msb-public
/
PaddleSpeech
mirror of https://github.com/PaddlePaddle/PaddleSpeech
1
0
Code Issues Projects Releases Wiki Activity
PaddleSpeech/third_party/nnAudio/nnAudio/Spectrogram.py

2444 lines
104 KiB
Raw Blame History

"""
Module containing all the spectrogram classes
"""
# 0.2.0
import torch
import torch.nn as nn
from torch.nn.functional import conv1d, conv2d, fold
import scipy # used only in CFP
import numpy as np
from time import time
# from nnAudio.librosa_functions import * # For debug purpose
# from nnAudio.utils import *
from .librosa_functions import *
from .utils import *
sz_float = 4 # size of a float
epsilon = 10e-8 # fudge factor for normalization
### --------------------------- Spectrogram Classes ---------------------------###
class STFT(torch.nn.Module):
"""This function is to calculate the short-time Fourier transform (STFT) of the input signal.
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
The correct shape will be inferred automatically if the input follows these 3 shapes.
Most of the arguments follow the convention from librosa.
This class inherits from ``torch.nn.Module``, therefore, the usage is same as ``torch.nn.Module``.
Parameters
----------
n_fft : int
Size of Fourier transform. Default value is 2048.
win_length : int
the size of window frame and STFT filter.
Default: None (treated as equal to n_fft)
freq_bins : int
Number of frequency bins. Default is ``None``, which means ``n_fft//2+1`` bins.
hop_length : int
The hop (or stride) size. Default value is ``None`` which is equivalent to ``n_fft//4``.
window : str
The windowing function for STFT. It uses ``scipy.signal.get_window``, please refer to
scipy documentation for possible windowing functions. The default value is 'hann'.
freq_scale : 'linear', 'log', or 'no'
Determine the spacing between each frequency bin. When `linear` or `log` is used,
the bin spacing can be controlled by ``fmin`` and ``fmax``. If 'no' is used, the bin will
start at 0Hz and end at Nyquist frequency with linear spacing.
center : bool
Putting the STFT keneral at the center of the time-step or not. If ``False``, the time
index is the beginning of the STFT kernel, if ``True``, the time index is the center of
the STFT kernel. Default value if ``True``.
pad_mode : str
The padding method. Default value is 'reflect'.
iSTFT : bool
To activate the iSTFT module or not. By default, it is False to save GPU memory.
Note: The iSTFT kernel is not trainable. If you want
a trainable iSTFT, use the iSTFT module.
fmin : int
The starting frequency for the lowest frequency bin. If freq_scale is ``no``, this argument
does nothing.
fmax : int
The ending frequency for the highest frequency bin. If freq_scale is ``no``, this argument
does nothing.
sr : int
The sampling rate for the input audio. It is used to calucate the correct ``fmin`` and ``fmax``.
Setting the correct sampling rate is very important for calculating the correct frequency.
trainable : bool
Determine if the STFT kenrels are trainable or not. If ``True``, the gradients for STFT
kernels will also be caluclated and the STFT kernels will be updated during model training.
Default value is ``False``
output_format : str
Control the spectrogram output type, either ``Magnitude``, ``Complex``, or ``Phase``.
The output_format can also be changed during the ``forward`` method.
verbose : bool
If ``True``, it shows layer information. If ``False``, it suppresses all prints
Returns
-------
spectrogram : torch.tensor
It returns a tensor of spectrograms.
``shape = (num_samples, freq_bins,time_steps)`` if ``output_format='Magnitude'``;
``shape = (num_samples, freq_bins,time_steps, 2)`` if ``output_format='Complex' or 'Phase'``;
Examples
--------
>>> spec_layer = Spectrogram.STFT()
>>> specs = spec_layer(x)
"""
def __init__(self, n_fft=2048, win_length=None, freq_bins=None, hop_length=None, window='hann',
freq_scale='no', center=True, pad_mode='reflect', iSTFT=False,
fmin=50, fmax=6000, sr=22050, trainable=False,
output_format="Complex", verbose=True):
super().__init__()
# Trying to make the default setting same as librosa
if win_length==None: win_length = n_fft
if hop_length==None: hop_length = int(win_length // 4)
self.output_format = output_format
self.trainable = trainable
self.stride = hop_length
self.center = center
self.pad_mode = pad_mode
self.n_fft = n_fft
self.freq_bins = freq_bins
self.trainable = trainable
self.pad_amount = self.n_fft // 2
self.window = window
self.win_length = win_length
self.iSTFT = iSTFT
self.trainable = trainable
start = time()
# Create filter windows for stft
kernel_sin, kernel_cos, self.bins2freq, self.bin_list, window_mask = create_fourier_kernels(n_fft,
win_length=win_length,
freq_bins=freq_bins,
window=window,
freq_scale=freq_scale,
fmin=fmin,
fmax=fmax,
sr=sr,
verbose=verbose)
kernel_sin = torch.tensor(kernel_sin, dtype=torch.float)
kernel_cos = torch.tensor(kernel_cos, dtype=torch.float)
# In this way, the inverse kernel and the forward kernel do not share the same memory...
kernel_sin_inv = torch.cat((kernel_sin, -kernel_sin[1:-1].flip(0)), 0)
kernel_cos_inv = torch.cat((kernel_cos, kernel_cos[1:-1].flip(0)), 0)
if iSTFT:
self.register_buffer('kernel_sin_inv', kernel_sin_inv.unsqueeze(-1))
self.register_buffer('kernel_cos_inv', kernel_cos_inv.unsqueeze(-1))
# Making all these variables nn.Parameter, so that the model can be used with nn.Parallel
# self.kernel_sin = torch.nn.Parameter(self.kernel_sin, requires_grad=self.trainable)
# self.kernel_cos = torch.nn.Parameter(self.kernel_cos, requires_grad=self.trainable)
# Applying window functions to the Fourier kernels
if window:
window_mask = torch.tensor(window_mask)
wsin = kernel_sin * window_mask
wcos = kernel_cos * window_mask
else:
wsin = kernel_sin
wcos = kernel_cos
if self.trainable==False:
self.register_buffer('wsin', wsin)
self.register_buffer('wcos', wcos)
if self.trainable==True:
wsin = torch.nn.Parameter(wsin, requires_grad=self.trainable)
wcos = torch.nn.Parameter(wcos, requires_grad=self.trainable)
self.register_parameter('wsin', wsin)
self.register_parameter('wcos', wcos)
# Prepare the shape of window mask so that it can be used later in inverse
self.register_buffer('window_mask', window_mask.unsqueeze(0).unsqueeze(-1))
if verbose==True:
print("STFT kernels created, time used = {:.4f} seconds".format(time()-start))
else:
pass
def forward(self, x, output_format=None):
"""
Convert a batch of waveforms to spectrograms.
Parameters
----------
x : torch tensor
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
It will be automatically broadcast to the right shape
output_format : str
Control the type of spectrogram to be return. Can be either ``Magnitude`` or ``Complex`` or ``Phase``.
Default value is ``Complex``.
"""
output_format = output_format or self.output_format
self.num_samples = x.shape[-1]
x = broadcast_dim(x)
if self.center:
if self.pad_mode == 'constant':
padding = nn.ConstantPad1d(self.pad_amount, 0)
elif self.pad_mode == 'reflect':
if self.num_samples < self.pad_amount:
raise AssertionError("Signal length shorter than reflect padding length (n_fft // 2).")
padding = nn.ReflectionPad1d(self.pad_amount)
x = padding(x)
spec_imag = conv1d(x, self.wsin, stride=self.stride)
spec_real = conv1d(x, self.wcos, stride=self.stride) # Doing STFT by using conv1d
# remove redundant parts
spec_real = spec_real[:, :self.freq_bins, :]
spec_imag = spec_imag[:, :self.freq_bins, :]
if output_format=='Magnitude':
spec = spec_real.pow(2) + spec_imag.pow(2)
if self.trainable==True:
return torch.sqrt(spec+1e-8) # prevent Nan gradient when sqrt(0) due to output=0
else:
return torch.sqrt(spec)
elif output_format=='Complex':
return torch.stack((spec_real,-spec_imag), -1) # Remember the minus sign for imaginary part
elif output_format=='Phase':
return torch.atan2(-spec_imag+0.0,spec_real) # +0.0 removes -0.0 elements, which leads to error in calculating phase
def inverse(self, X, onesided=True, length=None, refresh_win=True):
"""
This function is same as the :func:`~nnAudio.Spectrogram.iSTFT` class,
which is to convert spectrograms back to waveforms.
It only works for the complex value spectrograms. If you have the magnitude spectrograms,
please use :func:`~nnAudio.Spectrogram.Griffin_Lim`.
Parameters
----------
onesided : bool
If your spectrograms only have ``n_fft//2+1`` frequency bins, please use ``onesided=True``,
else use ``onesided=False``
length : int
To make sure the inverse STFT has the same output length of the original waveform, please
set `length` as your intended waveform length. By default, ``length=None``,
which will remove ``n_fft//2`` samples from the start and the end of the output.
refresh_win : bool
Recalculating the window sum square. If you have an input with fixed number of timesteps,
you can increase the speed by setting ``refresh_win=False``. Else please keep ``refresh_win=True``
"""
if (hasattr(self, 'kernel_sin_inv') != True) or (hasattr(self, 'kernel_cos_inv') != True):
raise NameError("Please activate the iSTFT module by setting `iSTFT=True` if you want to use `inverse`")
assert X.dim()==4 , "Inverse iSTFT only works for complex number," \
"make sure our tensor is in the shape of (batch, freq_bins, timesteps, 2)."\
"\nIf you have a magnitude spectrogram, please consider using Griffin-Lim."
if onesided:
X = extend_fbins(X) # extend freq
X_real, X_imag = X[:, :, :, 0], X[:, :, :, 1]
# broadcast dimensions to support 2D convolution
X_real_bc = X_real.unsqueeze(1)
X_imag_bc = X_imag.unsqueeze(1)
a1 = conv2d(X_real_bc, self.kernel_cos_inv, stride=(1,1))
b2 = conv2d(X_imag_bc, self.kernel_sin_inv, stride=(1,1))
# compute real and imag part. signal lies in the real part
real = a1 - b2
real = real.squeeze(-2)*self.window_mask
# Normalize the amplitude with n_fft
real /= (self.n_fft)
# Overlap and Add algorithm to connect all the frames
real = overlap_add(real, self.stride)
# Prepare the window sumsqure for division
# Only need to create this window once to save time
# Unless the input spectrograms have different time steps
if hasattr(self, 'w_sum')==False or refresh_win==True:
self.w_sum = torch_window_sumsquare(self.window_mask.flatten(), X.shape[2], self.stride, self.n_fft).flatten()
self.nonzero_indices = (self.w_sum>1e-10)
else:
pass
real[:, self.nonzero_indices] = real[:,self.nonzero_indices].div(self.w_sum[self.nonzero_indices])
# Remove padding
if length is None:
if self.center:
real = real[:, self.pad_amount:-self.pad_amount]
else:
if self.center:
real = real[:, self.pad_amount:self.pad_amount + length]
else:
real = real[:, :length]
return real
def extra_repr(self) -> str:
return 'n_fft={}, Fourier Kernel size={}, iSTFT={}, trainable={}'.format(
self.n_fft, (*self.wsin.shape,), self.iSTFT, self.trainable
)
class MelSpectrogram(torch.nn.Module):
"""This function is to calculate the Melspectrogram of the input signal.
