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Delete vc_infer_pipeline.py
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vc_infer_pipeline.py
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import numpy as np, parselmouth, torch, pdb, sys, os
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from time import time as ttime
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import torch.nn.functional as F
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import torchcrepe # Fork feature. Use the crepe f0 algorithm. New dependency (pip install torchcrepe)
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from torch import Tensor
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import scipy.signal as signal
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import pyworld, os, traceback, faiss, librosa, torchcrepe
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from scipy import signal
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from functools import lru_cache
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now_dir = os.getcwd()
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sys.path.append(now_dir)
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bh, ah = signal.butter(N=5, Wn=48, btype="high", fs=16000)
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input_audio_path2wav = {}
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#A fun little addition from my personal RVC branch.
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#You don't have to implement it if you don't have to
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from config import Config
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config=Config()
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from rmvpe import RMVPE
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print("Preloading RMVPE model...")
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model_rmvpe = RMVPE("rmvpe.pt", is_half=config.is_half, device=config.device)
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del config
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@lru_cache
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def cache_harvest_f0(input_audio_path, fs, f0max, f0min, frame_period):
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audio = input_audio_path2wav[input_audio_path]
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f0, t = pyworld.harvest(
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audio,
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fs=fs,
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f0_ceil=f0max,
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f0_floor=f0min,
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frame_period=frame_period,
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)
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f0 = pyworld.stonemask(audio, f0, t, fs)
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return f0
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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
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# print(data1.max(),data2.max())
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rms1 = librosa.feature.rms(
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y=data1, frame_length=sr1 // 2 * 2, hop_length=sr1 // 2
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) # 每半秒一个点
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rms2 = librosa.feature.rms(y=data2, frame_length=sr2 // 2 * 2, hop_length=sr2 // 2)
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rms1 = torch.from_numpy(rms1)
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rms1 = F.interpolate(
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rms1.unsqueeze(0), size=data2.shape[0], mode="linear"
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).squeeze()
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rms2 = torch.from_numpy(rms2)
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rms2 = F.interpolate(
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rms2.unsqueeze(0), size=data2.shape[0], mode="linear"
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).squeeze()
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rms2 = torch.max(rms2, torch.zeros_like(rms2) + 1e-6)
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data2 *= (
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torch.pow(rms1, torch.tensor(1 - rate))
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* torch.pow(rms2, torch.tensor(rate - 1))
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).numpy()
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return data2
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class VC(object):
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def __init__(self, tgt_sr, config):
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self.x_pad, self.x_query, self.x_center, self.x_max, self.is_half = (
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config.x_pad,
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config.x_query,
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config.x_center,
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config.x_max,
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config.is_half,
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)
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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 * self.x_pad # 每条前后pad时间
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self.t_pad_tgt = tgt_sr * self.x_pad
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self.t_pad2 = self.t_pad * 2
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self.t_query = self.sr * self.x_query # 查询切点前后查询时间
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self.t_center = self.sr * self.x_center # 查询切点位置
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self.t_max = self.sr * self.x_max # 免查询时长阈值
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self.device = config.device
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# Fork Feature: Get the best torch device to use for f0 algorithms that require a torch device. Will return the type (torch.device)
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def get_optimal_torch_device(self, index: int = 0) -> torch.device:
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# Get cuda device
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if torch.cuda.is_available():
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return torch.device(
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f"cuda:{index % torch.cuda.device_count()}"
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) # Very fast
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elif torch.backends.mps.is_available():
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return torch.device("mps")
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# Insert an else here to grab "xla" devices if available. TO DO later. Requires the torch_xla.core.xla_model library
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# Else wise return the "cpu" as a torch device,
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return torch.device("cpu")
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# Fork Feature: Compute f0 with the crepe method
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def get_f0_crepe_computation(
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self,
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x,
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f0_min,
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f0_max,
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p_len,
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hop_length=160, # 512 before. Hop length changes the speed that the voice jumps to a different dramatic pitch. Lower hop lengths means more pitch accuracy but longer inference time.
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model="full", # Either use crepe-tiny "tiny" or crepe "full". Default is full
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):
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x = x.astype(
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np.float32
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) # fixes the F.conv2D exception. We needed to convert double to float.
