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from typing import Dict, Tuple | |
import librosa | |
import numpy as np | |
import scipy.io.wavfile | |
import scipy.signal | |
import soundfile as sf | |
import torch | |
from torch import nn | |
# from TTS.tts.utils.helpers import StandardScaler | |
class StandardScaler: | |
"""StandardScaler for mean-scale normalization with the given mean and scale values.""" | |
def __init__(self, mean: np.ndarray = None, scale: np.ndarray = None) -> None: | |
self.mean_ = mean | |
self.scale_ = scale | |
def set_stats(self, mean, scale): | |
self.mean_ = mean | |
self.scale_ = scale | |
def reset_stats(self): | |
delattr(self, "mean_") | |
delattr(self, "scale_") | |
def transform(self, X): | |
X = np.asarray(X) | |
X -= self.mean_ | |
X /= self.scale_ | |
return X | |
def inverse_transform(self, X): | |
X = np.asarray(X) | |
X *= self.scale_ | |
X += self.mean_ | |
return X | |
class TorchSTFT(nn.Module): # pylint: disable=abstract-method | |
"""Some of the audio processing funtions using Torch for faster batch processing. | |
TODO: Merge this with audio.py | |
Args: | |
n_fft (int): | |
FFT window size for STFT. | |
hop_length (int): | |
number of frames between STFT columns. | |
win_length (int, optional): | |
STFT window length. | |
pad_wav (bool, optional): | |
If True pad the audio with (n_fft - hop_length) / 2). Defaults to False. | |
window (str, optional): | |
The name of a function to create a window tensor that is applied/multiplied to each frame/window. Defaults to "hann_window" | |
sample_rate (int, optional): | |
target audio sampling rate. Defaults to None. | |
mel_fmin (int, optional): | |
minimum filter frequency for computing melspectrograms. Defaults to None. | |
mel_fmax (int, optional): | |
maximum filter frequency for computing melspectrograms. Defaults to None. | |
n_mels (int, optional): | |
number of melspectrogram dimensions. Defaults to None. | |
use_mel (bool, optional): | |
If True compute the melspectrograms otherwise. Defaults to False. | |
do_amp_to_db_linear (bool, optional): | |
enable/disable amplitude to dB conversion of linear spectrograms. Defaults to False. | |
spec_gain (float, optional): | |
gain applied when converting amplitude to DB. Defaults to 1.0. | |
power (float, optional): | |
Exponent for the magnitude spectrogram, e.g., 1 for energy, 2 for power, etc. Defaults to None. | |
use_htk (bool, optional): | |
Use HTK formula in mel filter instead of Slaney. | |
mel_norm (None, 'slaney', or number, optional): | |
If 'slaney', divide the triangular mel weights by the width of the mel band | |
(area normalization). | |
If numeric, use `librosa.util.normalize` to normalize each filter by to unit l_p norm. | |
See `librosa.util.normalize` for a full description of supported norm values | |
(including `+-np.inf`). | |
Otherwise, leave all the triangles aiming for a peak value of 1.0. Defaults to "slaney". | |
""" | |
def __init__( | |
self, | |
n_fft, | |
hop_length, | |
win_length, | |
pad_wav=False, | |
window="hann_window", | |
sample_rate=None, | |
mel_fmin=0, | |
mel_fmax=None, | |
n_mels=80, | |
use_mel=False, | |
do_amp_to_db=False, | |
spec_gain=1.0, | |
power=None, | |
use_htk=False, | |
mel_norm="slaney", | |
): | |
super().__init__() | |
self.n_fft = n_fft | |
self.hop_length = hop_length | |
self.win_length = win_length | |
self.pad_wav = pad_wav | |
self.sample_rate = sample_rate | |
self.mel_fmin = mel_fmin | |
self.mel_fmax = mel_fmax | |
self.n_mels = n_mels | |
self.use_mel = use_mel | |
self.do_amp_to_db = do_amp_to_db | |
self.spec_gain = spec_gain | |
self.power = power | |
self.use_htk = use_htk | |
self.mel_norm = mel_norm | |
