#!/usr/bin/env python
# -*- coding: utf-8 -*-
'''Utilities for spectral processing'''
import numpy as np
import scipy.fftpack as fft
import scipy
import scipy.signal
import scipy.interpolate
import six
from . import time_frequency
from .. import cache
from .. import util
from ..util.decorators import moved
from ..util.deprecation import rename_kw, Deprecated
from ..util.exceptions import ParameterError
from ..filters import get_window
__all__ = ['stft', 'istft', 'magphase',
'ifgram', 'phase_vocoder',
'logamplitude', 'perceptual_weighting',
'power_to_db', 'db_to_power',
'amplitude_to_db', 'db_to_amplitude',
'fmt']
@cache(level=20)
def stft(y, n_fft=2048, hop_length=None, win_length=None, window='hann',
center=True, dtype=np.complex64, pad_mode='reflect'):
"""Short-time Fourier transform (STFT)
Returns a complex-valued matrix D such that
`np.abs(D[f, t])` is the magnitude of frequency bin `f`
at frame `t`
`np.angle(D[f, t])` is the phase of frequency bin `f`
at frame `t`
Parameters
----------
y : np.ndarray [shape=(n,)], real-valued
the input signal (audio time series)
n_fft : int > 0 [scalar]
FFT window size
#做n次FFT
hop_length : int > 0 [scalar]
number audio of frames between STFT columns.
If unspecified, defaults `win_length / 4`.
#也就是帧之间的overlap，默认为窗口长度的1/4，这点是我理解错误的
win_length : int <= n_fft [scalar]
Each frame of audio is windowed by `window()`.
The window will be of length `win_length` and then padded
with zeros to match `n_fft`.
If unspecified, defaults to ``win_length = n_fft``.
#窗口大小要小于等于N_FFT，默认N_FFT
window : string, tuple, number, function, or np.ndarray [shape=(n_fft,)]
- a window specification (string, tuple, or number);
see `scipy.signal.get_window`
- a window function, such as `scipy.signal.hanning`
- a vector or array of length `n_fft`
.. see also:: `filters.get_window`
center : boolean
- If `True`, the signal `y` is padded so that frame
`D[:, t]` is centered at `y[t * hop_length]`.
- If `False`, then `D[:, t]` begins at `y[t * hop_length]`
True是以窗的中心时刻作为时间标示，False以窗的开始位置作为时间标示
dtype : numeric type
Complex numeric type for `D`. Default is 64-bit complex.
#做fft得到64位的复数，有幅值（实部）和相位（虚部），个人纠结是使用实部还是都使用？很多应用说只使用了实部，相位可以忽略不计，but。。。
mode : string
If `center=True`, the padding mode to use at the edges of the signal.
By default, STFT uses reflection padding.
Returns
-------
D : np.ndarray [shape=(1 + n_fft/2, t), dtype=dtype]
STFT matrix
#需要进一步对FFT进行理解，只取信号FFT的一半进行分析，是因为实数信号的FFT具有对称性，
完整的FFT结果具有信息冗余。
See Also
--------
istft : Inverse STFT
ifgram : Instantaneous frequency spectrogram
np.pad : array padding
Notes
-----
This function caches at level 20.
Examples
--------
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> D = librosa.stft(y)
>>> D
array([[ 2.576e-03 -0.000e+00j, 4.327e-02 -0.000e+00j, ...,
3.189e-04 -0.000e+00j, -5.961e-06 -0.000e+00j],
[ 2.441e-03 +2.884e-19j, 5.145e-02 -5.076e-03j, ...,
-3.885e-04 -7.253e-05j, 7.334e-05 +3.868e-04j],
...,
[ -7.120e-06 -1.029e-19j, -1.951e-09 -3.568e-06j, ...,
-4.912e-07 -1.487e-07j, 4.438e-06 -1.448e-05j],
[ 7.136e-06 -0.000e+00j, 3.561e-06 -0.000e+00j, ...,
-5.144e-07 -0.000e+00j, -1.514e-05 -0.000e+00j]], dtype=complex64)
Use left-aligned frames, instead of centered frames
>>> D_left = librosa.stft(y, center=False)
Use a shorter hop length
>>> D_short = librosa.stft(y, hop_length=64)
Display a spectrogram
>>> import matplotlib.pyplot as plt
>>> librosa.display.specshow(librosa.amplitude_to_db(D,
... ref=np.max),
... y_axis='log', x_axis='time')
>>> plt.title('Power spectrogram')
>>> plt.colorbar(format='%+2.0f dB')
>>> plt.tight_layout()
"""
# By default, use the entire frame
if win_length is None:
win_length = n_fft
# Set the default hop, if it's not already specified
if hop_length is None:
hop_length = int(win_length // 4)
fft_window = get_window(window, win_length, fftbins=True)
# Pad the window out to n_fft size
fft_window = util.pad_center(fft_window, n_fft)
# Reshape so that the window can be broadcast
fft_window = fft_window.reshape((-1, 1))
# Pad the time series so that frames are centered
if center:
util.valid_audio(y)
y = np.pad(y, int(n_fft // 2), mode=pad_mode)