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x /= np.quantile(np.abs(x), 0.999)
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torch_device = self.get_optimal_torch_device()
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audio = torch.from_numpy(x).to(torch_device, copy=True)
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audio = torch.unsqueeze(audio, dim=0)
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if audio.ndim == 2 and audio.shape[0] > 1:
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audio = torch.mean(audio, dim=0, keepdim=True).detach()
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audio = audio.detach()
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print("Initiating prediction with a crepe_hop_length of: " + str(hop_length))
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pitch: Tensor = torchcrepe.predict(
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audio,
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self.sr,
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hop_length,
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f0_min,
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f0_max,
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model,
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batch_size=hop_length * 2,
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device=torch_device,
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pad=True,
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)
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p_len = p_len or x.shape[0] // hop_length
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# Resize the pitch for final f0
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source = np.array(pitch.squeeze(0).cpu().float().numpy())
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source[source < 0.001] = np.nan
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target = np.interp(
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np.arange(0, len(source) * p_len, len(source)) / p_len,
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np.arange(0, len(source)),
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source,
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)
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f0 = np.nan_to_num(target)
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return f0 # Resized f0
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def get_f0_official_crepe_computation(
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self,
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x,
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f0_min,
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f0_max,
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model="full",
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):
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# Pick a batch size that doesn't cause memory errors on your gpu
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batch_size = 512
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# Compute pitch using first gpu
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audio = torch.tensor(np.copy(x))[None].float()
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f0, pd = torchcrepe.predict(
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audio,
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self.sr,
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self.window,
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f0_min,
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f0_max,
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model,
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batch_size=batch_size,
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device=self.device,
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return_periodicity=True,
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)
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pd = torchcrepe.filter.median(pd, 3)
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f0 = torchcrepe.filter.mean(f0, 3)
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f0[pd < 0.1] = 0
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f0 = f0[0].cpu().numpy()
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return f0
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# Fork Feature: Compute pYIN f0 method
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def get_f0_pyin_computation(self, x, f0_min, f0_max):
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y, sr = librosa.load("saudio/Sidney.wav", self.sr, mono=True)
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f0, _, _ = librosa.pyin(y, sr=self.sr, fmin=f0_min, fmax=f0_max)
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f0 = f0[1:] # Get rid of extra first frame
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return f0
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# Fork Feature: Acquire median hybrid f0 estimation calculation
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def get_f0_hybrid_computation(
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self,
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methods_str,
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input_audio_path,
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x,
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f0_min,
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f0_max,
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p_len,
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filter_radius,
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crepe_hop_length,
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time_step,
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):
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# Get various f0 methods from input to use in the computation stack
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s = methods_str
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s = s.split("hybrid")[1]
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s = s.replace("[", "").replace("]", "")
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methods = s.split("+")
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f0_computation_stack = []
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print("Calculating f0 pitch estimations for methods: %s" % str(methods))
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x = x.astype(np.float32)
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x /= np.quantile(np.abs(x), 0.999)
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# Get f0 calculations for all methods specified
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for method in methods:
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f0 = None
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if method == "pm":
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f0 = (
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parselmouth.Sound(x, 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 method == "crepe":
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f0 = self.get_f0_official_crepe_computation(x, f0_min, f0_max)
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f0 = f0[1:] # Get rid of extra first frame
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elif method == "crepe-tiny":
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f0 = self.get_f0_official_crepe_computation(x, f0_min, f0_max, "tiny")
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f0 = f0[1:] # Get rid of extra first frame
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elif method == "mangio-crepe":
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f0 = self.get_f0_crepe_computation(
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x, f0_min, f0_max, p_len, crepe_hop_length
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)
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elif method == "mangio-crepe-tiny":
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f0 = self.get_f0_crepe_computation(
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x, f0_min, f0_max, p_len, crepe_hop_length, "tiny"
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)
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elif method == "harvest":
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f0 = cache_harvest_f0(input_audio_path, self.sr, f0_max, f0_min, 10)
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if filter_radius > 2:
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f0 = signal.medfilt(f0, 3)
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f0 = f0[1:] # Get rid of first frame.
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elif method == "dio": # Potentially buggy?