self.window = nn.Parameter(getattr(torch, window)(win_length), requires_grad=False) | |
self.mel_basis = None | |
if use_mel: | |
self._build_mel_basis() | |
def __call__(self, x): | |
"""Compute spectrogram frames by torch based stft. | |
Args: | |
x (Tensor): input waveform | |
Returns: | |
Tensor: spectrogram frames. | |
Shapes: | |
x: [B x T] or [:math:`[B, 1, T]`] | |
""" | |
if x.ndim == 2: | |
x = x.unsqueeze(1) | |
if self.pad_wav: | |
padding = int((self.n_fft - self.hop_length) / 2) | |
x = torch.nn.functional.pad(x, (padding, padding), mode="reflect") | |
# B x D x T x 2 | |
x_device = x.device | |
o = torch.stft( | |
x.squeeze(1).to(self.window.device), | |
self.n_fft, | |
self.hop_length, | |
self.win_length, | |
self.window, | |
center=True, | |
pad_mode="reflect", # compatible with audio.py | |
normalized=False, | |
onesided=True, | |
return_complex=False, | |
) | |
M = o[:, :, :, 0] | |
P = o[:, :, :, 1] | |
S = torch.sqrt(torch.clamp(M ** 2 + P ** 2, min=1e-8)) | |
if self.power is not None: | |
S = S ** self.power | |
if self.use_mel: | |
S = torch.matmul(self.mel_basis.to(self.window.device), S) | |
# S = torch.matmul(self.mel_basis, S) | |
if self.do_amp_to_db: | |
S = self._amp_to_db(S, spec_gain=self.spec_gain) | |
return S.to(x_device) | |
def _build_mel_basis(self): | |
mel_basis = librosa.filters.mel( | |
self.sample_rate, | |
self.n_fft, | |
n_mels=self.n_mels, | |
fmin=self.mel_fmin, | |
fmax=self.mel_fmax, | |
htk=self.use_htk, | |
norm=self.mel_norm, | |
) | |
self.mel_basis = torch.from_numpy(mel_basis).float() | |
def _amp_to_db(x, spec_gain=1.0): | |
return torch.log(torch.clamp(x, min=1e-5) * spec_gain) | |
def _db_to_amp(x, spec_gain=1.0): | |
return torch.exp(x) / spec_gain | |
# pylint: disable=too-many-public-methods | |
class AudioProcessor(object): | |
"""Audio Processor for TTS used by all the data pipelines. | |
TODO: Make this a dataclass to replace `BaseAudioConfig`. | |
Note: | |
All the class arguments are set to default values to enable a flexible initialization | |
of the class with the model config. They are not meaningful for all the arguments. | |
Args: | |
sample_rate (int, optional): | |
target audio sampling rate. Defaults to None. | |
resample (bool, optional): | |
enable/disable resampling of the audio clips when the target sampling rate does not match the original sampling rate. Defaults to False. | |
num_mels (int, optional): | |
number of melspectrogram dimensions. Defaults to None. | |
log_func (int, optional): | |
log exponent used for converting spectrogram aplitude to DB. | |
min_level_db (int, optional): | |
minimum db threshold for the computed melspectrograms. Defaults to None. | |
frame_shift_ms (int, optional): | |
milliseconds of frames between STFT columns. Defaults to None. | |
frame_length_ms (int, optional): | |
milliseconds of STFT window length. Defaults to None. | |
hop_length (int, optional): | |
number of frames between STFT columns. Used if ```frame_shift_ms``` is None. Defaults to None. | |
win_length (int, optional): | |
STFT window length. Used if ```frame_length_ms``` is None. Defaults to None. | |
ref_level_db (int, optional): | |
reference DB level to avoid background noise. In general <20DB corresponds to the air noise. Defaults to None. | |
fft_size (int, optional): | |
FFT window size for STFT. Defaults to 1024. | |
power (int, optional): | |
Exponent value applied to the spectrogram before GriffinLim. Defaults to None. | |
preemphasis (float, optional): | |
Preemphasis coefficient. Preemphasis is disabled if == 0.0. Defaults to 0.0. | |
signal_norm (bool, optional): | |
enable/disable signal normalization. Defaults to None. | |
symmetric_norm (bool, optional): | |
