ApplioC / rvc /infer /infer.py
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import os
import sys
import torch
import numpy as np
import soundfile as sf
from vc_infer_pipeline import VC
from rvc.lib.utils import load_audio
from rvc.lib.tools.split_audio import process_audio, merge_audio
from fairseq import checkpoint_utils
from rvc.lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from rvc.configs.config import Config
config = Config()
torch.manual_seed(114514)
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=None,
f0_file=None,
f0_method=None,
file_index=None,
index_rate=None,
resample_sr=0,
rms_mix_rate=1,
protect=0.33,
hop_length=None,
output_path=None,
split_audio=False,
):
global tgt_sr, net_g, vc, hubert_model, version
if input_audio_path is None:
return "Please, load an audio!", None
f0_up_key = int(f0_up_key)
try:
audio = load_audio(input_audio_path, 16000)
audio_max = np.abs(audio).max() / 0.95
if audio_max > 1:
audio /= audio_max
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 tgt_sr != resample_sr >= 16000:
tgt_sr = resample_sr
if split_audio == "True":
result, new_dir_path = process_audio(input_audio_path)
if result == "Error":
return "Error with Split Audio", None
dir_path = (
new_dir_path.strip(" ").strip('"').strip("\n").strip('"').strip(" ")
)
if dir_path != "":
paths = [
os.path.join(root, name)
for root, _, files in os.walk(dir_path, topdown=False)
for name in files
if name.endswith(".wav") and root == dir_path
]
try:
for path in paths:
info, opt = vc_single(
sid,
path,
f0_up_key,
None,
f0_method,
file_index,
index_rate,
resample_sr,
rms_mix_rate,
protect,
hop_length,
path,
False,
)
# new_dir_path
except Exception as error:
print(error)
return "Error", None
print("Finished processing segmented audio, now merging audio...")
merge_timestamps_file = os.path.join(
os.path.dirname(new_dir_path),
f"{os.path.basename(input_audio_path).split('.')[0]}_timestamps.txt",
)
tgt_sr, audio_opt = merge_audio(merge_timestamps_file)
else:
audio_opt = vc.pipeline(
hubert_model,
net_g,
sid,
audio,
input_audio_path,
f0_up_key,
f0_method,
file_index,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
hop_length,
f0_file=f0_file,
)
if output_path is not None:
sf.write(output_path, audio_opt, tgt_sr, format="WAV")
return (tgt_sr, audio_opt)
except Exception as error:
print(error)
def get_vc(weight_root, sid):
global n_spk, tgt_sr, net_g, vc, cpt, version
if sid == "" or sid == []:
global hubert_model
if hubert_model is not None:
print("clean_empty_cache")
del net_g, n_spk, vc, hubert_model, tgt_sr # ,cpt
hubert_model = net_g = n_spk = vc = hubert_model = tgt_sr = None
if torch.cuda.is_available():
torch.cuda.empty_cache()
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, cpt
if torch.cuda.is_available():
torch.cuda.empty_cache()
cpt = None
person = weight_root
cpt = torch.load(person, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0]
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(config.device)
if config.is_half:
net_g = net_g.half()
else:
net_g = net_g.float()
vc = VC(tgt_sr, config)
n_spk = cpt["config"][-3]
f0up_key = sys.argv[1]
filter_radius = sys.argv[2]
index_rate = float(sys.argv[3])
hop_length = sys.argv[4]
f0method = sys.argv[5]
audio_input_path = sys.argv[6]
audio_output_path = sys.argv[7]
model_path = sys.argv[8]
index_path = sys.argv[9]
try:
split_audio = sys.argv[10]
except IndexError:
split_audio = None
sid = f0up_key
input_audio = audio_input_path
f0_pitch = f0up_key
f0_file = None
f0_method = f0method
file_index = index_path
index_rate = index_rate
output_file = audio_output_path
split_audio = split_audio
get_vc(model_path, 0)
try:
result, audio_opt = vc_single(
sid=0,
input_audio_path=input_audio,
f0_up_key=f0_pitch,
f0_file=None,
f0_method=f0_method,
file_index=file_index,
index_rate=index_rate,
hop_length=hop_length,
output_path=output_file,
split_audio=split_audio,
)
if os.path.exists(output_file) and os.path.getsize(output_file) > 0:
message = result
else:
message = result
print(f"Conversion completed. Output file: '{output_file}'")
except Exception as error:
print(f"Voice conversion failed: {error}")