No OneTemporary
Actions

Size

29 KB

Subscribers

None

View Options

	diff --git a/AIMeiSheng/vc_infer_pipeline_org_embed.py b/AIMeiSheng/vc_infer_pipeline_org_embed.py
	index d53f4ae..cfa0675 100644
	--- a/AIMeiSheng/vc_infer_pipeline_org_embed.py
	+++ b/AIMeiSheng/vc_infer_pipeline_org_embed.py
	@@ -1,760 +1,766 @@
	import numpy as np, parselmouth, torch, pdb, sys, os
	from time import time as ttime
	import torch.nn.functional as F
	import scipy.signal as signal
	import pyworld, os, traceback, faiss, librosa, torchcrepe
	from scipy import signal
	from functools import lru_cache

	now_dir = os.getcwd()
	sys.path.append(now_dir)

	bh, ah = signal.butter(N=5, Wn=48, btype="high", fs=16000)

	input_audio_path2wav = {}
	fidx = 0

	import threading
	import concurrent.futures


	@lru_cache
	def cache_harvest_f0(input_audio_path, fs, f0max, f0min, frame_period):
	audio = input_audio_path2wav[input_audio_path]
	f0, t = pyworld.harvest(
	audio,
	fs=fs,
	f0_ceil=f0max,
	f0_floor=f0min,
	frame_period=frame_period,
	)
	f0 = pyworld.stonemask(audio, f0, t, fs)
	return f0


	def change_rms(data1, sr1, data2, sr2, rate): # 1是输入音频，2是输出音频,rate是2的占比
	# print(data1.max(),data2.max())
	rms1 = librosa.feature.rms(
	y=data1, frame_length=sr1 // 2 * 2, hop_length=sr1 // 2
	) # 每半秒一个点
	rms2 = librosa.feature.rms(y=data2, frame_length=sr2 // 2 * 2, hop_length=sr2 // 2)
	rms1 = torch.from_numpy(rms1)
	rms1 = F.interpolate(
	rms1.unsqueeze(0), size=data2.shape[0], mode="linear"
	).squeeze()
	rms2 = torch.from_numpy(rms2)
	rms2 = F.interpolate(
	rms2.unsqueeze(0), size=data2.shape[0], mode="linear"
	).squeeze()
	rms2 = torch.max(rms2, torch.zeros_like(rms2) + 1e-6)
	data2 *= (
	torch.pow(rms1, torch.tensor(1 - rate))
	* torch.pow(rms2, torch.tensor(rate - 1))
	).numpy()
	return data2


	class VC(object):
	def __init__(self, tgt_sr, config):
	self.x_pad, self.x_query, self.x_center, self.x_max, self.is_half = (
	config.x_pad, ##config会根据设备配置不通知如：3
	config.x_query, # 10 等于x_max-x_center)*2
	config.x_center, #60
	config.x_max, #65
	config.is_half,
	)
	self.sr = 16000 # hubert输入采样率
	self.window = 160 # 每帧点数
	self.t_pad = self.sr * self.x_pad # 每条前后pad时间
	self.t_pad_tgt = tgt_sr * self.x_pad
	self.t_pad2 = self.t_pad * 2
	self.t_query = self.sr * self.x_query # 查询切点前后查询时间,
	self.t_center = self.sr * self.x_center # 查询切点位置
	self.t_max = self.sr * self.x_max # 免查询时长阈值
	self.device = config.device

	def get_f0(
	self,
	input_audio_path,
	x,
	p_len,
	f0_up_key,
	f0_method,
	filter_radius,
	inp_f0=None,
	):
	global input_audio_path2wav
	time_step = self.window / self.sr * 1000
	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)
	if f0_method == "pm":
	f0 = (
	parselmouth.Sound(x, self.sr)
	.to_pitch_ac(
	time_step=time_step / 1000,
	voicing_threshold=0.6,
	pitch_floor=f0_min,
	pitch_ceiling=f0_max,
	)
	.selected_array["frequency"]
	)
	pad_size = (p_len - len(f0) + 1) // 2
	if pad_size > 0 or p_len - len(f0) - pad_size > 0:
	f0 = np.pad(
	f0, [[pad_size, p_len - len(f0) - pad_size]], mode="constant"
	)
	elif f0_method == "harvest":
	input_audio_path2wav[input_audio_path] = x.astype(np.double)
	f0 = cache_harvest_f0(input_audio_path, self.sr, f0_max, f0_min, 10)
	if filter_radius > 2:
	f0 = signal.medfilt(f0, 3)
	elif f0_method == "crepe":
	model = "full"
	# Pick a batch size that doesn't cause memory errors on your gpu
	batch_size = 512
	# Compute pitch using first gpu
	audio = torch.tensor(np.copy(x))[None].float()
	f0, pd = torchcrepe.predict(
	audio,
	self.sr,
	self.window,
	f0_min,
	f0_max,
	model,
	batch_size=batch_size,
	device=self.device,
	return_periodicity=True,
	)
	pd = torchcrepe.filter.median(pd, 3)
	f0 = torchcrepe.filter.mean(f0, 3)
	f0[pd < 0.1] = 0
	f0 = f0[0].cpu().numpy()
	elif f0_method == "rmvpe":
	if hasattr(self, "model_rmvpe") == False:
	from lib.rmvpe import RMVPE