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
The correct shape will be inferred automatically if the input follows these 3 shapes.
Most of the arguments follow the convention from librosa.
This class inherits from ``torch.nn.Module``, therefore, the usage is same as ``torch.nn.Module``.
Parameters
----------
sr : int
The sampling rate for the input audio.
It is used to calculate the correct ``fmin`` and ``fmax``.
Setting the correct sampling rate is very important for calculating the correct frequency.
n_fft : int
The window size for the STFT. Default value is 2048
win_length : int
the size of window frame and STFT filter.
Default: None (treated as equal to n_fft)
n_mels : int
The number of Mel filter banks. The filter banks maps the n_fft to mel bins.
Default value is 128.
hop_length : int
The hop (or stride) size. Default value is 512.
window : str
The windowing function for STFT. It uses ``scipy.signal.get_window``, please refer to
scipy documentation for possible windowing functions. The default value is 'hann'.
center : bool
Putting the STFT keneral at the center of the time-step or not. If ``False``,
the time index is the beginning of the STFT kernel, if ``True``, the time index is the
center of the STFT kernel. Default value if ``True``.
pad_mode : str
The padding method. Default value is 'reflect'.
htk : bool
When ``False`` is used, the Mel scale is quasi-logarithmic. When ``True`` is used, the
Mel scale is logarithmic. The default value is ``False``.
fmin : int
The starting frequency for the lowest Mel filter bank.
fmax : int
The ending frequency for the highest Mel filter bank.
norm :
if 1, divide the triangular mel weights by the width of the mel band
(area normalization, AKA 'slaney' default in librosa).
Otherwise, leave all the triangles aiming for
a peak value of 1.0
trainable_mel : bool
Determine if the Mel filter banks are trainable or not. If ``True``, the gradients for Mel
filter banks will also be calculated and the Mel filter banks will be updated during model
training. Default value is ``False``.
trainable_STFT : bool
Determine if the STFT kenrels are trainable or not. If ``True``, the gradients for STFT
kernels will also be caluclated and the STFT kernels will be updated during model training.
Default value is ``False``.
verbose : bool
If ``True``, it shows layer information. If ``False``, it suppresses all prints.
Returns
-------
spectrogram : torch.tensor
It returns a tensor of spectrograms. shape = ``(num_samples, freq_bins,time_steps)``.
Examples
--------
>>> spec_layer = Spectrogram.MelSpectrogram()
>>> specs = spec_layer(x)
"""
def __init__(self, sr=22050, n_fft=2048, win_length=None, n_mels=128, hop_length=512,
window='hann', center=True, pad_mode='reflect', power=2.0, htk=False,
fmin=0.0, fmax=None, norm=1, trainable_mel=False, trainable_STFT=False,
verbose=True, **kwargs):
super().__init__()
self.stride = hop_length
self.center = center
self.pad_mode = pad_mode
self.n_fft = n_fft
self.power = power
self.trainable_mel = trainable_mel
self.trainable_STFT = trainable_STFT
# Preparing for the stft layer. No need for center
self.stft = STFT(n_fft=n_fft, win_length=win_length, freq_bins=None,
hop_length=hop_length, window=window, freq_scale='no',
center=center, pad_mode=pad_mode, sr=sr, trainable=trainable_STFT,
output_format="Magnitude", verbose=verbose, **kwargs)
# Create filter windows for stft
start = time()
# Creating kernel for mel spectrogram
start = time()
mel_basis = mel(sr, n_fft, n_mels, fmin, fmax, htk=htk, norm=norm)
mel_basis = torch.tensor(mel_basis)
if verbose==True:
print("STFT filter created, time used = {:.4f} seconds".format(time()-start))
print("Mel filter created, time used = {:.4f} seconds".format(time()-start))
else:
pass
if trainable_mel:
# Making everything nn.Parameter, so that this model can support nn.DataParallel
mel_basis = torch.nn.Parameter(mel_basis, requires_grad=trainable_mel)
self.register_parameter('mel_basis', mel_basis)
else:
self.register_buffer('mel_basis', mel_basis)
# if trainable_mel==True:
# self.mel_basis = torch.nn.Parameter(self.mel_basis)
# if trainable_STFT==True:
# self.wsin = torch.nn.Parameter(self.wsin)
# self.wcos = torch.nn.Parameter(self.wcos)
def forward(self, x):
"""
Convert a batch of waveforms to Mel spectrograms.
Parameters
----------
x : torch tensor
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
It will be automatically broadcast to the right shape
"""
x = broadcast_dim(x)
spec = self.stft(x, output_format='Magnitude')**self.power
melspec = torch.matmul(self.mel_basis, spec)
return melspec
def extra_repr(self) -> str:
return 'Mel filter banks size = {}, trainable_mel={}'.format(
(*self.mel_basis.shape,), self.trainable_mel, self.trainable_STFT
)
class MFCC(torch.nn.Module):
"""This function is to calculate the Mel-frequency cepstral coefficients (MFCCs) of the input signal.
This algorithm first extracts Mel spectrograms from the audio clips,
then the discrete cosine transform is calcuated to obtain the final MFCCs.
Therefore, the Mel spectrogram part can be made trainable using
``trainable_mel`` and ``trainable_STFT``.
It only support type-II DCT at the moment. Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
The correct shape will be inferred autommatically if the input follows these 3 shapes.
Most of the arguments follow the convention from librosa.
This class inherits from ``torch.nn.Module``, therefore, the usage is same as ``torch.nn.Module``.
Parameters
----------
sr : int
The sampling rate for the input audio. It is used to calculate the correct ``fmin`` and ``fmax``.
Setting the correct sampling rate is very important for calculating the correct frequency.
n_mfcc : int
The number of Mel-frequency cepstral coefficients
norm : string
The default value is 'ortho'. Normalization for DCT basis
**kwargs
Other arguments for Melspectrogram such as n_fft, n_mels, hop_length, and window
Returns
-------
MFCCs : torch.tensor
It returns a tensor of MFCCs. shape = ``(num_samples, n_mfcc, time_steps)``.
Examples
--------
>>> spec_layer = Spectrogram.MFCC()
>>> mfcc = spec_layer(x)
"""
def __init__(self, sr=22050, n_mfcc=20, norm='ortho', verbose=True, ref=1.0, amin=1e-10, top_db=80.0, **kwargs):
super().__init__()
self.melspec_layer = MelSpectrogram(sr=sr, verbose=verbose, **kwargs)
self.m_mfcc = n_mfcc
# attributes that will be used for _power_to_db
if amin <= 0:
raise ParameterError('amin must be strictly positive')
amin = torch.tensor([amin])
ref = torch.abs(torch.tensor([ref]))
self.register_buffer('amin', amin)
self.register_buffer('ref', ref)
self.top_db = top_db
self.n_mfcc = n_mfcc
def _power_to_db(self, S):
'''
Refer to https://librosa.github.io/librosa/_modules/librosa/core/spectrum.html#power_to_db
for the original implmentation.
'''
log_spec = 10.0 * torch.log10(torch.max(S, self.amin))
log_spec -= 10.0 * torch.log10(torch.max(self.amin, self.ref))
if self.top_db is not None:
if self.top_db < 0:
raise ParameterError('top_db must be non-negative')
# make the dim same as log_spec so that it can be broadcasted
batch_wise_max = log_spec.flatten(1).max(1)[0].unsqueeze(1).unsqueeze(1)
log_spec = torch.max(log_spec, batch_wise_max - self.top_db)
return log_spec
def _dct(self, x, norm=None):
'''
Refer to https://github.com/zh217/torch-dct for the original implmentation.
'''
x = x.permute(0,2,1) # make freq the last axis, since dct applies to the frequency axis
x_shape = x.shape
N = x_shape[-1]
v = torch.cat([x[:, :, ::2], x[:, :, 1::2].flip([2])], dim=2)
Vc = torch.rfft(v, 1, onesided=False)
# TODO: Can make the W_r and W_i trainable here
k = - torch.arange(N, dtype=x.dtype, device=x.device)[None, :] * np.pi / (2 * N)
W_r = torch.cos(k)
W_i = torch.sin(k)
V = Vc[:, :, :, 0] * W_r - Vc[:, :, :, 1] * W_i
if norm == 'ortho':
V[:, :, 0] /= np.sqrt(N) * 2
V[:, :, 1:] /= np.sqrt(N / 2) * 2
V = 2 * V
return V.permute(0,2,1) # swapping back the time axis and freq axis
def forward(self, x):
"""
Convert a batch of waveforms to MFCC.
Parameters
----------
x : torch tensor
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
It will be automatically broadcast to the right shape
"""
x = self.melspec_layer(x)
x = self._power_to_db(x)
x = self._dct(x, norm='ortho')[:,:self.m_mfcc,:]
return x
def extra_repr(self) -> str:
return 'n_mfcc = {}'.format(
(self.n_mfcc)
)
class Gammatonegram(torch.nn.Module):
"""
This function is to calculate the Gammatonegram of the input signal. Input signal should be in either of the following shapes. 1. ``(len_audio)``, 2. ``(num_audio, len_audio)``, 3. ``(num_audio, 1, len_audio)``. The correct shape will be inferred autommatically if the input follows these 3 shapes. This class inherits from ``torch.nn.Module``, therefore, the usage is same as ``torch.nn.Module``.
Parameters
----------
sr : int
The sampling rate for the input audio. It is used to calucate the correct ``fmin`` and ``fmax``. Setting the correct sampling rate is very important for calculating the correct frequency.
n_fft : int
The window size for the STFT. Default value is 2048
n_mels : int
The number of Gammatonegram filter banks. The filter banks maps the n_fft to Gammatone bins. Default value is 64
hop_length : int
The hop (or stride) size. Default value is 512.
window : str
The windowing function for STFT. It uses ``scipy.signal.get_window``, please refer to scipy documentation for possible windowing functions. The default value is 'hann'
center : bool
Putting the STFT keneral at the center of the time-step or not. If ``False``, the time index is the beginning of the STFT kernel, if ``True``, the time index is the center of the STFT kernel. Default value if ``True``.
pad_mode : str
The padding method. Default value is 'reflect'.
htk : bool
When ``False`` is used, the Mel scale is quasi-logarithmic. When ``True`` is used, the Mel scale is logarithmic. The default value is ``False``
fmin : int
The starting frequency for the lowest Gammatone filter bank
fmax : int
The ending frequency for the highest Gammatone filter bank
trainable_mel : bool
Determine if the Gammatone filter banks are trainable or not. If ``True``, the gradients for Mel filter banks will also be caluclated and the Mel filter banks will be updated during model training. Default value is ``False``
trainable_STFT : bool
Determine if the STFT kenrels are trainable or not. If ``True``, the gradients for STFT kernels will also be caluclated and the STFT kernels will be updated during model training. Default value is ``False``
verbose : bool
If ``True``, it shows layer information. If ``False``, it suppresses all prints
Returns
-------
spectrogram : torch.tensor
It returns a tensor of spectrograms. shape = ``(num_samples, freq_bins,time_steps)``.
