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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
# LICENSE is in incl_licenses directory.
import math
import os
import random
import torch
import torch.utils.data
import numpy as np
from librosa.util import normalize
from scipy.io.wavfile import read
from librosa.filters import mel as librosa_mel_fn
import pathlib
from tqdm import tqdm
MAX_WAV_VALUE = 32767.0 # NOTE: 32768.0 -1 to prevent int16 overflow (results in popping sound in corner cases)
def load_wav(full_path, sr_target):
sampling_rate, data = read(full_path)
if sampling_rate != sr_target:
raise RuntimeError(
f"Sampling rate of the file {full_path} is {sampling_rate} Hz, but the model requires {sr_target} Hz"
)
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):
return dynamic_range_compression_torch(magnitudes)
def spectral_de_normalize_torch(magnitudes):
return dynamic_range_decompression_torch(magnitudes)
mel_basis_cache = {}
hann_window_cache = {}
def mel_spectrogram(
y: torch.Tensor,
n_fft: int,
num_mels: int,
sampling_rate: int,
hop_size: int,
win_size: int,
fmin: int,
fmax: int = None,
center: bool = False,
) -> torch.Tensor:
"""
Calculate the mel spectrogram of an input signal.
This function uses slaney norm for the librosa mel filterbank (using librosa.filters.mel) and uses Hann window for STFT (using torch.stft).
Args:
y (torch.Tensor): Input signal.
n_fft (int): FFT size.
num_mels (int): Number of mel bins.
sampling_rate (int): Sampling rate of the input signal.
hop_size (int): Hop size for STFT.
win_size (int): Window size for STFT.
fmin (int): Minimum frequency for mel filterbank.
fmax (int): Maximum frequency for mel filterbank. If None, defaults to half the sampling rate (fmax = sr / 2.0) inside librosa_mel_fn
center (bool): Whether to pad the input to center the frames. Default is False.
Returns:
torch.Tensor: Mel spectrogram.
"""
if torch.min(y) < -1.0:
print(f"[WARNING] Min value of input waveform signal is {torch.min(y)}")
if torch.max(y) > 1.0:
print(f"[WARNING] Max value of input waveform signal is {torch.max(y)}")
device = y.device
key = f"{n_fft}_{num_mels}_{sampling_rate}_{hop_size}_{win_size}_{fmin}_{fmax}_{device}"
if key not in mel_basis_cache:
mel = librosa_mel_fn(
sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax
)
mel_basis_cache[key] = torch.from_numpy(mel).float().to(device)
hann_window_cache[key] = torch.hann_window(win_size).to(device)
mel_basis = mel_basis_cache[key]
hann_window = hann_window_cache[key]
padding = (n_fft - hop_size) // 2
y = torch.nn.functional.pad(
y.unsqueeze(1), (padding, padding), mode="reflect"
).squeeze(1)
spec = torch.stft(
y,
n_fft,
hop_length=hop_size,
win_length=win_size,
window=hann_window,
center=center,
pad_mode="reflect",
normalized=False,
onesided=True,
return_complex=True,
)
spec = torch.sqrt(torch.view_as_real(spec).pow(2).sum(-1) + 1e-9)
mel_spec = torch.matmul(mel_basis, spec)
mel_spec = spectral_normalize_torch(mel_spec)
return mel_spec
def get_mel_spectrogram(wav, h):
"""
Generate mel spectrogram from a waveform using given hyperparameters.
Args:
wav (torch.Tensor): Input waveform.
h: Hyperparameters object with attributes n_fft, num_mels, sampling_rate, hop_size, win_size, fmin, fmax.
Returns:
torch.Tensor: Mel spectrogram.
