voice-changer/server/voice_changer/RVC/custom_vc_infer_pipeline.py

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import numpy as np
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# import parselmouth
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import torch
import torch.nn.functional as F
import scipy.signal as signal
import pyworld
class VC(object):
def __init__(self, tgt_sr, device, is_half, x_pad):
self.sr = 16000 # hubert输入采样率
self.window = 160 # 每帧点数
self.t_pad = self.sr * x_pad # 每条前后pad时间
self.t_pad_tgt = tgt_sr * x_pad
self.device = device
self.is_half = is_half
def get_f0(self, audio, p_len, f0_up_key, f0_method, silence_front=0):
n_frames = int(len(audio) // self.window) + 1
start_frame = int(silence_front * self.sr / self.window)
real_silence_front = start_frame * self.window / self.sr
silence_front_offset = int(np.round(real_silence_front * self.sr))
audio = audio[silence_front_offset:]
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# time_step = self.window / self.sr * 1000
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f0_min = 50
f0_max = 1100
f0_mel_min = 1127 * np.log(1 + f0_min / 700)
f0_mel_max = 1127 * np.log(1 + f0_max / 700)
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if f0_method == "dio":
_f0, t = pyworld.dio(
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audio.astype(np.double),
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self.sr,
f0_floor=f0_min,
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f0_ceil=f0_max,
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channels_in_octave=2,
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frame_period=10,
)
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f0 = pyworld.stonemask(audio.astype(np.double), _f0, t, self.sr)
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f0 = np.pad(
f0.astype("float"), (start_frame, n_frames - len(f0) - start_frame)
)
else:
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f0, t = pyworld.harvest(
audio.astype(np.double),
fs=self.sr,
f0_ceil=f0_max,
frame_period=10,
)
f0 = pyworld.stonemask(audio.astype(np.double), f0, t, self.sr)
f0 = signal.medfilt(f0, 3)
f0 = np.pad(
f0.astype("float"), (start_frame, n_frames - len(f0) - start_frame)
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)
f0 *= pow(2, f0_up_key / 12)
f0bak = f0.copy()
f0_mel = 1127 * np.log(1 + f0 / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (
f0_mel_max - f0_mel_min
) + 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > 255] = 255
f0_coarse = np.rint(f0_mel).astype(np.int)
# Volume Extract
# volume = self.extractVolume(audio, 512)
# volume = np.pad(
# volume.astype("float"), (start_frame, n_frames - len(volume) - start_frame)
# )
# return f0_coarse, f0bak, volume # 1-0
return f0_coarse, f0bak
# def extractVolume(self, audio, hopsize):
# n_frames = int(len(audio) // hopsize) + 1
# audio2 = audio**2
# audio2 = np.pad(
# audio2,
# (int(hopsize // 2), int((hopsize + 1) // 2)),
# mode="reflect",
# )
# volume = np.array(
# [
# np.mean(audio2[int(n * hopsize) : int((n + 1) * hopsize)]) # noqa:E203
# for n in range(n_frames)
# ]
# )
# volume = np.sqrt(volume)
# return volume
def pipeline(
self,
embedder,
model,
sid,
audio,
f0_up_key,
f0_method,
index,
big_npy,
index_rate,
if_f0,
silence_front=0,
embChannels=256,
):
audio_pad = np.pad(audio, (self.t_pad, self.t_pad), mode="reflect")
p_len = audio_pad.shape[0] // self.window
sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
# ピッチ検出
pitch, pitchf = None, None
if if_f0 == 1:
pitch, pitchf = self.get_f0(
audio_pad,
p_len,
f0_up_key,
f0_method,
silence_front=silence_front,
)
pitch = pitch[:p_len]
pitchf = pitchf[:p_len]
pitch = torch.tensor(pitch, device=self.device).unsqueeze(0).long()
pitchf = torch.tensor(
pitchf, device=self.device, dtype=torch.float
).unsqueeze(0)
# tensor
feats = torch.from_numpy(audio_pad)
if self.is_half is True:
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)
# embedding
padding_mask = torch.BoolTensor(feats.shape).to(self.device).fill_(False)
if embChannels == 256:
inputs = {
"source": feats.to(self.device),
"padding_mask": padding_mask,
"output_layer": 9, # layer 9
}
else:
inputs = {
"source": feats.to(self.device),
"padding_mask": padding_mask,
}
with torch.no_grad():
logits = embedder.extract_features(**inputs)
if embChannels == 256:
feats = embedder.final_proj(logits[0])
else:
feats = logits[0]
# Index - feature抽出
if (
isinstance(index, type(None)) is False
and isinstance(big_npy, type(None)) is False
and index_rate != 0
):
npy = feats[0].cpu().numpy()
if self.is_half is True:
npy = npy.astype("float32")
D, I = index.search(npy, 1)
npy = big_npy[I.squeeze()]
if self.is_half is True:
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)
# ピッチ抽出
p_len = audio_pad.shape[0] // self.window
if feats.shape[1] < p_len:
p_len = feats.shape[1]
if pitch is not None and pitchf is not None:
pitch = pitch[:, :p_len]
pitchf = pitchf[:, :p_len]
p_len = torch.tensor([p_len], device=self.device).long()
# 推論実行
with torch.no_grad():
if pitch is not None:
audio1 = (
(model.infer(feats, p_len, pitch, pitchf, sid)[0][0, 0] * 32768)
.data.cpu()
.float()
.numpy()
.astype(np.int16)
)
else:
if hasattr(model, "infer_pitchless"):
audio1 = (
(model.infer_pitchless(feats, p_len, sid)[0][0, 0] * 32768)
.data.cpu()
.float()
.numpy()
.astype(np.int16)
)
else:
audio1 = (
(model.infer(feats, p_len, sid)[0][0, 0] * 32768)
.data.cpu()
.float()
.numpy()
.astype(np.int16)
)
del feats, p_len, padding_mask
torch.cuda.empty_cache()
if self.t_pad_tgt != 0:
offset = self.t_pad_tgt
end = -1 * self.t_pad_tgt
audio1 = audio1[offset:end]
del pitch, pitchf, sid
torch.cuda.empty_cache()
return audio1