# Window the time series.
y_frames = util.frame(y, frame_length=n_fft, hop_length=hop_length)
# Pre-allocate the STFT matrix
stft_matrix = np.empty((int(1 + n_fft // 2), y_frames.shape[1]),
dtype=dtype,
order='F')
# how many columns can we fit within MAX_MEM_BLOCK?
n_columns = int(util.MAX_MEM_BLOCK / (stft_matrix.shape[0] *
stft_matrix.itemsize))
for bl_s in range(0, stft_matrix.shape[1], n_columns):
bl_t = min(bl_s + n_columns, stft_matrix.shape[1])
# RFFT and Conjugate here to match phase from DPWE code
stft_matrix[:, bl_s:bl_t] = fft.fft(fft_window *
y_frames[:, bl_s:bl_t],
axis=0)[:stft_matrix.shape[0]].conj()
return stft_matrix
@cache(level=30)
def istft(stft_matrix, hop_length=None, win_length=None, window='hann',
center=True, dtype=np.float32, length=None):
"""
Inverse short-time Fourier transform (ISTFT).
Converts a complex-valued spectrogram `stft_matrix` to time-series `y`
by minimizing the mean squared error between `stft_matrix` and STFT of
`y` as described in [1]_.
In general, window function, hop length and other parameters should be same
as in stft, which mostly leads to perfect reconstruction of a signal from
unmodified `stft_matrix`.
.. [1] D. W. Griffin and J. S. Lim,
"Signal estimation from modified short-time Fourier transform,"
IEEE Trans. ASSP, vol.32, no.2, pp.236–243, Apr. 1984.
Parameters
----------
stft_matrix : np.ndarray [shape=(1 + n_fft/2, t)]
STFT matrix from `stft`
hop_length : int > 0 [scalar]
Number of frames between STFT columns.
If unspecified, defaults to `win_length / 4`.
win_length : int <= n_fft = 2 * (stft_matrix.shape[0] - 1)
When reconstructing the time series, each frame is windowed
and each sample is normalized by the sum of squared window
according to the `window` function (see below).
If unspecified, defaults to `n_fft`.
window : string, tuple, number, function, np.ndarray [shape=(n_fft,)]
- a window specification (string, tuple, or number);
see `scipy.signal.get_window`
- a window function, such as `scipy.signal.hanning`
- a user-specified window vector of length `n_fft`
.. see also:: `filters.get_window`
center : boolean
- If `True`, `D` is assumed to have centered frames.
- If `False`, `D` is assumed to have left-aligned frames.
dtype : numeric type
Real numeric type for `y`. Default is 32-bit float.
length : int > 0, optional
If provided, the output `y` is zero-padded or clipped to exactly
`length` samples.
Returns
-------
y : np.ndarray [shape=(n,)]
time domain signal reconstructed from `stft_matrix`
See Also
--------
stft : Short-time Fourier Transform
Notes
-----
This function caches at level 30.
Examples
--------
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> D = librosa.stft(y)
>>> y_hat = librosa.istft(D)
>>> y_hat
array([ -4.812e-06, -4.267e-06, ..., 6.271e-06, 2.827e-07], dtype=float32)
Exactly preserving length of the input signal requires explicit padding.
Otherwise, a partial frame at the end of `y` will not be represented.