	print("loading rmvpe model")
	self.model_rmvpe = RMVPE(
	"rmvpe.pt", is_half=self.is_half, device=self.device
	)
	f0 = self.model_rmvpe.infer_from_audio(x, thred=0.03)
	else: ##for meisheng
	self.model_rmvpe = f0_method
	f0 = self.model_rmvpe.infer_from_audio(x, thred=0.03)

	##这里读文件，更改pitch st fang
	valid_f0 = f0[f0 > 50]
	mean_pitch_cur = np.mean(valid_f0[:min(len(valid_f0),500)])


	#print("@@f0_up_key:",f0_up_key)
	deta = 0
	if(f0_up_key > 50 ):
	deta = -mean_pitch_cur + f0_up_key

	#print("$$$$$$$$$fangxxxxx pitch shift: ",deta)
	f0_up_key = int(np.log2(deta/(mean_pitch_cur + 1) + 1) * 12)##方法2 fang
	+ if( abs(f0_up_key) <= 8 ):
	+ f0_up_key = 0
	+ elif f0_up_key > 8:
	+ f0_up_key = 12
	+ elif f0_up_key < -8:
	+ f0_up_key = -12
	#if( abs(f0_up_key) < 3 ):
	# f0_up_key = 0
	f0_up_key = max(min(12,f0_up_key),-12)
	#print("f0_up_key: ",f0_up_key)

	f0 *= pow(2, f0_up_key / 12)#这块是音调更改 fang 我设置的0
	# with open("test.txt","w")as f:f.write("\n".join([str(i)for i in f0.tolist()]))
	tf0 = self.sr // self.window # 每秒f0点数
	if inp_f0 is not None:
	delta_t = np.round(
	(inp_f0[:, 0].max() - inp_f0[:, 0].min()) * tf0 + 1
	).astype("int16")
	replace_f0 = np.interp(
	list(range(delta_t)), inp_f0[:, 0] * 100, inp_f0[:, 1]
	)
	shape = f0[self.x_pad * tf0 : self.x_pad * tf0 + len(replace_f0)].shape[0]
	f0[self.x_pad * tf0 : self.x_pad * tf0 + len(replace_f0)] = replace_f0[
	:shape
	]
	# with open("test_opt.txt","w")as f:f.write("\n".join([str(i)for i in f0.tolist()]))


	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(int)
	return f0_coarse, f0bak # 1-0

	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)
	#print("@@@feats: ",feats.shape)
	#print("@@@padding_mask: ",padding_mask.shape)
	inputs = {
	"source": feats.to(self.device),
	"padding_mask": padding_mask,
	"output_layer": 9 if version == "v1" else 12,
	#"output_layer": 6 if version == "v1" else 12,
	}
	t0 = ttime()
	#'''
	with torch.no_grad():
	logits = model.extract_features(**inputs)
	feats = model.final_proj(logits[0]) if version == "v1" else logits[0]#为何v1要转化，维度问题??? fang
	#'''

	#print("@@@feats: ",feats.shape)
	'''
	global fidx
	feats_name = f"./feats_{fidx}.pt"
	fidx += 1
	torch.save(feats, feats_name)
	feats = torch.load(feats_name)
	#'''

	if protect < 0.5 and pitch != None and pitchf != None:
	feats0 = feats.clone()
	if (
	isinstance(index, type(None)) == False
	and isinstance(big_npy, type(None)) == False
	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
	)##基于index和实际音频的特征进行组合，作为输入 fang

