def __early_downsample(y, sr, hop_length, res_type, n_octaves, nyquist,
filter_cutoff, scale):
'''Perform early downsampling on an audio signal, if it applies.'''
downsample_count = __early_downsample_count(nyquist, filter_cutoff,
hop_length, n_octaves)
if downsample_count > 0 and res_type == 'kaiser_fast':
downsample_factor = 2**(downsample_count)
hop_length //= downsample_factor
if len(y) < downsample_factor:
raise ParameterError('Input signal length={:d} is too short for '
'{:d}-octave CQT'.format(len(y), n_octaves))
new_sr = sr / float(downsample_factor)
y = audio.resample(y, sr, new_sr, res_type=res_type, scale=True)
# If we're not going to length-scale after CQT, we
# need to compensate for the downsampling factor here
if not scale:
y *= np.sqrt(downsample_factor)
sr = new_sr
return y, sr, hop_length
def main():
from nara_wpe import project_root
import soundfile as sf
from nara_wpe.utils import stft
from nara_wpe.utils import istft as istft
from nara_wpe.utils import get_stft_center_frequencies
from tqdm import tqdm
from librosa.core.audio import resample
channels = 8
parameter_set = 'Katka'
if parameter_set == 'Katka':
sampling_rate = 16000
stft_size, stft_shift = 512, 128
delay = 3
iterations = 5
def get_K(f):
return 10
elif parameter_set == 'Yoshioka2012GeneralWPE':
sampling_rate = 8000
stft_size, stft_shift = 128, 64
delay = 2
iterations = 2
def get_K(f):
if center_frequencies[f] < 800:
K = 18
elif center_frequencies[f] < 1500:
K = 15
else:
K = 12
return K
else:
raise ValueError
file_template = 'AMI_WSJ20-Array1-{}_T10c0201.wav'
signal_list = [
sf.read(str(project_root / 'data' / file_template.format(d + 1)))[0]
for d in range(channels)
]
signal_list = [resample(x_, 16000, sampling_rate) for x_ in signal_list]
y = np.stack(signal_list, axis=0)
center_frequencies = get_stft_center_frequencies(stft_size, sampling_rate)
Y = stft(y, size=stft_size, shift=stft_shift)
X = np.copy(Y)
D, T, F = Y.shape
for f in tqdm(range(F), total=F):
K = get_K(f)
X[:, :, f] = wpe_v5(Y[:, :, f],
K=K,
delay=delay,
iterations=iterations)
x = istft(X, size=stft_size, shift=stft_shift)
sf.write(str(project_root / 'data' / 'wpe_out.wav'),
x[0],
samplerate=sampling_rate)
def main(channels, sampling_rate, file_template, taps_frequency_dependent,
delay, iterations):
"""
User interface for WPE. The defaults of the command line interface are
suited for example audio files of nara_wpe.
'Yoshioka2012GeneralWPE'
sampling_rate = 8000
delay = 2
iterations = 2
"""
from nara_wpe import project_root
import soundfile as sf
from nara_wpe.utils import stft
from nara_wpe.utils import istft
from nara_wpe.utils import get_stft_center_frequencies
from tqdm import tqdm
from librosa.core.audio import resample
stft_options = dict(size=512,
shift=128,
window_length=None,
fading=True,
pad=True,
symmetric_window=False)
def get_taps(f, mode=taps_frequency_dependent):
if mode:
if center_frequencies[f] < 800:
taps = 18
elif center_frequencies[f] < 1500:
taps = 15
else:
taps = 12
else:
taps = 10
return taps
if file_template == 'AMI_WSJ20-Array1-{}_T10c0201.wav':
signal_list = [
sf.read(str(project_root / 'data' /
file_template.format(d + 1)))[0]
for d in range(channels)
]
else:
signal = sf.read(file_template)[0].transpose(1, 0)
signal_list = list(signal)
signal_list = [resample(x_, 16000, sampling_rate) for x_ in signal_list]
y = np.stack(signal_list, axis=0)
center_frequencies = get_stft_center_frequencies(stft_options['size'],
sampling_rate)
Y = stft(y, **stft_options)
X = np.copy(Y)
D, T, F = Y.shape
for f in tqdm(range(F), total=F):
taps = get_taps(f)
X[:, :, f] = wpe_v7(Y[:, :, f],
taps=taps,
delay=delay,
iterations=iterations)
x = istft(X, size=stft_options['size'], shift=stft_options['shift'])
sf.write(str(project_root / 'data' / 'wpe_out.wav'),
x[0],
samplerate=sampling_rate)
print('Output in {}'.format(str(project_root / 'data' / 'wpe_out.wav')))
def icqt(C,
sr=22050,
hop_length=512,
fmin=None,
n_bins=84,
bins_per_octave=12,
filter_scale=1,
norm=1,
sparsity=0.01,
window='hann',
scale=True,
pad_mode='reflect',
use_smoothing=True):