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f0, t = pyworld.dio(
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x.astype(np.double),
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fs=self.sr,
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f0_ceil=f0_max,
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f0_floor=f0_min,
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frame_period=10,
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)
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f0 = pyworld.stonemask(x.astype(np.double), f0, t, self.sr)
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f0 = signal.medfilt(f0, 3)
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f0 = f0[1:]
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# elif method == "pyin": Not Working just yet
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# f0 = self.get_f0_pyin_computation(x, f0_min, f0_max)
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# Push method to the stack
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f0_computation_stack.append(f0)
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for fc in f0_computation_stack:
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print(len(fc))
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print("Calculating hybrid median f0 from the stack of: %s" % str(methods))
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f0_median_hybrid = None
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if len(f0_computation_stack) == 1:
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f0_median_hybrid = f0_computation_stack[0]
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else:
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f0_median_hybrid = np.nanmedian(f0_computation_stack, axis=0)
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return f0_median_hybrid
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def get_f0(
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self,
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input_audio_path,
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x,
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p_len,
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f0_up_key,
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f0_method,
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filter_radius,
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crepe_hop_length,
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inp_f0=None,
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):
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global input_audio_path2wav
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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(x, 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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input_audio_path2wav[input_audio_path] = x.astype(np.double)
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f0 = cache_harvest_f0(input_audio_path, self.sr, f0_max, f0_min, 10)
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if filter_radius > 2:
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f0 = signal.medfilt(f0, 3)
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elif f0_method == "dio": # Potentially Buggy?
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f0, t = pyworld.dio(
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x.astype(np.double),
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fs=self.sr,
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f0_ceil=f0_max,
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f0_floor=f0_min,
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frame_period=10,
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)
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f0 = pyworld.stonemask(x.astype(np.double), f0, t, self.sr)
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f0 = signal.medfilt(f0, 3)