enable/disable symmetric normalization. If set True normalization is performed in the range [-k, k] else [0, k], Defaults to None. | |
max_norm (float, optional): | |
```k``` defining the normalization range. Defaults to None. | |
mel_fmin (int, optional): | |
minimum filter frequency for computing melspectrograms. Defaults to None. | |
mel_fmax (int, optional): | |
maximum filter frequency for computing melspectrograms. Defaults to None. | |
spec_gain (int, optional): | |
gain applied when converting amplitude to DB. Defaults to 20. | |
stft_pad_mode (str, optional): | |
Padding mode for STFT. Defaults to 'reflect'. | |
clip_norm (bool, optional): | |
enable/disable clipping the our of range values in the normalized audio signal. Defaults to True. | |
griffin_lim_iters (int, optional): | |
Number of GriffinLim iterations. Defaults to None. | |
do_trim_silence (bool, optional): | |
enable/disable silence trimming when loading the audio signal. Defaults to False. | |
trim_db (int, optional): | |
DB threshold used for silence trimming. Defaults to 60. | |
do_sound_norm (bool, optional): | |
enable/disable signal normalization. Defaults to False. | |
do_amp_to_db_linear (bool, optional): | |
enable/disable amplitude to dB conversion of linear spectrograms. Defaults to True. | |
do_amp_to_db_mel (bool, optional): | |
enable/disable amplitude to dB conversion of mel spectrograms. Defaults to True. | |
do_rms_norm (bool, optional): | |
enable/disable RMS volume normalization when loading an audio file. Defaults to False. | |
db_level (int, optional): | |
dB level used for rms normalization. The range is -99 to 0. Defaults to None. | |
stats_path (str, optional): | |
Path to the computed stats file. Defaults to None. | |
verbose (bool, optional): | |
enable/disable logging. Defaults to True. | |
""" | |
def __init__( | |
self, | |
# sample_rate=None, | |
sample_rate=22050, | |
resample=False, | |
num_mels=None, | |
log_func="np.log10", | |
min_level_db=None, | |
frame_shift_ms=None, | |
frame_length_ms=None, | |
# hop_length=None, | |
hop_length=256, | |
# win_length=None, | |
win_length=1024, | |
ref_level_db=None, | |
fft_size=1024, | |
power=None, | |
preemphasis=0.0, | |
signal_norm=None, | |
symmetric_norm=None, | |
max_norm=None, | |
mel_fmin=None, | |
mel_fmax=None, | |
spec_gain=20, | |
stft_pad_mode="reflect", | |
clip_norm=True, | |
griffin_lim_iters=None, | |
do_trim_silence=False, | |
trim_db=60, | |
do_sound_norm=False, | |
do_amp_to_db_linear=True, | |
do_amp_to_db_mel=True, | |
do_rms_norm=False, | |
db_level=None, | |
stats_path=None, | |
verbose=True, | |
**_, | |
): | |
# setup class attributed | |
self.sample_rate = sample_rate | |
self.resample = resample | |
self.num_mels = num_mels | |
self.log_func = log_func | |
self.min_level_db = min_level_db or 0 | |
self.frame_shift_ms = frame_shift_ms | |
self.frame_length_ms = frame_length_ms | |
self.ref_level_db = ref_level_db | |
self.fft_size = fft_size | |
self.power = power | |
self.preemphasis = preemphasis | |
self.griffin_lim_iters = griffin_lim_iters | |
self.signal_norm = signal_norm | |
self.symmetric_norm = symmetric_norm | |
self.mel_fmin = mel_fmin or 0 | |
self.mel_fmax = mel_fmax | |
self.spec_gain = float(spec_gain) | |
self.stft_pad_mode = stft_pad_mode | |
self.max_norm = 1.0 if max_norm is None else float(max_norm) | |
self.clip_norm = clip_norm | |
self.do_trim_silence = do_trim_silence | |
self.trim_db = trim_db | |
self.do_sound_norm = do_sound_norm | |
self.do_amp_to_db_linear = do_amp_to_db_linear | |
self.do_amp_to_db_mel = do_amp_to_db_mel | |
self.do_rms_norm = do_rms_norm | |
self.db_level = db_level | |
self.stats_path = stats_path | |
