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WIP: refactoring
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wataru
parent
37bb25682e
commit
234595eeeb
@ -1,315 +0,0 @@
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import numpy as np
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import parselmouth
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import torch
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import pdb
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from time import time as ttime
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import torch.nn.functional as F
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from config import x_pad, x_query, x_center, x_max
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import scipy.signal as signal
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import pyworld
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import os
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import traceback
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import faiss
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from .const import RVC_MODEL_TYPE_RVC, RVC_MODEL_TYPE_WEBUI
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class VC(object):
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def __init__(self, tgt_sr, device, is_half, x_pad):
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self.sr = 16000 # hubert输入采样率
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self.window = 160 # 每帧点数
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self.t_pad = self.sr * x_pad # 每条前后pad时间
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self.t_pad_tgt = tgt_sr * x_pad
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self.t_pad2 = self.t_pad * 2
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self.t_query = self.sr * x_query # 查询切点前后查询时间
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self.t_center = self.sr * x_center # 查询切点位置
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self.t_max = self.sr * x_max # 免查询时长阈值
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self.device = device
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self.is_half = is_half
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def get_f0(self, audio, p_len, f0_up_key, f0_method, inp_f0=None, silence_front=0):
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n_frames = int(len(audio) // self.window) + 1
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start_frame = int(silence_front * self.sr / self.window)
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real_silence_front = start_frame * self.window / self.sr
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audio = audio[int(np.round(real_silence_front * self.sr)) :]
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time_step = self.window / self.sr * 1000
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f0_min = 50
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f0_max = 1100
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f0_mel_min = 1127 * np.log(1 + f0_min / 700)
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f0_mel_max = 1127 * np.log(1 + f0_max / 700)
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if f0_method == "pm":
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f0 = (
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parselmouth.Sound(audio, self.sr)
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.to_pitch_ac(
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time_step=time_step / 1000,
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voicing_threshold=0.6,
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pitch_floor=f0_min,
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pitch_ceiling=f0_max,
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)
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.selected_array["frequency"]
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)
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pad_size = (p_len - len(f0) + 1) // 2
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if pad_size > 0 or p_len - len(f0) - pad_size > 0:
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f0 = np.pad(
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f0, [[pad_size, p_len - len(f0) - pad_size]], mode="constant"
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)
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elif f0_method == "harvest":
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f0, t = pyworld.harvest(
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audio.astype(np.double),
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fs=self.sr,
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f0_ceil=f0_max,
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frame_period=10,
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)
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f0 = pyworld.stonemask(audio.astype(np.double), f0, t, self.sr)
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f0 = signal.medfilt(f0, 3)
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f0 = np.pad(
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f0.astype("float"), (start_frame, n_frames - len(f0) - start_frame)
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)
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else:
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print("[Voice Changer] invalid f0 detector, use pm.", f0_method)
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f0 = (
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parselmouth.Sound(audio, self.sr)
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.to_pitch_ac(
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time_step=time_step / 1000,
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voicing_threshold=0.6,
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pitch_floor=f0_min,
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pitch_ceiling=f0_max,
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)
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.selected_array["frequency"]
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)
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pad_size = (p_len - len(f0) + 1) // 2
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if pad_size > 0 or p_len - len(f0) - pad_size > 0:
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f0 = np.pad(
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f0, [[pad_size, p_len - len(f0) - pad_size]], mode="constant"
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)
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f0 *= pow(2, f0_up_key / 12)
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# with open("test.txt","w")as f:f.write("\n".join([str(i)for i in f0.tolist()]))
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tf0 = self.sr // self.window # 每秒f0点数
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if inp_f0 is not None:
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delta_t = np.round(
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(inp_f0[:, 0].max() - inp_f0[:, 0].min()) * tf0 + 1
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).astype("int16")
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replace_f0 = np.interp(
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list(range(delta_t)), inp_f0[:, 0] * 100, inp_f0[:, 1]
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)
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shape = f0[x_pad * tf0 : x_pad * tf0 + len(replace_f0)].shape[0]
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f0[x_pad * tf0 : x_pad * tf0 + len(replace_f0)] = replace_f0[:shape]
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# with open("test_opt.txt","w")as f:f.write("\n".join([str(i)for i in f0.tolist()]))
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f0bak = f0.copy()
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f0_mel = 1127 * np.log(1 + f0 / 700)
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f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (
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f0_mel_max - f0_mel_min
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) + 1
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f0_mel[f0_mel <= 1] = 1
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f0_mel[f0_mel > 255] = 255
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f0_coarse = np.rint(f0_mel).astype(np.int)
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return f0_coarse, f0bak # 1-0
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def vc(
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self,
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model,
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net_g,
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sid,
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audio0,
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pitch,
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pitchf,
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times,
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index,
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big_npy,
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index_rate,
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embChannels=256,
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): # ,file_index,file_big_npy
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feats = torch.from_numpy(audio0)
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if self.is_half == True:
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feats = feats.half()
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else:
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feats = feats.float()
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if feats.dim() == 2: # double channels
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feats = feats.mean(-1)
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assert feats.dim() == 1, feats.dim()
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feats = feats.view(1, -1)
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padding_mask = torch.BoolTensor(feats.shape).to(self.device).fill_(False)
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if embChannels == 256:
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inputs = {
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"source": feats.to(self.device),
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"padding_mask": padding_mask,
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"output_layer": 9, # layer 9
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}
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else:
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inputs = {
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"source": feats.to(self.device),
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"padding_mask": padding_mask,
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}
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t0 = ttime()
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with torch.no_grad():
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logits = model.extract_features(**inputs)