>>> n = len(y)
>>> n_fft = 2048
>>> y_pad = librosa.util.fix_length(y, n + n_fft // 2)
>>> D = librosa.stft(y_pad, n_fft=n_fft)
>>> y_out = librosa.istft(D, length=n)
>>> np.max(np.abs(y - y_out))
1.4901161e-07
"""
n_fft = 2 * (stft_matrix.shape[0] - 1)
# By default, use the entire frame
if win_length is None:
win_length = n_fft
# Set the default hop, if it's not already specified
if hop_length is None:
hop_length = int(win_length // 4)
ifft_window = get_window(window, win_length, fftbins=True)
# Pad out to match n_fft
ifft_window = util.pad_center(ifft_window, n_fft)
n_frames = stft_matrix.shape[1]
expected_signal_len = n_fft + hop_length * (n_frames - 1)
y = np.zeros(expected_signal_len, dtype=dtype)
ifft_window_sum = np.zeros(expected_signal_len, dtype=dtype)
ifft_window_square = ifft_window * ifft_window
for i in range(n_frames):
sample = i * hop_length
spec = stft_matrix[:, i].flatten()
spec = np.concatenate((spec.conj(), spec[-2:0:-1]), 0)
ytmp = ifft_window * fft.ifft(spec).real
y[sample:(sample + n_fft)] = y[sample:(sample + n_fft)] + ytmp
ifft_window_sum[sample:(sample + n_fft)] += ifft_window_square
# Normalize by sum of squared window
approx_nonzero_indices = ifft_window_sum > util.tiny(ifft_window_sum)
y[approx_nonzero_indices] /= ifft_window_sum[approx_nonzero_indices]
if length is None:
# If we don't need to control length, just do the usual center trimming
# to eliminate padded data
if center:
y = y[int(n_fft // 2):-int(n_fft // 2)]
else:
if center:
# If we're centering, crop off the first n_fft//2 samples
# and then trim/pad to the target length.
# We don't trim the end here, so that if the signal is zero-padded
# to a longer duration, the decay is smooth by windowing
start = int(n_fft // 2)
else:
# If we're not centering, start at 0 and trim/pad as necessary
start = 0
y = util.fix_length(y[start:], length)
return y
def ifgram(y, sr=22050, n_fft=2048, hop_length=None, win_length=None,
window='hann', norm=False, center=True, ref_power=1e-6,
clip=True, dtype=np.complex64, pad_mode='reflect'):
'''Compute the instantaneous frequency (as a proportion of the sampling rate)
obtained as the time-derivative of the phase of the complex spectrum as
described by [1]_.
Calculates regular STFT as a side effect.
.. [1] Abe, Toshihiko, Takao Kobayashi, and Satoshi Imai.
"Harmonics tracking and pitch extraction based on instantaneous
frequency."
International Conference on Acoustics, Speech, and Signal Processing,
ICASSP-95., Vol. 1. IEEE, 1995.
Parameters
----------
y : np.ndarray [shape=(n,)]
audio time series
sr : number > 0 [scalar]
sampling rate of `y`
n_fft : int > 0 [scalar]
FFT window size
hop_length : int > 0 [scalar]
hop length, number samples between subsequent frames.
If not supplied, defaults to `win_length / 4`.
win_length : int > 0, <= n_fft
Window length. Defaults to `n_fft`.
See `stft` for details.
window : string, tuple, number, function, or np.ndarray [shape=(n_fft,)]
- a window specification (string, tuple, number);
see `scipy.signal.get_window`
- a window function, such as `scipy.signal.hanning`
- a user-specified window vector of length `n_fft`
See `stft` for details.
.. see also:: `filters.get_window`
norm : bool
Normalize the STFT.
center : boolean
- If `True`, the signal `y` is padded so that frame
`D[:, t]` (and `if_gram`) is centered at `y[t * hop_length]`.
- If `False`, then `D[:, t]` at `y[t * hop_length]`
ref_power : float >= 0 or callable
Minimum power threshold for estimating instantaneous frequency.
Any bin with `np.abs(D[f, t])**2 < ref_power` will receive the
default frequency estimate.
If callable, the threshold is set to `ref_power(np.abs(D)**2)`.
clip : boolean
- If `True`, clip estimated frequencies to the range `[0, 0.5 * sr]`.
- If `False`, estimated frequencies can be negative or exceed
`0.5 * sr`.
dtype : numeric type
Complex numeric type for `D`. Default is 64-bit complex.
mode : string
If `center=True`, the padding mode to use at the edges of the signal.
By default, STFT uses reflection padding.
Returns
-------
if_gram : np.ndarray [shape=(1 + n_fft/2, t), dtype=real]
Instantaneous frequency spectrogram:
`if_gram[f, t]` is the frequency at bin `f`, time `t`
D : np.ndarray [shape=(1 + n_fft/2, t), dtype=complex]
Short-time Fourier transform
See Also
--------
stft : Short-time Fourier Transform
Examples
--------
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> frequencies, D = librosa.ifgram(y, sr=sr)
>>> frequencies
array([[ 0.000e+00, 0.000e+00, ..., 0.000e+00, 0.000e+00],
[ 3.150e+01, 3.070e+01, ..., 1.077e+01, 1.077e+01],
...,
[ 1.101e+04, 1.101e+04, ..., 1.101e+04, 1.101e+04],
[ 1.102e+04, 1.102e+04, ..., 1.102e+04, 1.102e+04]])
'''
if win_length is None:
win_length = n_fft
if hop_length is None:
hop_length = int(win_length // 4)
# Construct a padded hann window
fft_window = util.pad_center(get_window(window, win_length,
fftbins=True),
n_fft)
# Window for discrete differentiation
freq_angular = np.linspace(0, 2 * np.pi, n_fft, endpoint=False)
d_window = np.sin(-freq_angular) * np.pi / n_fft
stft_matrix = stft(y, n_fft=n_fft, hop_length=hop_length,
win_length=win_length,
window=window, center=center,
dtype=dtype, pad_mode=pad_mode)
diff_stft = stft(y, n_fft=n_fft, hop_length=hop_length,
window=d_window, center=center,
dtype=dtype, pad_mode=pad_mode).conj()
