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mirror of https://github.com/fumiama/Retrieval-based-Voice-Conversion-WebUI.git synced 2026-06-08 12:00:49 +08:00

optimize: some training optimizations (#95)

* optimzie(train&uvr5): rm sf & simp. AudioPre

* fix(audio): too many mallocs

* feat(audio): load_audio support stereo

* fix(audio): float32 wav saving

* fix(train): missing ckpt var
This commit is contained in:
源文雨
2024-11-28 03:20:14 +09:00
committed by GitHub
parent f4644ec1ec
commit a8783c6639
19 changed files with 163 additions and 433 deletions

View File

@@ -5,12 +5,10 @@ logger = logging.getLogger(__name__)
import librosa
import numpy as np
import soundfile as sf
import torch
from tqdm import tqdm
import av
from infer.lib.audio import downsample_audio
from infer.lib.audio import downsample_audio, save_audio
cpu = torch.device("cpu")
@@ -210,15 +208,13 @@ class Predictor:
sources = self.demix(mix.T)
opt = sources[0].T
if format in ["wav", "flac"]:
sf.write(
"%s/%s_main_vocal.%s" % (vocal_root, basename, format), mix - opt, rate
)
sf.write("%s/%s_others.%s" % (others_root, basename, format), opt, rate)
save_audio("%s/vocal_%s.%s" % (vocal_root, basename, format), mix - opt, rate)
save_audio("%s/instrument_%s.%s" % (others_root, basename, format), opt, rate)
else:
path_vocal = "%s/%s_main_vocal.wav" % (vocal_root, basename)
path_other = "%s/%s_others.wav" % (others_root, basename)
sf.write(path_vocal, mix - opt, rate)
sf.write(path_other, opt, rate)
path_vocal = "%s/vocal_%s.wav" % (vocal_root, basename)
path_other = "%s/instrument_%s.wav" % (others_root, basename)
save_audio(path_vocal, opt, rate)
save_audio(path_other, opt, rate)
opt_path_vocal = path_vocal[:-4] + ".%s" % format
opt_path_other = path_other[:-4] + ".%s" % format
downsample_audio(path_vocal, opt_path_vocal, format)

View File

@@ -9,7 +9,7 @@ import torch
from configs import Config
from infer.modules.uvr5.mdxnet import MDXNetDereverb
from infer.modules.uvr5.vr import AudioPre, AudioPreDeEcho
from infer.modules.uvr5.vr import AudioPre
config = Config()
@@ -27,8 +27,7 @@ def uvr(model_name, inp_root, save_root_vocal, paths, save_root_ins, agg, format
if model_name == "onnx_dereverb_By_FoxJoy":
pre_fun = MDXNetDereverb(15, config.device)
else:
func = AudioPre if "DeEcho" not in model_name else AudioPreDeEcho
pre_fun = func(
pre_fun = AudioPre(
agg=int(agg),
model_path=os.path.join(
os.getenv("weight_uvr5_root"), model_name + ".pth"
@@ -72,18 +71,10 @@ def uvr(model_name, inp_root, save_root_vocal, paths, save_root_ins, agg, format
infos.append("%s->Success" % (os.path.basename(inp_path)))
yield "\n".join(infos)
except:
try:
if done == 0:
pre_fun._path_audio_(
inp_path, save_root_ins, save_root_vocal, format0
)
infos.append("%s->Success" % (os.path.basename(inp_path)))
yield "\n".join(infos)
except:
infos.append(
"%s->%s" % (os.path.basename(inp_path), traceback.format_exc())
)
yield "\n".join(infos)
infos.append(
"%s->%s" % (os.path.basename(inp_path), traceback.format_exc())
)
yield "\n".join(infos)
except:
infos.append(traceback.format_exc())
yield "\n".join(infos)

