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* ruff check --fix * ruff format --line-length 120 --target-version py39 * Change the link for G2PW Model * update pytorch version and colab
109 lines
3.8 KiB
Python
109 lines
3.8 KiB
Python
import os
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import random
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import torch
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import torchaudio
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import torch.utils.data
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import torchaudio.functional as aF
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def amp_pha_stft(audio, n_fft, hop_size, win_size, center=True):
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hann_window = torch.hann_window(win_size).to(audio.device)
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stft_spec = torch.stft(
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audio,
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n_fft,
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hop_length=hop_size,
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win_length=win_size,
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window=hann_window,
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center=center,
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pad_mode="reflect",
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normalized=False,
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return_complex=True,
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)
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log_amp = torch.log(torch.abs(stft_spec) + 1e-4)
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pha = torch.angle(stft_spec)
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com = torch.stack((torch.exp(log_amp) * torch.cos(pha), torch.exp(log_amp) * torch.sin(pha)), dim=-1)
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return log_amp, pha, com
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def amp_pha_istft(log_amp, pha, n_fft, hop_size, win_size, center=True):
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amp = torch.exp(log_amp)
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com = torch.complex(amp * torch.cos(pha), amp * torch.sin(pha))
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hann_window = torch.hann_window(win_size).to(com.device)
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audio = torch.istft(com, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window, center=center)
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return audio
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def get_dataset_filelist(a):
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with open(a.input_training_file, "r", encoding="utf-8") as fi:
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training_indexes = [x.split("|")[0] for x in fi.read().split("\n") if len(x) > 0]
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with open(a.input_validation_file, "r", encoding="utf-8") as fi:
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validation_indexes = [x.split("|")[0] for x in fi.read().split("\n") if len(x) > 0]
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return training_indexes, validation_indexes
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class Dataset(torch.utils.data.Dataset):
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def __init__(
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self,
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training_indexes,
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wavs_dir,
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segment_size,
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hr_sampling_rate,
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lr_sampling_rate,
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split=True,
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shuffle=True,
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n_cache_reuse=1,
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device=None,
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):
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self.audio_indexes = training_indexes
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random.seed(1234)
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if shuffle:
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random.shuffle(self.audio_indexes)
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self.wavs_dir = wavs_dir
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self.segment_size = segment_size
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self.hr_sampling_rate = hr_sampling_rate
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self.lr_sampling_rate = lr_sampling_rate
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self.split = split
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self.cached_wav = None
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self.n_cache_reuse = n_cache_reuse
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self._cache_ref_count = 0
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self.device = device
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def __getitem__(self, index):
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filename = self.audio_indexes[index]
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if self._cache_ref_count == 0:
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audio, orig_sampling_rate = torchaudio.load(os.path.join(self.wavs_dir, filename + ".wav"))
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self.cached_wav = audio
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self._cache_ref_count = self.n_cache_reuse
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else:
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audio = self.cached_wav
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self._cache_ref_count -= 1
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if orig_sampling_rate == self.hr_sampling_rate:
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audio_hr = audio
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else:
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audio_hr = aF.resample(audio, orig_freq=orig_sampling_rate, new_freq=self.hr_sampling_rate)
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audio_lr = aF.resample(audio, orig_freq=orig_sampling_rate, new_freq=self.lr_sampling_rate)
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audio_lr = aF.resample(audio_lr, orig_freq=self.lr_sampling_rate, new_freq=self.hr_sampling_rate)
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audio_lr = audio_lr[:, : audio_hr.size(1)]
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if self.split:
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if audio_hr.size(1) >= self.segment_size:
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max_audio_start = audio_hr.size(1) - self.segment_size
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audio_start = random.randint(0, max_audio_start)
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audio_hr = audio_hr[:, audio_start : audio_start + self.segment_size]
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audio_lr = audio_lr[:, audio_start : audio_start + self.segment_size]
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else:
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audio_hr = torch.nn.functional.pad(audio_hr, (0, self.segment_size - audio_hr.size(1)), "constant")
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audio_lr = torch.nn.functional.pad(audio_lr, (0, self.segment_size - audio_lr.size(1)), "constant")
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return (audio_hr.squeeze(), audio_lr.squeeze())
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def __len__(self):
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return len(self.audio_indexes)
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