# Compute power normalization. Suppress zeros.
mag, phase = magphase(stft_matrix)
if six.callable(ref_power):
ref_power = ref_power(mag**2)
elif ref_power < 0:
raise ParameterError('ref_power must be non-negative or callable.')
# Pylint does not correctly infer the type here, but it's correct.
# pylint: disable=maybe-no-member
freq_angular = freq_angular.reshape((-1, 1))
bin_offset = (phase * diff_stft).imag / mag
bin_offset[mag < ref_power**0.5] = 0
if_gram = freq_angular[:n_fft//2 + 1] + bin_offset
if norm:
stft_matrix = stft_matrix * 2.0 / fft_window.sum()
if clip:
np.clip(if_gram, 0, np.pi, out=if_gram)
if_gram *= float(sr) * 0.5 / np.pi
return if_gram, stft_matrix
def magphase(D):
"""Separate a complex-valued spectrogram D into its magnitude (S)
and phase (P) components, so that `D = S * P`.
Parameters
----------
D : np.ndarray [shape=(d, t), dtype=complex]
complex-valued spectrogram
Returns
-------
D_mag : np.ndarray [shape=(d, t), dtype=real]
magnitude of `D`
D_phase : np.ndarray [shape=(d, t), dtype=complex]
`exp(1.j * phi)` where `phi` is the phase of `D`
Examples
--------
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> D = librosa.stft(y)
>>> magnitude, phase = librosa.magphase(D)
>>> magnitude
array([[ 2.524e-03, 4.329e-02, ..., 3.217e-04, 3.520e-05],
[ 2.645e-03, 5.152e-02, ..., 3.283e-04, 3.432e-04],
...,
[ 1.966e-05, 9.828e-06, ..., 3.164e-07, 9.370e-06],
[ 1.966e-05, 9.830e-06, ..., 3.161e-07, 9.366e-06]], dtype=float32)
>>> phase
array([[ 1.000e+00 +0.000e+00j, 1.000e+00 +0.000e+00j, ...,
-1.000e+00 +8.742e-08j, -1.000e+00 +8.742e-08j],
[ 1.000e+00 +1.615e-16j, 9.950e-01 -1.001e-01j, ...,
9.794e-01 +2.017e-01j, 1.492e-02 -9.999e-01j],
...,
[ 1.000e+00 -5.609e-15j, -5.081e-04 +1.000e+00j, ...,
-9.549e-01 -2.970e-01j, 2.938e-01 -9.559e-01j],
[ -1.000e+00 +8.742e-08j, -1.000e+00 +8.742e-08j, ...,
-1.000e+00 +8.742e-08j, -1.000e+00 +8.742e-08j]], dtype=complex64)
Or get the phase angle (in radians)
>>> np.angle(phase)
array([[ 0.000e+00, 0.000e+00, ..., 3.142e+00, 3.142e+00],
[ 1.615e-16, -1.003e-01, ..., 2.031e-01, -1.556e+00],
...,
[ -5.609e-15, 1.571e+00, ..., -2.840e+00, -1.273e+00],
[ 3.142e+00, 3.142e+00, ..., 3.142e+00, 3.142e+00]], dtype=float32)
"""
mag = np.abs(D)
phase = np.exp(1.j * np.angle(D))
return mag, phase
def phase_vocoder(D, rate, hop_length=None):
"""Phase vocoder. Given an STFT matrix D, speed up by a factor of `rate`
Based on the implementation provided by [1]_.
.. [1] Ellis, D. P. W. "A phase vocoder in Matlab."