tuning = 0.0
# How many octaves are we dealing with?
n_octaves = int(np.ceil(float(n_bins) / bins_per_octave))
n_filters = min(bins_per_octave, n_bins)
if scale:
lengths = filters.constant_q_lengths(sr,
fmin,
n_bins=n_bins,
bins_per_octave=bins_per_octave,
tuning=tuning,
window=window,
filter_scale=filter_scale)
C *= np.sqrt(lengths[:, np.newaxis])
if fmin is None:
# C1 by default
fmin = note_to_hz('C1')
# First thing, get the freqs of the top octave
freqs = cqt_frequencies(n_bins, fmin,
bins_per_octave=bins_per_octave)[-bins_per_octave:]
fmin_t = np.min(freqs)
fmax_t = np.max(freqs)
# Determine required resampling quality
Q = float(filter_scale) / (2.0**(1. / bins_per_octave) - 1)
filter_cutoff = fmax_t * (1 + 0.5 * filters.window_bandwidth(window) / Q)
nyquist = sr / 2.0
if filter_cutoff < audio.BW_FASTEST * nyquist:
res_type = 'kaiser_fast'
else:
res_type = 'kaiser_best'
y = np.zeros((1000, ))
y, sr, hop_length = __early_downsample(y, sr, hop_length, res_type,
n_octaves, nyquist, filter_cutoff,
scale)
cqt_resp = []
for i in range(n_octaves):
cqt_resp += [
C[i * bins_per_octave:i * bins_per_octave + bins_per_octave, :]
]
cqt_resp = cqt_resp[::-1]
if res_type != 'kaiser_fast':
# Do the top octave before resampling to allow for fast resampling
fft_basis, n_fft, _ = __cqt_filter_fft(sr,
fmin_t,
n_filters,
bins_per_octave,
tuning,
filter_scale,
norm,
sparsity,
window=window)
fft_basis = np.linalg.pinv(fft_basis)
# Compute the CQT filter response and append it to the stack
y = __icqt_response(cqt_resp[0], n_fft, hop_length, fft_basis,
pad_mode)
fmin_t /= 2
fmax_t /= 2
n_octaves -= 1
filter_cutoff = fmax_t * (1 +
0.5 * filters.window_bandwidth(window) / Q)
res_type = 'kaiser_fast'
# Make sure our hop is long enough to support the bottom octave
num_twos = __num_two_factors(hop_length)
if num_twos < n_octaves - 1:
raise ParameterError('hop_length must be a positive integer '
'multiple of 2^{0:d} for {1:d}-octave CQT'.format(
n_octaves - 1, n_octaves))
# Now do the recursive bit
fft_basis, n_fft, _ = __cqt_filter_fft(sr,
fmin_t,
n_filters,
bins_per_octave,
tuning,
filter_scale,
norm,
sparsity,
window=window)
fft_basis = np.linalg.pinv(fft_basis)
my_y, my_sr, my_hop = y, sr, hop_length
y = 0.0
# Iterate down the octaves
for i in range(n_octaves):
# Resample (except first time)
if i > 0:
#my_y = audio_resample_tf(my_y, my_sr, my_sr/2.0,
# res_type=res_type,
# scale=True, use_smoothing=use_smoothing)
# The re-scale the filters to compensate for downsampling
fft_basis /= np.sqrt(2)
my_sr /= 2.0
my_hop //= 2
# Compute the cqt filter response and append to the stack
my_y = __icqt_response(cqt_resp[i], n_fft, my_hop, fft_basis,
pad_mode)
my_y = audio.resample(my_y,
my_sr,
sr,
res_type=res_type,
scale=True)
y += my_y
else:
my_y = __icqt_response(cqt_resp[i], n_fft, my_hop, fft_basis,
pad_mode)
y += my_y
print('Octave:', i)
print('y.size:', my_y.size)
print('SR:', my_sr)
print('Hop:', my_hop)
print('New SR:', sr)
return y
def icqt(C,
sr=22050,
hop_length=512,
fmin=None,
bins_per_octave=12,
tuning=0.0,
filter_scale=1,
norm=1,
sparsity=0.01,
window='hann',
scale=True,
amin=1e-6):
'''Compute the inverse constant-Q transform.