Examples
--------
>>> spec_layer = Spectrogram.Gammatonegram()
>>> specs = spec_layer(x)
"""
def __init__(self, sr=44100, n_fft=2048, n_bins=64, hop_length=512, window='hann', center=True, pad_mode='reflect',
power=2.0, htk=False, fmin=20.0, fmax=None, norm=1, trainable_bins=False, trainable_STFT=False,
verbose=True):
super(Gammatonegram, self).__init__()
self.stride = hop_length
self.center = center
self.pad_mode = pad_mode
self.n_fft = n_fft
self.power = power
# Create filter windows for stft
start = time()
wsin, wcos, self.bins2freq, _, _ = create_fourier_kernels(n_fft, freq_bins=None, window=window, freq_scale='no',
sr=sr)
wsin = torch.tensor(wsin, dtype=torch.float)
wcos = torch.tensor(wcos, dtype=torch.float)
if trainable_STFT:
wsin = torch.nn.Parameter(wsin, requires_grad=trainable_STFT)
wcos = torch.nn.Parameter(wcos, requires_grad=trainable_STFT)
self.register_parameter('wsin', wsin)
self.register_parameter('wcos', wcos)
else:
self.register_buffer('wsin', wsin)
self.register_buffer('wcos', wcos)
# Creating kenral for Gammatone spectrogram
start = time()
gammatone_basis = gammatone(sr, n_fft, n_bins, fmin, fmax)
gammatone_basis = torch.tensor(gammatone_basis)
if verbose == True:
print("STFT filter created, time used = {:.4f} seconds".format(time() - start))
print("Gammatone filter created, time used = {:.4f} seconds".format(time() - start))
else:
pass
# Making everything nn.Prarmeter, so that this model can support nn.DataParallel
if trainable_bins:
gammatone_basis = torch.nn.Parameter(gammatone_basis, requires_grad=trainable_bins)
self.register_parameter('gammatone_basis', gammatone_basis)
else:
self.register_buffer('gammatone_basis', gammatone_basis)
# if trainable_mel==True:
# self.mel_basis = torch.nn.Parameter(self.mel_basis)
# if trainable_STFT==True:
# self.wsin = torch.nn.Parameter(self.wsin)
# self.wcos = torch.nn.Parameter(self.wcos)
def forward(self, x):
x = broadcast_dim(x)
if self.center:
if self.pad_mode == 'constant':
padding = nn.ConstantPad1d(self.n_fft // 2, 0)
elif self.pad_mode == 'reflect':
padding = nn.ReflectionPad1d(self.n_fft // 2)
x = padding(x)
spec = torch.sqrt(conv1d(x, self.wsin, stride=self.stride).pow(2) \
+ conv1d(x, self.wcos, stride=self.stride).pow(2)) ** self.power # Doing STFT by using conv1d
gammatonespec = torch.matmul(self.gammatone_basis, spec)
return gammatonespec
class CQT1992(torch.nn.Module):
"""
This alogrithm uses the method proposed in [1], which would run extremely slow if low frequencies (below 220Hz)
are included in the frequency bins.
Please refer to :func:`~nnAudio.Spectrogram.CQT1992v2` for a more
computational and memory efficient version.
[1] Brown, Judith C.C. and Miller Puckette. “An efficient algorithm for the calculation of a
constant Q transform.” (1992).
This function is to calculate the CQT of the input signal.
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
The correct shape will be inferred autommatically if the input follows these 3 shapes.
Most of the arguments follow the convention from librosa.
This class inherits from ``torch.nn.Module``, therefore, the usage is same as ``torch.nn.Module``.
Parameters
----------
sr : int
The sampling rate for the input audio. It is used to calucate the correct ``fmin`` and ``fmax``.
Setting the correct sampling rate is very important for calculating the correct frequency.
hop_length : int
The hop (or stride) size. Default value is 512.
fmin : float
The frequency for the lowest CQT bin. Default is 32.70Hz, which coresponds to the note C0.
fmax : float
The frequency for the highest CQT bin. Default is ``None``, therefore the higest CQT bin is
inferred from the ``n_bins`` and ``bins_per_octave``.
If ``fmax`` is not ``None``, then the argument ``n_bins`` will be ignored and ``n_bins``
will be calculated automatically. Default is ``None``
n_bins : int
The total numbers of CQT bins. Default is 84. Will be ignored if ``fmax`` is not ``None``.
bins_per_octave : int
Number of bins per octave. Default is 12.
trainable_STFT : bool
Determine if the time to frequency domain transformation kernel for the input audio is trainable or not.
Default is ``False``
trainable_CQT : bool
Determine if the frequency domain CQT kernel is trainable or not.
Default is ``False``
norm : int
Normalization for the CQT kernels. ``1`` means L1 normalization, and ``2`` means L2 normalization.
Default is ``1``, which is same as the normalization used in librosa.
window : str
The windowing function for CQT. It uses ``scipy.signal.get_window``, please refer to
scipy documentation for possible windowing functions. The default value is 'hann'.
center : bool
Putting the CQT keneral at the center of the time-step or not. If ``False``, the time index is
the beginning of the CQT kernel, if ``True``, the time index is the center of the CQT kernel.
Default value if ``True``.
pad_mode : str
The padding method. Default value is 'reflect'.
trainable : bool
Determine if the CQT kernels are trainable or not. If ``True``, the gradients for CQT kernels
will also be caluclated and the CQT kernels will be updated during model training.
Default value is ``False``.
output_format : str
Determine the return type.
``Magnitude`` will return the magnitude of the STFT result, shape = ``(num_samples, freq_bins,time_steps)``;
``Complex`` will return the STFT result in complex number, shape = ``(num_samples, freq_bins,time_steps, 2)``;
``Phase`` will return the phase of the STFT reuslt, shape = ``(num_samples, freq_bins,time_steps, 2)``.
The complex number is stored as ``(real, imag)`` in the last axis. Default value is 'Magnitude'.
verbose : bool
If ``True``, it shows layer information. If ``False``, it suppresses all prints
Returns
-------
spectrogram : torch.tensor
It returns a tensor of spectrograms.
shape = ``(num_samples, freq_bins,time_steps)`` if ``output_format='Magnitude'``;
shape = ``(num_samples, freq_bins,time_steps, 2)`` if ``output_format='Complex' or 'Phase'``;
Examples
--------
>>> spec_layer = Spectrogram.CQT1992v2()
>>> specs = spec_layer(x)
"""
def __init__(self, sr=22050, hop_length=512, fmin=220, fmax=None, n_bins=84,
trainable_STFT=False, trainable_CQT=False, bins_per_octave=12, filter_scale=1,
output_format='Magnitude', norm=1, window='hann', center=True, pad_mode='reflect'):
super().__init__()
# norm arg is not functioning
self.hop_length = hop_length
self.center = center
self.pad_mode = pad_mode
self.norm = norm
self.output_format = output_format
# creating kernels for CQT
Q = float(filter_scale)/(2**(1/bins_per_octave)-1)
print("Creating CQT kernels ...", end='\r')
start = time()
cqt_kernels, self.kernel_width, lenghts, freqs = create_cqt_kernels(Q,
sr,
fmin,
n_bins,
bins_per_octave,
norm,
window,
fmax)
self.register_buffer('lenghts', lenghts)
self.frequencies = freqs
cqt_kernels = fft(cqt_kernels)[:,:self.kernel_width//2+1]
print("CQT kernels created, time used = {:.4f} seconds".format(time()-start))
# creating kernels for stft
# self.cqt_kernels_real*=lenghts.unsqueeze(1)/self.kernel_width # Trying to normalize as librosa
# self.cqt_kernels_imag*=lenghts.unsqueeze(1)/self.kernel_width
print("Creating STFT kernels ...", end='\r')
start = time()
kernel_sin, kernel_cos, self.bins2freq, _, window = create_fourier_kernels(self.kernel_width,
window='ones',
freq_scale='no')
# Converting kernels from numpy arrays to torch tensors
wsin = torch.tensor(kernel_sin * window)
wcos = torch.tensor(kernel_cos * window)
cqt_kernels_real = torch.tensor(cqt_kernels.real.astype(np.float32))
cqt_kernels_imag = torch.tensor(cqt_kernels.imag.astype(np.float32))
if trainable_STFT:
wsin = torch.nn.Parameter(wsin, requires_grad=trainable_STFT)
wcos = torch.nn.Parameter(wcos, requires_grad=trainable_STFT)
self.register_parameter('wsin', wsin)
self.register_parameter('wcos', wcos)
else:
self.register_buffer('wsin', wsin)
self.register_buffer('wcos', wcos)
if trainable_CQT:
cqt_kernels_real = torch.nn.Parameter(cqt_kernels_real, requires_grad=trainable_CQT)
cqt_kernels_imag = torch.nn.Parameter(cqt_kernels_imag, requires_grad=trainable_CQT)
self.register_parameter('cqt_kernels_real', cqt_kernels_real)
self.register_parameter('cqt_kernels_imag', cqt_kernels_imag)
else:
self.register_buffer('cqt_kernels_real', cqt_kernels_real)
self.register_buffer('cqt_kernels_imag', cqt_kernels_imag)
print("STFT kernels created, time used = {:.4f} seconds".format(time()-start))
def forward(self, x, output_format=None, normalization_type='librosa'):
"""
Convert a batch of waveforms to CQT spectrograms.
Parameters
----------
x : torch tensor
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
It will be automatically broadcast to the right shape
"""
output_format = output_format or self.output_format
x = broadcast_dim(x)
if self.center:
if self.pad_mode == 'constant':
padding = nn.ConstantPad1d(self.kernel_width//2, 0)
elif self.pad_mode == 'reflect':
padding = nn.ReflectionPad1d(self.kernel_width//2)
x = padding(x)
# STFT
fourier_real = conv1d(x, self.wcos, stride=self.hop_length)
fourier_imag = conv1d(x, self.wsin, stride=self.hop_length)
# CQT
CQT_real, CQT_imag = complex_mul((self.cqt_kernels_real, self.cqt_kernels_imag),
(fourier_real, fourier_imag))
CQT = torch.stack((CQT_real,-CQT_imag),-1)
if normalization_type == 'librosa':
CQT *= torch.sqrt(self.lenghts.view(-1,1,1))/self.kernel_width
elif normalization_type == 'convolutional':
pass
elif normalization_type == 'wrap':
CQT *= 2/self.kernel_width
else:
raise ValueError("The normalization_type %r is not part of our current options." % normalization_type)
# if self.norm:
# CQT = CQT/self.kernel_width*torch.sqrt(self.lenghts.view(-1,1,1))
# else:
# CQT = CQT*torch.sqrt(self.lenghts.view(-1,1,1))
if output_format=='Magnitude':
# Getting CQT Amplitude
return torch.sqrt(CQT.pow(2).sum(-1))
elif output_format=='Complex':
return CQT
elif output_format=='Phase':
phase_real = torch.cos(torch.atan2(CQT_imag,CQT_real))
phase_imag = torch.sin(torch.atan2(CQT_imag,CQT_real))
return torch.stack((phase_real,phase_imag), -1)
def extra_repr(self) -> str:
return 'STFT kernel size = {}, CQT kernel size = {}'.format(
(*self.wcos.shape,), (*self.cqt_kernels_real.shape,)
)
class CQT2010(torch.nn.Module):
"""
This algorithm is using the resampling method proposed in [1].
