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import math
import os
import json
import random
import torch
# from torchvision.transforms.functional import resize
import torch.utils.data
import numpy as np
import librosa
from librosa.util import normalize
from scipy.io.wavfile import read
from librosa.filters import mel as librosa_mel_fn
# from speechbrain.lobes.models.FastSpeech2 import mel_spectogram
MAX_WAV_VALUE = 32768.0
def load_wav(full_path):
sampling_rate, data = read(full_path)
return data, sampling_rate
def dynamic_range_compression(x, C=1, clip_val=1e-5):
return np.log(np.clip(x, a_min=clip_val, a_max=None) * C)
def dynamic_range_decompression(x, C=1):
return np.exp(x) / C
def dynamic_range_compression_torch(x, C=1, clip_val=1e-5):
return torch.log(torch.clamp(x, min=clip_val) * C)
def dynamic_range_decompression_torch(x, C=1):
return torch.exp(x) / C
def spectral_normalize_torch(magnitudes):
output = dynamic_range_compression_torch(magnitudes)
return output
def spectral_de_normalize_torch(magnitudes):
output = dynamic_range_decompression_torch(magnitudes)
return output
mel_basis = {}
hann_window = {}
def mel_spectrogram(y, n_fft, num_mels, sampling_rate, hop_size, win_size, fmin, fmax, center=False):
# if torch.min(y) < -1.:
# print('min value is ', torch.min(y))
# if torch.max(y) > 1.:
# print('max value is ', torch.max(y))
global mel_basis, hann_window
if fmax not in mel_basis:
mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax)
mel_basis[str(fmax)+'_'+str(y.device)] = torch.from_numpy(mel).float().to(y.device)
hann_window[str(y.device)] = torch.hann_window(win_size).to(y.device)
y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_size)/2), int((n_fft-hop_size)/2)), mode='reflect')
y = y.squeeze(1)
# complex tensor as default, then use view_as_real for future pytorch compatibility
spec = torch.stft(y, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window[str(y.device)],
center=center, pad_mode='reflect', normalized=False, onesided=True, return_complex=True)
spec = torch.view_as_real(spec)
spec = torch.sqrt(spec.pow(2).sum(-1)+(1e-9))
spec = torch.matmul(mel_basis[str(fmax)+'_'+str(y.device)], spec)
spec = spectral_normalize_torch(spec)
return spec
def get_dataset_filelist(a):
training_files =[]
validation_files =[]
total_files = 0
audio_dir = "dataset/audio"
with open("filelists/train.txt") as f:
training_files = f.readlines()
for i, line in enumerate(training_files):
spk, basename = line.strip().split('|')
training_files[i] = f"{audio_dir}/{spk}/{basename}.wav"
with open("filelists/val.txt") as f:
validation_files = f.readlines()
for i, line in enumerate(validation_files):
spk, basename = line.strip().split('|')
validation_files[i] = f"{audio_dir}/{spk}/{basename}.wav"
random.seed(1234)
random.shuffle(training_files)
random.shuffle(validation_files)
return training_files, validation_files
class MelDataset(torch.utils.data.Dataset):
def __init__(self, training_files, segment_size, n_fft, num_mels,
hop_size, win_size, sampling_rate, fmin, fmax, shuffle=True, n_cache_reuse=1,
device=None, fmax_loss=None, use_aug=False):
self.audio_files = training_files
random.seed(1234)
if shuffle:
random.shuffle(self.audio_files)
self.segment_size = segment_size
self.sampling_rate = sampling_rate
self.n_fft = n_fft
self.num_mels = num_mels
self.hop_size = hop_size
self.win_size = win_size
self.fmin = fmin
self.fmax = fmax
self.fmax_loss = fmax_loss
self.cached_wav = None
self.n_cache_reuse = n_cache_reuse
self._cache_ref_count = 0
self.device = device
self.use_aug = use_aug
with open("filelists/spk2id.json") as f:
self.spk2id = json.load(f)
def __getitem__(self, index):
filename = self.audio_files[index]
if self._cache_ref_count == 0:
audio, sampling_rate = load_wav(filename)
audio = audio / MAX_WAV_VALUE
audio = normalize(audio) * 0.95
self.cached_wav = audio
if sampling_rate != self.sampling_rate:
raise ValueError("{} SR doesn't match target {} SR".format(
sampling_rate, self.sampling_rate))
self._cache_ref_count = self.n_cache_reuse
else:
audio = self.cached_wav
self._cache_ref_count -= 1
audio = torch.FloatTensor(audio)
audio = audio.unsqueeze(0)
if audio.size(1) >= self.segment_size:
max_audio_start = audio.size(1) - self.segment_size
audio_start = random.randint(0, max_audio_start)
audio = audio[:, audio_start:audio_start+self.segment_size]
else:
audio = torch.nn.functional.pad(audio, (0, self.segment_size - audio.size(1)), 'constant')
mel = mel_spectrogram(audio, self.n_fft, self.num_mels,
self.sampling_rate, self.hop_size, self.win_size, self.fmin, self.fmax,
center=False)
mel_loss = mel_spectrogram(audio, self.n_fft, self.num_mels,
self.sampling_rate, self.hop_size, self.win_size, self.fmin, self.fmax_loss,
center=False)
spk_path = filename.replace("audio", "spk").replace(".wav", ".npy")
spk_emb = torch.from_numpy(np.load(spk_path)) # (256)
spk = filename.split("/")[-2]
spk_id = self.spk2id[spk]
spk_id = torch.LongTensor([spk_id])
if not self.use_aug:
return (mel.squeeze(), audio.squeeze(0), filename, mel_loss.squeeze(), spk_emb, spk_id)
mel_aug, _ = mel_spectogram(
audio=audio.squeeze(),
sample_rate=16000,
hop_length=256,
win_length=1024,
n_mels=80,
n_fft=1024,
f_min=0.0,
f_max=8000.0,
power=1,
normalized=False,
min_max_energy_norm=True,
norm="slaney",
mel_scale="slaney",
compression=True
)
mel_aug = self.resize_mel(mel_aug.unsqueeze(0)).squeeze(0)
return (mel_aug.squeeze(), mel.squeeze(), audio.squeeze(0), filename, mel_loss.squeeze(), spk_emb, spk_id)
def __len__(self):
return len(self.audio_files)
def resize_mel(self, mel):
ratio = 0.85 + 0.3 * torch.rand(1) # 0.85 ~ 1.15
height = int(mel.size(-2) * ratio)
width = mel.size(-1)
mel_r = resize(mel, (height, width), antialias=True)
if height >= mel.size(-2):
mel_r = mel_r[:, :mel.size(-2), :]
else:
pad = mel_r[:, -1:, :].repeat(1, mel.size(-2) - height, 1)
pad += torch.randn_like(pad) / 1e3
mel_r = torch.cat((mel_r, pad), 1)
return mel_r
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