"""Basic feature pipelines.
Authors
* Mirco Ravanelli 2020
* Peter Plantinga 2020
* Sarthak Yadav 2020
"""
import torch
from speechbrain.processing.features import (
STFT,
spectral_magnitude,
Filterbank,
DCT,
Deltas,
ContextWindow,
)
from speechbrain.nnet.CNN import GaborConv1d
from speechbrain.nnet.normalization import PCEN
from speechbrain.nnet.pooling import GaussianLowpassPooling
[docs]class Fbank(torch.nn.Module):
"""Generate features for input to the speech pipeline.
Arguments
---------
deltas : bool (default: False)
Whether or not to append derivatives and second derivatives
to the features.
context : bool (default: False)
Whether or not to append forward and backward contexts to
the features.
requires_grad : bool (default: False)
Whether to allow parameters (i.e. fbank centers and
spreads) to update during training.
sample_rate : int (default: 160000)
Sampling rate for the input waveforms.
f_min : int (default: 0)
Lowest frequency for the Mel filters.
f_max : int (default: None)
Highest frequency for the Mel filters. Note that if f_max is not
specified it will be set to sample_rate // 2.
win_length : float (default: 25)
Length (in ms) of the sliding window used to compute the STFT.
hop_length : float (default: 10)
Length (in ms) of the hop of the sliding window used to compute
the STFT.
n_fft : int (default: 400)
Number of samples to use in each stft.
n_mels : int (default: 40)
Number of Mel filters.
filter_shape : str (default: triangular)
Shape of the filters ('triangular', 'rectangular', 'gaussian').
param_change_factor : float (default: 1.0)
If freeze=False, this parameter affects the speed at which the filter
parameters (i.e., central_freqs and bands) can be changed. When high
(e.g., param_change_factor=1) the filters change a lot during training.
When low (e.g. param_change_factor=0.1) the filter parameters are more
stable during training.
param_rand_factor : float (default: 0.0)
This parameter can be used to randomly change the filter parameters
(i.e, central frequencies and bands) during training. It is thus a
sort of regularization. param_rand_factor=0 does not affect, while
param_rand_factor=0.15 allows random variations within +-15% of the
standard values of the filter parameters (e.g., if the central freq
is 100 Hz, we can randomly change it from 85 Hz to 115 Hz).
left_frames : int (default: 5)
Number of frames of left context to add.
right_frames : int (default: 5)
Number of frames of right context to add.
Example
-------
>>> import torch
>>> inputs = torch.randn([10, 16000])
>>> feature_maker = Fbank()
>>> feats = feature_maker(inputs)
>>> feats.shape
torch.Size([10, 101, 40])
"""
def __init__(
self,
deltas=False,
context=False,
requires_grad=False,
sample_rate=16000,
f_min=0,
f_max=None,
n_fft=400,
n_mels=40,
filter_shape="triangular",
param_change_factor=1.0,
param_rand_factor=0.0,
left_frames=5,
right_frames=5,
win_length=25,
hop_length=10,
):
super().__init__()
self.deltas = deltas
self.context = context
self.requires_grad = requires_grad
if f_max is None:
f_max = sample_rate / 2
self.compute_STFT = STFT(
sample_rate=sample_rate,
n_fft=n_fft,
win_length=win_length,
hop_length=hop_length,
)
self.compute_fbanks = Filterbank(
sample_rate=sample_rate,
n_fft=n_fft,
n_mels=n_mels,
f_min=f_min,
f_max=f_max,
freeze=not requires_grad,
filter_shape=filter_shape,
param_change_factor=param_change_factor,
param_rand_factor=param_rand_factor,
)
self.compute_deltas = Deltas(input_size=n_mels)
self.context_window = ContextWindow(
left_frames=left_frames, right_frames=right_frames,
)
[docs] def forward(self, wav):
"""Returns a set of features generated from the input waveforms.
Arguments
---------
wav : tensor
A batch of audio signals to transform to features.
"""
STFT = self.compute_STFT(wav)
mag = spectral_magnitude(STFT)
fbanks = self.compute_fbanks(mag)
if self.deltas:
delta1 = self.compute_deltas(fbanks)
delta2 = self.compute_deltas(delta1)
fbanks = torch.cat([fbanks, delta1, delta2], dim=2)
if self.context:
fbanks = self.context_window(fbanks)
return fbanks
[docs]class MFCC(torch.nn.Module):
"""Generate features for input to the speech pipeline.