Instead of convoluting the STFT results with a gigantic CQT kernel covering the full frequency
spectrum, we make a small CQT kernel covering only the top octave.
Then we keep downsampling the input audio by a factor of 2 to convoluting it with the
small CQT kernel. Everytime the input audio is downsampled, the CQT relative to the downsampled
input is equavalent to the next lower octave.
The kernel creation process is still same as the 1992 algorithm. Therefore, we can reuse the code
from the 1992 alogrithm [2]
[1] Schörkhuber, Christian. “CONSTANT-Q TRANSFORM TOOLBOX FOR MUSIC PROCESSING.” (2010).
[2] Brown, Judith C.C. and Miller Puckette. “An efficient algorithm for the calculation of a
constant Q transform.” (1992).
early downsampling factor is to downsample the input audio to reduce the CQT kernel size.
The result with and without early downsampling are more or less the same except in the very low
frequency region where freq < 40Hz.
"""
def __init__(self, sr=22050, hop_length=512, fmin=32.70, fmax=None, n_bins=84, bins_per_octave=12,
norm=True, basis_norm=1, window='hann', pad_mode='reflect', trainable_STFT=False, filter_scale=1,
trainable_CQT=False, output_format='Magnitude', earlydownsample=True, verbose=True):
super().__init__()
self.norm = norm # Now norm is used to normalize the final CQT result by dividing n_fft
# basis_norm is for normalizing basis
self.hop_length = hop_length
self.pad_mode = pad_mode
self.n_bins = n_bins
self.output_format = output_format
self.earlydownsample = earlydownsample # TODO: activate early downsampling later if possible
# This will be used to calculate filter_cutoff and creating CQT kernels
Q = float(filter_scale)/(2**(1/bins_per_octave)-1)
# Creating lowpass filter and make it a torch tensor
if verbose==True:
print("Creating low pass filter ...", end='\r')
start = time()
lowpass_filter = torch.tensor(create_lowpass_filter(
band_center = 0.5,
kernelLength=256,
transitionBandwidth=0.001
)
)
# Broadcast the tensor to the shape that fits conv1d
self.register_buffer('lowpass_filter', lowpass_filter[None,None,:])
if verbose==True:
print("Low pass filter created, time used = {:.4f} seconds".format(time()-start))
# Calculate num of filter requires for the kernel
# n_octaves determines how many resampling requires for the CQT
n_filters = min(bins_per_octave, n_bins)
self.n_octaves = int(np.ceil(float(n_bins) / bins_per_octave))
# print("n_octaves = ", self.n_octaves)
# Calculate the lowest frequency bin for the top octave kernel
self.fmin_t = fmin*2**(self.n_octaves-1)
remainder = n_bins % bins_per_octave
# print("remainder = ", remainder)
if remainder==0:
# Calculate the top bin frequency
fmax_t = self.fmin_t*2**((bins_per_octave-1)/bins_per_octave)
else:
# Calculate the top bin frequency
fmax_t = self.fmin_t*2**((remainder-1)/bins_per_octave)
self.fmin_t = fmax_t/2**(1-1/bins_per_octave) # Adjusting the top minium bins
if fmax_t > sr/2:
raise ValueError('The top bin {}Hz has exceeded the Nyquist frequency, \
please reduce the n_bins'.format(fmax_t))
if self.earlydownsample == True: # Do early downsampling if this argument is True
if verbose==True:
print("Creating early downsampling filter ...", end='\r')
start = time()
sr, self.hop_length, self.downsample_factor, early_downsample_filter, \
self.earlydownsample = get_early_downsample_params(sr,
hop_length,
fmax_t,
Q,
self.n_octaves,
verbose)
self.register_buffer('early_downsample_filter', early_downsample_filter)
if verbose==True:
print("Early downsampling filter created, \
time used = {:.4f} seconds".format(time()-start))
else:
self.downsample_factor=1.
# Preparing CQT kernels
if verbose==True:
print("Creating CQT kernels ...", end='\r')
start = time()
# print("Q = {}, fmin_t = {}, n_filters = {}".format(Q, self.fmin_t, n_filters))
basis, self.n_fft, _, _ = create_cqt_kernels(Q,
sr,
self.fmin_t,
n_filters,
bins_per_octave,
norm=basis_norm,
topbin_check=False)
# This is for the normalization in the end
freqs = fmin * 2.0 ** (np.r_[0:n_bins] / np.float(bins_per_octave))
self.frequencies = freqs
lenghts = np.ceil(Q * sr / freqs)
lenghts = torch.tensor(lenghts).float()
self.register_buffer('lenghts', lenghts)
self.basis=basis
fft_basis = fft(basis)[:,:self.n_fft//2+1] # Convert CQT kenral from time domain to freq domain
# These cqt_kernel is already in the frequency domain
cqt_kernels_real = torch.tensor(fft_basis.real.astype(np.float32))
cqt_kernels_imag = torch.tensor(fft_basis.imag.astype(np.float32))
if verbose==True:
print("CQT kernels created, time used = {:.4f} seconds".format(time()-start))
# print("Getting cqt kernel done, n_fft = ",self.n_fft)
# Preparing kernels for Short-Time Fourier Transform (STFT)
# We set the frequency range in the CQT filter instead of here.
if verbose==True:
print("Creating STFT kernels ...", end='\r')
start = time()
kernel_sin, kernel_cos, self.bins2freq, _, window = create_fourier_kernels(self.n_fft, window='ones', freq_scale='no')
wsin = kernel_sin * window
wcos = kernel_cos * window
wsin = torch.tensor(wsin)
wcos = torch.tensor(wcos)
if verbose==True:
print("STFT kernels created, time used = {:.4f} seconds".format(time()-start))
if trainable_STFT:
wsin = torch.nn.Parameter(wsin, requires_grad=trainable_STFT)
wcos = torch.nn.Parameter(wcos, requires_grad=trainable_STFT)
self.register_parameter('wsin', wsin)
self.register_parameter('wcos', wcos)
else:
self.register_buffer('wsin', wsin)
self.register_buffer('wcos', wcos)
if trainable_CQT:
cqt_kernels_real = torch.nn.Parameter(cqt_kernels_real, requires_grad=trainable_CQT)
cqt_kernels_imag = torch.nn.Parameter(cqt_kernels_imag, requires_grad=trainable_CQT)
self.register_parameter('cqt_kernels_real', cqt_kernels_real)
self.register_parameter('cqt_kernels_imag', cqt_kernels_imag)
else:
self.register_buffer('cqt_kernels_real', cqt_kernels_real)
self.register_buffer('cqt_kernels_imag', cqt_kernels_imag)
# If center==True, the STFT window will be put in the middle, and paddings at the beginning
# and ending are required.
if self.pad_mode == 'constant':
self.padding = nn.ConstantPad1d(self.n_fft//2, 0)
elif self.pad_mode == 'reflect':
self.padding = nn.ReflectionPad1d(self.n_fft//2)
def forward(self,x, output_format=None, normalization_type='librosa'):
"""
Convert a batch of waveforms to CQT spectrograms.
Parameters
----------
x : torch tensor
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
It will be automatically broadcast to the right shape
"""
output_format = output_format or self.output_format
x = broadcast_dim(x)
if self.earlydownsample==True:
x = downsampling_by_n(x, self.early_downsample_filter, self.downsample_factor)
hop = self.hop_length
CQT = get_cqt_complex2(x, self.cqt_kernels_real, self.cqt_kernels_imag, hop, self.padding,
wcos=self.wcos, wsin=self.wsin)
x_down = x # Preparing a new variable for downsampling
for i in range(self.n_octaves-1):
hop = hop//2
x_down = downsampling_by_2(x_down, self.lowpass_filter)
CQT1 = get_cqt_complex2(x_down, self.cqt_kernels_real, self.cqt_kernels_imag, hop, self.padding,
wcos=self.wcos, wsin=self.wsin)
CQT = torch.cat((CQT1, CQT),1)
CQT = CQT[:,-self.n_bins:,:] # Removing unwanted top bins
if normalization_type == 'librosa':
CQT *= torch.sqrt(self.lenghts.view(-1,1,1))/self.n_fft
elif normalization_type == 'convolutional':
pass
elif normalization_type == 'wrap':
CQT *= 2/self.n_fft
else:
raise ValueError("The normalization_type %r is not part of our current options." % normalization_type)
if output_format=='Magnitude':
# Getting CQT Amplitude
return torch.sqrt(CQT.pow(2).sum(-1))
elif output_format=='Complex':
return CQT
elif output_format=='Phase':
phase_real = torch.cos(torch.atan2(CQT[:,:,:,1],CQT[:,:,:,0]))
phase_imag = torch.sin(torch.atan2(CQT[:,:,:,1],CQT[:,:,:,0]))
return torch.stack((phase_real,phase_imag), -1)
def extra_repr(self) -> str:
return 'STFT kernel size = {}, CQT kernel size = {}'.format(
(*self.wcos.shape,), (*self.cqt_kernels_real.shape,)
)
class CQT1992v2(torch.nn.Module):
"""This function is to calculate the CQT of the input signal.
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
The correct shape will be inferred autommatically if the input follows these 3 shapes.
Most of the arguments follow the convention from librosa.
This class inherits from ``torch.nn.Module``, therefore, the usage is same as ``torch.nn.Module``.
This alogrithm uses the method proposed in [1]. I slightly modify it so that it runs faster
than the original 1992 algorithm, that is why I call it version 2.
[1] Brown, Judith C.C. and Miller Puckette. “An efficient algorithm for the calculation of a
constant Q transform.” (1992).
Parameters
----------
sr : int
The sampling rate for the input audio. It is used to calucate the correct ``fmin`` and ``fmax``.
Setting the correct sampling rate is very important for calculating the correct frequency.
hop_length : int
The hop (or stride) size. Default value is 512.
fmin : float
The frequency for the lowest CQT bin. Default is 32.70Hz, which coresponds to the note C0.
fmax : float
The frequency for the highest CQT bin. Default is ``None``, therefore the higest CQT bin is
inferred from the ``n_bins`` and ``bins_per_octave``.
If ``fmax`` is not ``None``, then the argument ``n_bins`` will be ignored and ``n_bins``
will be calculated automatically. Default is ``None``
n_bins : int
The total numbers of CQT bins. Default is 84. Will be ignored if ``fmax`` is not ``None``.
bins_per_octave : int
Number of bins per octave. Default is 12.
filter_scale : float > 0
Filter scale factor. Values of filter_scale smaller than 1 can be used to improve the time resolution at the
cost of degrading the frequency resolution. Important to note is that setting for example filter_scale = 0.5 and
bins_per_octave = 48 leads to exactly the same time-frequency resolution trade-off as setting filter_scale = 1
and bins_per_octave = 24, but the former contains twice more frequency bins per octave. In this sense, values
filter_scale < 1 can be seen to implement oversampling of the frequency axis, analogously to the use of zero
padding when calculating the DFT.
norm : int
Normalization for the CQT kernels. ``1`` means L1 normalization, and ``2`` means L2 normalization.
Default is ``1``, which is same as the normalization used in librosa.
window : string, float, or tuple
The windowing function for CQT. If it is a string, It uses ``scipy.signal.get_window``. If it is a
tuple, only the gaussian window wanrantees constant Q factor. Gaussian window should be given as a
tuple ('gaussian', att) where att is the attenuation in the border given in dB.
