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https://github.com/w-okada/voice-changer.git
synced 2025-01-23 13:35:12 +03:00
WIP: docker support v1.5.x trial 4
This commit is contained in:
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commit
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@ -28,18 +28,24 @@ RUN python3 warmup.py
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ADD dummy /
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ADD dummy /
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RUN git clone --depth 1 -b v1.5.0.0_SiFiGAN https://github.com/isletennos/MMVC_Trainer.git
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RUN git clone -b v1.5.0.0_SiFiGAN https://github.com/isletennos/MMVC_Trainer.git
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WORKDIR /MMVC_Trainer/
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WORKDIR /MMVC_Trainer/
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RUN git checkout c242d3d1cf7f768af70d9735082ca2bdd90c45f3
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#RUN git checkout 1109f1201e1ee9008ee61dc221d620eb1c93d7d0
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RUN git clone https://github.com/isletennos/MMVC_Client.git
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RUN git checkout f28a1514b6731c36bd68989c600bff0a3b217d75
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WORKDIR /MMVC_Trainer/MMVC_Client
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RUN git checkout 3374a1177b73e3f6d600e5dbe93af033c36ee120
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WORKDIR /
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# ↓ テストスクリプトはTrainerのrootに置くとmodelsがconflictする。
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ADD /scripts /MMVC_Trainer/MMVC_Client/python/
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ADD /model/D_v15_best.pth /MMVC_Trainer/fine_model/
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ADD /model/D_v15_best.pth /MMVC_Trainer/fine_model/
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ADD /model/G_v15_best.pth /MMVC_Trainer/fine_model/
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ADD /model/G_v15_best.pth /MMVC_Trainer/fine_model/
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RUN cp -r /MMVC_Trainer/configs /MMVC_Trainer/configs_org
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RUN cp -r /MMVC_Trainer/configs /MMVC_Trainer/configs_org
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WORKDIR /MMVC_Trainer/
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@ -58,3 +58,8 @@ $ python3 train_ms.py -c configs/train_config.json -m 20220306_24000 -fg fine_mo
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$ python3 train_ms.py -c configs/train_config.json -m 20220306_24000
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$ python3 train_ms.py -c configs/train_config.json -m 20220306_24000
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```
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```
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(x) テスト
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```
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$ python3 MMVC_Client/python/conver_test.py -m logs/G_40000.pth -c configs/train_config.json -s 0 -t 101 --input dataset/00_myvoice/wav/emotion011.wav --output dataset/test.wav --f0_scale 3
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```
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208
docker_trainer/scripts/client_modules.py
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208
docker_trainer/scripts/client_modules.py
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@ -0,0 +1,208 @@
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from features import SignalGenerator, dilated_factor
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from scipy.interpolate import interp1d
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import torch
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import numpy as np
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import json
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import os
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hann_window = {}
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class TextAudioSpeakerCollate():
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""" Zero-pads model inputs and targets
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"""
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def __init__(
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self,
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sample_rate,
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hop_size,
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f0_factor=1.0,
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dense_factors=[0.5, 1, 4, 8],
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upsample_scales=[8, 4, 2, 2],
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sine_amp=0.1,
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noise_amp=0.003,
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signal_types=["sine"],
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):
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self.dense_factors = dense_factors
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self.prod_upsample_scales = np.cumprod(upsample_scales)
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self.sample_rate = sample_rate
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self.signal_generator = SignalGenerator(
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sample_rate=sample_rate,
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hop_size=hop_size,
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sine_amp=sine_amp,
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noise_amp=noise_amp,
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signal_types=signal_types,
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)
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self.f0_factor = f0_factor
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def __call__(self, batch):
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"""Collate's training batch from normalized text, audio and speaker identities
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PARAMS
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------
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batch: [text_normalized, spec_normalized, wav_normalized, sid, note]
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"""
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spec_lengths = torch.LongTensor(len(batch))
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sid = torch.LongTensor(len(batch))
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spec_padded = torch.FloatTensor(len(batch), batch[0][0].size(0), batch[0][0].size(1))
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f0_padded = torch.FloatTensor(len(batch), 1, batch[0][2].size(0))
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# 返り値の初期化
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spec_padded.zero_()
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f0_padded.zero_()
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# dfs
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dfs_batch = [[] for _ in range(len(self.dense_factors))]
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# row spec, sid, f0
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for i in range(len(batch)):
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row = batch[i]
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spec = row[0]
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spec_padded[i, :, :spec.size(1)] = spec
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spec_lengths[i] = spec.size(1)
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sid[i] = row[1]
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# 推論時 f0/cf0にf0の倍率を乗算してf0/cf0を求める
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f0 = row[2] * self.f0_factor
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f0_padded[i, :, :f0.size(0)] = f0
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# dfs
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dfs = []
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# dilated_factor の入力はnumpy!!