Columbia University, 2002.
http://www.ee.columbia.edu/~dpwe/resources/matlab/pvoc/
Examples
--------
>>> # Play at double speed
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> D = librosa.stft(y, n_fft=2048, hop_length=512)
>>> D_fast = librosa.phase_vocoder(D, 2.0, hop_length=512)
>>> y_fast = librosa.istft(D_fast, hop_length=512)
>>> # Or play at 1/3 speed
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> D = librosa.stft(y, n_fft=2048, hop_length=512)
>>> D_slow = librosa.phase_vocoder(D, 1./3, hop_length=512)
>>> y_slow = librosa.istft(D_slow, hop_length=512)
Parameters
----------
D : np.ndarray [shape=(d, t), dtype=complex]
STFT matrix
rate : float > 0 [scalar]
Speed-up factor: `rate > 1` is faster, `rate < 1` is slower.
hop_length : int > 0 [scalar] or None
The number of samples between successive columns of `D`.
If None, defaults to `n_fft/4 = (D.shape[0]-1)/2`
Returns
-------
D_stretched : np.ndarray [shape=(d, t / rate), dtype=complex]
time-stretched STFT
"""
n_fft = 2 * (D.shape[0] - 1)
if hop_length is None:
hop_length = int(n_fft // 4)
time_steps = np.arange(0, D.shape[1], rate, dtype=np.float)
# Create an empty output array
d_stretch = np.zeros((D.shape[0], len(time_steps)), D.dtype, order='F')
# Expected phase advance in each bin
phi_advance = np.linspace(0, np.pi * hop_length, D.shape[0])
# Phase accumulator; initialize to the first sample
phase_acc = np.angle(D[:, 0])
# Pad 0 columns to simplify boundary logic
D = np.pad(D, [(0, 0), (0, 2)], mode='constant')
for (t, step) in enumerate(time_steps):
columns = D[:, int(step):int(step + 2)]
# Weighting for linear magnitude interpolation
alpha = np.mod(step, 1.0)
mag = ((1.0 - alpha) * np.abs(columns[:, 0])
+ alpha * np.abs(columns[:, 1]))
# Store to output array
d_stretch[:, t] = mag * np.exp(1.j * phase_acc)
# Compute phase advance
dphase = (np.angle(columns[:, 1])
- np.angle(columns[:, 0])
- phi_advance)
# Wrap to -pi:pi range
dphase = dphase - 2.0 * np.pi * np.round(dphase / (2.0 * np.pi))
# Accumulate phase
phase_acc += phi_advance + dphase
return d_stretch
@cache(level=30)
def power_to_db(S, ref=1.0, amin=1e-10, top_db=80.0, ref_power=Deprecated()):
"""Convert a power spectrogram (amplitude squared) to decibel (dB) units
This computes the scaling ``10 * log10(S / ref)`` in a numerically
stable way.
Parameters
----------
S : np.ndarray
input power
ref : scalar or callable
If scalar, the amplitude `abs(S)` is scaled relative to `ref`:
`10 * log10(S / ref)`.
Zeros in the output correspond to positions where `S == ref`.
If callable, the reference value is computed as `ref(S)`.
amin : float > 0 [scalar]
minimum threshold for `abs(S)` and `ref`
top_db : float >= 0 [scalar]
threshold the output at `top_db` below the peak:
``max(10 * log10(S)) - top_db``
ref_power : scalar or callable
.. warning:: This parameter name was deprecated in librosa 0.5.0.
Use the `ref` parameter instead.
The `ref_power` parameter will be removed in librosa 0.6.0.
Returns
-------
S_db : np.ndarray
``S_db ~= 10 * log10(S) - 10 * log10(ref)``
See Also
--------
perceptual_weighting
db_to_power
amplitude_to_db
db_to_amplitude
Notes
-----
This function caches at level 30.