Given a constant-Q transform representation `C` of an audio signal `y`,
this function produces an approximation `y_hat`.
.. warning:: This implementation is unstable, and subject to change in
future versions of librosa. We recommend that its use be
limited to sonification and diagnostic applications.
Parameters
----------
C : np.ndarray, [shape=(n_bins, n_frames)]
Constant-Q representation as produced by `core.cqt`
hop_length : int > 0 [scalar]
number of samples between successive frames
fmin : float > 0 [scalar]
Minimum frequency. Defaults to C1 ~= 32.70 Hz
tuning : float in `[-0.5, 0.5)` [scalar]
Tuning offset in fractions of a bin (cents).
filter_scale : float > 0 [scalar]
Filter scale factor. Small values (<1) use shorter windows
for improved time resolution.
norm : {inf, -inf, 0, float > 0}
Type of norm to use for basis function normalization.
See `librosa.util.normalize`.
sparsity : float in [0, 1)
Sparsify the CQT basis by discarding up to `sparsity`
fraction of the energy in each basis.
Set `sparsity=0` to disable sparsification.
window : str, tuple, number, or function
Window specification for the basis filters.
See `filters.get_window` for details.
scale : bool
If `True`, scale the CQT response by square-root the length
of each channel's filter. This is analogous to `norm='ortho'` in FFT.
If `False`, do not scale the CQT. This is analogous to `norm=None`
in FFT.
amin : float or None
When applying squared window normalization, sample positions with
coefficients below `amin` will left as is.
If `None`, then `amin` is inferred as the smallest valid floating
point value.
Returns
-------
y : np.ndarray, [shape=(n_samples), dtype=np.float]
Audio time-series reconstructed from the CQT representation.
See Also
--------
cqt
Notes
-----
This function caches at level 40.
Examples
--------
Using default parameters
>>> y, sr = librosa.load(librosa.util.example_audio_file(), duration=15)
>>> C = librosa.cqt(y=y, sr=sr)
>>> y_hat = librosa.icqt(C=C, sr=sr)
Or with a different hop length and frequency resolution:
>>> hop_length = 256
>>> bins_per_octave = 12 * 3
>>> C = librosa.cqt(y=y, sr=sr, hop_length=256, n_bins=7*bins_per_octave,
... bins_per_octave=bins_per_octave)
>>> y_hat = librosa.icqt(C=C, sr=sr, hop_length=hop_length,
... bins_per_octave=bins_per_octave)
'''
warnings.warn(
'librosa.icqt is unstable, and subject to change in future versions. '
'Please use with caution.')
n_bins, n_frames = C.shape
n_octaves = int(np.ceil(float(n_bins) / bins_per_octave))
if amin is None:
amin = util.tiny(C)
if fmin is None:
fmin = note_to_hz('C1')
freqs = cqt_frequencies(n_bins,
fmin,
bins_per_octave=bins_per_octave,
tuning=tuning)[-bins_per_octave:]
fmin_t = np.min(freqs)
# Make the filter bank
basis, lengths = filters.constant_q(sr=sr,
fmin=fmin_t,
n_bins=bins_per_octave,
bins_per_octave=bins_per_octave,
filter_scale=filter_scale,
tuning=tuning,
norm=norm,
window=window,
pad_fft=True)
n_fft = basis.shape[1]
# The extra factor of lengths**0.5 corrects for within-octave tapering
basis = basis * np.sqrt(lengths[:, np.newaxis])
# Estimate the gain per filter
bdot = basis.conj().dot(basis.T)
bscale = np.sum(np.abs(bdot), axis=1)
n_trim = basis.shape[1] // 2
if scale:
Cnorm = np.ones(n_bins)[:, np.newaxis]
else:
Cnorm = filters.constant_q_lengths(sr=sr,
fmin=fmin,
n_bins=n_bins,
bins_per_octave=bins_per_octave,
filter_scale=filter_scale,
tuning=tuning,
window=window)[:, np.newaxis]**0.5
y = None
# Revised algorithm:
# for each octave
# upsample old octave
# @--numba accelerate this loop?
# for each basis
# convolve with activation (valid-mode)
# divide by window sumsquare
# trim and add to total
for octave in range(n_octaves - 1, -1, -1):
# Compute the slice index for the current octave
slice_ = slice(-(octave + 1) * bins_per_octave - 1,
-(octave) * bins_per_octave - 1)
# Project onto the basis
C_oct = C[slice_] / Cnorm[slice_]
basis_oct = basis[-C_oct.shape[0]:]
y_oct = None
# Make a dummy activation
oct_hop = hop_length // 2**octave
n = n_fft + (C_oct.shape[1] - 1) * oct_hop
for i in range(basis_oct.shape[0] - 1, -1, -1):
wss = filters.window_sumsquare(window,
n_frames,
hop_length=oct_hop,
win_length=int(lengths[i]),
n_fft=n_fft,
norm=norm)
wss *= lengths[i]**2
# Construct the response for this filter
y_oct_i = np.zeros(n, dtype=C_oct.dtype)
__activation_fill(y_oct_i, basis_oct[i], C_oct[i], oct_hop)
# Retain only the real part
# Only do window normalization for sufficiently large window
# coefficients
y_oct_i = y_oct_i.real / np.maximum(amin, wss)
if y_oct is None:
y_oct = y_oct_i
else:
y_oct += y_oct_i
# Remove the effects of zero-padding
y_oct = y_oct[n_trim:-n_trim] * bscale[i]
if y is None:
y = y_oct
else:
# Up-sample the previous buffer and add in the new one
# Scipy-resampling is fast here, since it's a power-of-two relation
y = audio.resample(y, 1, 2, scale=True, res_type='scipy') + y_oct
return y
def cqt(y,
sr=22050,
hop_length=512,
fmin=None,
n_bins=84,
bins_per_octave=12,
tuning=0.0,
filter_scale=1,
norm=1,
sparsity=0.01,
window='hann',
scale=True,
pad_mode='reflect',
res_type='scipy'):
'''Compute the constant-Q transform of an audio signal.
This implementation is based on the recursive sub-sampling method
described by [1]_.
.. [1] Schoerkhuber, Christian, and Anssi Klapuri.
"Constant-Q transform toolbox for music processing."
7th Sound and Music Computing Conference, Barcelona, Spain. 2010.
Parameters
----------
y : np.ndarray [shape=(n,)]
audio time series
sr : number > 0 [scalar]
sampling rate of `y`
hop_length : int > 0 [scalar]
number of samples between successive CQT columns.
fmin : float > 0 [scalar]
Minimum frequency. Defaults to C1 ~= 32.70 Hz
n_bins : int > 0 [scalar]
Number of frequency bins, starting at `fmin`
bins_per_octave : int > 0 [scalar]
Number of bins per octave
tuning : None or float in `[-0.5, 0.5)`
Tuning offset in fractions of a bin (cents).
If `None`, tuning will be automatically estimated from the signal.
filter_scale : float > 0
Filter scale factor. Small values (<1) use shorter windows
for improved time resolution.
norm : {inf, -inf, 0, float > 0}
Type of norm to use for basis function normalization.
See `librosa.util.normalize`.
sparsity : float in [0, 1)
Sparsify the CQT basis by discarding up to `sparsity`
fraction of the energy in each basis.
Set `sparsity=0` to disable sparsification.
window : str, tuple, number, or function
Window specification for the basis filters.
See `filters.get_window` for details.
scale : bool
If `True`, scale the CQT response by square-root the length of
each channel's filter. This is analogous to `norm='ortho'` in FFT.
If `False`, do not scale the CQT. This is analogous to
`norm=None` in FFT.
pad_mode : string
Padding mode for centered frame analysis.
See also: `librosa.core.stft` and `np.pad`.
Returns
-------
CQT : np.ndarray [shape=(n_bins, t), dtype=np.complex or np.float]
Constant-Q value each frequency at each time.
Raises
------
ParameterError
If `hop_length` is not an integer multiple of
`2**(n_bins / bins_per_octave)`
Or if `y` is too short to support the frequency range of the CQT.
See Also
--------
librosa.core.resample
librosa.util.normalize
Notes
-----
This function caches at level 20.
