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DatasetLoader.py
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#! /usr/bin/python
# -*- encoding: utf-8 -*-
import torch
import torchaudio
import numpy
import sys
import random
import pdb
import os
import threading
import time
import math
import glob
import soundfile
from scipy import signal
from scipy.io import wavfile
from torch.utils.data import Dataset, DataLoader
import torch.distributed as dist
import h5py
def round_down(num, divisor):
return num - (num%divisor)
# TODO: implement for multi
def worker_init_fn(worker_id):
numpy.random.seed(numpy.random.get_state()[1][0] + worker_id)
def loadWAV(index, max_frames, evalmode=True, num_eval=10):
# Maximum audio length
max_audio = max_frames * 160 + 240
if evalmode:
filename = index
audio, sample_rate = soundfile.read(filename)
audiosize = audio.shape[0]
startframe = numpy.linspace(0,audiosize-max_audio,num=num_eval)
speaker_id = 0
else:
# Read wav file and convert to torch tensor
h5_file = h5py.File('/home/ubuntu/Documents/vt1code/voxceleb_trainer/data/h5_files/vox2.raw.h5','r') # TODO: change this
#sample_size = 30000
(start, end, speaker_id) = h5_file["y"][int(index), :]
duration = end - start
""" if duration < sample_size:
sample_size = duration
start = random.randint(0,duration-sample_size) + start
audio = h5_file["x"][int(start):int(start+sample_size)] """
audio = h5_file["x"][int(start):int(end)]
audiosize = audio.shape[0]
if audiosize <= max_audio:
shortage = max_audio - audiosize + 1
audio = numpy.pad(audio, (0, shortage), 'wrap')
audiosize = audio.shape[0]
startframe = numpy.array([numpy.int64(random.random()*(audiosize-max_audio))])
feats = []
if evalmode and max_frames == 0:
feats.append(audio)
else:
for asf in startframe:
feats.append(audio[int(asf):int(asf)+max_audio])
feat = numpy.stack(feats,axis=0).astype(numpy.float)
return feat, speaker_id
class train_dataset_loader(Dataset):
def __init__(self, train_list, augment, musan_path, rir_path, max_frames, train_path, **kwargs):
self.augment_wav = AugmentWAV(musan_path=musan_path, rir_path=rir_path, max_frames = max_frames)
self.train_list = train_list
self.max_frames = max_frames
self.musan_path = musan_path
self.rir_path = rir_path
self.augment = augment
self.train_path = train_path
self.data_label = []
h5_file = h5py.File('/home/ubuntu/Documents/vt1code/voxceleb_trainer/data/h5_files/vox2.raw.h5','r') # TODO: change this
for index in range(len(h5_file["y"])):
(_, _, speaker_id) = h5_file["y"][int(index), :]
self.data_label.append(speaker_id)
""" # Read training files
with open(train_list) as dataset_file:
lines = dataset_file.readlines()
# Make a dictionary of ID names and ID indices
dictkeys = list(set([x.split()[0] for x in lines]))
dictkeys.sort()
dictkeys = { key : ii for ii, key in enumerate(dictkeys) }
# Parse the training list into file names and ID indices
self.data_list = []
self.data_label = []
for lidx, line in enumerate(lines):
data = line.strip().split()
speaker_label = dictkeys[data[0]]
filename = os.path.join(train_path,data[1])
self.data_label.append(speaker_label)
self.data_list.append(filename) """
def __getitem__(self, indices):
feat = []
#indices = random.sample(self.batch_size, len(self.data_list))
#for index in range(len(data_label)): # in indices TODO
for index in indices:
audio, speaker_id = loadWAV(index, self.max_frames, evalmode=False)
""" if self.augment:
augtype = random.randint(0,4)
if augtype == 1:
audio = self.augment_wav.reverberate(audio)
elif augtype == 2:
audio = self.augment_wav.additive_noise('music',audio)
elif augtype == 3:
audio = self.augment_wav.additive_noise('speech',audio)
elif augtype == 4:
audio = self.augment_wav.additive_noise('noise',audio) """
feat.append(audio)
return torch.FloatTensor(numpy.array(feat)), speaker_id # X, y
def __len__(self):
return len(self.data_list)
class test_dataset_loader(Dataset):
def __init__(self, test_list, test_path, eval_frames, num_eval, **kwargs):
self.max_frames = eval_frames
self.num_eval = num_eval
self.test_path = test_path
self.test_list = test_list