Examples
--------
Get a power spectrogram from a waveform ``y``
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> S = np.abs(librosa.stft(y))
>>> librosa.power_to_db(S**2)
array([[-33.293, -27.32 , ..., -33.293, -33.293],
[-33.293, -25.723, ..., -33.293, -33.293],
...,
[-33.293, -33.293, ..., -33.293, -33.293],
[-33.293, -33.293, ..., -33.293, -33.293]], dtype=float32)
Compute dB relative to peak power
>>> librosa.power_to_db(S**2, ref=np.max)
array([[-80. , -74.027, ..., -80. , -80. ],
[-80. , -72.431, ..., -80. , -80. ],
...,
[-80. , -80. , ..., -80. , -80. ],
[-80. , -80. , ..., -80. , -80. ]], dtype=float32)
Or compare to median power
>>> librosa.power_to_db(S**2, ref=np.median)
array([[-0.189, 5.784, ..., -0.189, -0.189],
[-0.189, 7.381, ..., -0.189, -0.189],
...,
[-0.189, -0.189, ..., -0.189, -0.189],
[-0.189, -0.189, ..., -0.189, -0.189]], dtype=float32)
And plot the results
>>> import matplotlib.pyplot as plt
>>> plt.figure()
>>> plt.subplot(2, 1, 1)
>>> librosa.display.specshow(S**2, sr=sr, y_axis='log')
>>> plt.colorbar()
>>> plt.title('Power spectrogram')
>>> plt.subplot(2, 1, 2)
>>> librosa.display.specshow(librosa.power_to_db(S**2, ref=np.max),
... sr=sr, y_axis='log', x_axis='time')
>>> plt.colorbar(format='%+2.0f dB')
>>> plt.title('Log-Power spectrogram')
>>> plt.tight_layout()
"""
if amin <= 0:
raise ParameterError('amin must be strictly positive')
magnitude = np.abs(S)
ref = rename_kw('ref_power', ref_power,
'ref', ref,
'0.5', '0.6')
if six.callable(ref):
# User supplied a function to calculate reference power
ref_value = ref(magnitude)
else:
ref_value = np.abs(ref)
log_spec = 10.0 * np.log10(np.maximum(amin, magnitude))
log_spec -= 10.0 * np.log10(np.maximum(amin, ref_value))
if top_db is not None:
if top_db < 0:
raise ParameterError('top_db must be non-negative')
log_spec = np.maximum(log_spec, log_spec.max() - top_db)
return log_spec
logamplitude = moved('librosa.logamplitude', '0.5', '0.6')(power_to_db)
@cache(level=30)
def db_to_power(S_db, ref=1.0):
'''Convert a dB-scale spectrogram to a power spectrogram.
This effectively inverts `power_to_db`:
`db_to_power(S_db) ~= ref * 10.0**(S_db / 10)`
Parameters
----------
S_db : np.ndarray
dB-scaled spectrogram
ref : number > 0
Reference power: output will be scaled by this value
Returns
-------
S : np.ndarray
Power spectrogram
Notes
-----
This function caches at level 30.
'''
return ref * np.power(10.0, 0.1 * S_db)
@cache(level=30)
def amplitude_to_db(S, ref=1.0, amin=1e-5, top_db=80.0):
'''Convert an amplitude spectrogram to dB-scaled spectrogram.
This is equivalent to ``power_to_db(S**2)``, but is provided for convenience.
Parameters
----------
S : np.ndarray
input amplitude
ref : scalar or callable
If scalar, the amplitude `abs(S)` is scaled relative to `ref`:
`20 * log10(S / ref)`.
Zeros in the output correspond to positions where `S == ref`.
If callable, the reference value is computed as `ref(S)`.
amin : float > 0 [scalar]
minimum threshold for `S` and `ref`
top_db : float >= 0 [scalar]
threshold the output at `top_db` below the peak:
``max(20 * log10(S)) - top_db``
Returns
-------
S_db : np.ndarray
``S`` measured in dB
See Also
--------
logamplitude, power_to_db, db_to_amplitude
Notes
-----
This function caches at level 30.
'''
magnitude = np.abs(S)
if six.callable(ref):
# User supplied a function to calculate reference power
ref_value = ref(magnitude)
else:
ref_value = np.abs(ref)
magnitude **= 2
return power_to_db(magnitude, ref=ref_value**2, amin=amin**2,
top_db=top_db)
@cache(level=30)
def db_to_amplitude(S_db, ref=1.0):
'''Convert a dB-scaled spectrogram to an amplitude spectrogram.
This effectively inverts `amplitude_to_db`:
`db_to_amplitude(S_db) ~= 10.0**(0.5 * (S_db + log10(ref)/10))`
Parameters
----------
S_db : np.ndarray
dB-scaled spectrogram
ref: number > 0
Optional reference power.
Returns
-------
S : np.ndarray
Linear magnitude spectrogram
Notes
-----
This function caches at level 30.
'''
return db_to_power(S_db, ref=ref**2)**0.5
@cache(level=30)
def perceptual_weighting(S, frequencies, **kwargs):
'''Perceptual weighting of a power spectrogram:
`S_p[f] = A_weighting(f) + 10*log(S[f] / ref)`
Parameters
----------
S : np.ndarray [shape=(d, t)]
Power spectrogram
frequencies : np.ndarray [shape=(d,)]
Center frequency for each row of `S`
kwargs : additional keyword arguments
Additional keyword arguments to `logamplitude`.
Returns
-------
S_p : np.ndarray [shape=(d, t)]
perceptually weighted version of `S`
See Also
--------
logamplitude
Notes
-----
This function caches at level 30.