View File

@@ -5,8 +5,7 @@ logger = logging.getLogger(__name__)
import librosa
import numpy as np
import soundfile as sf
from infer.lib.audio import downsample_audio
from infer.lib.audio import downsample_audio, save_audio
import torch
from infer.lib.uvr5_pack.lib_v5 import nets_123821KB as Nets
@@ -20,6 +19,8 @@ class AudioPre:
def __init__(self, agg, model_path, device, is_half, tta=False):
self.model_path = model_path
self.device = device
self.is_de_echo = "DeEcho" in model_path
self.is_reverse = self.is_de_echo or "HP3" in model_path
self.data = {
# Processing Options
"postprocess": False,
@@ -29,8 +30,13 @@ class AudioPre:
"agg": agg,
"high_end_process": "mirroring",
}
mp = ModelParameters("infer/lib/uvr5_pack/lib_v5/modelparams/4band_v2.json")
model = Nets.CascadedASPPNet(mp.param["bins"] * 2)
if self.is_de_echo:
mp = ModelParameters("infer/lib/uvr5_pack/lib_v5/modelparams/4band_v3.json")
nout = 64 if "DeReverb" in model_path else 48
model = CascadedNet(mp.param["bins"] * 2, nout)
else:
mp = ModelParameters("infer/lib/uvr5_pack/lib_v5/modelparams/4band_v2.json")
model = Nets.CascadedASPPNet(mp.param["bins"] * 2)
cpk = torch.load(model_path, map_location="cpu")
model.load_state_dict(cpk)
model.eval()
@@ -123,30 +129,28 @@ class AudioPre:
else:
wav_instrument = spec_utils.cmb_spectrogram_to_wave(y_spec_m, self.mp)
logger.info("%s instruments done" % name)
head = "instrument_"
if self.is_reverse:
head = "vocal_"
else:
head = "instrument_"
if format in ["wav", "flac"]:
sf.write(
os.path.join(
save_audio(os.path.join(
ins_root,
head + "{}_{}.{}".format(name, self.data["agg"], format),
),
(np.array(wav_instrument) * 32768).astype("int16"),
self.mp.param["sr"],
) #
), wav_instrument, self.mp.param["sr"])
else:
path = os.path.join(
ins_root, head + "{}_{}.wav".format(name, self.data["agg"])
)
sf.write(
path,
(np.array(wav_instrument) * 32768).astype("int16"),
self.mp.param["sr"],
)
save_audio(path, wav_instrument, self.mp.param["sr"])
if os.path.exists(path):
opt_format_path = path[:-4] + ".%s" % format
downsample_audio(path, opt_format_path, format)
if vocal_root is not None:
head = "vocal_"
if self.is_reverse:
head = "instrument_"
else:
head = "vocal_"
if self.data["high_end_process"].startswith("mirroring"):
input_high_end_ = spec_utils.mirroring(
self.data["high_end_process"], v_spec_m, input_high_end, self.mp
@@ -158,185 +162,15 @@ class AudioPre:
wav_vocals = spec_utils.cmb_spectrogram_to_wave(v_spec_m, self.mp)
logger.info("%s vocals done" % name)
if format in ["wav", "flac"]:
sf.write(
os.path.join(
save_audio(os.path.join(
vocal_root,
head + "{}_{}.{}".format(name, self.data["agg"], format),
),
(np.array(wav_vocals) * 32768).astype("int16"),
self.mp.param["sr"],
)
), wav_vocals, self.mp.param["sr"])
else:
path = os.path.join(
vocal_root, head + "{}_{}.wav".format(name, self.data["agg"])
)
sf.write(
path,
(np.array(wav_vocals) * 32768).astype("int16"),
self.mp.param["sr"],
)
opt_format_path = path[:-4] + ".%s" % format
downsample_audio(path, opt_format_path, format)
class AudioPreDeEcho:
def __init__(self, agg, model_path, device, is_half, tta=False):
self.model_path = model_path
self.device = device
self.data = {
# Processing Options
"postprocess": False,
"tta": tta,
# Constants
"window_size": 512,
"agg": agg,
"high_end_process": "mirroring",
}
mp = ModelParameters("infer/lib/uvr5_pack/lib_v5/modelparams/4band_v3.json")
nout = 64 if "DeReverb" in model_path else 48
model = CascadedNet(mp.param["bins"] * 2, nout)
cpk = torch.load(model_path, map_location="cpu")
model.load_state_dict(cpk)
model.eval()
if is_half:
model = model.half().to(device)
else:
model = model.to(device)
self.mp = mp
self.model = model
def _path_audio_(
self, music_file, vocal_root=None, ins_root=None, format="flac"