# setup exp_func for db to amp conversion | |
if log_func == "np.log": | |
self.base = np.e | |
elif log_func == "np.log10": | |
self.base = 10 | |
else: | |
raise ValueError(" [!] unknown `log_func` value.") | |
# setup stft parameters | |
if hop_length is None: | |
# compute stft parameters from given time values | |
self.hop_length, self.win_length = self._stft_parameters() | |
else: | |
# use stft parameters from config file | |
self.hop_length = hop_length | |
self.win_length = win_length | |
assert min_level_db != 0.0, " [!] min_level_db is 0" | |
assert self.win_length <= self.fft_size, " [!] win_length cannot be larger than fft_size" | |
# members = vars(self) | |
# if verbose: | |
# print(" > Setting up Audio Processor...") | |
# for key, value in members.items(): | |
# print(" | > {}:{}".format(key, value)) | |
# create spectrogram utils | |
self.mel_basis = self._build_mel_basis() | |
self.inv_mel_basis = np.linalg.pinv(self._build_mel_basis()) | |
# setup scaler | |
if stats_path and signal_norm: | |
mel_mean, mel_std, linear_mean, linear_std, _ = self.load_stats(stats_path) | |
self.setup_scaler(mel_mean, mel_std, linear_mean, linear_std) | |
self.signal_norm = True | |
self.max_norm = None | |
self.clip_norm = None | |
self.symmetric_norm = None | |
### setting up the parameters ### | |
def _build_mel_basis( | |
self, | |
) -> np.ndarray: | |
"""Build melspectrogram basis. | |
Returns: | |
np.ndarray: melspectrogram basis. | |
""" | |
if self.mel_fmax is not None: | |
assert self.mel_fmax <= self.sample_rate // 2 | |
return librosa.filters.mel( | |
self.sample_rate, self.fft_size, n_mels=self.num_mels, fmin=self.mel_fmin, fmax=self.mel_fmax | |
) | |
def _stft_parameters( | |
self, | |
) -> Tuple[int, int]: | |
"""Compute the real STFT parameters from the time values. | |
Returns: | |
Tuple[int, int]: hop length and window length for STFT. | |
""" | |
factor = self.frame_length_ms / self.frame_shift_ms | |
assert (factor).is_integer(), " [!] frame_shift_ms should divide frame_length_ms" | |
hop_length = int(self.frame_shift_ms / 1000.0 * self.sample_rate) | |
win_length = int(hop_length * factor) | |
return hop_length, win_length | |
### normalization ### | |
def normalize(self, S: np.ndarray) -> np.ndarray: | |
"""Normalize values into `[0, self.max_norm]` or `[-self.max_norm, self.max_norm]` | |
Args: | |
S (np.ndarray): Spectrogram to normalize. | |
Raises: | |
RuntimeError: Mean and variance is computed from incompatible parameters. | |
Returns: | |
np.ndarray: Normalized spectrogram. | |
""" | |
# pylint: disable=no-else-return | |
return S | |
# S = S.copy() | |
# if self.signal_norm: | |
# # mean-var scaling | |
# if hasattr(self, "mel_scaler"): | |
# if S.shape[0] == self.num_mels: | |
# return self.mel_scaler.transform(S.T).T | |
# elif S.shape[0] == self.fft_size / 2: | |
# return self.linear_scaler.transform(S.T).T | |
# else: | |
# raise RuntimeError(" [!] Mean-Var stats does not match the given feature dimensions.") | |
# # range normalization | |
# S -= self.ref_level_db # discard certain range of DB assuming it is air noise | |
# S_norm = (S - self.min_level_db) / (-self.min_level_db) | |
# if self.symmetric_norm: | |
# S_norm = ((2 * self.max_norm) * S_norm) - self.max_norm | |
# if self.clip_norm: | |
# S_norm = np.clip( | |
# S_norm, -self.max_norm, self.max_norm # pylint: disable=invalid-unary-operand-type | |
# ) | |
# return S_norm | |
# else: | |
# S_norm = self.max_norm * S_norm | |
# if self.clip_norm: | |
# S_norm = np.clip(S_norm, 0, self.max_norm) | |
# return S_norm | |
# else: | |
# return S | |
def denormalize(self, S: np.ndarray) -> np.ndarray: | |
"""Denormalize spectrogram values. | |
Args: | |