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for df, us in zip(self.dense_factors, self.prod_upsample_scales):
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dfs += [
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np.repeat(dilated_factor(torch.unsqueeze(f0, dim=1).to('cpu').detach().numpy(), self.sample_rate, df), us)
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]
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# よくわからないけど、後で論文ちゃんと読む
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for i in range(len(self.dense_factors)):
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dfs_batch[i] += [
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dfs[i].astype(np.float32).reshape(-1, 1)
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] # [(T', 1), ...]
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# よくわからないdfsを転置
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for i in range(len(self.dense_factors)):
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dfs_batch[i] = torch.FloatTensor(np.array(dfs_batch[i])).transpose(
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2, 1
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) # (B, 1, T')
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# f0/cf0を実際に使うSignalに変換する
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in_batch = self.signal_generator(f0_padded)
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return spec_padded, spec_lengths, sid, in_batch, dfs_batch
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def convert_continuos_f0(f0, f0_size):
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# get start and end of f0
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if (f0 == 0).all():
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return np.zeros((f0_size,))
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start_f0 = f0[f0 != 0][0]
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end_f0 = f0[f0 != 0][-1]
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# padding start and end of f0 sequence
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cf0 = f0
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start_idx = np.where(cf0 == start_f0)[0][0]
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end_idx = np.where(cf0 == end_f0)[0][-1]
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cf0[:start_idx] = start_f0
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cf0[end_idx:] = end_f0
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# get non-zero frame index
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nz_frames = np.where(cf0 != 0)[0]
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# perform linear interpolation
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f = interp1d(nz_frames, cf0[nz_frames], bounds_error=False, fill_value=0.0)
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cf0_ = f(np.arange(0, f0_size))
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# print(cf0.shape, cf0_.shape, f0.shape, f0_size)
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# print(cf0_)
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return f(np.arange(0, f0_size))
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def spectrogram_torch(y, n_fft, sampling_rate, hop_size, win_size, center=False):
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if torch.min(y) < -1.:
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print('min value is ', torch.min(y))
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if torch.max(y) > 1.:
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print('max value is ', torch.max(y))
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dtype_device = str(y.dtype) + '_' + str(y.device)
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wnsize_dtype_device = str(win_size) + '_' + dtype_device
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if wnsize_dtype_device not in hann_window:
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hann_window[wnsize_dtype_device] = torch.hann_window(win_size).to(dtype=y.dtype, device=y.device)
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y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft - hop_size) / 2), int((n_fft - hop_size) / 2)), mode='reflect')
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y = y.squeeze(1)
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spec = torch.stft(y, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window[wnsize_dtype_device],
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center=center, pad_mode='reflect', normalized=False, onesided=True, return_complex=True)
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spec = torch.view_as_real(spec)
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spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-6)
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return spec
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def get_hparams_from_file(config_path):
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with open(config_path, "r", encoding="utf-8") as f:
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data = f.read()
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config = json.loads(data)
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hparams = HParams(**config)
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return hparams
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class HParams():
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def __init__(self, **kwargs):
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for k, v in kwargs.items():
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if type(v) == dict:
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v = HParams(**v)
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self[k] = v
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def keys(self):
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return self.__dict__.keys()
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def items(self):
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return self.__dict__.items()
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def values(self):
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return self.__dict__.values()
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def __len__(self):
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return len(self.__dict__)
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def __getitem__(self, key):
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return getattr(self, key)
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def __setitem__(self, key, value):
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return setattr(self, key, value)
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def __contains__(self, key):
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return key in self.__dict__
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def __repr__(self):
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return self.__dict__.__repr__()
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def load_checkpoint(checkpoint_path, model, optimizer=None):
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assert os.path.isfile(checkpoint_path), f"No such file or directory: {checkpoint_path}"
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checkpoint_dict = torch.load(checkpoint_path, map_location='cpu')
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iteration = checkpoint_dict['iteration']
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learning_rate = checkpoint_dict['learning_rate']
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if optimizer is not None:
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optimizer.load_state_dict(checkpoint_dict['optimizer'])
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saved_state_dict = {
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**checkpoint_dict['pe'],
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**checkpoint_dict['flow'],
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**checkpoint_dict['text_enc'],
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**checkpoint_dict['dec'],
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**checkpoint_dict['emb_g']
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}
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if hasattr(model, 'module'):
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state_dict = model.module.state_dict()
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else:
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state_dict = model.state_dict()
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new_state_dict = {}
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for k, v in state_dict.items():
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try:
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new_state_dict[k] = saved_state_dict[k]
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except:
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new_state_dict[k] = v
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if hasattr(model, 'module'):
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model.module.load_state_dict(new_state_dict)
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else:
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model.load_state_dict(new_state_dict)
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return model, optimizer, learning_rate, iteration
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104
docker_trainer/scripts/conver_test.py
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104
docker_trainer/scripts/conver_test.py
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import sys
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sys.path.append(".") # sifiganへのパスが必要。
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import argparse
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import torch
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import numpy as np
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from scipy.io.wavfile import write, read
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import pyworld as pw
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from logging import getLogger
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# import utils
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from models import SynthesizerTrn
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# from mmvc_client import Hyperparameters # <- pyaudioなどが必要になるため必要なロジックのみコピペ
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from client_modules import convert_continuos_f0, spectrogram_torch, TextAudioSpeakerCollate, get_hparams_from_file, load_checkpoint
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logger = getLogger(__name__)
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def setupArgParser():
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parser = argparse.ArgumentParser()
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parser.add_argument("-c", type=str, required=True, help="path for the config.json")
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parser.add_argument("-m", type=str, help="path for the pytorch model file")
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parser.add_argument("-o", type=str, help="path for the onnx model file")
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parser.add_argument("-s", type=int, required=True, help="source speaker id")
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parser.add_argument("-t", type=int, required=True, help="target speaker id")
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parser.add_argument("--input", type=str, required=True, help="input wav file")
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parser.add_argument("--output", type=str, required=True, help="outpu wav file")
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parser.add_argument("--f0_scale", type=float, required=True, help="f0 scale")
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return parser
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def create_model(hps, pytorch_model_file):
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net_g = SynthesizerTrn(
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spec_channels=hps.data.filter_length // 2 + 1,
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segment_size=hps.train.segment_size // hps.data.hop_length,
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inter_channels=hps.model.inter_channels,
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hidden_channels=hps.model.hidden_channels,
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upsample_rates=hps.model.upsample_rates,
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upsample_initial_channel=hps.model.upsample_initial_channel,
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upsample_kernel_sizes=hps.model.upsample_kernel_sizes,
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n_flow=hps.model.n_flow,
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dec_out_channels=1,
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dec_kernel_size=7,
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n_speakers=hps.data.n_speakers,
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gin_channels=hps.model.gin_channels,
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requires_grad_pe=hps.requires_grad.pe,
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requires_grad_flow=hps.requires_grad.flow,
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requires_grad_text_enc=hps.requires_grad.text_enc,
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requires_grad_dec=hps.requires_grad.dec
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)
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_ = net_g.eval()
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_ = load_checkpoint(pytorch_model_file, net_g, None)
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return net_g
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def convert(hps, ssid, tsid, input, output, f0_scale):
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sr, signal = read(input)
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signal = signal / hps.data.max_wav_value
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_f0, _time = pw.dio(signal, hps.data.sampling_rate, frame_period=5.5)
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f0 = pw.stonemask(signal, _f0, _time, hps.data.sampling_rate)
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f0 = convert_continuos_f0(f0, int(signal.shape[0] / hps.data.hop_length))
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f0 = torch.from_numpy(f0.astype(np.float32))
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signal = torch.from_numpy(signal.astype(np.float32)).clone()
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signal = signal.unsqueeze(0)
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spec = spectrogram_torch(signal, hps.data.filter_length, hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length, center=False)
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spec = torch.squeeze(spec, 0)
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sid = torch.LongTensor([int(ssid)])
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data = TextAudioSpeakerCollate(
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sample_rate=hps.data.sampling_rate,
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hop_size=hps.data.hop_length,
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f0_factor=f0_scale
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)([(spec, sid, f0)])
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spec, spec_lengths, sid_src, sin, d = data
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spec = spec.cuda()
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spec_lengths = spec_lengths.cuda()
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sid_src = sid_src.cuda()
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sin = sin.cuda()
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d = tuple([d[:1].cuda() for d in d])
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sid_target = torch.LongTensor([tsid]).cuda()
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audio = net_g.cuda().voice_conversion(spec, spec_lengths, sin, d, sid_src, sid_target)[0, 0].data.cpu().float().numpy()
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# print(audio)
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write(output, 24000, audio)
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if __name__ == '__main__':
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print("main")
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parser = setupArgParser()
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||||||
|
args = parser.parse_args()
|
||||||
|
|
||||||
|
CONFIG_PATH = args.c
|
||||||
|
hps = get_hparams_from_file(CONFIG_PATH)
|
||||||
|
pytorch_model_file = args.m
|
||||||
|
ssid = args.s
|
||||||
|
tsid = args.t
|
||||||
|
input = args.input
|
||||||
|
output = args.output
|
||||||
|
f0_scale = args.f0_scale
|
||||||
|
|
||||||
|
net_g = create_model(hps, pytorch_model_file)
|
||||||
|
convert(hps, ssid, tsid, input, output, f0_scale)
|
@ -1,7 +1,7 @@
|
|||||||
#!/bin/bash
|
#!/bin/bash
|
||||||
set -eu
|
set -eu
|
||||||
|
|
||||||
DOCKER_IMAGE=dannadori/trainer:20230209_214044
|
DOCKER_IMAGE=dannadori/trainer:20230210_153105
|
||||||
# DOCKER_IMAGE=trainer
|
# DOCKER_IMAGE=trainer
|
||||||
|
|
||||||
docker run --gpus all --rm -ti \
|
docker run --gpus all --rm -ti \
|
||||||
|
Loading…
Reference in New Issue
Block a user