Please refer to scipy documentation for possible windowing functions. The default value is 'hann'.
center : bool
Putting the CQT keneral at the center of the time-step or not. If ``False``, the time index is
the beginning of the CQT kernel, if ``True``, the time index is the center of the CQT kernel.
Default value if ``True``.
pad_mode : str
The padding method. Default value is 'reflect'.
trainable : bool
Determine if the CQT kernels are trainable or not. If ``True``, the gradients for CQT kernels
will also be caluclated and the CQT kernels will be updated during model training.
Default value is ``False``.
output_format : str
Determine the return type.
``Magnitude`` will return the magnitude of the STFT result, shape = ``(num_samples, freq_bins,time_steps)``;
``Complex`` will return the STFT result in complex number, shape = ``(num_samples, freq_bins,time_steps, 2)``;
``Phase`` will return the phase of the STFT reuslt, shape = ``(num_samples, freq_bins,time_steps, 2)``.
The complex number is stored as ``(real, imag)`` in the last axis. Default value is 'Magnitude'.
verbose : bool
If ``True``, it shows layer information. If ``False``, it suppresses all prints
Returns
-------
spectrogram : torch.tensor
It returns a tensor of spectrograms.
shape = ``(num_samples, freq_bins,time_steps)`` if ``output_format='Magnitude'``;
shape = ``(num_samples, freq_bins,time_steps, 2)`` if ``output_format='Complex' or 'Phase'``;
Examples
--------
>>> spec_layer = Spectrogram.CQT1992v2()
>>> specs = spec_layer(x)
"""
def __init__(self, sr=22050, hop_length=512, fmin=32.70, fmax=None, n_bins=84,
bins_per_octave=12, filter_scale=1, norm=1, window='hann', center=True, pad_mode='reflect',
trainable=False, output_format='Magnitude', verbose=True):
super().__init__()
self.trainable = trainable
self.hop_length = hop_length
self.center = center
self.pad_mode = pad_mode
self.output_format = output_format
# creating kernels for CQT
Q = float(filter_scale)/(2**(1/bins_per_octave)-1)
if verbose==True:
print("Creating CQT kernels ...", end='\r')
start = time()
cqt_kernels, self.kernel_width, lenghts, freqs = create_cqt_kernels(Q,
sr,
fmin,
n_bins,
bins_per_octave,
norm,
window,
fmax)
self.register_buffer('lenghts', lenghts)
self.frequencies = freqs
cqt_kernels_real = torch.tensor(cqt_kernels.real).unsqueeze(1)
cqt_kernels_imag = torch.tensor(cqt_kernels.imag).unsqueeze(1)
if trainable:
cqt_kernels_real = torch.nn.Parameter(cqt_kernels_real, requires_grad=trainable)
cqt_kernels_imag = torch.nn.Parameter(cqt_kernels_imag, requires_grad=trainable)
self.register_parameter('cqt_kernels_real', cqt_kernels_real)
self.register_parameter('cqt_kernels_imag', cqt_kernels_imag)
else:
self.register_buffer('cqt_kernels_real', cqt_kernels_real)
self.register_buffer('cqt_kernels_imag', cqt_kernels_imag)
if verbose==True:
print("CQT kernels created, time used = {:.4f} seconds".format(time()-start))
def forward(self,x, output_format=None, normalization_type='librosa'):
"""
Convert a batch of waveforms to CQT spectrograms.
Parameters
----------
x : torch tensor
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
It will be automatically broadcast to the right shape
normalization_type : str
Type of the normalisation. The possible options are: \n
'librosa' : the output fits the librosa one \n
'convolutional' : the output conserves the convolutional inequalities of the wavelet transform:\n
for all p ϵ [1, inf] \n
- || CQT ||_p <= || f ||_p || g ||_1 \n
- || CQT ||_p <= || f ||_1 || g ||_p \n
- || CQT ||_2 = || f ||_2 || g ||_2 \n
'wrap' : wraps positive and negative frequencies into positive frequencies. This means that the CQT of a
sinus (or a cosinus) with a constant amplitude equal to 1 will have the value 1 in the bin corresponding to
its frequency.
"""
output_format = output_format or self.output_format
x = broadcast_dim(x)
if self.center:
if self.pad_mode == 'constant':
padding = nn.ConstantPad1d(self.kernel_width//2, 0)
elif self.pad_mode == 'reflect':
padding = nn.ReflectionPad1d(self.kernel_width//2)
x = padding(x)
# CQT
CQT_real = conv1d(x, self.cqt_kernels_real, stride=self.hop_length)
CQT_imag = -conv1d(x, self.cqt_kernels_imag, stride=self.hop_length)
if normalization_type == 'librosa':
CQT_real *= torch.sqrt(self.lenghts.view(-1, 1))
CQT_imag *= torch.sqrt(self.lenghts.view(-1, 1))
elif normalization_type == 'convolutional':
pass
elif normalization_type == 'wrap':
CQT_real *= 2
CQT_imag *= 2
else:
raise ValueError("The normalization_type %r is not part of our current options." % normalization_type)
if output_format=='Magnitude':
if self.trainable==False:
# Getting CQT Amplitude
CQT = torch.sqrt(CQT_real.pow(2)+CQT_imag.pow(2))
else:
CQT = torch.sqrt(CQT_real.pow(2)+CQT_imag.pow(2)+1e-8)
return CQT
elif output_format=='Complex':
return torch.stack((CQT_real,CQT_imag),-1)
elif output_format=='Phase':
phase_real = torch.cos(torch.atan2(CQT_imag,CQT_real))
phase_imag = torch.sin(torch.atan2(CQT_imag,CQT_real))
return torch.stack((phase_real,phase_imag), -1)
def forward_manual(self,x):
"""
Method for debugging
"""
x = broadcast_dim(x)
if self.center:
if self.pad_mode == 'constant':
padding = nn.ConstantPad1d(self.kernel_width//2, 0)
elif self.pad_mode == 'reflect':
padding = nn.ReflectionPad1d(self.kernel_width//2)
x = padding(x)
# CQT
CQT_real = conv1d(x, self.cqt_kernels_real, stride=self.hop_length)
CQT_imag = conv1d(x, self.cqt_kernels_imag, stride=self.hop_length)
# Getting CQT Amplitude
CQT = torch.sqrt(CQT_real.pow(2)+CQT_imag.pow(2))
return CQT*torch.sqrt(self.lenghts.view(-1,1))
class CQT2010v2(torch.nn.Module):
"""This function is to calculate the CQT of the input signal.
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
The correct shape will be inferred autommatically if the input follows these 3 shapes.
Most of the arguments follow the convention from librosa.
This class inherits from ``torch.nn.Module``, therefore, the usage is same as ``torch.nn.Module``.
This alogrithm uses the resampling method proposed in [1].
Instead of convoluting the STFT results with a gigantic CQT kernel covering the full frequency
spectrum, we make a small CQT kernel covering only the top octave. Then we keep downsampling the
input audio by a factor of 2 to convoluting it with the small CQT kernel.
Everytime the input audio is downsampled, the CQT relative to the downsampled input is equivalent
to the next lower octave.
The kernel creation process is still same as the 1992 algorithm. Therefore, we can reuse the
code from the 1992 alogrithm [2]
[1] Schörkhuber, Christian. “CONSTANT-Q TRANSFORM TOOLBOX FOR MUSIC PROCESSING.” (2010).
[2] Brown, Judith C.C. and Miller Puckette. “An efficient algorithm for the calculation of a
constant Q transform.” (1992).
Early downsampling factor is to downsample the input audio to reduce the CQT kernel size.
The result with and without early downsampling are more or less the same except in the very low
frequency region where freq < 40Hz.
Parameters
----------
sr : int
The sampling rate for the input audio. It is used to calucate the correct ``fmin`` and ``fmax``.
Setting the correct sampling rate is very important for calculating the correct frequency.
hop_length : int
The hop (or stride) size. Default value is 512.
fmin : float
The frequency for the lowest CQT bin. Default is 32.70Hz, which coresponds to the note C0.
fmax : float
The frequency for the highest CQT bin. Default is ``None``, therefore the higest CQT bin is
inferred from the ``n_bins`` and ``bins_per_octave``. If ``fmax`` is not ``None``, then the
argument ``n_bins`` will be ignored and ``n_bins`` will be calculated automatically.
Default is ``None``
n_bins : int
The total numbers of CQT bins. Default is 84. Will be ignored if ``fmax`` is not ``None``.
bins_per_octave : int
Number of bins per octave. Default is 12.
norm : bool
Normalization for the CQT result.
basis_norm : int
Normalization for the CQT kernels. ``1`` means L1 normalization, and ``2`` means L2 normalization.
Default is ``1``, which is same as the normalization used in librosa.
window : str
The windowing function for CQT. It uses ``scipy.signal.get_window``, please refer to
scipy documentation for possible windowing functions. The default value is 'hann'
pad_mode : str
The padding method. Default value is 'reflect'.
trainable : bool
Determine if the CQT kernels are trainable or not. If ``True``, the gradients for CQT kernels
will also be caluclated and the CQT kernels will be updated during model training.
Default value is ``False``
output_format : str
Determine the return type.
'Magnitude' will return the magnitude of the STFT result, shape = ``(num_samples, freq_bins, time_steps)``;
'Complex' will return the STFT result in complex number, shape = ``(num_samples, freq_bins, time_steps, 2)``;
'Phase' will return the phase of the STFT reuslt, shape = ``(num_samples, freq_bins,time_steps, 2)``.
The complex number is stored as ``(real, imag)`` in the last axis. Default value is 'Magnitude'.
verbose : bool
If ``True``, it shows layer information. If ``False``, it suppresses all prints.