	#print("@@@feats: ",feats.shape)
	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
	)#feats0的维度1 插值增加一倍 fang
	t1 = ttime()
	p_len = audio0.shape[0] // self.window ##分帧求pitch fang
	if feats.shape[1] < p_len:
	p_len = feats.shape[1]
	if pitch != None and pitchf != None:
	pitch = pitch[:, :p_len]
	pitchf = pitchf[:, :p_len]

	if protect < 0.5 and pitch != None and pitchf != None:
	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()
	#print("###feats:",feats.shape,"pitch:",pitch.shape,"p_len:",p_len)
	with torch.no_grad():
	if pitch != None and pitchf != None:
	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()
	t2 = ttime()
	times[0] += t1 - t0
	times[2] += t2 - t1
	return audio1

	def pipeline(
	self,
	model,
	net_g,
	sid,
	audio,## input wav
	input_audio_path, #input wav name
	times,
	f0_up_key,
	f0_method,# f0 meathod
	file_index, #index 路径
	# file_big_npy,
	index_rate,
	if_f0,
	filter_radius,
	tgt_sr,
	resample_sr,
	rms_mix_rate,
	version,
	protect,
	f0_file=None,
	):
	if (
	file_index != "" #.index文件不为空 fang
	# and file_big_npy != ""
	# and os.path.exists(file_big_npy) == True
	and os.path.exists(file_index) == True
	and index_rate != 0
	):
	try:
	index = faiss.read_index(file_index)
	# big_npy = np.load(file_big_npy)
	big_npy = index.reconstruct_n(0, index.ntotal)
	except:
	traceback.print_exc()
	index = big_npy = None
	else:
	index = big_npy = None
	#print("####audio 1:",audio.shape)
	audio = signal.filtfilt(bh, ah, audio)
	#print("####audio 2:",audio.shape)
	audio_pad = np.pad(audio, (self.window // 2, self.window // 2), mode="reflect")
	opt_ts = []

	#print("###t_max:",self.t_max)
	#print("###window:",self.window,"self.t_query:",self.t_query,"self.t_pad2:",self.t_pad2)
	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]#这样算循环了,每个idx是过去一帧的值的和 fang
	for t in range(self.t_center, audio.shape[0], self.t_center):#一分钟一帧？？ fang
	opt_ts.append(
	t
	- self.t_query
	+ np.where(
	np.abs(audio_sum[t - self.t_query : t + self.t_query])
	== np.abs(audio_sum[t - self.t_query : t + self.t_query]).min()
	)[0][0]
	)#返回[ t - self.t_query, t+self.t_query] 区间最小值位置的索引保存，fang
	s = 0
	audio_opt = []
	t = None
	t1 = ttime()
	audio_pad = np.pad(audio, (self.t_pad, self.t_pad), mode="reflect")
	p_len = audio_pad.shape[0] // self.window
	inp_f0 = None
	if hasattr(f0_file, "name") == True:
	try:
	with open(f0_file.name, "r") as f:
	lines = f.read().strip("\n").split("\n")
	inp_f0 = []
	for line in lines:
	inp_f0.append([float(i) for i in line.split(",")])
	inp_f0 = np.array(inp_f0, dtype="float32")
	except:
	traceback.print_exc()
	#sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
	sid_embed = np.load(sid)
	sid = torch.FloatTensor(sid_embed).to(self.device).half()
	pitch, pitchf = None, None
	if if_f0 == 1:
	pitch, pitchf = self.get_f0(
	input_audio_path,
	audio_pad,
	p_len,
	f0_up_key,
	f0_method,
	filter_radius,
	inp_f0,
	)
	pitch = pitch[:p_len]
	pitchf = pitchf[:p_len]
	if self.device == "mps":
	pitchf = pitchf.astype(np.float32)
	pitch = torch.tensor(pitch, device=self.device).unsqueeze(0).long()
	pitchf = torch.tensor(pitchf, device=self.device).unsqueeze(0).float()