S (np.ndarray): Spectrogram to denormalize. | |
Raises: | |
RuntimeError: Mean and variance are incompatible. | |
Returns: | |
np.ndarray: Denormalized spectrogram. | |
""" | |
# pylint: disable=no-else-return | |
S_denorm = S.copy() | |
if self.signal_norm: | |
# mean-var scaling | |
if hasattr(self, "mel_scaler"): | |
if S_denorm.shape[0] == self.num_mels: | |
return self.mel_scaler.inverse_transform(S_denorm.T).T | |
elif S_denorm.shape[0] == self.fft_size / 2: | |
return self.linear_scaler.inverse_transform(S_denorm.T).T | |
else: | |
raise RuntimeError(" [!] Mean-Var stats does not match the given feature dimensions.") | |
if self.symmetric_norm: | |
if self.clip_norm: | |
S_denorm = np.clip( | |
S_denorm, -self.max_norm, self.max_norm # pylint: disable=invalid-unary-operand-type | |
) | |
S_denorm = ((S_denorm + self.max_norm) * -self.min_level_db / (2 * self.max_norm)) + self.min_level_db | |
return S_denorm + self.ref_level_db | |
else: | |
if self.clip_norm: | |
S_denorm = np.clip(S_denorm, 0, self.max_norm) | |
S_denorm = (S_denorm * -self.min_level_db / self.max_norm) + self.min_level_db | |
return S_denorm + self.ref_level_db | |
else: | |
return S_denorm | |
### Mean-STD scaling ### | |
def load_stats(self, stats_path: str) -> Tuple[np.array, np.array, np.array, np.array, Dict]: | |
"""Loading mean and variance statistics from a `npy` file. | |
Args: | |
stats_path (str): Path to the `npy` file containing | |
Returns: | |
Tuple[np.array, np.array, np.array, np.array, Dict]: loaded statistics and the config used to | |
compute them. | |
""" | |
stats = np.load(stats_path, allow_pickle=True).item() # pylint: disable=unexpected-keyword-arg | |
mel_mean = stats["mel_mean"] | |
mel_std = stats["mel_std"] | |
linear_mean = stats["linear_mean"] | |
linear_std = stats["linear_std"] | |
stats_config = stats["audio_config"] | |
# check all audio parameters used for computing stats | |
skip_parameters = ["griffin_lim_iters", "stats_path", "do_trim_silence", "ref_level_db", "power"] | |
for key in stats_config.keys(): | |
if key in skip_parameters: | |
continue | |
if key not in ["sample_rate", "trim_db"]: | |
assert ( | |
stats_config[key] == self.__dict__[key] | |
), f" [!] Audio param {key} does not match the value used for computing mean-var stats. {stats_config[key]} vs {self.__dict__[key]}" | |
return mel_mean, mel_std, linear_mean, linear_std, stats_config | |
# pylint: disable=attribute-defined-outside-init | |
def setup_scaler( | |
self, mel_mean: np.ndarray, mel_std: np.ndarray, linear_mean: np.ndarray, linear_std: np.ndarray | |
) -> None: | |
"""Initialize scaler objects used in mean-std normalization. | |
Args: | |
mel_mean (np.ndarray): Mean for melspectrograms. | |
mel_std (np.ndarray): STD for melspectrograms. | |
linear_mean (np.ndarray): Mean for full scale spectrograms. | |
linear_std (np.ndarray): STD for full scale spectrograms. | |
""" | |
self.mel_scaler = StandardScaler() | |
self.mel_scaler.set_stats(mel_mean, mel_std) | |
self.linear_scaler = StandardScaler() | |
self.linear_scaler.set_stats(linear_mean, linear_std) | |
### DB and AMP conversion ### | |
# pylint: disable=no-self-use | |
def _amp_to_db(self, x: np.ndarray) -> np.ndarray: | |
"""Convert amplitude values to decibels. | |
Args: | |
x (np.ndarray): Amplitude spectrogram. | |
Returns: | |
np.ndarray: Decibels spectrogram. | |
""" | |
return self.spec_gain * _log(np.maximum(1e-5, x), self.base) | |
# pylint: disable=no-self-use | |
def _db_to_amp(self, x: np.ndarray) -> np.ndarray: | |
"""Convert decibels spectrogram to amplitude spectrogram. | |
Args: | |
x (np.ndarray): Decibels spectrogram. | |
Returns: | |
np.ndarray: Amplitude spectrogram. | |