Arguments
---------
deltas : bool (default: True)
Whether or not to append derivatives and second derivatives
to the features.
context : bool (default: True)
Whether or not to append forward and backward contexts to
the features.
requires_grad : bool (default: False)
Whether to allow parameters (i.e. fbank centers and
spreads) to update during training.
sample_rate : int (default: 16000)
Sampling rate for the input waveforms.
f_min : int (default: 0)
Lowest frequency for the Mel filters.
f_max : int (default: None)
Highest frequency for the Mel filters. Note that if f_max is not
specified it will be set to sample_rate // 2.
win_length : float (default: 25)
Length (in ms) of the sliding window used to compute the STFT.
hop_length : float (default: 10)
Length (in ms) of the hop of the sliding window used to compute
the STFT.
n_fft : int (default: 400)
Number of samples to use in each stft.
n_mels : int (default: 23)
Number of filters to use for creating filterbank.
n_mfcc : int (default: 20)
Number of output coefficients
filter_shape : str (default 'triangular')
Shape of the filters ('triangular', 'rectangular', 'gaussian').
param_change_factor: bool (default 1.0)
If freeze=False, this parameter affects the speed at which the filter
parameters (i.e., central_freqs and bands) can be changed. When high
(e.g., param_change_factor=1) the filters change a lot during training.
When low (e.g. param_change_factor=0.1) the filter parameters are more
stable during training.
param_rand_factor: float (default 0.0)
This parameter can be used to randomly change the filter parameters
(i.e, central frequencies and bands) during training. It is thus a
sort of regularization. param_rand_factor=0 does not affect, while
param_rand_factor=0.15 allows random variations within +-15% of the
standard values of the filter parameters (e.g., if the central freq
is 100 Hz, we can randomly change it from 85 Hz to 115 Hz).
left_frames : int (default 5)
Number of frames of left context to add.
right_frames : int (default 5)
Number of frames of right context to add.
Example
-------
>>> import torch
>>> inputs = torch.randn([10, 16000])
>>> feature_maker = MFCC()
>>> feats = feature_maker(inputs)
>>> feats.shape
torch.Size([10, 101, 660])
"""
def __init__(
self,
deltas=True,
context=True,
requires_grad=False,
sample_rate=16000,
f_min=0,
f_max=None,
n_fft=400,
n_mels=23,
n_mfcc=20,
filter_shape="triangular",
param_change_factor=1.0,
param_rand_factor=0.0,
left_frames=5,
right_frames=5,
win_length=25,
hop_length=10,
):
super().__init__()
self.deltas = deltas
self.context = context
self.requires_grad = requires_grad
if f_max is None:
f_max = sample_rate / 2
self.compute_STFT = STFT(
sample_rate=sample_rate,
n_fft=n_fft,
win_length=win_length,
hop_length=hop_length,
)
self.compute_fbanks = Filterbank(
sample_rate=sample_rate,
n_fft=n_fft,
n_mels=n_mels,
f_min=f_min,
f_max=f_max,
freeze=not requires_grad,
filter_shape=filter_shape,
param_change_factor=param_change_factor,
param_rand_factor=param_rand_factor,
)
self.compute_dct = DCT(input_size=n_mels, n_out=n_mfcc)
self.compute_deltas = Deltas(input_size=n_mfcc)
self.context_window = ContextWindow(
left_frames=left_frames, right_frames=right_frames,
)
[docs] def forward(self, wav):
"""Returns a set of mfccs generated from the input waveforms.
Arguments
---------
wav : tensor
A batch of audio signals to transform to features.