Examples
--------
Re-weight a CQT power spectrum, using peak power as reference
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> CQT = librosa.cqt(y, sr=sr, fmin=librosa.note_to_hz('A1'))
>>> freqs = librosa.cqt_frequencies(CQT.shape[0],
... fmin=librosa.note_to_hz('A1'))
>>> perceptual_CQT = librosa.perceptual_weighting(CQT**2,
... freqs,
... ref=np.max)
>>> perceptual_CQT
array([[ -80.076, -80.049, ..., -104.735, -104.735],
[ -78.344, -78.555, ..., -103.725, -103.725],
...,
[ -76.272, -76.272, ..., -76.272, -76.272],
[ -76.485, -76.485, ..., -76.485, -76.485]])
>>> import matplotlib.pyplot as plt
>>> plt.figure()
>>> plt.subplot(2, 1, 1)
>>> librosa.display.specshow(librosa.amplitude_to_db(CQT,
... ref=np.max),
... fmin=librosa.note_to_hz('A1'),
... y_axis='cqt_hz')
>>> plt.title('Log CQT power')
>>> plt.colorbar(format='%+2.0f dB')
>>> plt.subplot(2, 1, 2)
>>> librosa.display.specshow(perceptual_CQT, y_axis='cqt_hz',
... fmin=librosa.note_to_hz('A1'),
... x_axis='time')
>>> plt.title('Perceptually weighted log CQT')
>>> plt.colorbar(format='%+2.0f dB')
>>> plt.tight_layout()
'''
offset = time_frequency.A_weighting(frequencies).reshape((-1, 1))
return offset + logamplitude(S, **kwargs)
@cache(level=30)
def fmt(y, t_min=0.5, n_fmt=None, kind='cubic', beta=0.5, over_sample=1, axis=-1):
"""The fast Mellin transform (FMT) [1]_ of a uniformly sampled signal y.
When the Mellin parameter (beta) is 1/2, it is also known as the scale transform [2]_.
The scale transform can be useful for audio analysis because its magnitude is invariant
to scaling of the domain (e.g., time stretching or compression). This is analogous
to the magnitude of the Fourier transform being invariant to shifts in the input domain.
.. [1] De Sena, Antonio, and Davide Rocchesso.
"A fast Mellin and scale transform."
EURASIP Journal on Applied Signal Processing 2007.1 (2007): 75-75.
.. [2] Cohen, L.
"The scale representation."
IEEE Transactions on Signal Processing 41, no. 12 (1993): 3275-3292.
Parameters
----------
y : np.ndarray, real-valued
The input signal(s). Can be multidimensional.
The target axis must contain at least 3 samples.
t_min : float > 0
The minimum time spacing (in samples).
This value should generally be less than 1 to preserve as much information as
possible.
n_fmt : int > 2 or None
The number of scale transform bins to use.
If None, then `n_bins = over_sample * ceil(n * log((n-1)/t_min))` is taken,
where `n = y.shape[axis]`
kind : str
The type of interpolation to use when re-sampling the input.
See `scipy.interpolate.interp1d` for possible values.
Note that the default is to use high-precision (cubic) interpolation.
This can be slow in practice; if speed is preferred over accuracy,
then consider using `kind='linear'`.
beta : float
The Mellin parameter. `beta=0.5` provides the scale transform.
over_sample : float >= 1
Over-sampling factor for exponential resampling.
axis : int
The axis along which to transform `y`
Returns
-------
x_scale : np.ndarray [dtype=complex]
The scale transform of `y` along the `axis` dimension.
Raises
------
ParameterError
if `n_fmt < 2` or `t_min <= 0`
or if `y` is not finite
or if `y.shape[axis] < 3`.
Notes
-----
This function caches at level 30.
