161 lines
5.5 KiB
Python
161 lines
5.5 KiB
Python
import hashlib
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import json
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import logging
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import os
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import time
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from pathlib import Path
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import io
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import librosa
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import maad
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import numpy as np
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from inference import slicer
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import parselmouth
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import soundfile
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import torch
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import torchaudio
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from hubert import hubert_model
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import utils
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from models import SynthesizerTrn
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logging.getLogger('numba').setLevel(logging.WARNING)
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logging.getLogger('matplotlib').setLevel(logging.WARNING)
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def resize2d_f0(x, target_len):
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source = np.array(x)
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source[source < 0.001] = np.nan
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target = np.interp(np.arange(0, len(source) * target_len, len(source)) / target_len, np.arange(0, len(source)),
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source)
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res = np.nan_to_num(target)
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return res
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def get_f0(x, p_len,f0_up_key=0):
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time_step = 160 / 16000 * 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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f0 = parselmouth.Sound(x, 16000).to_pitch_ac(
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time_step=time_step / 1000, voicing_threshold=0.6,
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pitch_floor=f0_min, pitch_ceiling=f0_max).selected_array['frequency']
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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(f0,[[pad_size,p_len - len(f0) - pad_size]], mode='constant')
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f0 *= pow(2, f0_up_key / 12)
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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 / (f0_mel_max - f0_mel_min) + 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, f0
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def clean_pitch(input_pitch):
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num_nan = np.sum(input_pitch == 1)
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if num_nan / len(input_pitch) > 0.9:
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input_pitch[input_pitch != 1] = 1
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return input_pitch
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def plt_pitch(input_pitch):
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input_pitch = input_pitch.astype(float)
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input_pitch[input_pitch == 1] = np.nan
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return input_pitch
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def f0_to_pitch(ff):
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f0_pitch = 69 + 12 * np.log2(ff / 440)
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return f0_pitch
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def fill_a_to_b(a, b):
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if len(a) < len(b):
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for _ in range(0, len(b) - len(a)):
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a.append(a[0])
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def mkdir(paths: list):
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for path in paths:
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if not os.path.exists(path):
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os.mkdir(path)
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class VitsSvc(object):
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def __init__(self):
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self.device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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self.SVCVITS = None
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self.hps = None
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self.speakers = None
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self.hubert_soft = utils.get_hubert_model()
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def set_device(self, device):
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self.device = torch.device(device)
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self.hubert_soft.to(self.device)
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if self.SVCVITS != None:
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self.SVCVITS.to(self.device)
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def loadCheckpoint(self, path):
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self.hps = utils.get_hparams_from_file(f"checkpoints/{path}/config.json")
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self.SVCVITS = SynthesizerTrn(
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self.hps.data.filter_length // 2 + 1,
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self.hps.train.segment_size // self.hps.data.hop_length,
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**self.hps.model)
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_ = utils.load_checkpoint(f"checkpoints/{path}/model.pth", self.SVCVITS, None)
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_ = self.SVCVITS.eval().to(self.device)
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self.speakers = self.hps.spk
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def get_units(self, source, sr):
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source = source.unsqueeze(0).to(self.device)
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with torch.inference_mode():
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units = self.hubert_soft.units(source)
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return units
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def get_unit_pitch(self, in_path, tran):
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source, sr = torchaudio.load(in_path)
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source = torchaudio.functional.resample(source, sr, 16000)
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if len(source.shape) == 2 and source.shape[1] >= 2:
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source = torch.mean(source, dim=0).unsqueeze(0)
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soft = self.get_units(source, sr).squeeze(0).cpu().numpy()
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f0_coarse, f0 = get_f0(source.cpu().numpy()[0], soft.shape[0]*2, tran)
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return soft, f0
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def infer(self, speaker_id, tran, raw_path):
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speaker_id = self.speakers[speaker_id]
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sid = torch.LongTensor([int(speaker_id)]).to(self.device).unsqueeze(0)
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soft, pitch = self.get_unit_pitch(raw_path, tran)
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f0 = torch.FloatTensor(clean_pitch(pitch)).unsqueeze(0).to(self.device)
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stn_tst = torch.FloatTensor(soft)
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with torch.no_grad():
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x_tst = stn_tst.unsqueeze(0).to(self.device)
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x_tst = torch.repeat_interleave(x_tst, repeats=2, dim=1).transpose(1, 2)
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audio = self.SVCVITS.infer(x_tst, f0=f0, g=sid)[0,0].data.float()
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return audio, audio.shape[-1]
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def inference(self,srcaudio,chara,tran,slice_db):
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sampling_rate, audio = srcaudio
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audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
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if len(audio.shape) > 1:
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audio = librosa.to_mono(audio.transpose(1, 0))
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if sampling_rate != 16000:
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audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000)
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soundfile.write("tmpwav.wav", audio, 16000, format="wav")
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chunks = slicer.cut("tmpwav.wav", db_thresh=slice_db)
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audio_data, audio_sr = slicer.chunks2audio("tmpwav.wav", chunks)
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audio = []
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for (slice_tag, data) in audio_data:
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length = int(np.ceil(len(data) / audio_sr * self.hps.data.sampling_rate))
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raw_path = io.BytesIO()
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soundfile.write(raw_path, data, audio_sr, format="wav")
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raw_path.seek(0)
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if slice_tag:
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_audio = np.zeros(length)
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else:
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out_audio, out_sr = self.infer(chara, tran, raw_path)
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_audio = out_audio.cpu().numpy()
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audio.extend(list(_audio))
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audio = (np.array(audio) * 32768.0).astype('int16')
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return (self.hps.data.sampling_rate,audio)
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