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elif f0_method == "crepe":
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f0 = self.get_f0_official_crepe_computation(x, f0_min, f0_max)
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elif f0_method == "crepe-tiny":
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f0 = self.get_f0_official_crepe_computation(x, f0_min, f0_max, "tiny")
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elif f0_method == "mangio-crepe":
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f0 = self.get_f0_crepe_computation(
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x, f0_min, f0_max, p_len, crepe_hop_length
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)
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elif f0_method == "mangio-crepe-tiny":
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f0 = self.get_f0_crepe_computation(
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x, f0_min, f0_max, p_len, crepe_hop_length, "tiny"
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)
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elif f0_method == "rmvpe":
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f0 = model_rmvpe.infer_from_audio(x, thred=0.03)
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elif "hybrid" in f0_method:
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# Perform hybrid median pitch estimation
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input_audio_path2wav[input_audio_path] = x.astype(np.double)
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f0 = self.get_f0_hybrid_computation(
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f0_method,
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input_audio_path,
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x,
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f0_min,
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f0_max,
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p_len,
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filter_radius,
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crepe_hop_length,
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time_step,
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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[self.x_pad * tf0 : self.x_pad * tf0 + len(replace_f0)].shape[0]
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f0[self.x_pad * tf0 : self.x_pad * tf0 + len(replace_f0)] = replace_f0[
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:shape
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]
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# with open("test_opt.txt","w")as f:f.write("\n".join([str(i)for i in f0.tolist()]))
|
355 |
-
f0bak = f0.copy()
|
356 |
-
f0_mel = 1127 * np.log(1 + f0 / 700)
|
357 |
-
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (
|
358 |
-
f0_mel_max - f0_mel_min
|
359 |
-
) + 1
|
360 |
-
f0_mel[f0_mel <= 1] = 1
|
361 |
-
f0_mel[f0_mel > 255] = 255
|
362 |
-
f0_coarse = np.rint(f0_mel).astype(int)
|
363 |
-
|
364 |
-
return f0_coarse, f0bak # 1-0
|
365 |
-
|
366 |
-
def vc(
|
367 |
-
self,
|
368 |
-
model,
|
369 |
-
net_g,
|
370 |
-
sid,
|
371 |
-
audio0,
|
372 |
-
pitch,
|
373 |
-
pitchf,
|
374 |
-
times,
|
375 |
-
index,
|
376 |
-
big_npy,
|
377 |
-
index_rate,
|
378 |
-
version,
|
379 |
-
protect,
|
380 |
-
): # ,file_index,file_big_npy
|
381 |
-
feats = torch.from_numpy(audio0)
|
382 |
-
if self.is_half:
|
383 |
-
feats = feats.half()
|
384 |
-
else:
|
385 |
-
feats = feats.float()
|
386 |
-
if feats.dim() == 2: # double channels
|
387 |
-
feats = feats.mean(-1)
|
388 |
-
assert feats.dim() == 1, feats.dim()
|
389 |
-
feats = feats.view(1, -1)
|
390 |
-
padding_mask = torch.BoolTensor(feats.shape).to(self.device).fill_(False)
|
391 |
-
|
392 |
-
inputs = {
|
393 |
-
"source": feats.to(self.device),
|
394 |
-
"padding_mask": padding_mask,
|
395 |
-
"output_layer": 9 if version == "v1" else 12,
|
396 |
-
}
|
397 |
-
t0 = ttime()
|
398 |
-