""" | |
return _exp(x / self.spec_gain, self.base) | |
### Preemphasis ### | |
def apply_preemphasis(self, x: np.ndarray) -> np.ndarray: | |
"""Apply pre-emphasis to the audio signal. Useful to reduce the correlation between neighbouring signal values. | |
Args: | |
x (np.ndarray): Audio signal. | |
Raises: | |
RuntimeError: Preemphasis coeff is set to 0. | |
Returns: | |
np.ndarray: Decorrelated audio signal. | |
""" | |
if self.preemphasis == 0: | |
raise RuntimeError(" [!] Preemphasis is set 0.0.") | |
return scipy.signal.lfilter([1, -self.preemphasis], [1], x) | |
def apply_inv_preemphasis(self, x: np.ndarray) -> np.ndarray: | |
"""Reverse pre-emphasis.""" | |
if self.preemphasis == 0: | |
raise RuntimeError(" [!] Preemphasis is set 0.0.") | |
return scipy.signal.lfilter([1], [1, -self.preemphasis], x) | |
### SPECTROGRAMs ### | |
def _linear_to_mel(self, spectrogram: np.ndarray) -> np.ndarray: | |
"""Project a full scale spectrogram to a melspectrogram. | |
Args: | |
spectrogram (np.ndarray): Full scale spectrogram. | |
Returns: | |
np.ndarray: Melspectrogram | |
""" | |
return np.dot(self.mel_basis, spectrogram) | |
def _mel_to_linear(self, mel_spec: np.ndarray) -> np.ndarray: | |
"""Convert a melspectrogram to full scale spectrogram.""" | |
return np.maximum(1e-10, np.dot(self.inv_mel_basis, mel_spec)) | |
def spectrogram(self, y: np.ndarray) -> np.ndarray: | |
"""Compute a spectrogram from a waveform. | |
Args: | |
y (np.ndarray): Waveform. | |
Returns: | |
np.ndarray: Spectrogram. | |
""" | |
# if self.preemphasis != 0: | |
# D = self._stft(self.apply_preemphasis(y)) | |
# else: | |
# D = self._stft(y) | |
D = self._stft(y) | |
# if self.do_amp_to_db_linear: | |
# S = self._amp_to_db(np.abs(D)) | |
# else: | |
# S = np.abs(D) | |
S = np.abs(D) | |
return self.normalize(S).astype(np.float32) | |
def melspectrogram(self, y: np.ndarray) -> np.ndarray: | |
"""Compute a melspectrogram from a waveform.""" | |
# if self.preemphasis != 0: | |
# D = self._stft(self.apply_preemphasis(y)) | |
# else: | |
# D = self._stft(y) | |
D = self._stft(y) | |
# if self.do_amp_to_db_mel: | |
# S = self._amp_to_db(self._linear_to_mel(np.abs(D))) | |
# else: | |
# S = self._linear_to_mel(np.abs(D)) | |
S = self._amp_to_db(self._linear_to_mel(np.abs(D))) | |
return self.normalize(S).astype(np.float32) | |
def inv_spectrogram(self, spectrogram: np.ndarray) -> np.ndarray: | |
"""Convert a spectrogram to a waveform using Griffi-Lim vocoder.""" | |
S = self.denormalize(spectrogram) | |
S = self._db_to_amp(S) | |
# Reconstruct phase | |
if self.preemphasis != 0: | |
return self.apply_inv_preemphasis(self._griffin_lim(S ** self.power)) | |
return self._griffin_lim(S ** self.power) | |
def inv_melspectrogram(self, mel_spectrogram: np.ndarray) -> np.ndarray: | |
"""Convert a melspectrogram to a waveform using Griffi-Lim vocoder.""" | |
D = self.denormalize(mel_spectrogram) | |
S = self._db_to_amp(D) | |
S = self._mel_to_linear(S) # Convert back to linear | |
if self.preemphasis != 0: | |
return self.apply_inv_preemphasis(self._griffin_lim(S ** self.power)) | |
return self._griffin_lim(S ** self.power) | |
def out_linear_to_mel(self, linear_spec: np.ndarray) -> np.ndarray: | |
"""Convert a full scale linear spectrogram output of a network to a melspectrogram. | |
Args: | |
linear_spec (np.ndarray): Normalized full scale linear spectrogram. | |
Returns: | |
np.ndarray: Normalized melspectrogram. | |
""" | |
S = self.denormalize(linear_spec) | |
S = self._db_to_amp(S) | |
S = self._linear_to_mel(np.abs(S)) | |
S = self._amp_to_db(S) | |
mel = self.normalize(S) | |
return mel | |
### STFT and ISTFT ### | |
def _stft(self, y: np.ndarray) -> np.ndarray: | |