Returns
-------
spectrogram : torch.tensor
It returns a tensor of spectrograms.
shape = ``(num_samples, freq_bins,time_steps)`` if ``output_format='Magnitude'``;
shape = ``(num_samples, freq_bins,time_steps, 2)`` if ``output_format='Complex' or 'Phase'``;
Examples
--------
>>> spec_layer = Spectrogram.CQT2010v2()
>>> specs = spec_layer(x)
"""
# To DO:
# need to deal with the filter and other tensors
def __init__(self, sr=22050, hop_length=512, fmin=32.70, fmax=None, n_bins=84, filter_scale=1,
bins_per_octave=12, norm=True, basis_norm=1, window='hann', pad_mode='reflect',
earlydownsample=True, trainable=False, output_format='Magnitude', verbose=True):
super().__init__()
self.norm = norm # Now norm is used to normalize the final CQT result by dividing n_fft
# basis_norm is for normalizing basis
self.hop_length = hop_length
self.pad_mode = pad_mode
self.n_bins = n_bins
self.earlydownsample = earlydownsample # We will activate early downsampling later if possible
self.trainable = trainable
self.output_format = output_format
# It will be used to calculate filter_cutoff and creating CQT kernels
Q = float(filter_scale)/(2**(1/bins_per_octave)-1)
# Creating lowpass filter and make it a torch tensor
if verbose==True:
print("Creating low pass filter ...", end='\r')
start = time()
# self.lowpass_filter = torch.tensor(
# create_lowpass_filter(
# band_center = 0.50,
# kernelLength=256,
# transitionBandwidth=0.001))
lowpass_filter = torch.tensor(create_lowpass_filter(
band_center = 0.50,
kernelLength=256,
transitionBandwidth=0.001)
)
# Broadcast the tensor to the shape that fits conv1d
self.register_buffer('lowpass_filter', lowpass_filter[None,None,:])
if verbose==True:
print("Low pass filter created, time used = {:.4f} seconds".format(time()-start))
# Caluate num of filter requires for the kernel
# n_octaves determines how many resampling requires for the CQT
n_filters = min(bins_per_octave, n_bins)
self.n_octaves = int(np.ceil(float(n_bins) / bins_per_octave))
if verbose==True:
print("num_octave = ", self.n_octaves)
# Calculate the lowest frequency bin for the top octave kernel
self.fmin_t = fmin*2**(self.n_octaves-1)
remainder = n_bins % bins_per_octave
# print("remainder = ", remainder)
if remainder==0:
# Calculate the top bin frequency
fmax_t = self.fmin_t*2**((bins_per_octave-1)/bins_per_octave)
else:
# Calculate the top bin frequency
fmax_t = self.fmin_t*2**((remainder-1)/bins_per_octave)
self.fmin_t = fmax_t/2**(1-1/bins_per_octave) # Adjusting the top minium bins
if fmax_t > sr/2:
raise ValueError('The top bin {}Hz has exceeded the Nyquist frequency, \
please reduce the n_bins'.format(fmax_t))
if self.earlydownsample == True: # Do early downsampling if this argument is True
if verbose==True:
print("Creating early downsampling filter ...", end='\r')
start = time()
sr, self.hop_length, self.downsample_factor, early_downsample_filter, \
self.earlydownsample = get_early_downsample_params(sr,
hop_length,
fmax_t,
Q,
self.n_octaves,
verbose)
self.register_buffer('early_downsample_filter', early_downsample_filter)
if verbose==True:
print("Early downsampling filter created, \
time used = {:.4f} seconds".format(time()-start))
else:
self.downsample_factor=1.
# Preparing CQT kernels
if verbose==True:
print("Creating CQT kernels ...", end='\r')
start = time()
basis, self.n_fft, lenghts, _ = create_cqt_kernels(Q,
sr,
self.fmin_t,
n_filters,
bins_per_octave,
norm=basis_norm,
topbin_check=False)
# For normalization in the end
# The freqs returned by create_cqt_kernels cannot be used
# Since that returns only the top octave bins
# We need the information for all freq bin
freqs = fmin * 2.0 ** (np.r_[0:n_bins] / np.float(bins_per_octave))
self.frequencies = freqs
lenghts = np.ceil(Q * sr / freqs)
lenghts = torch.tensor(lenghts).float()
self.register_buffer('lenghts', lenghts)
self.basis = basis
# These cqt_kernel is already in the frequency domain
cqt_kernels_real = torch.tensor(basis.real.astype(np.float32)).unsqueeze(1)
cqt_kernels_imag = torch.tensor(basis.imag.astype(np.float32)).unsqueeze(1)
if trainable:
cqt_kernels_real = torch.nn.Parameter(cqt_kernels_real, requires_grad=trainable)
cqt_kernels_imag = torch.nn.Parameter(cqt_kernels_imag, requires_grad=trainable)
self.register_parameter('cqt_kernels_real', cqt_kernels_real)
self.register_parameter('cqt_kernels_imag', cqt_kernels_imag)
else:
self.register_buffer('cqt_kernels_real', cqt_kernels_real)
self.register_buffer('cqt_kernels_imag', cqt_kernels_imag)
if verbose==True:
print("CQT kernels created, time used = {:.4f} seconds".format(time()-start))
# print("Getting cqt kernel done, n_fft = ",self.n_fft)
# If center==True, the STFT window will be put in the middle, and paddings at the beginning
# and ending are required.
if self.pad_mode == 'constant':
self.padding = nn.ConstantPad1d(self.n_fft//2, 0)
elif self.pad_mode == 'reflect':
self.padding = nn.ReflectionPad1d(self.n_fft//2)
def forward(self,x,output_format=None, normalization_type='librosa'):
"""
Convert a batch of waveforms to CQT spectrograms.
Parameters
----------
x : torch tensor
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
It will be automatically broadcast to the right shape
"""
output_format = output_format or self.output_format
x = broadcast_dim(x)
if self.earlydownsample==True:
x = downsampling_by_n(x, self.early_downsample_filter, self.downsample_factor)
hop = self.hop_length
CQT = get_cqt_complex(x, self.cqt_kernels_real, self.cqt_kernels_imag, hop, self.padding) # Getting the top octave CQT
x_down = x # Preparing a new variable for downsampling
for i in range(self.n_octaves-1):
hop = hop//2
x_down = downsampling_by_2(x_down, self.lowpass_filter)
CQT1 = get_cqt_complex(x_down, self.cqt_kernels_real, self.cqt_kernels_imag, hop, self.padding)
CQT = torch.cat((CQT1, CQT),1)
CQT = CQT[:,-self.n_bins:,:] # Removing unwanted bottom bins
# print("downsample_factor = ",self.downsample_factor)
# print(CQT.shape)
# print(self.lenghts.view(-1,1).shape)
# Normalizing the output with the downsampling factor, 2**(self.n_octaves-1) is make it
# same mag as 1992
CQT = CQT*self.downsample_factor
# Normalize again to get same result as librosa
if normalization_type == 'librosa':
CQT = CQT*torch.sqrt(self.lenghts.view(-1,1,1))
elif normalization_type == 'convolutional':
pass
elif normalization_type == 'wrap':
CQT *= 2
else:
raise ValueError("The normalization_type %r is not part of our current options." % normalization_type)
if output_format=='Magnitude':
if self.trainable==False:
# Getting CQT Amplitude
return torch.sqrt(CQT.pow(2).sum(-1))
else:
return torch.sqrt(CQT.pow(2).sum(-1)+1e-8)
elif output_format=='Complex':
return CQT
elif output_format=='Phase':
phase_real = torch.cos(torch.atan2(CQT[:,:,:,1],CQT[:,:,:,0]))
phase_imag = torch.sin(torch.atan2(CQT[:,:,:,1],CQT[:,:,:,0]))
return torch.stack((phase_real,phase_imag), -1)
class CQT(CQT1992v2):
"""An abbreviation for :func:`~nnAudio.Spectrogram.CQT1992v2`. Please refer to the :func:`~nnAudio.Spectrogram.CQT1992v2` documentation"""
pass
# The section below is for developing purpose
# Please don't use the following classes
#
class DFT(torch.nn.Module):
"""
Experimental feature before `torch.fft` was made avaliable.
The inverse function only works for 1 single frame. i.e. input shape = (batch, n_fft, 1)
"""
def __init__(self, n_fft=2048, freq_bins=None, hop_length=512,
window='hann', freq_scale='no', center=True, pad_mode='reflect',
fmin=50, fmax=6000, sr=22050):
super().__init__()
self.stride = hop_length
self.center = center
self.pad_mode = pad_mode
self.n_fft = n_fft
# Create filter windows for stft
wsin, wcos, self.bins2freq = create_fourier_kernels(n_fft=n_fft,
freq_bins=n_fft,
window=window,
freq_scale=freq_scale,
fmin=fmin,
fmax=fmax,
sr=sr)
self.wsin = torch.tensor(wsin, dtype=torch.float)
self.wcos = torch.tensor(wcos, dtype=torch.float)
def forward(self,x):
"""
Convert a batch of waveforms to spectrums.
Parameters
----------
x : torch tensor
Input signal should be in either of the following shapes.\n
1. ``(len_audio)``\n
2. ``(num_audio, len_audio)``\n
3. ``(num_audio, 1, len_audio)``
It will be automatically broadcast to the right shape
"""
x = broadcast_dim(x)
if self.center:
if self.pad_mode == 'constant':
padding = nn.ConstantPad1d(self.n_fft//2, 0)
elif self.pad_mode == 'reflect':
padding = nn.ReflectionPad1d(self.n_fft//2)
x = padding(x)
imag = conv1d(x, self.wsin, stride=self.stride)
real = conv1d(x, self.wcos, stride=self.stride)
return (real, -imag)
def inverse(self,x_real,x_imag):
"""
Convert a batch of waveforms to CQT spectrograms.
Parameters
----------
x_real : torch tensor
Real part of the signal.
x_imag : torch tensor
Imaginary part of the signal.
"""
x_real = broadcast_dim(x_real)
x_imag = broadcast_dim(x_imag)
x_real.transpose_(1,2) # Prepare the right shape to do inverse
x_imag.transpose_(1,2) # Prepare the right shape to do inverse
# if self.center:
# if self.pad_mode == 'constant':
# padding = nn.ConstantPad1d(self.n_fft//2, 0)
# elif self.pad_mode == 'reflect':
# padding = nn.ReflectionPad1d(self.n_fft//2)
# x_real = padding(x_real)
# x_imag = padding(x_imag)
# Watch out for the positive and negative signs
# ifft = e^(+2\pi*j)*X
# ifft(X_real) = (a1, a2)
# ifft(X_imag)*1j = (b1, b2)*1j
# = (-b2, b1)
a1 = conv1d(x_real, self.wcos, stride=self.stride)
a2 = conv1d(x_real, self.wsin, stride=self.stride)
b1 = conv1d(x_imag, self.wcos, stride=self.stride)
b2 = conv1d(x_imag, self.wsin, stride=self.stride)
imag = a2+b1
real = a1-b2
return (real/self.n_fft, imag/self.n_fft)
class iSTFT(torch.nn.Module):
"""This class is to convert spectrograms back to waveforms. It only works for the complex value spectrograms.
If you have the magnitude spectrograms, please use :func:`~nnAudio.Spectrogram.Griffin_Lim`.
The parameters (e.g. n_fft, window) need to be the same as the STFT in order to obtain the correct inverse.
If trainability is not required, it is recommended to use the ``inverse`` method under the ``STFT`` class
to save GPU/RAM memory.
When ``trainable=True`` and ``freq_scale!='no'``, there is no guarantee that the inverse is perfect, please
use with extra care.
Parameters
----------
n_fft : int
The window size. Default value is 2048.
freq_bins : int
Number of frequency bins. Default is ``None``, which means ``n_fft//2+1`` bins
Please make sure the value is the same as the forward STFT.
hop_length : int
The hop (or stride) size. Default value is ``None`` which is equivalent to ``n_fft//4``.
Please make sure the value is the same as the forward STFT.
window : str
The windowing function for iSTFT. It uses ``scipy.signal.get_window``, please refer to
scipy documentation for possible windowing functions. The default value is 'hann'.
Please make sure the value is the same as the forward STFT.
freq_scale : 'linear', 'log', or 'no'
Determine the spacing between each frequency bin. When `linear` or `log` is used,
the bin spacing can be controlled by ``fmin`` and ``fmax``. If 'no' is used, the bin will
start at 0Hz and end at Nyquist frequency with linear spacing.
Please make sure the value is the same as the forward STFT.
center : bool
Putting the iSTFT keneral at the center of the time-step or not. If ``False``, the time
index is the beginning of the iSTFT kernel, if ``True``, the time index is the center of
the iSTFT kernel. Default value if ``True``.