with torch.no_grad():
|
399 |
-
logits = model.extract_features(**inputs)
|
400 |
-
feats = model.final_proj(logits[0]) if version == "v1" else logits[0]
|
401 |
-
if protect < 0.5 and pitch != None and pitchf != None:
|
402 |
-
feats0 = feats.clone()
|
403 |
-
if (
|
404 |
-
isinstance(index, type(None)) == False
|
405 |
-
and isinstance(big_npy, type(None)) == False
|
406 |
-
and index_rate != 0
|
407 |
-
):
|
408 |
-
npy = feats[0].cpu().numpy()
|
409 |
-
if self.is_half:
|
410 |
-
npy = npy.astype("float32")
|
411 |
-
|
412 |
-
# _, I = index.search(npy, 1)
|
413 |
-
# npy = big_npy[I.squeeze()]
|
414 |
-
|
415 |
-
score, ix = index.search(npy, k=8)
|
416 |
-
weight = np.square(1 / score)
|
417 |
-
weight /= weight.sum(axis=1, keepdims=True)
|
418 |
-
npy = np.sum(big_npy[ix] * np.expand_dims(weight, axis=2), axis=1)
|
419 |
-
|
420 |
-
if self.is_half:
|
421 |
-
npy = npy.astype("float16")
|
422 |
-
feats = (
|
423 |
-
torch.from_numpy(npy).unsqueeze(0).to(self.device) * index_rate
|
424 |
-
+ (1 - index_rate) * feats
|
425 |
-
)
|
426 |
-
|
427 |
-
feats = F.interpolate(feats.permute(0, 2, 1), scale_factor=2).permute(0, 2, 1)
|
428 |
-
if protect < 0.5 and pitch != None and pitchf != None:
|
429 |
-
feats0 = F.interpolate(feats0.permute(0, 2, 1), scale_factor=2).permute(
|
430 |
-
0, 2, 1
|
431 |
-
)
|
432 |
-
t1 = ttime()
|
433 |
-
p_len = audio0.shape[0] // self.window
|
434 |
-
if feats.shape[1] < p_len:
|
435 |
-
p_len = feats.shape[1]
|
436 |
-
if pitch != None and pitchf != None:
|
437 |
-
pitch = pitch[:, :p_len]
|
438 |
-
pitchf = pitchf[:, :p_len]
|
439 |
-
|
440 |
-
if protect < 0.5 and pitch != None and pitchf != None:
|
441 |
-
pitchff = pitchf.clone()
|
442 |
-
pitchff[pitchf > 0] = 1
|
443 |
-
pitchff[pitchf < 1] = protect
|
444 |
-
pitchff = pitchff.unsqueeze(-1)
|
445 |
-
feats = feats * pitchff + feats0 * (1 - pitchff)
|
446 |
-
feats = feats.to(feats0.dtype)
|
447 |
-
p_len = torch.tensor([p_len], device=self.device).long()
|
448 |
-
with torch.no_grad():
|
449 |
-
if pitch != None and pitchf != None:
|
450 |
-
audio1 = (
|
451 |
-
(net_g.infer(feats, p_len, pitch, pitchf, sid)[0][0, 0])
|
452 |
-
.data.cpu()
|
453 |
-
.float()
|
454 |
-
.numpy()
|
455 |
-
)
|
456 |
-
else:
|
457 |
-
audio1 = (
|
458 |
-
(net_g.infer(feats, p_len, sid)[0][0, 0]).data.cpu().float().numpy()
|
459 |
-
)
|
460 |
-
del feats, p_len, padding_mask
|
461 |
-
if torch.cuda.is_available():
|
462 |
-
torch.cuda.empty_cache()
|
463 |
-
t2 = ttime()
|
464 |
-
times[0] += t1 - t0
|
465 |
-
times[2] += t2 - t1
|
466 |
-
return audio1
|
467 |
-
|
468 |
-
def pipeline(
|
469 |
-
self,
|
470 |
-
model,
|
471 |
-
net_g,
|
472 |
-
sid,
|
473 |
-
audio,
|
474 |
-
input_audio_path,
|
475 |
-
times,
|
476 |
-
f0_up_key,
|
477 |
-
f0_method,
|
478 |
-
file_index,
|
479 |
-
# file_big_npy,
|
480 |
-
index_rate,
|
481 |
-
if_f0,
|
482 |
-
filter_radius,
|
483 |
-
tgt_sr,
|
484 |
-
resample_sr,
|
485 |
-
rms_mix_rate,
|
486 |
-
version,
|
487 |
-
protect,
|
488 |
-
crepe_hop_length,
|
489 |
-
f0_file=None,
|
490 |
-
):
|
491 |
-
if (
|
492 |
-
file_index != ""
|
493 |
-
# and file_big_npy != ""
|
494 |
-
# and os.path.exists(file_big_npy) == True
|
495 |
-
and os.path.exists(file_index) == True
|
496 |
-
and index_rate != 0
|
497 |
-
):
|
498 |
-
try:
|
499 |
-
index = faiss.read_index(file_index)
|
500 |
-
# big_npy = np.load(file_big_npy)
|
501 |
-
big_npy = index.reconstruct_n(0, index.ntotal)
|
502 |
-
except:
|
503 |
-
traceback.print_exc()
|
504 |
-
index = big_npy = None
|
505 |
-
else:
|
506 |
-
index = big_npy = None
|
507 |
-
audio = signal.filtfilt(bh, ah, audio)
|
508 |
-
audio_pad = np.pad(audio, (self.window // 2, self.window // 2), mode="reflect")
|
509 |
-
opt_ts = []
|
510 |
-
if audio_pad.shape[0] > self.t_max:
|
511 |
-
audio_sum = np.zeros_like(audio)
|
512 |
-
for i in range(self.window):
|
513 |
-
audio_sum += audio_pad[i : i - self.window]
|
514 |
-
for t in range(self.t_center, audio.shape[0], self.t_center):
|
515 |
-
opt_ts.append(
|
516 |
-