"""Librosa STFT wrapper. | |
Args: | |
y (np.ndarray): Audio signal. | |
Returns: | |
np.ndarray: Complex number array. | |
""" | |
return librosa.stft( | |
y=y, | |
n_fft=self.fft_size, | |
hop_length=self.hop_length, | |
win_length=self.win_length, | |
pad_mode=self.stft_pad_mode, | |
window="hann", | |
center=True, | |
) | |
def _istft(self, y: np.ndarray) -> np.ndarray: | |
"""Librosa iSTFT wrapper.""" | |
return librosa.istft(y, hop_length=self.hop_length, win_length=self.win_length) | |
def _griffin_lim(self, S): | |
angles = np.exp(2j * np.pi * np.random.rand(*S.shape)) | |
S_complex = np.abs(S).astype(np.complex) | |
y = self._istft(S_complex * angles) | |
if not np.isfinite(y).all(): | |
print(" [!] Waveform is not finite everywhere. Skipping the GL.") | |
return np.array([0.0]) | |
for _ in range(self.griffin_lim_iters): | |
angles = np.exp(1j * np.angle(self._stft(y))) | |
y = self._istft(S_complex * angles) | |
return y | |
def compute_stft_paddings(self, x, pad_sides=1): | |
"""Compute paddings used by Librosa's STFT. Compute right padding (final frame) or both sides padding | |
(first and final frames)""" | |
assert pad_sides in (1, 2) | |
pad = (x.shape[0] // self.hop_length + 1) * self.hop_length - x.shape[0] | |
if pad_sides == 1: | |
return 0, pad | |
return pad // 2, pad // 2 + pad % 2 | |
def compute_f0(self, x: np.ndarray) -> np.ndarray: | |
import pyworld as pw | |
"""Compute pitch (f0) of a waveform using the same parameters used for computing melspectrogram. | |
Args: | |
x (np.ndarray): Waveform. | |
Returns: | |
np.ndarray: Pitch. | |
Examples: | |
>>> WAV_FILE = filename = librosa.util.example_audio_file() | |
>>> from TTS.config import BaseAudioConfig | |
>>> from TTS.utils.audio import AudioProcessor | |
>>> conf = BaseAudioConfig(mel_fmax=8000) | |
>>> ap = AudioProcessor(**conf) | |
>>> wav = ap.load_wav(WAV_FILE, sr=22050)[:5 * 22050] | |
>>> pitch = ap.compute_f0(wav) | |
""" | |
# align F0 length to the spectrogram length | |
# if len(x) % self.hop_length == 0: | |
# x = np.pad(x, (0, self.hop_length // 2), mode="reflect") | |
# f0, t = pw.dio( | |
# x.astype(np.double), | |
# fs=self.sample_rate, | |
# f0_ceil=self.mel_fmax, | |
# frame_period=1000 * self.hop_length / self.sample_rate, | |
# ) | |
# f0 = pw.stonemask(x.astype(np.double), f0, t, self.sample_rate) | |
# return f0 | |
pass | |
### Audio Processing ### | |
def find_endpoint(self, wav: np.ndarray, min_silence_sec=0.8) -> int: | |
"""Find the last point without silence at the end of a audio signal. | |
Args: | |
wav (np.ndarray): Audio signal. | |
threshold_db (int, optional): Silence threshold in decibels. Defaults to -40. | |
min_silence_sec (float, optional): Ignore silences that are shorter then this in secs. Defaults to 0.8. | |
Returns: | |
int: Last point without silence. | |
""" | |
window_length = int(self.sample_rate * min_silence_sec) | |
hop_length = int(window_length / 4) | |
threshold = self._db_to_amp(-self.trim_db) | |
for x in range(hop_length, len(wav) - window_length, hop_length): | |
if np.max(wav[x : x + window_length]) < threshold: | |
return x + hop_length | |
return len(wav) | |
def trim_silence(self, wav): | |
"""Trim silent parts with a threshold and 0.01 sec margin""" | |
margin = int(self.sample_rate * 0.01) | |
wav = wav[margin:-margin] | |
return librosa.effects.trim(wav, top_db=self.trim_db, frame_length=self.win_length, hop_length=self.hop_length)[ | |
0 | |
] | |
def sound_norm(x: np.ndarray) -> np.ndarray: | |
"""Normalize the volume of an audio signal. | |
Args: | |
x (np.ndarray): Raw waveform. | |
Returns: | |
np.ndarray: Volume normalized waveform. | |
""" | |
return x / abs(x).max() * 0.95 | |