Examples
--------
Generate and plot a constant-Q power spectrum
>>> import matplotlib.pyplot as plt
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> C = np.abs(librosa.cqt(y, sr=sr))
>>> librosa.display.specshow(librosa.amplitude_to_db(C, ref=np.max),
... sr=sr, x_axis='time', y_axis='cqt_note')
>>> plt.colorbar(format='%+2.0f dB')
>>> plt.title('Constant-Q power spectrum')
>>> plt.tight_layout()
Limit the frequency range
>>> C = np.abs(librosa.cqt(y, sr=sr, fmin=librosa.note_to_hz('C2'),
... n_bins=60))
>>> C
array([[ 8.827e-04, 9.293e-04, ..., 3.133e-07, 2.942e-07],
[ 1.076e-03, 1.068e-03, ..., 1.153e-06, 1.148e-06],
...,
[ 1.042e-07, 4.087e-07, ..., 1.612e-07, 1.928e-07],
[ 2.363e-07, 5.329e-07, ..., 1.294e-07, 1.611e-07]])
Using a higher frequency resolution
>>> C = np.abs(librosa.cqt(y, sr=sr, fmin=librosa.note_to_hz('C2'),
... n_bins=60 * 2, bins_per_octave=12 * 2))
>>> C
array([[ 1.536e-05, 5.848e-05, ..., 3.241e-07, 2.453e-07],
[ 1.856e-03, 1.854e-03, ..., 2.397e-08, 3.549e-08],
...,
[ 2.034e-07, 4.245e-07, ..., 6.213e-08, 1.463e-07],
[ 4.896e-08, 5.407e-07, ..., 9.176e-08, 1.051e-07]])
'''
# How many octaves are we dealing with?
n_octaves = int(np.ceil(float(n_bins) / bins_per_octave))
n_filters = min(bins_per_octave, n_bins)
len_orig = len(y)
if fmin is None:
# C1 by default
fmin = note_to_hz('C1')
if tuning is None:
tuning = estimate_tuning(y=y, sr=sr)
# First thing, get the freqs of the top octave
freqs = cqt_frequencies(n_bins, fmin,
bins_per_octave=bins_per_octave)[-bins_per_octave:]
fmin_t = np.min(freqs)
fmax_t = np.max(freqs)
# Determine required resampling quality
Q = float(filter_scale) / (2.0**(1. / bins_per_octave) - 1)
filter_cutoff = fmax_t * (1 + 0.5 * filters.window_bandwidth(window) / Q)
nyquist = sr / 2.0
y, sr, hop_length = __early_downsample(y, sr, hop_length, res_type,
n_octaves, nyquist, filter_cutoff,
scale)
cqt_resp = []
if res_type != 'kaiser_fast':
# Do the top octave before resampling to allow for fast resampling
fft_basis, n_fft, _ = __cqt_filter_fft(sr,
fmin_t,
n_filters,
bins_per_octave,
tuning,
filter_scale,
norm,
sparsity,
window=window)
# Compute the CQT filter response and append it to the stack
cqt_resp.append(
__cqt_response(y, n_fft, hop_length, fft_basis, pad_mode))
fmin_t /= 2
fmax_t /= 2
n_octaves -= 1
filter_cutoff = fmax_t * (1 +
0.5 * filters.window_bandwidth(window) / Q)
res_type = 'kaiser_fast'
# Make sure our hop is long enough to support the bottom octave
num_twos = __num_two_factors(hop_length)
if num_twos < n_octaves - 1:
raise ParameterError('hop_length must be a positive integer '
'multiple of 2^{0:d} for {1:d}-octave CQT'.format(
n_octaves - 1, n_octaves))
# Now do the recursive bit
fft_basis, n_fft, _ = __cqt_filter_fft(sr,
fmin_t,
n_filters,
bins_per_octave,
tuning,
filter_scale,
norm,
sparsity,
window=window)
my_y, my_sr, my_hop = y, sr, hop_length
# Iterate down the octaves
for i in range(n_octaves):
# Resample (except first time)
if i > 0:
if len(my_y) < 2:
raise ParameterError('Input signal length={} is too short for '
'{:d}-octave CQT'.format(
len_orig, n_octaves))
my_y = audio.resample(my_y,
my_sr,
my_sr / 2.0,
res_type=res_type,
scale=True)
# The re-scale the filters to compensate for downsampling
fft_basis[:] *= np.sqrt(2)
my_sr /= 2.0
my_hop //= 2
# Compute the cqt filter response and append to the stack
cqt_resp.append(
__cqt_response(my_y, n_fft, my_hop, fft_basis, pad_mode))
C = __trim_stack(cqt_resp, n_bins)
if scale:
lengths = filters.constant_q_lengths(sr,
fmin,
n_bins=n_bins,
bins_per_octave=bins_per_octave,
tuning=tuning,
window=window,
filter_scale=filter_scale)
C /= np.sqrt(lengths[:, np.newaxis])
return C