t
|
517 |
-
- self.t_query
|
518 |
-
+ np.where(
|
519 |
-
np.abs(audio_sum[t - self.t_query : t + self.t_query])
|
520 |
-
== np.abs(audio_sum[t - self.t_query : t + self.t_query]).min()
|
521 |
-
)[0][0]
|
522 |
-
)
|
523 |
-
s = 0
|
524 |
-
audio_opt = []
|
525 |
-
t = None
|
526 |
-
t1 = ttime()
|
527 |
-
audio_pad = np.pad(audio, (self.t_pad, self.t_pad), mode="reflect")
|
528 |
-
p_len = audio_pad.shape[0] // self.window
|
529 |
-
inp_f0 = None
|
530 |
-
if hasattr(f0_file, "name") == True:
|
531 |
-
try:
|
532 |
-
with open(f0_file.name, "r") as f:
|
533 |
-
lines = f.read().strip("\n").split("\n")
|
534 |
-
inp_f0 = []
|
535 |
-
for line in lines:
|
536 |
-
inp_f0.append([float(i) for i in line.split(",")])
|
537 |
-
inp_f0 = np.array(inp_f0, dtype="float32")
|
538 |
-
except:
|
539 |
-
traceback.print_exc()
|
540 |
-
sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
|
541 |
-
pitch, pitchf = None, None
|
542 |
-
if if_f0 == 1:
|
543 |
-
pitch, pitchf = self.get_f0(
|
544 |
-
input_audio_path,
|
545 |
-
audio_pad,
|
546 |
-
p_len,
|
547 |
-
f0_up_key,
|
548 |
-
f0_method,
|
549 |
-
filter_radius,
|
550 |
-
crepe_hop_length,
|
551 |
-
inp_f0,
|
552 |
-
)
|
553 |
-
pitch = pitch[:p_len]
|
554 |
-
pitchf = pitchf[:p_len]
|
555 |
-
if self.device == "mps":
|
556 |
-
pitchf = pitchf.astype(np.float32)
|
557 |
-
pitch = torch.tensor(pitch, device=self.device).unsqueeze(0).long()
|
558 |
-
pitchf = torch.tensor(pitchf, device=self.device).unsqueeze(0).float()
|
559 |
-
t2 = ttime()
|
560 |
-
times[1] += t2 - t1
|
561 |
-
for t in opt_ts:
|
562 |
-
t = t // self.window * self.window
|
563 |
-
if if_f0 == 1:
|
564 |
-
audio_opt.append(
|
565 |
-
self.vc(
|
566 |
-
model,
|
567 |
-
net_g,
|
568 |
-
sid,
|
569 |
-
audio_pad[s : t + self.t_pad2 + self.window],
|
570 |
-
pitch[:, s // self.window : (t + self.t_pad2) // self.window],
|
571 |
-
pitchf[:, s // self.window : (t + self.t_pad2) // self.window],
|
572 |
-
times,
|
573 |
-
index,
|
574 |
-
big_npy,
|
575 |
-
index_rate,
|
576 |
-
version,
|
577 |
-
protect,
|
578 |
-
)[self.t_pad_tgt : -self.t_pad_tgt]
|
579 |
-
)
|
580 |
-
else:
|
581 |
-
audio_opt.append(
|
582 |
-
self.vc(
|
583 |
-
model,
|
584 |
-
net_g,
|
585 |
-
sid,
|
586 |
-
audio_pad[s : t + self.t_pad2 + self.window],
|
587 |
-
None,
|
588 |
-
None,
|
589 |
-
times,
|
590 |
-
index,
|
591 |
-
big_npy,
|
592 |
-
index_rate,
|
593 |
-
version,
|
594 |
-
protect,
|
595 |
-
)[self.t_pad_tgt : -self.t_pad_tgt]
|
596 |
-
)
|
597 |
-
s = t
|
598 |
-
if if_f0 == 1:
|
599 |
-
audio_opt.append(
|
600 |
-
self.vc(
|
601 |
-
model,
|
602 |
-
net_g,
|
603 |
-
sid,
|
604 |
-
audio_pad[t:],
|
605 |
-
pitch[:, t // self.window :] if t is not None else pitch,
|
606 |
-
pitchf[:, t // self.window :] if t is not None else pitchf,
|
607 |
-
times,
|
608 |
-
index,
|
609 |
-
big_npy,
|
610 |
-
index_rate,
|
611 |
-
version,
|
612 |
-
protect,
|
613 |
-
)[self.t_pad_tgt : -self.t_pad_tgt]
|
614 |
-
)
|
615 |
-
else:
|
616 |
-
audio_opt.append(
|
617 |
-
self.vc(
|
618 |
-
model,
|
619 |
-
net_g,
|
620 |
-
sid,
|
621 |
-
audio_pad[t:],
|
622 |
-
None,
|
623 |
-
None,
|
624 |
-
times,
|
625 |
-
index,
|
626 |
-
big_npy,
|
627 |
-
index_rate,
|
628 |
-
version,
|
629 |
-
protect,
|
630 |
-
)[self.t_pad_tgt : -self.t_pad_tgt]
|
631 |
-
)
|
632 |
-
audio_opt = np.concatenate(audio_opt)
|
633 |
-
if rms_mix_rate != 1:
|
634 |
-
audio_opt = change_rms(audio, 16000, audio_opt, tgt_sr, rms_mix_rate)
|
635 |
-
if resample_sr >= 16000 and tgt_sr != resample_sr:
|
636 |
-
audio_opt = librosa.resample(
|
637 |
-
audio_opt, orig_sr=tgt_sr, target_sr=resample_sr
|
638 |
-
)
|
639 |
-
audio_max = np.abs(audio_opt).max() / 0.99
|
640 |
-
max_int16 = 32768
|
641 |
-
if audio_max > 1:
|
642 |
-
max_int16 /= audio_max
|
643 |
-
audio_opt = (audio_opt * max_int16).astype(np.int16)
|
644 |
-
del pitch, pitchf, sid
|
645 |
-
if torch.cuda.is_available():
|
646 |
-
torch.cuda.empty_cache()
|
647 |
-
return audio_opt
|
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