Please make sure the value is the same as the forward STFT.
fmin : int
The starting frequency for the lowest frequency bin. If freq_scale is ``no``, this argument
does nothing. Please make sure the value is the same as the forward STFT.
fmax : int
The ending frequency for the highest frequency bin. If freq_scale is ``no``, this argument
does nothing. Please make sure the value is the same as the forward STFT.
sr : int
The sampling rate for the input audio. It is used to calucate the correct ``fmin`` and ``fmax``.
Setting the correct sampling rate is very important for calculating the correct frequency.
trainable_kernels : bool
Determine if the STFT kenrels are trainable or not. If ``True``, the gradients for STFT
kernels will also be caluclated and the STFT kernels will be updated during model training.
Default value is ``False``.
trainable_window : bool
Determine if the window function is trainable or not.
Default value is ``False``.
verbose : bool
If ``True``, it shows layer information. If ``False``, it suppresses all prints.
Returns
-------
spectrogram : torch.tensor
It returns a batch of waveforms.
Examples
--------
>>> spec_layer = Spectrogram.iSTFT()
>>> specs = spec_layer(x)
"""
def __init__(self, n_fft=2048, win_length=None, freq_bins=None, hop_length=None, window='hann',
freq_scale='no', center=True, fmin=50, fmax=6000, sr=22050, trainable_kernels=False,
trainable_window=False, verbose=True, refresh_win=True):
super().__init__()
# Trying to make the default setting same as librosa
if win_length==None: win_length = n_fft
if hop_length==None: hop_length = int(win_length // 4)
self.n_fft = n_fft
self.win_length = win_length
self.stride = hop_length
self.center = center
self.pad_amount = self.n_fft // 2
self.refresh_win = refresh_win
start = time()
# Create the window function and prepare the shape for batch-wise-time-wise multiplication
# Create filter windows for inverse
kernel_sin, kernel_cos, _, _, window_mask = create_fourier_kernels(n_fft,
win_length=win_length,
freq_bins=n_fft,
window=window,
freq_scale=freq_scale,
fmin=fmin,
fmax=fmax,
sr=sr,
verbose=False)
window_mask = get_window(window,int(win_length), fftbins=True)
# For inverse, the Fourier kernels do not need to be windowed
window_mask = torch.tensor(window_mask).unsqueeze(0).unsqueeze(-1)
# kernel_sin and kernel_cos have the shape (freq_bins, 1, n_fft, 1) to support 2D Conv
kernel_sin = torch.tensor(kernel_sin, dtype=torch.float).unsqueeze(-1)
kernel_cos = torch.tensor(kernel_cos, dtype=torch.float).unsqueeze(-1)
# Decide if the Fourier kernels are trainable
if trainable_kernels:
# Making all these variables trainable
kernel_sin = torch.nn.Parameter(kernel_sin, requires_grad=trainable_kernels)
kernel_cos = torch.nn.Parameter(kernel_cos, requires_grad=trainable_kernels)
self.register_parameter('kernel_sin', kernel_sin)
self.register_parameter('kernel_cos', kernel_cos)
else:
self.register_buffer('kernel_sin', kernel_sin)
self.register_buffer('kernel_cos', kernel_cos)
# Decide if the window function is trainable
if trainable_window:
window_mask = torch.nn.Parameter(window_mask, requires_grad=trainable_window)
self.register_parameter('window_mask', window_mask)
else:
self.register_buffer('window_mask', window_mask)
if verbose==True:
print("iSTFT kernels created, time used = {:.4f} seconds".format(time()-start))
else:
pass
def forward(self, X, onesided=False, length=None, refresh_win=None):
"""
If your spectrograms only have ``n_fft//2+1`` frequency bins, please use ``onesided=True``,
else use ``onesided=False``
To make sure the inverse STFT has the same output length of the original waveform, please
set `length` as your intended waveform length. By default, ``length=None``,
which will remove ``n_fft//2`` samples from the start and the end of the output.
If your input spectrograms X are of the same length, please use ``refresh_win=None`` to increase
computational speed.
"""
if refresh_win==None:
refresh_win=self.refresh_win
assert X.dim()==4 , "Inverse iSTFT only works for complex number," \
"make sure our tensor is in the shape of (batch, freq_bins, timesteps, 2)"
# If the input spectrogram contains only half of the n_fft
# Use extend_fbins function to get back another half
if onesided:
X = extend_fbins(X) # extend freq
X_real, X_imag = X[:, :, :, 0], X[:, :, :, 1]
# broadcast dimensions to support 2D convolution
X_real_bc = X_real.unsqueeze(1)
X_imag_bc = X_imag.unsqueeze(1)
a1 = conv2d(X_real_bc, self.kernel_cos, stride=(1,1))
b2 = conv2d(X_imag_bc, self.kernel_sin, stride=(1,1))
# compute real and imag part. signal lies in the real part
real = a1 - b2
real = real.squeeze(-2)*self.window_mask
# Normalize the amplitude with n_fft
real /= (self.n_fft)
# Overlap and Add algorithm to connect all the frames
real = overlap_add(real, self.stride)
# Prepare the window sumsqure for division
# Only need to create this window once to save time
# Unless the input spectrograms have different time steps
if hasattr(self, 'w_sum')==False or refresh_win==True:
self.w_sum = torch_window_sumsquare(self.window_mask.flatten(), X.shape[2], self.stride, self.n_fft).flatten()
self.nonzero_indices = (self.w_sum>1e-10)
else:
pass
real[:, self.nonzero_indices] = real[:,self.nonzero_indices].div(self.w_sum[self.nonzero_indices])
# Remove padding
if length is None:
if self.center:
real = real[:, self.pad_amount:-self.pad_amount]
else:
if self.center:
real = real[:, self.pad_amount:self.pad_amount + length]
else:
real = real[:, :length]
return real
class Griffin_Lim(torch.nn.Module):
"""
Converting Magnitude spectrograms back to waveforms based on the "fast Griffin-Lim"[1].
This Griffin Lim is a direct clone from librosa.griffinlim.
[1] Perraudin, N., Balazs, P., & Søndergaard, P. L. “A fast Griffin-Lim algorithm,”
IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (pp. 1-4), Oct. 2013.
Parameters
----------
n_fft : int
The window size. Default value is 2048.
n_iter=32 : int
The number of iterations for Griffin-Lim. The default value is ``32``
hop_length : int
The hop (or stride) size. Default value is ``None`` which is equivalent to ``n_fft//4``.
Please make sure the value is the same as the forward STFT.
window : str
The windowing function for iSTFT. It uses ``scipy.signal.get_window``, please refer to
scipy documentation for possible windowing functions. The default value is 'hann'.
Please make sure the value is the same as the forward STFT.
center : bool
Putting the iSTFT keneral at the center of the time-step or not. If ``False``, the time
index is the beginning of the iSTFT kernel, if ``True``, the time index is the center of
the iSTFT kernel. Default value if ``True``.
Please make sure the value is the same as the forward STFT.
momentum : float
The momentum for the update rule. The default value is ``0.99``.
device : str
Choose which device to initialize this layer. Default value is 'cpu'
"""
def __init__(self,
n_fft,
n_iter=32,
hop_length=None,
win_length=None,
window='hann',
center=True,
pad_mode='reflect',
momentum=0.99,
device='cpu'):
super().__init__()
self.n_fft = n_fft
self.win_length = win_length
self.n_iter = n_iter
self.center = center
self.pad_mode = pad_mode
self.momentum = momentum
self.device = device
if win_length==None:
self.win_length=n_fft
else:
self.win_length=win_length
if hop_length==None:
self.hop_length = n_fft//4
else:
self.hop_length = hop_length
# Creating window function for stft and istft later
self.w = torch.tensor(get_window(window,
int(self.win_length),
fftbins=True),
device=device).float()
def forward(self, S):
"""
Convert a batch of magnitude spectrograms to waveforms.
Parameters
----------
S : torch tensor
Spectrogram of the shape ``(batch, n_fft//2+1, timesteps)``
"""
assert S.dim()==3 , "Please make sure your input is in the shape of (batch, freq_bins, timesteps)"
# Initializing Random Phase
rand_phase = torch.randn(*S.shape, device=self.device)
angles = torch.empty((*S.shape,2), device=self.device)
angles[:, :,:,0] = torch.cos(2 * np.pi * rand_phase)
angles[:,:,:,1] = torch.sin(2 * np.pi * rand_phase)
# Initializing the rebuilt magnitude spectrogram
rebuilt = torch.zeros(*angles.shape, device=self.device)
for _ in range(self.n_iter):
tprev = rebuilt # Saving previous rebuilt magnitude spec
# spec2wav conversion
# print(f'win_length={self.win_length}\tw={self.w.shape}')
inverse = torch.istft(S.unsqueeze(-1) * angles,
self.n_fft,
self.hop_length,
win_length=self.win_length,
window=self.w,
center=self.center)
# wav2spec conversion
rebuilt = torch.stft(inverse,
self.n_fft,
self.hop_length,
win_length=self.win_length,
window=self.w,
pad_mode=self.pad_mode)
# Phase update rule
angles[:,:,:] = rebuilt[:,:,:] - (self.momentum / (1 + self.momentum)) * tprev[:,:,:]
# Phase normalization
angles = angles.div(torch.sqrt(angles.pow(2).sum(-1)).unsqueeze(-1) + 1e-16) # normalizing the phase
# Using the final phase to reconstruct the waveforms
inverse = torch.istft(S.unsqueeze(-1) * angles,
self.n_fft,
self.hop_length,
win_length=self.win_length,
window=self.w,
center=self.center)
return inverse
class Combined_Frequency_Periodicity(nn.Module):
"""
Vectorized version of the code in https://github.com/leo-so/VocalMelodyExtPatchCNN/blob/master/MelodyExt.py.