def __getitem__(self, index):
audio, _ = loadWAV(os.path.join(self.test_path,self.test_list[index]), self.max_frames, evalmode=True, num_eval=self.num_eval)
return torch.FloatTensor(audio), self.test_list[index]
def __len__(self):
return len(self.test_list)
class train_dataset_sampler(torch.utils.data.Sampler):
def __init__(self, data_source, nPerSpeaker, max_seg_per_spk, batch_size, distributed, seed, **kwargs):
self.data_label = data_source.data_label
self.nPerSpeaker = nPerSpeaker
self.max_seg_per_spk = max_seg_per_spk
self.batch_size = batch_size
self.epoch = 0
self.seed = seed
self.distributed = distributed
def __iter__(self):
g = torch.Generator()
g.manual_seed(self.seed + self.epoch)
indices = torch.randperm(len(self.data_label), generator=g).tolist()
data_dict = {}
# Sort into dictionary of file indices for each ID
for index in indices:
speaker_label = self.data_label[index]
if not (speaker_label in data_dict):
data_dict[speaker_label] = []
data_dict[speaker_label].append(index)
## Group file indices for each class
dictkeys = list(data_dict.keys())
dictkeys.sort()
lol = lambda lst, sz: [lst[i:i+sz] for i in range(0, len(lst), sz)]
flattened_list = []
flattened_label = []
for findex, key in enumerate(dictkeys):
data = data_dict[key]
numSeg = round_down(min(len(data),self.max_seg_per_spk),self.nPerSpeaker)
rp = lol(numpy.arange(numSeg),self.nPerSpeaker)
flattened_label.extend([findex] * (len(rp)))
for indices in rp:
flattened_list.append([data[i] for i in indices])
## Mix data in random order
mixid = torch.randperm(len(flattened_label), generator=g).tolist()
mixlabel = []
mixmap = []
## Prevent two pairs of the same speaker in the same batch
for ii in mixid:
startbatch = round_down(len(mixlabel), self.batch_size)
if flattened_label[ii] not in mixlabel[startbatch:]:
mixlabel.append(flattened_label[ii])
mixmap.append(ii)
mixed_list = [flattened_list[i] for i in mixmap]
## Divide data to each GPU
if self.distributed:
total_size = round_down(len(mixed_list), self.batch_size * dist.get_world_size())
start_index = int ( ( dist.get_rank() ) / dist.get_world_size() * total_size )
end_index = int ( ( dist.get_rank() + 1 ) / dist.get_world_size() * total_size )
self.num_samples = end_index - start_index
return iter(mixed_list[start_index:end_index])
else:
total_size = round_down(len(mixed_list), self.batch_size)
self.num_samples = total_size
return iter(mixed_list[:total_size])
def __len__(self) -> int:
return self.num_samples
def set_epoch(self, epoch: int) -> None:
self.epoch = epoch
class AugmentWAV(object):
def __init__(self, musan_path, rir_path, max_frames):
self.max_frames = max_frames
self.max_audio = max_audio = max_frames * 160 + 240
self.noisetypes = ['noise','speech','music']
self.noisesnr = {'noise':[0,15],'speech':[13,20],'music':[5,15]}
self.numnoise = {'noise':[1,1], 'speech':[3,7], 'music':[1,1] }
self.noiselist = {}
augment_files = glob.glob(os.path.join(musan_path,'*/*/*/*.wav'));
for file in augment_files:
if not file.split('/')[-4] in self.noiselist:
self.noiselist[file.split('/')[-4]] = []
self.noiselist[file.split('/')[-4]].append(file)
self.rir_files = glob.glob(os.path.join(rir_path,'*/*/*.wav'));
def additive_noise(self, noisecat, audio):
clean_db = 10 * numpy.log10(numpy.mean(audio ** 2)+1e-4)
numnoise = self.numnoise[noisecat]
noiselist = random.sample(self.noiselist[noisecat], random.randint(numnoise[0],numnoise[1]))
noises = []
for noise in noiselist:
noiseaudio, _ = loadWAV(noise, self.max_frames, evalmode=False)
noise_snr = random.uniform(self.noisesnr[noisecat][0],self.noisesnr[noisecat][1])
noise_db = 10 * numpy.log10(numpy.mean(noiseaudio[0] ** 2)+1e-4)
noises.append(numpy.sqrt(10 ** ((clean_db - noise_db - noise_snr) / 10)) * noiseaudio)
return numpy.sum(numpy.concatenate(noises,axis=0),axis=0,keepdims=True) + audio
def reverberate(self, audio):
rir_file = random.choice(self.rir_files)
rir, fs = soundfile.read(rir_file)
rir = numpy.expand_dims(rir.astype(numpy.float),0)
rir = rir / numpy.sqrt(numpy.sum(rir**2))
return signal.convolve(audio, rir, mode='full')[:,:self.max_audio]