"""
return mel_spectrogram(
wav,
h.n_fft,
h.num_mels,
h.sampling_rate,
h.hop_size,
h.win_size,
h.fmin,
h.fmax,
)
def get_dataset_filelist(a):
training_files = []
validation_files = []
list_unseen_validation_files = []
with open(a.input_training_file, "r", encoding="utf-8") as fi:
training_files = [
os.path.join(a.input_wavs_dir, x.split("|")[0] + ".wav")
for x in fi.read().split("\n")
if len(x) > 0
]
print(f"first training file: {training_files[0]}")
with open(a.input_validation_file, "r", encoding="utf-8") as fi:
validation_files = [
os.path.join(a.input_wavs_dir, x.split("|")[0] + ".wav")
for x in fi.read().split("\n")
if len(x) > 0
]
print(f"first validation file: {validation_files[0]}")
for i in range(len(a.list_input_unseen_validation_file)):
with open(a.list_input_unseen_validation_file[i], "r", encoding="utf-8") as fi:
unseen_validation_files = [
os.path.join(a.list_input_unseen_wavs_dir[i], x.split("|")[0] + ".wav")
for x in fi.read().split("\n")
if len(x) > 0
]
print(
f"first unseen {i}th validation fileset: {unseen_validation_files[0]}"
)
list_unseen_validation_files.append(unseen_validation_files)
return training_files, validation_files, list_unseen_validation_files
class MelDataset(torch.utils.data.Dataset):
def __init__(
self,
training_files,
hparams,
segment_size,
n_fft,
num_mels,
hop_size,
win_size,
sampling_rate,
fmin,
fmax,
split=True,
shuffle=True,
n_cache_reuse=1,
device=None,
fmax_loss=None,
fine_tuning=False,
base_mels_path=None,
is_seen=True,
):
self.audio_files = training_files
random.seed(1234)
if shuffle:
random.shuffle(self.audio_files)
self.hparams = hparams
self.is_seen = is_seen
if self.is_seen:
self.name = pathlib.Path(self.audio_files[0]).parts[0]
else:
self.name = "-".join(pathlib.Path(self.audio_files[0]).parts[:2]).strip("/")
self.segment_size = segment_size
self.sampling_rate = sampling_rate
self.split = split
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.fine_tuning = fine_tuning
self.base_mels_path = base_mels_path
print("[INFO] checking dataset integrity...")
for i in tqdm(range(len(self.audio_files))):
assert os.path.exists(
self.audio_files[i]
), f"{self.audio_files[i]} not found"
def __getitem__(self, index):
filename = self.audio_files[index]
if self._cache_ref_count == 0:
audio, sampling_rate = load_wav(filename, self.sampling_rate)
audio = audio / MAX_WAV_VALUE
if not self.fine_tuning:
audio = normalize(audio) * 0.95
self.cached_wav = audio
if sampling_rate != self.sampling_rate:
raise ValueError(
f"{sampling_rate} SR doesn't match target {self.sampling_rate} SR"
)
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 not self.fine_tuning:
if self.split:
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,
)
else: # Validation step
# Match audio length to self.hop_size * n for evaluation
if (audio.size(1) % self.hop_size) != 0:
audio = audio[:, : -(audio.size(1) % self.hop_size)]
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,
)
assert (
audio.shape[1] == mel.shape[2] * self.hop_size
), f"audio shape {audio.shape} mel shape {mel.shape}"
else:
mel = np.load(
os.path.join(
self.base_mels_path,
os.path.splitext(os.path.split(filename)[-1])[0] + ".npy",
)
)
mel = torch.from_numpy(mel)
if len(mel.shape) < 3:
mel = mel.unsqueeze(0)
if self.split:
frames_per_seg = math.ceil(self.segment_size / self.hop_size)
if audio.size(1) >= self.segment_size:
mel_start = random.randint(0, mel.size(2) - frames_per_seg - 1)
mel = mel[:, :, mel_start : mel_start + frames_per_seg]
audio = audio[
:,
mel_start
* self.hop_size : (mel_start + frames_per_seg)
* self.hop_size,
]
else:
mel = torch.nn.functional.pad(
mel, (0, frames_per_seg - mel.size(2)), "constant"
)
audio = torch.nn.functional.pad(
audio, (0, self.segment_size - audio.size(1)), "constant"
)
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,
)
return (mel.squeeze(), audio.squeeze(0), filename, mel_loss.squeeze())
def __len__(self):
return len(self.audio_files)
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