"""
STFT = self.compute_STFT(wav)
mag = spectral_magnitude(STFT)
fbanks = self.compute_fbanks(mag)
mfccs = self.compute_dct(fbanks)
if self.deltas:
delta1 = self.compute_deltas(mfccs)
delta2 = self.compute_deltas(delta1)
mfccs = torch.cat([mfccs, delta1, delta2], dim=2)
if self.context:
mfccs = self.context_window(mfccs)
return mfccs
[docs]class Leaf(torch.nn.Module):
"""
This class implements the LEAF audio frontend from
Neil Zeghidour, Olivier Teboul, F{\'e}lix de Chaumont Quitry & Marco Tagliasacchi, "LEAF: A LEARNABLE FRONTEND
FOR AUDIO CLASSIFICATION", in Proc. of ICLR 2021 (https://arxiv.org/abs/2101.08596)
Arguments
---------
out_channels : int
It is the number of output channels.
window_len: float
length of filter window in milliseconds
window_stride : float
Stride factor of the filters in milliseconds
sample_rate : int,
Sampling rate of the input signals. It is only used for sinc_conv.
min_freq : float
Lowest possible frequency (in Hz) for a filter
max_freq : float
Highest possible frequency (in Hz) for a filter
use_pcen: bool
If True (default), a per-channel energy normalization layer is used
learnable_pcen: bool:
If True (default), the per-channel energy normalization layer is learnable
use_legacy_complex: bool
If False, torch.complex64 data type is used for gabor impulse responses
If True, computation is performed on two real-valued tensors
skip_transpose: bool
If False, uses batch x time x channel convention of speechbrain.
If True, uses batch x channel x time convention.
Example
-------
>>> inp_tensor = torch.rand([10, 8000])
>>> leaf = Leaf(
... out_channels=40, window_len=25., window_stride=10., in_channels=1
... )
>>> out_tensor = leaf(inp_tensor)
>>> out_tensor.shape
torch.Size([10, 50, 40])
"""
def __init__(
self,
out_channels,
window_len: float = 25.0,
window_stride: float = 10.0,
sample_rate: int = 16000,
input_shape=None,
in_channels=None,
min_freq=60.0,
max_freq=None,
use_pcen=True,
learnable_pcen=True,
use_legacy_complex=False,
skip_transpose=False,
n_fft=512,
):
super(Leaf, self).__init__()
self.out_channels = out_channels
window_size = int(sample_rate * window_len // 1000 + 1)
window_stride = int(sample_rate * window_stride // 1000)
if input_shape is None and in_channels is None:
raise ValueError("Must provide one of input_shape or in_channels")
if in_channels is None:
in_channels = self._check_input_shape(input_shape)
self.complex_conv = GaborConv1d(
out_channels=2 * out_channels,
in_channels=in_channels,
kernel_size=window_size,
stride=1,
padding="same",
bias=False,
n_fft=n_fft,
sample_rate=sample_rate,
min_freq=min_freq,
max_freq=max_freq,
use_legacy_complex=use_legacy_complex,
skip_transpose=True,
)
self.pooling = GaussianLowpassPooling(
in_channels=self.out_channels,
kernel_size=window_size,
stride=window_stride,
skip_transpose=True,
)
if use_pcen:
self.compression = PCEN(
self.out_channels,
alpha=0.96,
smooth_coef=0.04,
delta=2.0,
floor=1e-12,
trainable=learnable_pcen,
per_channel_smooth_coef=True,
skip_transpose=True,
)
else:
self.compression = None
self.skip_transpose = skip_transpose
[docs] def forward(self, x):
"""
Returns the learned LEAF features
Arguments
---------
x : torch.Tensor of shape (batch, time, 1) or (batch, time)
batch of input signals. 2d or 3d tensors are expected.
"""
if not self.skip_transpose:
x = x.transpose(1, -1)
unsqueeze = x.ndim == 2
if unsqueeze:
x = x.unsqueeze(1)
outputs = self.complex_conv(x)
outputs = self._squared_modulus_activation(outputs)
outputs = self.pooling(outputs)
outputs = torch.maximum(
outputs, torch.tensor(1e-5, device=outputs.device)
)
if self.compression:
outputs = self.compression(outputs)
if not self.skip_transpose:
outputs = outputs.transpose(1, -1)
return outputs
def _squared_modulus_activation(self, x):
x = x.transpose(1, 2)
output = 2 * torch.nn.functional.avg_pool1d(
x ** 2.0, kernel_size=2, stride=2
)
output = output.transpose(1, 2)
return output
def _check_input_shape(self, shape):
"""Checks the input shape and returns the number of input channels.
"""
if len(shape) == 2:
in_channels = 1
elif len(shape) == 3:
in_channels = 1
else:
raise ValueError(
"Leaf expects 2d or 3d inputs. Got " + str(len(shape))
)
return in_channels