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# Copyright 2024 The YourMT3 Authors. | |
# | |
# Licensed under the Apache License, Version 2.0 (the "License"); | |
# you may not use this file except in compliance with the License. | |
# You may obtain a copy of the License at | |
# | |
# http://www.apache.org/licenses/LICENSE-2.0 | |
# | |
# Please see the details in the LICENSE file. | |
from typing import Dict, Tuple | |
from copy import deepcopy | |
import soundfile as sf | |
import torch | |
from utils.data_modules import AMTDataModule | |
from config.data_presets import data_preset_single_cfg, data_preset_multi_cfg | |
from utils.augment import intra_stem_augment_processor | |
def get_ds(data_preset_multi: Dict, train_num_samples_per_epoch: int = 90000): | |
dm = AMTDataModule(data_preset_multi=data_preset_multi, train_num_samples_per_epoch=train_num_samples_per_epoch) | |
dm.setup('fit') | |
dl = dm.train_dataloader() | |
ds = dl.flattened[0].dataset | |
return ds | |
def debug_func(num_segments: int = 10): | |
sampled_data, sampled_ids = ds._get_rand_segments_from_cache(num_segments) | |
ux_sampled_data, _ = ds._get_rand_segments_from_cache(ux_count_sum, False, sampled_ids) | |
s = deepcopy(sampled_data) | |
intra_stem_augment_processor(sampled_data, submix_audio=False) | |
def gen_audio(index: int = 0): | |
# audio_arr: (b, 1, nframe), note_token_arr: (b, l), task_token_arr: (b, task_l) | |
audio_arr, note_token_arr, task_token_arr = ds.__getitem__(index) | |
# merge all the segments into one audio file | |
audio = audio_arr.permute(0, 2, 1).reshape(-1).squeeze().numpy() | |
# save the audio file | |
sf.write('xaug_demo_audio.wav', audio, 16000, subtype='PCM_16') | |
data_preset_multi = data_preset_multi_cfg["all_cross_rebal5"] | |
ds = get_ds(data_preset_multi) | |
ds.random_amp_range = [0.8, 1.1] | |
ds.stem_xaug_policy = { | |
"max_k": 5, | |
"tau": 0.3, | |
"alpha": 1.0, | |
"max_subunit_stems": 12, | |
"no_instr_overlap": True, | |
"no_drum_overlap": True, | |
"uhat_intra_stem_augment": True, | |
} | |
gen_audio(3) | |
# for k in ds.cache.keys(): | |
# arr = ds.cache[k]['audio_array'] | |
# arr = np.sum(arr, axis=1).reshape(-1) | |
# # sf.write(f'xxx/{k}.wav', arr, 16000, subtype='PCM_16') | |
# if np.min(arr) > -0.5: | |
# print(k) | |
# arr = ds.cache[52]['audio_array'] | |
# for i in range(arr.shape[1]): | |
# a = arr[:, i, :].reshape(-1) | |
# sf.write(f'xxx52/52_{i}.wav', a, 16000, subtype='PCM_16') | |