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vc_infer_pipeline.py
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vc_infer_pipeline.py
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import numpy as np, torch, sys, os
from time import time as ttime
import torch.nn.functional as F
import scipy.signal as signal
import os, traceback, faiss, librosa
from scipy import signal
from lib.model_utils import load_hubert, change_rms
# from tqdm import tqdm
from pitch_extraction import FeatureExtractor
from lib.audio import MAX_INT16, load_input_audio, remix_audio
from lib import config, BASE_MODELS_DIR
from lib.utils import gc_collect, get_filenames
# torchcrepe = lazyload("torchcrepe") # Fork Feature. Crepe algo for training and preprocess
# torch = lazyload("torch")
# rmvpe = lazyload("rmvpe")
bh, ah = signal.butter(N=5, Wn=48, btype="high", fs=16000)
class VC(FeatureExtractor):
def vc(
self,
model,
net_g,
sid,
audio0,
pitch,
pitchf,
times,
index,
big_npy,
index_rate,
version,
protect,
): # ,file_index,file_big_npy
feats = torch.from_numpy(audio0)
if self.is_half:
feats = feats.half()
else:
feats = feats.float()
if feats.dim() == 2: # double channels
feats = feats.mean(-1)
assert feats.dim() == 1, feats.dim()
feats = feats.view(1, -1)
padding_mask = torch.BoolTensor(feats.shape).to(self.device).fill_(False)
inputs = {
"source": feats.to(self.device),
"padding_mask": padding_mask,
"output_layer": 9 if version == "v1" else 12,
}
with torch.no_grad():
logits = model.extract_features(**inputs)
feats = model.final_proj(logits[0]) if version == "v1" else logits[0]
if protect < 0.5 and pitch is not None and pitchf is not None:
feats0 = feats.clone()
if index is not None and big_npy is not None and index_rate > 0:
npy = feats[0].cpu().numpy()
if self.is_half:
npy = npy.astype("float32")
# _, I = index.search(npy, 1)
# npy = big_npy[I.squeeze()]
score, ix = index.search(npy, k=8)
weight = np.square(1 / score)
weight /= weight.sum(axis=1, keepdims=True)
npy = np.sum(big_npy[ix] * np.expand_dims(weight, axis=2), axis=1)
if self.is_half:
npy = npy.astype("float16")
feats = (
torch.from_numpy(npy).unsqueeze(0).to(self.device) * index_rate
+ (1 - index_rate) * feats
)
feats = F.interpolate(feats.permute(0, 2, 1), scale_factor=2).permute(0, 2, 1)
if protect < 0.5 and pitch != None and pitchf != None:
feats0 = F.interpolate(feats0.permute(0, 2, 1), scale_factor=2).permute(
0, 2, 1
)
p_len = min(audio0.shape[0] // self.window, feats.shape[1])
if pitch is not None and pitchf is not None:
pitch = pitch[:, :p_len]
pitchf = pitchf[:, :p_len]
if protect < 0.5:
pitchff = pitchf.clone()
pitchff[pitchf > 0] = 1
pitchff[pitchf < 1] = protect
pitchff = pitchff.unsqueeze(-1)
feats = feats * pitchff + feats0 * (1 - pitchff)
feats = feats.to(feats0.dtype)
p_len = torch.tensor([p_len], device=self.device).long()
with torch.no_grad():
if pitch != None and pitchf != None:
print("vc",feats.shape,pitch.shape,pitchf.shape)
audio1 = (
(net_g.infer(feats, p_len, pitch, pitchf, sid)[0][0, 0])
.data.cpu()
.float()
.numpy()
)
else:
audio1 = (
(net_g.infer(feats, p_len, sid)[0][0, 0]).data.cpu().float().numpy()
)
del feats, p_len, padding_mask
if torch.cuda.is_available():
torch.cuda.empty_cache()
return audio1
def pipeline(self, model, net_g, sid, audio, times, f0_up_key, f0_method, merge_type,
file_index, index_rate, if_f0, filter_radius, tgt_sr, resample_sr, rms_mix_rate,
version, protect, crepe_hop_length, f0_autotune, rmvpe_onnx, f0_file=None, f0_min=50, f0_max=1100):
index, big_npy = self.load_index(file_index)
audio = signal.filtfilt(bh, ah, audio)
audio_pad = np.pad(audio, (self.window // 2, self.window // 2), mode="reflect")
opt_ts = []
if audio_pad.shape[0] > self.t_max:
audio_sum = np.zeros_like(audio)
for i in range(self.window):
audio_sum += audio_pad[i : i - self.window]
for t in range(self.t_center, audio.shape[0], self.t_center):
abs_audio_sum = np.abs(audio_sum[t - self.t_query : t + self.t_query])
min_abs_audio_sum = abs_audio_sum.min()
opt_ts.append(t - self.t_query + np.where(abs_audio_sum == min_abs_audio_sum)[0][0])
s = 0
audio_opt = []
t = None
t1 = ttime()
audio_pad = np.pad(audio, (self.t_pad, self.t_pad), mode="reflect")
inp_f0 = None
if f0_file is not None:
try:
with open(f0_file.name, "r") as f:
inp_f0 = np.array([list(map(float, line.split(","))) for line in f.read().strip("\n").split("\n")], dtype="float32")
except:
traceback.print_exc()
sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
pitch, pitchf = None, None
if if_f0:
pitch, pitchf = self.get_f0(
audio_pad, f0_up_key, f0_method, merge_type,
filter_radius, crepe_hop_length, f0_autotune, rmvpe_onnx, inp_f0, f0_min, f0_max)
p_len = min(pitch.shape[0], pitchf.shape[0])
pitch = pitch[:p_len].astype(np.int64 if self.device != 'mps' else np.float32)
pitchf = pitchf[:p_len].astype(np.float32)
pitch = torch.from_numpy(pitch).to(self.device).unsqueeze(0)