def _rms_norm(wav, db_level=-27): | |
r = 10 ** (db_level / 20) | |
a = np.sqrt((len(wav) * (r ** 2)) / np.sum(wav ** 2)) | |
return wav * a | |
def rms_volume_norm(self, x: np.ndarray, db_level: float = None) -> np.ndarray: | |
"""Normalize the volume based on RMS of the signal. | |
Args: | |
x (np.ndarray): Raw waveform. | |
Returns: | |
np.ndarray: RMS normalized waveform. | |
""" | |
if db_level is None: | |
db_level = self.db_level | |
assert -99 <= db_level <= 0, " [!] db_level should be between -99 and 0" | |
wav = self._rms_norm(x, db_level) | |
return wav | |
### save and load ### | |
def load_wav(self, filename: str, sr: int = None) -> np.ndarray: | |
"""Read a wav file using Librosa and optionally resample, silence trim, volume normalize. | |
Resampling slows down loading the file significantly. Therefore it is recommended to resample the file before. | |
Args: | |
filename (str): Path to the wav file. | |
sr (int, optional): Sampling rate for resampling. Defaults to None. | |
Returns: | |
np.ndarray: Loaded waveform. | |
""" | |
if self.resample: | |
# loading with resampling. It is significantly slower. | |
x, sr = librosa.load(filename, sr=self.sample_rate) | |
elif sr is None: | |
# SF is faster than librosa for loading files | |
x, sr = sf.read(filename) | |
assert self.sample_rate == sr, "%s vs %s (%s)" % (self.sample_rate, sr, filename) | |
else: | |
x, sr = librosa.load(filename, sr=sr) | |
if self.do_trim_silence: | |
try: | |
x = self.trim_silence(x) | |
except ValueError as e: | |
print(f" [!] File cannot be trimmed for silence - {filename}:", e) | |
return None#"==DEL_BAD_FILE==" | |
if self.do_sound_norm: | |
x = self.sound_norm(x) | |
if self.do_rms_norm: | |
x = self.rms_volume_norm(x, self.db_level) | |
return x | |
def save_wav(self, wav: np.ndarray, path: str, sr: int = None) -> None: | |
"""Save a waveform to a file using Scipy. | |
Args: | |
wav (np.ndarray): Waveform to save. | |
path (str): Path to a output file. | |
sr (int, optional): Sampling rate used for saving to the file. Defaults to None. | |
""" | |
wav_norm = wav * (32767 / max(0.01, np.max(np.abs(wav)))) | |
scipy.io.wavfile.write(path, sr if sr else self.sample_rate, wav_norm.astype(np.int16)) | |
def get_duration(self, filename: str) -> float: | |
"""Get the duration of a wav file using Librosa. | |
Args: | |
filename (str): Path to the wav file. | |
""" | |
return librosa.get_duration(filename) | |
def mulaw_encode(wav: np.ndarray, qc: int) -> np.ndarray: | |
mu = 2 ** qc - 1 | |
# wav_abs = np.minimum(np.abs(wav), 1.0) | |
signal = np.sign(wav) * np.log(1 + mu * np.abs(wav)) / np.log(1.0 + mu) | |
# Quantize signal to the specified number of levels. | |
signal = (signal + 1) / 2 * mu + 0.5 | |
return np.floor( | |
signal, | |
) | |
def mulaw_decode(wav, qc): | |
"""Recovers waveform from quantized values.""" | |
mu = 2 ** qc - 1 | |
x = np.sign(wav) / mu * ((1 + mu) ** np.abs(wav) - 1) | |
return x | |
def encode_16bits(x): | |
return np.clip(x * 2 ** 15, -(2 ** 15), 2 ** 15 - 1).astype(np.int16) | |
def quantize(x: np.ndarray, bits: int) -> np.ndarray: | |
"""Quantize a waveform to a given number of bits. | |
Args: | |
x (np.ndarray): Waveform to quantize. Must be normalized into the range `[-1, 1]`. | |
bits (int): Number of quantization bits. | |
Returns: | |
np.ndarray: Quantized waveform. | |
""" | |
return (x + 1.0) * (2 ** bits - 1) / 2 | |
def dequantize(x, bits): | |
"""Dequantize a waveform from the given number of bits.""" | |
return 2 * x / (2 ** bits - 1) - 1 | |
def _log(x, base): | |
if base == 10: | |
return np.log10(x) | |
return np.log(x) | |
def _exp(x, base): | |
if base == 10: | |
return np.power(10, x) | |
return np.exp(x) | |