63.1
tornado>=6.1