	#print("&&&&pitch: ",pitchf)
	t2 = ttime()
	times[1] += t2 - t1
	#print("####len(audio_pad):",len(audio_pad))
	#print("###pitch:", pitch.shape)
	for t in opt_ts: #分段推理每段音频，一段这里设置60s左右 fang
	t = t // self.window * self.window
	if if_f0 == 1:
	audio_opt.append(
	self.vc(
	model,
	net_g,
	sid,
	audio_pad[s : t + self.t_pad2 + self.window],
	pitch[:, s // self.window : (t + self.t_pad2) // self.window],
	pitchf[:, s // self.window : (t + self.t_pad2) // self.window],
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect,
	)[self.t_pad_tgt : -self.t_pad_tgt]
	)
	else:
	audio_opt.append(
	self.vc(
	model,
	net_g,
	sid,
	audio_pad[s : t + self.t_pad2 + self.window],
	None,
	None,
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect,
	)[self.t_pad_tgt : -self.t_pad_tgt]
	)
	s = t
	if if_f0 == 1: ##后面是最后一段处理 fang
	audio_opt.append(
	self.vc(
	model,
	net_g,
	sid,
	audio_pad[t:],
	pitch[:, t // self.window :] if t is not None else pitch,
	pitchf[:, t // self.window :] if t is not None else pitchf,
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect,
	)[self.t_pad_tgt : -self.t_pad_tgt]
	)
	else:
	audio_opt.append(
	self.vc(
	model,
	net_g,
	sid,
	audio_pad[t:],
	None,
	None,
	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
	max_int16 = 32768
	if audio_max > 1:
	max_int16 /= audio_max
	audio_opt = (audio_opt * max_int16).astype(np.int16)
	del pitch, pitchf, sid
	if torch.cuda.is_available():
	torch.cuda.empty_cache()
	return audio_opt

	def infer_core_fang(self,para1,para2,para3,idx,
	model,
	net_g,
	sid,
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect):
	return [ self.vc(
	model,
	net_g,
	sid,
	para1, para2, para3,
	# audio_pad[s: t + self.t_pad2 + self.window],
	# pitch[:, s // self.window: (t + self.t_pad2) // self.window],
	# pitchf[:, s // self.window: (t + self.t_pad2) // self.window],
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect,
	)[self.t_pad_tgt: -self.t_pad_tgt], idx]

	def ThreadPool_process_core(self, func_process,params1,params2,params3,
	model,
	net_g,
	sid,
	# audio_pad[s: t + self.t_pad2 + self.window],
	# pitch[:, s // self.window: (t + self.t_pad2) // self.window],
	# pitchf[:, s // self.window: (t + self.t_pad2) // self.window],
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect
	):
	num_threads = 2
	futures = []
	sort_ret = {}
	with concurrent.futures.ThreadPoolExecutor(max_workers=num_threads) as executor:
	for idx in range(len(params1)):
	para1 = params1[idx]
	para2 = params2[idx]
	para3 = params3[idx]
	ret = executor.submit(self.infer_core_fang,para1,para2,para3,idx,
	model,
	net_g,
	sid,
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect)
	futures.append(ret)

	cnt = 0
	for future in concurrent.futures.as_completed(futures):
	cnt += 1
	#print(f"process finised {cnt}, and index :{future.result()[1]}")

	#print(future.result()) # result
	# print(future.result()[1]) ##index
	sort_ret[str(future.result()[1])] = future.result()[0]


	fea_list = []
	for idx in range(len(sort_ret)):
	fea_list.append(sort_ret[str(idx)])

	return fea_list

	def pipeline_mulprocess(
	self,
	model,
	net_g,
	sid,
	audio, ## input wav
	input_audio_path, # input wav name
	times,
	f0_up_key,
	f0_method, # f0 meathod
	file_index, # index 路径
	# file_big_npy,
	index_rate,
	if_f0,
	filter_radius,
	tgt_sr,
	resample_sr,
	rms_mix_rate,
	version,
	protect,
	f0_file=None,
	):
	if (
	file_index != "" # .index文件不为空 fang
	# and file_big_npy != ""
	# and os.path.exists(file_big_npy) == True
	and os.path.exists(file_index) == True
	and index_rate != 0
	):
	try:
	index = faiss.read_index(file_index)
	# big_npy = np.load(file_big_npy)
	big_npy = index.reconstruct_n(0, index.ntotal)
	except:
	traceback.print_exc()
	index = big_npy = None
	else:
	index = big_npy = None
	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] # 这样算循环了,每个idx是过去一帧的值的和 fang
	for t in range(self.t_center, audio.shape[0], self.t_center): # 一分钟一帧？？ fang
	opt_ts.append(
	t
	- self.t_query
	+ np.where(
	np.abs(audio_sum[t - self.t_query: t + self.t_query])
	== np.abs(audio_sum[t - self.t_query: t + self.t_query]).min()
	)[0][0]
	) # 返回[ t - self.t_query, t+self.t_query] 区间最小值位置的索引保存，fang
	s = 0

