import numpy as np import av import torch as t import jukebox.utils.dist_adapter as dist def get_duration_sec(file, cache=False): try: with open(file + '.dur', 'r') as f: duration = float(f.readline().strip('\n')) return duration except: container = av.open(file) audio = container.streams.get(audio=0)[0] duration = audio.duration * float(audio.time_base) if cache: with open(file + '.dur', 'w') as f: f.write(str(duration) + '\n') return duration def load_audio(file, sr, offset, duration, resample=True, approx=False, time_base='samples', check_duration=True): if time_base == 'sec': offset = offset * sr duration = duration * sr # Loads at target sr, stereo channels, seeks from offset, and stops after duration container = av.open(file) audio = container.streams.get(audio=0)[0] # Only first audio stream audio_duration = audio.duration * float(audio.time_base) if approx: if offset + duration > audio_duration*sr: # Move back one window. Cap at audio_duration offset = np.min(audio_duration*sr - duration, offset - duration) else: if check_duration: assert offset + duration <= audio_duration*sr, f'End {offset + duration} beyond duration {audio_duration*sr}' if resample: resampler = av.AudioResampler(format='fltp',layout='stereo', rate=sr) else: assert sr == audio.sample_rate offset = int(offset / sr / float(audio.time_base)) #int(offset / float(audio.time_base)) # Use units of time_base for seeking duration = int(duration) #duration = int(duration * sr) # Use units of time_out ie 1/sr for returning sig = np.zeros((2, duration), dtype=np.float32) container.seek(offset, stream=audio) total_read = 0 for frame in container.decode(audio=0): # Only first audio stream if resample: frame.pts = None frame = resampler.resample(frame) frame = frame.to_ndarray(format='fltp') # Convert to floats and not int16 read = frame.shape[-1] if total_read + read > duration: read = duration - total_read sig[:, total_read:total_read + read] = frame[:, :read] total_read += read if total_read == duration: break assert total_read <= duration, f'Expected {duration} frames, got {total_read}' return sig, sr def test_simple_loader(): import librosa from tqdm import tqdm collate_fn = lambda batch: t.stack([t.from_numpy(b) for b in batch], dim=0) def get_batch(file, loader): y1, sr = loader(file, sr=44100, offset=0.0, duration=6.0, time_base='sec') y2, sr = loader(file, sr=44100, offset=20.0, duration=6.0, time_base='sec') return [y1, y2] def load(file, loader): batch = get_batch(file, loader) # np x = collate_fn(batch) # torch cpu x = x.to('cuda', non_blocking=True) # torch gpu return x files = librosa.util.find_files('/root/data/', ['mp3', 'm4a', 'opus']) print(files[:10]) loader = load_audio print("Loader", loader.__name__) x = t.randn(2, 2).cuda() x = load(files[0], loader) for i,file in enumerate(tqdm(files)): x = load(file, loader) if i == 100: break def test_dataset_loader(): from tqdm import tqdm from torch.utils.data import DataLoader from torch.utils.data.distributed import DistributedSampler from jukebox.utils.audio_utils import audio_preprocess, audio_postprocess from jukebox.hparams import setup_hparams from jukebox.data.files_dataset import FilesAudioDataset hps = setup_hparams("teeny", {}) hps.sr = 22050 # 44100 hps.hop_length = 512 hps.labels = False hps.channels = 2 hps.aug_shift = False hps.bs = 2 hps.nworkers = 2 # Getting 20 it/s with 2 workers, 10 it/s with 1 worker print(hps) dataset = hps.dataset root = hps.root from tensorboardX import SummaryWriter sr = {22050: '22k', 44100: '44k', 48000: '48k'}[hps.sr] writer = SummaryWriter(f'{root}/{dataset}/logs/{sr}/logs') dataset = FilesAudioDataset(hps) print("Length of dataset", len(dataset)) # Torch Loader collate_fn = lambda batch: t.stack([t.from_numpy(b) for b in batch], 0) sampler = DistributedSampler(dataset) train_loader = DataLoader(dataset, batch_size=hps.bs, num_workers=hps.nworkers, pin_memory=False, sampler=sampler, drop_last=True, collate_fn=collate_fn) dist.barrier() sampler.set_epoch(0) for i, x in enumerate(tqdm(train_loader)): x = x.to('cuda', non_blocking=True) for j, aud in enumerate(x): writer.add_audio('in_' + str(i*hps.bs + j), aud, 1, hps.sr) print("Wrote in") x = audio_preprocess(x, hps) x = audio_postprocess(x, hps) for j, aud in enumerate(x): writer.add_audio('out_' + str(i*hps.bs + j), aud, 1, hps.sr) print("Wrote out") dist.barrier() break if __name__ == '__main__': from jukebox.utils.dist_utils import setup_dist_from_mpi setup_dist_from_mpi(port=29500) test_dataset_loader()