pitchf = torch.from_numpy(pitchf).to(self.device).unsqueeze(0)
t2 = ttime()
times[1] += t2 - t1
# with tqdm(total=len(opt_ts), desc="Processing", unit="window") as pbar:
for i, t in enumerate(opt_ts):
t = t // self.window * self.window
start = s
end = t + self.t_pad2 + self.window
audio_slice = audio_pad[start:end]
pitch_slice = pitch[:, start // self.window:end // self.window] if if_f0 else None
pitchf_slice = pitchf[:, start // self.window:end // self.window] if if_f0 else None
audio_opt.append(self.vc(model, net_g, sid, audio_slice, pitch_slice, pitchf_slice, times, index, big_npy, index_rate, version, protect)[self.t_pad_tgt : -self.t_pad_tgt])
s = t
# pbar.update(1)
# pbar.refresh()
audio_slice = audio_pad[t:]
pitch_slice = pitch[:, t // self.window:] if if_f0 and t is not None else pitch
pitchf_slice = pitchf[:, t // self.window:] if if_f0 and t is not None else pitchf
audio_opt.append(self.vc(model, net_g, sid, audio_slice, pitch_slice, pitchf_slice, times, index, big_npy, index_rate, version, protect)[self.t_pad_tgt : -self.t_pad_tgt])
audio_opt = np.concatenate(audio_opt)
if rms_mix_rate < 1:
audio_opt = change_rms(audio, 16000, audio_opt, tgt_sr, rms_mix_rate)
if resample_sr >= 16000 and tgt_sr != resample_sr:
audio_opt = librosa.resample(audio_opt, orig_sr=tgt_sr, target_sr=resample_sr)
audio_max = np.abs(audio_opt).max() / 0.99
audio_opt = (audio_opt * MAX_INT16 / audio_max).astype(np.int16)
gc_collect()
print("Returning completed audio...")
print("-------------------")
return audio_opt
def get_vc(model_path,config,device=None):
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:
from lib.infer_pack.models import SynthesizerTrnMs256NSFsid
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=config.is_half)
else:
from lib.infer_pack.models import SynthesizerTrnMs256NSFsid_nono
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1:
from lib.infer_pack.models import SynthesizerTrnMs768NSFsid
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=config.is_half)
else:
from lib.infer_pack.models import SynthesizerTrnMs768NSFsid_nono
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del net_g.enc_q
net_g.load_state_dict(cpt["weight"], strict=False)
net_g.eval().to(device if device else config.device)
if config.is_half:
net_g = net_g.half()
else:
net_g = net_g.float()
vc = VC(tgt_sr, config)
hubert_model = load_hubert(config)
model_name = os.path.basename(model_path).split(".")[0]
index_files = get_filenames(root=os.path.join(BASE_MODELS_DIR,"RVC"),folder=".index",exts=["index"],name_filters=[model_name])
try: #preload file_index
if len(index_files)==0:
print("File index was empty.")
file_index = None
else:
file_index = index_files.pop()
if os.path.exists(file_index):
sys.stdout.write(f"Attempting to load {file_index}....\n")
sys.stdout.flush()
else:
sys.stdout.write(f"Attempting to load {file_index}.... (despite it not existing)\n")
sys.stdout.flush()
file_index = faiss.read_index(file_index)
sys.stdout.write(f"loaded index: {file_index}\n")
except Exception as e:
print(f"Could not open Faiss index file for reading. {e}")
file_index = None
return {"vc": vc, "cpt": cpt, "net_g": net_g, "hubert_model": hubert_model,"model_name": model_name,
"file_index": file_index, "sr": cpt["config"][-1]}
def vc_single(
cpt=None,
net_g=None,
vc=None,
hubert_model=None,
sid=0,
input_audio=None,
input_audio_path=None,
f0_up_key=0,
f0_file=None,
f0_method="crepe",
merge_type="median",
file_index="", # .index file
index_rate=.75,
filter_radius=3,
resample_sr=0,
rms_mix_rate=.25,
protect=0.33,
crepe_hop_length=160,
f0_autotune=False,
is_onnx=False,
config=config,
**kwargs #prevents function from breaking
):
print(f"vc_single unused args: {kwargs}")
if hubert_model == None:
hubert_model = load_hubert(config)
if not (cpt and net_g and vc and hubert_model):
return None
tgt_sr = cpt["config"][-1]
version = cpt.get("version", "v1")
if input_audio is None and input_audio_path is None:
return None
f0_up_key = int(f0_up_key)
try:
audio = input_audio[0] if input_audio is not None else load_input_audio(input_audio_path, 16000)
audio,_ = remix_audio((audio,input_audio[1] if input_audio is not None else 16000), target_sr=16000)
times = [0, 0, 0]
if_f0 = cpt.get("f0", 1)
"""
model, net_g, sid, audio, times, f0_up_key, f0_method,
file_index, index_rate, if_f0, filter_radius, tgt_sr, resample_sr, rms_mix_rate,
version, protect, crepe_hop_length, f0_autotune, rmvpe_onnx
"""
audio_opt = vc.pipeline(
hubert_model,
net_g,
sid,
audio,
times,
f0_up_key,
f0_method if len(f0_method)>1 else f0_method[0], # more than 1 f0_method in list means hybrid
merge_type,
file_index,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
crepe_hop_length, f0_autotune, is_onnx,
f0_file=f0_file,
)
return (audio_opt, resample_sr if resample_sr >= 16000 and tgt_sr != resample_sr else tgt_sr)
except Exception as error:
print(error)
return None