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if embChannels == 256:
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feats = model.final_proj(logits[0])
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else:
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feats = logits[0]
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if (
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isinstance(index, type(None)) == False
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and isinstance(big_npy, type(None)) == False
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and index_rate != 0
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):
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npy = feats[0].cpu().numpy()
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if self.is_half == True:
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npy = npy.astype("float32")
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D, I = index.search(npy, 1)
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npy = big_npy[I.squeeze()]
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if self.is_half == True:
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npy = npy.astype("float16")
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feats = (
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torch.from_numpy(npy).unsqueeze(0).to(self.device) * index_rate
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+ (1 - index_rate) * feats
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)
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feats = F.interpolate(feats.permute(0, 2, 1), scale_factor=2).permute(0, 2, 1)
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t1 = ttime()
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p_len = audio0.shape[0] // self.window
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if feats.shape[1] < p_len:
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p_len = feats.shape[1]
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if pitch != None and pitchf != None:
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pitch = pitch[:, :p_len]
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pitchf = pitchf[:, :p_len]
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p_len = torch.tensor([p_len], device=self.device).long()
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with torch.no_grad():
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if pitch != None:
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audio1 = (
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(net_g.infer(feats, p_len, pitch, pitchf, sid)[0][0, 0] * 32768)
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.data.cpu()
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.float()
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.numpy()
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.astype(np.int16)
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)
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else:
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if hasattr(net_g, "infer_pitchless"):
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audio1 = (
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(net_g.infer_pitchless(feats, p_len, sid)[0][0, 0] * 32768)
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.data.cpu()
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.float()
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.numpy()
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.astype(np.int16)
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)
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else:
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audio1 = (
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(net_g.infer(feats, p_len, sid)[0][0, 0] * 32768)
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.data.cpu()
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.float()
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.numpy()
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.astype(np.int16)
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)
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# audio1 = (net_g.infer(feats, p_len, None, pitchf, sid)[0][0, 0] * 32768).data.cpu().float().numpy().astype(np.int16)
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del feats, p_len, padding_mask
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torch.cuda.empty_cache()
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t2 = ttime()
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times[0] += t1 - t0
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times[2] += t2 - t1
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return audio1
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def pipeline(
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self,
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model,
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net_g,
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sid,
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audio,
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times,
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f0_up_key,
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f0_method,
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file_index,
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file_big_npy,
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index_rate,
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if_f0,
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f0_file=None,
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silence_front=0,
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embChannels=256,
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):
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if (
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file_big_npy != ""
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and file_index != ""
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and os.path.exists(file_big_npy) == True
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and os.path.exists(file_index) == True
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and index_rate != 0
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):
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try:
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index = faiss.read_index(file_index)
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big_npy = np.load(file_big_npy)
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except:
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traceback.print_exc()
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index = big_npy = None
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else:
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index = big_npy = None
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audio_opt = []
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t = None
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t1 = ttime()
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audio_pad = np.pad(audio, (self.t_pad, self.t_pad), mode="reflect")
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p_len = audio_pad.shape[0] // self.window
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inp_f0 = None
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sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
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pitch, pitchf = None, None
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if if_f0 == 1:
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pitch, pitchf = self.get_f0(
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audio_pad,
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p_len,
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f0_up_key,
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f0_method,
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inp_f0,
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silence_front=silence_front,
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)
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pitch = pitch[:p_len]
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pitchf = pitchf[:p_len]
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pitch = torch.tensor(pitch, device=self.device).unsqueeze(0).long()
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pitchf = torch.tensor(
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pitchf, device=self.device, dtype=torch.float
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).unsqueeze(0)
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t2 = ttime()
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times[1] += t2 - t1
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if self.t_pad_tgt == 0:
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audio_opt.append(
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self.vc(
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model,
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net_g,
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sid,
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audio_pad[t:],
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pitch[:, t // self.window :] if t is not None else pitch,
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pitchf[:, t // self.window :] if t is not None else pitchf,
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times,
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index,
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big_npy,
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index_rate,
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embChannels,
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)
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)
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else:
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audio_opt.append(
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self.vc(
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model,
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net_g,
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sid,
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audio_pad[t:],
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pitch[:, t // self.window :] if t is not None else pitch,
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pitchf[:, t // self.window :] if t is not None else pitchf,
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times,
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index,
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big_npy,
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index_rate,
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embChannels,
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)[self.t_pad_tgt : -self.t_pad_tgt]
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)
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audio_opt = np.concatenate(audio_opt)
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del pitch, pitchf, sid
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torch.cuda.empty_cache()
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return audio_opt
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