	t = None
	t1 = ttime()
	audio_pad = np.pad(audio, (self.t_pad, self.t_pad), mode="reflect")
	p_len = audio_pad.shape[0] // self.window
	inp_f0 = None
	if hasattr(f0_file, "name") == True:
	try:
	with open(f0_file.name, "r") as f:
	lines = f.read().strip("\n").split("\n")
	inp_f0 = []
	for line in lines:
	inp_f0.append([float(i) for i in line.split(",")])
	inp_f0 = np.array(inp_f0, dtype="float32")
	except:
	traceback.print_exc()
	# sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
	sid_embed = np.load(sid)
	sid = torch.FloatTensor(sid_embed).to(self.device).half()
	pitch, pitchf = None, None
	#'''
	if if_f0 == 1:
	pitch, pitchf = self.get_f0(
	input_audio_path,
	audio_pad,
	p_len,
	f0_up_key,
	f0_method,
	filter_radius,
	inp_f0,
	)
	pitch = pitch[:p_len]
	pitchf = pitchf[:p_len]
	if self.device == "mps":
	pitchf = pitchf.astype(np.float32)
	pitch = torch.tensor(pitch, device=self.device).unsqueeze(0).long()
	pitchf = torch.tensor(pitchf, device=self.device).unsqueeze(0).float()
	#'''

	'''
	pitch_name = "./pitch_pitchf.npz"
	#np.savez(pitch_name, pitch = pitch.detach().cpu().numpy(), pitchf = pitchf.detach().cpu().numpy())
	npz_obj = np.load(pitch_name) #文件名的后缀为npz
	pitch, pitchf = npz_obj['pitch'], npz_obj['pitchf']
	pitch = torch.tensor(pitch, device=self.device).long()
	pitchf = torch.tensor(pitchf, device=self.device).float()
	#'''

	t2 = ttime()
	times[1] += t2 - t1

	audio_opt = []
	audio_pad_list = []
	pitch_list = []
	pitchf_list = []


	for t in opt_ts: # 分段推理每段音频，一段这里设置60s左右 fang
	t = t // self.window * self.window
	audio_pad_list.append(audio_pad[s: t + self.t_pad2 + self.window])
	pitch_list.append(pitch[:, s // self.window: (t + self.t_pad2) // self.window])
	pitchf_list.append(pitchf[:, s // self.window: (t + self.t_pad2) // self.window])
	s = t

	audio_pad_list.append(audio_pad[t:])
	pitch_list.append(pitch[:, t // self.window:] if t is not None else pitch)
	pitchf_list.append(pitchf[:, t // self.window:] if t is not None else pitchf)

	audio_opt = self.ThreadPool_process_core(self.infer_core_fang, audio_pad_list, pitch_list, pitchf_list,
	model,
	net_g,
	sid,
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect
	)
	'''
	if if_f0 == 1: ##后面是最后一段处理 fang
	audio_opt.append(
	self.vc(
	model,
	net_g,
	sid,
	audio_pad[t:],
	pitch[:, t // self.window:] if t is not None else pitch,
	pitchf[:, t // self.window:] if t is not None else pitchf,
	times,
	index,
	big_npy,
	index_rate,
	version,
	protect,
	)[self.t_pad_tgt: -self.t_pad_tgt]
	)
	else:
	audio_opt.append(
	self.vc(
	model,
	net_g,
	sid,
	audio_pad[t:],
	None,
	None,
	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
	max_int16 = 32768
	if audio_max > 1:
	max_int16 /= audio_max
	audio_opt = (audio_opt * max_int16).astype(np.int16)
	del pitch, pitchf, sid
	if torch.cuda.is_available():
	torch.cuda.empty_cache()
	return audio_opt

File Metadata

Mime Type: text/x-diff
Expires: Sun, Jan 12, 08:34 (1 d, 15 h)
Storage Engine: blob
Storage Format: Raw Data
Storage Handle: 1347208
Default Alt Text: (29 KB)

No OneTemporaryActions

View Options

File Metadata

Event Timeline

No OneTemporary
Actions