): # 3个VR模型vocal和ins是反的
if ins_root is None and vocal_root is None:
return "No save root."
name = os.path.basename(music_file)
if ins_root is not None:
os.makedirs(ins_root, exist_ok=True)
if vocal_root is not None:
os.makedirs(vocal_root, exist_ok=True)
X_wave, y_wave, X_spec_s, y_spec_s = {}, {}, {}, {}
bands_n = len(self.mp.param["band"])
# print(bands_n)
for d in range(bands_n, 0, -1):
bp = self.mp.param["band"][d]
if d == bands_n: # high-end band
(
X_wave[d],
_,
) = librosa.load( # 理论上librosa读取可能对某些音频有bug应该上ffmpeg读取但是太麻烦了弃坑
music_file,
sr=bp["sr"],
mono=False,
dtype=np.float32,
res_type=bp["res_type"],
)
if X_wave[d].ndim == 1:
X_wave[d] = np.asfortranarray([X_wave[d], X_wave[d]])
else: # lower bands
X_wave[d] = librosa.resample(
X_wave[d + 1],
orig_sr=self.mp.param["band"][d + 1]["sr"],
target_sr=bp["sr"],
res_type=bp["res_type"],
)
# Stft of wave source
X_spec_s[d] = spec_utils.wave_to_spectrogram_mt(
X_wave[d],
bp["hl"],
bp["n_fft"],
self.mp.param["mid_side"],
self.mp.param["mid_side_b2"],
self.mp.param["reverse"],
)
# pdb.set_trace()
if d == bands_n and self.data["high_end_process"] != "none":
input_high_end_h = (bp["n_fft"] // 2 - bp["crop_stop"]) + (
self.mp.param["pre_filter_stop"] - self.mp.param["pre_filter_start"]
)
input_high_end = X_spec_s[d][
:, bp["n_fft"] // 2 - input_high_end_h : bp["n_fft"] // 2, :
]
X_spec_m = spec_utils.combine_spectrograms(X_spec_s, self.mp)
aggresive_set = float(self.data["agg"] / 100)
aggressiveness = {
"value": aggresive_set,
"split_bin": self.mp.param["band"][1]["crop_stop"],
}
with torch.no_grad():
pred, X_mag, X_phase = inference(
X_spec_m, self.device, self.model, aggressiveness, self.data
)
# Postprocess
if self.data["postprocess"]:
pred_inv = np.clip(X_mag - pred, 0, np.inf)
pred = spec_utils.mask_silence(pred, pred_inv)
y_spec_m = pred * X_phase
v_spec_m = X_spec_m - y_spec_m
if ins_root is not None:
if self.data["high_end_process"].startswith("mirroring"):
input_high_end_ = spec_utils.mirroring(
self.data["high_end_process"], y_spec_m, input_high_end, self.mp
)
wav_instrument = spec_utils.cmb_spectrogram_to_wave(
y_spec_m, self.mp, input_high_end_h, input_high_end_
)
else:
wav_instrument = spec_utils.cmb_spectrogram_to_wave(y_spec_m, self.mp)
logger.info("%s instruments done" % name)
if format in ["wav", "flac"]:
sf.write(
os.path.join(
ins_root,
"vocal_{}_{}.{}".format(name, self.data["agg"], format),
),
(np.array(wav_instrument) * 32768).astype("int16"),
self.mp.param["sr"],
) #
else:
path = os.path.join(
ins_root, "vocal_{}_{}.wav".format(name, self.data["agg"])
)
sf.write(
path,
(np.array(wav_instrument) * 32768).astype("int16"),
self.mp.param["sr"],
)
if os.path.exists(path):
opt_format_path = path[:-4] + ".%s" % format
downsample_audio(path, opt_format_path, format)
if vocal_root is not None:
if self.data["high_end_process"].startswith("mirroring"):
input_high_end_ = spec_utils.mirroring(
self.data["high_end_process"], v_spec_m, input_high_end, self.mp
)
wav_vocals = spec_utils.cmb_spectrogram_to_wave(
v_spec_m, self.mp, input_high_end_h, input_high_end_
)
else:
wav_vocals = spec_utils.cmb_spectrogram_to_wave(v_spec_m, self.mp)
logger.info("%s vocals done" % name)
if format in ["wav", "flac"]:
sf.write(
os.path.join(
vocal_root,
"instrument_{}_{}.{}".format(name, self.data["agg"], format),
),
(np.array(wav_vocals) * 32768).astype("int16"),
self.mp.param["sr"],
)
else:
path = os.path.join(
vocal_root, "instrument_{}_{}.wav".format(name, self.data["agg"])
)
sf.write(
path,
(np.array(wav_vocals) * 32768).astype("int16"),
self.mp.param["sr"],
)
save_audio(path, wav_vocals, self.mp.param["sr"])
if os.path.exists(path):
opt_format_path = path[:-4] + ".%s" % format
downsample_audio(path, opt_format_path, format)