This feature is described in 'Combining Spectral and Temporal Representations for Multipitch Estimation of Polyphonic Music'
https://ieeexplore.ieee.org/document/7118691
Under development, please report any bugs you found
"""
def __init__(self,fr=2, fs=16000, hop_length=320,
window_size=2049, fc=80, tc=1/1000,
g=[0.24, 0.6, 1], NumPerOct=48):
super().__init__()
self.window_size = window_size
self.hop_length = hop_length
# variables for STFT part
self.N = int(fs/float(fr)) # Will be used to calculate padding
self.f = fs*np.linspace(0, 0.5, np.round(self.N//2), endpoint=True) # it won't be used but will be returned
self.pad_value = ((self.N-window_size))
# Create window function, always blackmanharris?
h = scipy.signal.blackmanharris(window_size).astype(np.float32) # window function for STFT
self.register_buffer('h',torch.tensor(h))
# variables for CFP
self.NumofLayer = np.size(g)
self.g = g
self.tc_idx = round(fs*tc) # index to filter out top tc_idx and bottom tc_idx bins
self.fc_idx = round(fc/fr) # index to filter out top fc_idx and bottom fc_idx bins
self.HighFreqIdx = int(round((1/tc)/fr)+1)
self.HighQuefIdx = int(round(fs/fc)+1)
# attributes to be returned
self.f = self.f[:self.HighFreqIdx]
self.q = np.arange(self.HighQuefIdx)/float(fs)
# filters for the final step
freq2logfreq_matrix, quef2logfreq_matrix = self.create_logfreq_matrix(self.f, self.q, fr, fc, tc, NumPerOct, fs)
self.register_buffer('freq2logfreq_matrix',torch.tensor(freq2logfreq_matrix.astype(np.float32)))
self.register_buffer('quef2logfreq_matrix',torch.tensor(quef2logfreq_matrix.astype(np.float32)))
def _CFP(self, spec):
spec = torch.relu(spec).pow(self.g[0])
if self.NumofLayer >= 2:
for gc in range(1, self.NumofLayer):
if np.remainder(gc, 2) == 1:
ceps = torch.rfft(spec, 1, onesided=False)[:,:,:,0]/np.sqrt(self.N)
ceps = self.nonlinear_func(ceps, self.g[gc], self.tc_idx)
else:
spec = torch.rfft(ceps, 1, onesided=False)[:,:,:,0]/np.sqrt(self.N)
spec = self.nonlinear_func(spec, self.g[gc], self.fc_idx)
return spec, ceps
def forward(self, x):
tfr0 = torch.stft(x, self.N, hop_length=self.hop_length, win_length=self.window_size,
window=self.h, onesided=False, pad_mode='constant')
tfr0 = torch.sqrt(tfr0.pow(2).sum(-1))/torch.norm(self.h) # calcuate magnitude
tfr0 = tfr0.transpose(1,2)[:,1:-1] #transpose F and T axis and discard first and last frames
# The transpose is necessary for rfft later
# (batch, timesteps, n_fft)
tfr, ceps = self._CFP(tfr0)
# return tfr0
# removing duplicate bins
tfr0 = tfr0[:,:,:int(round(self.N/2))]
tfr = tfr[:,:,:int(round(self.N/2))]
ceps = ceps[:,:,:int(round(self.N/2))]
# Crop up to the highest frequency
tfr0 = tfr0[:,:,:self.HighFreqIdx]
tfr = tfr[:,:,:self.HighFreqIdx]
ceps = ceps[:,:,:self.HighQuefIdx]
tfrL0 = torch.matmul(self.freq2logfreq_matrix, tfr0.transpose(1,2))
tfrLF = torch.matmul(self.freq2logfreq_matrix, tfr.transpose(1,2))
tfrLQ = torch.matmul(self.quef2logfreq_matrix, ceps.transpose(1,2))
Z = tfrLF * tfrLQ
# Only need to calculate this once
self.t = np.arange(self.hop_length,
np.ceil(len(x)/float(self.hop_length))*self.hop_length,
self.hop_length) # it won't be used but will be returned
return Z, tfrL0, tfrLF, tfrLQ
def nonlinear_func(self, X, g, cutoff):
cutoff = int(cutoff)
if g!=0:
X = torch.relu(X)
X[:, :, :cutoff] = 0
X[:, :, -cutoff:] = 0
X = X.pow(g)
else: # when g=0, it converges to log
X = torch.log(X)
X[:, :, :cutoff] = 0
X[:, :, -cutoff:] = 0
return X
def create_logfreq_matrix(self, f, q, fr, fc, tc, NumPerOct, fs):
StartFreq = fc
StopFreq = 1/tc
Nest = int(np.ceil(np.log2(StopFreq/StartFreq))*NumPerOct)
central_freq = [] # A list holding the frequencies in log scale
for i in range(0, Nest):
CenFreq = StartFreq*pow(2, float(i)/NumPerOct)
if CenFreq < StopFreq:
central_freq.append(CenFreq)
else:
break
Nest = len(central_freq)
freq_band_transformation = np.zeros((Nest-1, len(f)), dtype=np.float)
# Calculating the freq_band_transformation
for i in range(1, Nest-1):
l = int(round(central_freq[i-1]/fr))
r = int(round(central_freq[i+1]/fr)+1)
#rounding1
if l >= r-1:
freq_band_transformation[i, l] = 1
else:
for j in range(l, r):
if f[j] > central_freq[i-1] and f[j] < central_freq[i]:
freq_band_transformation[i, j] = (f[j] - central_freq[i-1]) / (central_freq[i] - central_freq[i-1])
elif f[j] > central_freq[i] and f[j] < central_freq[i+1]:
freq_band_transformation[i, j] = (central_freq[i + 1] - f[j]) / (central_freq[i + 1] - central_freq[i])
# Calculating the quef_band_transformation
f = 1/q # divide by 0, do I need to fix this?
quef_band_transformation = np.zeros((Nest-1, len(f)), dtype=np.float)
for i in range(1, Nest-1):
for j in range(int(round(fs/central_freq[i+1])), int(round(fs/central_freq[i-1])+1)):
if f[j] > central_freq[i-1] and f[j] < central_freq[i]:
quef_band_transformation[i, j] = (f[j] - central_freq[i-1])/(central_freq[i] - central_freq[i-1])
elif f[j] > central_freq[i] and f[j] < central_freq[i+1]:
quef_band_transformation[i, j] = (central_freq[i + 1] - f[j]) / (central_freq[i + 1] - central_freq[i])
return freq_band_transformation, quef_band_transformation
class CFP(nn.Module):
"""
This is the modified version so that the number of timesteps fits with other classes
Under development, please report any bugs you found
"""
def __init__(self,fr=2, fs=16000, hop_length=320,
window_size=2049, fc=80, tc=1/1000,
g=[0.24, 0.6, 1], NumPerOct=48):
super().__init__()
self.window_size = window_size
self.hop_length = hop_length
# variables for STFT part
self.N = int(fs/float(fr)) # Will be used to calculate padding
self.f = fs*np.linspace(0, 0.5, np.round(self.N//2), endpoint=True) # it won't be used but will be returned
self.pad_value = ((self.N-window_size))
# Create window function, always blackmanharris?
h = scipy.signal.blackmanharris(window_size).astype(np.float32) # window function for STFT
self.register_buffer('h',torch.tensor(h))
# variables for CFP
self.NumofLayer = np.size(g)
self.g = g
self.tc_idx = round(fs*tc) # index to filter out top tc_idx and bottom tc_idx bins
self.fc_idx = round(fc/fr) # index to filter out top fc_idx and bottom fc_idx bins
self.HighFreqIdx = int(round((1/tc)/fr)+1)
self.HighQuefIdx = int(round(fs/fc)+1)
# attributes to be returned
self.f = self.f[:self.HighFreqIdx]
self.q = np.arange(self.HighQuefIdx)/float(fs)
# filters for the final step
freq2logfreq_matrix, quef2logfreq_matrix = self.create_logfreq_matrix(self.f, self.q, fr, fc, tc, NumPerOct, fs)
self.register_buffer('freq2logfreq_matrix',torch.tensor(freq2logfreq_matrix.astype(np.float32)))
self.register_buffer('quef2logfreq_matrix',torch.tensor(quef2logfreq_matrix.astype(np.float32)))
def _CFP(self, spec):
spec = torch.relu(spec).pow(self.g[0])
if self.NumofLayer >= 2:
for gc in range(1, self.NumofLayer):
if np.remainder(gc, 2) == 1:
ceps = torch.rfft(spec, 1, onesided=False)[:,:,:,0]/np.sqrt(self.N)
ceps = self.nonlinear_func(ceps, self.g[gc], self.tc_idx)
else:
spec = torch.rfft(ceps, 1, onesided=False)[:,:,:,0]/np.sqrt(self.N)
spec = self.nonlinear_func(spec, self.g[gc], self.fc_idx)
return spec, ceps
def forward(self, x):
tfr0 = torch.stft(x, self.N, hop_length=self.hop_length, win_length=self.window_size,
window=self.h, onesided=False, pad_mode='constant')
tfr0 = torch.sqrt(tfr0.pow(2).sum(-1))/torch.norm(self.h) # calcuate magnitude
tfr0 = tfr0.transpose(1,2) #transpose F and T axis and discard first and last frames
# The transpose is necessary for rfft later
# (batch, timesteps, n_fft)
tfr, ceps = self._CFP(tfr0)
# return tfr0
# removing duplicate bins
tfr0 = tfr0[:,:,:int(round(self.N/2))]
tfr = tfr[:,:,:int(round(self.N/2))]
ceps = ceps[:,:,:int(round(self.N/2))]
# Crop up to the highest frequency
tfr0 = tfr0[:,:,:self.HighFreqIdx]
tfr = tfr[:,:,:self.HighFreqIdx]
ceps = ceps[:,:,:self.HighQuefIdx]
tfrL0 = torch.matmul(self.freq2logfreq_matrix, tfr0.transpose(1,2))
tfrLF = torch.matmul(self.freq2logfreq_matrix, tfr.transpose(1,2))
tfrLQ = torch.matmul(self.quef2logfreq_matrix, ceps.transpose(1,2))
Z = tfrLF * tfrLQ
# Only need to calculate this once
self.t = np.arange(self.hop_length,
np.ceil(len(x)/float(self.hop_length))*self.hop_length,
self.hop_length) # it won't be used but will be returned
return Z#, tfrL0, tfrLF, tfrLQ
def nonlinear_func(self, X, g, cutoff):
cutoff = int(cutoff)
if g!=0:
X = torch.relu(X)
X[:, :, :cutoff] = 0
X[:, :, -cutoff:] = 0
X = X.pow(g)
else: # when g=0, it converges to log
X = torch.log(X)
X[:, :, :cutoff] = 0
X[:, :, -cutoff:] = 0
return X
def create_logfreq_matrix(self, f, q, fr, fc, tc, NumPerOct, fs):
StartFreq = fc
StopFreq = 1/tc
Nest = int(np.ceil(np.log2(StopFreq/StartFreq))*NumPerOct)
central_freq = [] # A list holding the frequencies in log scale
for i in range(0, Nest):
CenFreq = StartFreq*pow(2, float(i)/NumPerOct)
if CenFreq < StopFreq:
central_freq.append(CenFreq)
else:
break
Nest = len(central_freq)
freq_band_transformation = np.zeros((Nest-1, len(f)), dtype=np.float)
# Calculating the freq_band_transformation
for i in range(1, Nest-1):
l = int(round(central_freq[i-1]/fr))
r = int(round(central_freq[i+1]/fr)+1)
#rounding1
if l >= r-1:
freq_band_transformation[i, l] = 1
else:
for j in range(l, r):
if f[j] > central_freq[i-1] and f[j] < central_freq[i]:
freq_band_transformation[i, j] = (f[j] - central_freq[i-1]) / (central_freq[i] - central_freq[i-1])
elif f[j] > central_freq[i] and f[j] < central_freq[i+1]:
freq_band_transformation[i, j] = (central_freq[i + 1] - f[j]) / (central_freq[i + 1] - central_freq[i])
# Calculating the quef_band_transformation
f = 1/q # divide by 0, do I need to fix this?
quef_band_transformation = np.zeros((Nest-1, len(f)), dtype=np.float)
for i in range(1, Nest-1):
for j in range(int(round(fs/central_freq[i+1])), int(round(fs/central_freq[i-1])+1)):
if f[j] > central_freq[i-1] and f[j] < central_freq[i]:
quef_band_transformation[i, j] = (f[j] - central_freq[i-1])/(central_freq[i] - central_freq[i-1])
elif f[j] > central_freq[i] and f[j] < central_freq[i+1]:
quef_band_transformation[i, j] = (central_freq[i + 1] - f[j]) / (central_freq[i + 1] - central_freq[i])
return freq_band_transformation, quef_band_transformation
Reference in new issue View Git Blame Copy Permalink