import os import pathlib import random import torch import torch.utils.data import torch.nn.functional as F import torchaudio from audio2numpy import open_audio from tqdm import tqdm from data.audio.wav_aug import WavAugmentor from data.util import find_files_of_type, is_wav_file, is_audio_file from models.tacotron2.taco_utils import load_wav_to_torch from utils.util import opt_get def load_audio(audiopath, sampling_rate): if audiopath[-4:] == '.wav': audio, lsr = load_wav_to_torch(audiopath) else: audio, lsr = open_audio(audiopath) audio = torch.FloatTensor(audio) # Remove any channel data. if len(audio.shape) > 1: if audio.shape[0] < 5: audio = audio[0] else: assert audio.shape[1] < 5 audio = audio[:, 0] if lsr != sampling_rate: if lsr < sampling_rate: print(f'{audiopath} has a sample rate of {sampling_rate} which is lower than the requested sample rate of {sampling_rate}. This is not a good idea.') audio = torch.nn.functional.interpolate(audio.unsqueeze(0).unsqueeze(1), scale_factor=sampling_rate/lsr, mode='nearest', recompute_scale_factor=False).squeeze() # Check some assumptions about audio range. This should be automatically fixed in load_wav_to_torch, but might not be in some edge cases, where we should squawk. # '2' is arbitrarily chosen since it seems like audio will often "overdrive" the [-1,1] bounds. if torch.any(audio > 2) or not torch.any(audio < 0): print(f"Error with {audiopath}. Max={audio.max()} min={audio.min()}") audio.clip_(-1, 1) return audio.unsqueeze(0) class UnsupervisedAudioDataset(torch.utils.data.Dataset): def __init__(self, opt): path = opt['path'] cache_path = opt['cache_path'] # Will fail when multiple paths specified, must be specified in this case. if not isinstance(path, list): path = [path] if os.path.exists(cache_path): self.audiopaths = torch.load(cache_path) else: print("Building cache..") self.audiopaths = [] for p in path: self.audiopaths.extend(find_files_of_type('img', p, qualifier=is_audio_file)[0]) torch.save(self.audiopaths, cache_path) # Parse options self.sampling_rate = opt_get(opt, ['sampling_rate'], 22050) self.pad_to = opt_get(opt, ['pad_to_seconds'], None) if self.pad_to is not None: self.pad_to *= self.sampling_rate self.pad_to = opt_get(opt, ['pad_to_samples'], self.pad_to) self.extra_samples = opt_get(opt, ['extra_samples'], 0) self.extra_sample_len = opt_get(opt, ['extra_sample_length'], 2) self.extra_sample_len *= self.sampling_rate def get_audio_for_index(self, index): audiopath = self.audiopaths[index] audio = load_audio(audiopath, self.sampling_rate) return audio, audiopath def get_related_audio_for_index(self, index): if self.extra_samples <= 0: return None audiopath = self.audiopaths[index] related_files = find_files_of_type('img', os.path.dirname(audiopath), qualifier=is_audio_file)[0] assert audiopath in related_files assert len(related_files) < 50000 # Sanity check to ensure we aren't loading "related files" that aren't actually related. if len(related_files) == 0: print(f"No related files for {audiopath}") related_files.remove(audiopath) related_clips = [] random.shuffle(related_clips) i = 0 for related_file in related_files: rel_clip = load_audio(related_file, self.sampling_rate) gap = rel_clip.shape[-1] - self.extra_sample_len if gap < 0: rel_clip = F.pad(rel_clip, pad=(0, abs(gap))) elif gap > 0: rand_start = random.randint(0, gap) rel_clip = rel_clip[:, rand_start:rand_start+self.extra_sample_len] related_clips.append(rel_clip) i += 1 if i >= self.extra_samples: break actual_extra_samples = i while i < self.extra_samples: related_clips.append(torch.zeros(1, self.extra_sample_len)) i += 1 return torch.stack(related_clips, dim=0), actual_extra_samples def __getitem__(self, index): # Split audio_norm into two tensors of equal size. audio_norm, filename = self.get_audio_for_index(index) alt_files, actual_samples = self.get_related_audio_for_index(index) # This is required when training to make sure all clips align. if self.pad_to is not None: if audio_norm.shape[-1] <= self.pad_to: audio_norm = torch.nn.functional.pad(audio_norm, (0, self.pad_to - audio_norm.shape[-1])) else: gap = audio_norm.shape[-1] - self.pad_to start = random.randint(0, gap-1) audio_norm = audio_norm[:, start:start+self.pad_to] output = { 'clip': audio_norm, 'alt_clips': alt_files, 'num_alt_clips': actual_samples, # We need to pad so that the dataloader can combine these. 'path': filename, } return output def __len__(self): return len(self.audiopaths) if __name__ == '__main__': params = { 'mode': 'unsupervised_audio', 'path': ['Z:\\split\\cleaned\\books0'], 'cache_path': 'E:\\audio\\remote-cache.pth', 'sampling_rate': 22050, 'pad_to_seconds': 5, 'phase': 'train', 'n_workers': 0, 'batch_size': 16, 'extra_samples': 4, } from data import create_dataset, create_dataloader, util ds = create_dataset(params) dl = create_dataloader(ds, params) i = 0 for b in tqdm(dl): for b_ in range(16): pass #torchaudio.save(f'{i}_clip1_{b_}.wav', b['clip1'][b_], ds.sampling_rate) #torchaudio.save(f'{i}_clip2_{b_}.wav', b['clip2'][b_], ds.sampling_rate) #i += 1