Examples
--------
>>> # Generate a signal and time-stretch it (with energy normalization)
>>> scale = 1.25
>>> freq = 3.0
>>> x1 = np.linspace(0, 1, num=1024, endpoint=False)
>>> x2 = np.linspace(0, 1, num=scale * len(x1), endpoint=False)
>>> y1 = np.sin(2 * np.pi * freq * x1)
>>> y2 = np.sin(2 * np.pi * freq * x2) / np.sqrt(scale)
>>> # Verify that the two signals have the same energy
>>> np.sum(np.abs(y1)**2), np.sum(np.abs(y2)**2)
(255.99999999999997, 255.99999999999969)
>>> scale1 = librosa.fmt(y1, n_fmt=512)
>>> scale2 = librosa.fmt(y2, n_fmt=512)
>>> # And plot the results
>>> import matplotlib.pyplot as plt
>>> plt.figure(figsize=(8, 4))
>>> plt.subplot(1, 2, 1)
>>> plt.plot(y1, label='Original')
>>> plt.plot(y2, linestyle='--', label='Stretched')
>>> plt.xlabel('time (samples)')
>>> plt.title('Input signals')
>>> plt.legend(frameon=True)
>>> plt.axis('tight')
>>> plt.subplot(1, 2, 2)
>>> plt.semilogy(np.abs(scale1), label='Original')
>>> plt.semilogy(np.abs(scale2), linestyle='--', label='Stretched')
>>> plt.xlabel('scale coefficients')
>>> plt.title('Scale transform magnitude')
>>> plt.legend(frameon=True)
>>> plt.axis('tight')
>>> plt.tight_layout()
>>> # Plot the scale transform of an onset strength autocorrelation
>>> y, sr = librosa.load(librosa.util.example_audio_file(),
... offset=10.0, duration=30.0)
>>> odf = librosa.onset.onset_strength(y=y, sr=sr)
>>> # Auto-correlate with up to 10 seconds lag
>>> odf_ac = librosa.autocorrelate(odf, max_size=10 * sr // 512)
>>> # Normalize
>>> odf_ac = librosa.util.normalize(odf_ac, norm=np.inf)
>>> # Compute the scale transform
>>> odf_ac_scale = librosa.fmt(librosa.util.normalize(odf_ac), n_fmt=512)
>>> # Plot the results
>>> plt.figure()
>>> plt.subplot(3, 1, 1)
>>> plt.plot(odf, label='Onset strength')
>>> plt.axis('tight')
>>> plt.xlabel('Time (frames)')
>>> plt.xticks([])
>>> plt.legend(frameon=True)
>>> plt.subplot(3, 1, 2)
>>> plt.plot(odf_ac, label='Onset autocorrelation')
>>> plt.axis('tight')
>>> plt.xlabel('Lag (frames)')
>>> plt.xticks([])
>>> plt.legend(frameon=True)
>>> plt.subplot(3, 1, 3)
>>> plt.semilogy(np.abs(odf_ac_scale), label='Scale transform magnitude')
>>> plt.axis('tight')
>>> plt.xlabel('scale coefficients')
>>> plt.legend(frameon=True)
>>> plt.tight_layout()
"""
n = y.shape[axis]
if n < 3:
raise ParameterError('y.shape[{:}]=={:} < 3'.format(axis, n))
if t_min <= 0:
raise ParameterError('t_min must be a positive number')
if n_fmt is None:
if over_sample < 1:
raise ParameterError('over_sample must be >= 1')
# The base is the maximum ratio between adjacent samples
# Since the sample spacing is increasing, this is simply the
# ratio between the positions of the last two samples: (n-1)/(n-2)
log_base = np.log(n - 1) - np.log(n - 2)
n_fmt = int(np.ceil(over_sample * (np.log(n - 1) - np.log(t_min)) / log_base))
elif n_fmt < 3:
raise ParameterError('n_fmt=={:} < 3'.format(n_fmt))
else:
log_base = (np.log(n_fmt - 1) - np.log(n_fmt - 2)) / over_sample
if not np.all(np.isfinite(y)):
raise ParameterError('y must be finite everywhere')
base = np.exp(log_base)
# original grid: signal covers [0, 1). This range is arbitrary, but convenient.
# The final sample is positioned at (n-1)/n, so we omit the endpoint
x = np.linspace(0, 1, num=n, endpoint=False)
# build the interpolator
f_interp = scipy.interpolate.interp1d(x, y, kind=kind, axis=axis)
# build the new sampling grid
# exponentially spaced between t_min/n and 1 (exclusive)
# we'll go one past where we need, and drop the last sample
# When over-sampling, the last input sample contributions n_over samples.
# To keep the spacing consistent, we over-sample by n_over, and then
# trim the final samples.
n_over = int(np.ceil(over_sample))
x_exp = np.logspace((np.log(t_min) - np.log(n)) / log_base,
0,
num=n_fmt + n_over,
endpoint=False,
base=base)[:-n_over]
# Clean up any rounding errors at the boundaries of the interpolation
# The interpolator gets angry if we try to extrapolate, so clipping is necessary here.
if x_exp[0] < t_min or x_exp[-1] > float(n - 1.0) / n:
x_exp = np.clip(x_exp, float(t_min) / n, x[-1])
# Make sure that all sample points are unique
assert len(np.unique(x_exp)) == len(x_exp)
# Resample the signal
y_res = f_interp(x_exp)
# Broadcast the window correctly
shape = [1] * y_res.ndim
shape[axis] = -1
# Apply the window and fft
result = fft.fft(y_res * (x_exp**beta).reshape(shape),
axis=axis, overwrite_x=True)
# Slice out the positive-scale component
idx = [slice(None)] * result.ndim
idx[axis] = slice(0, 1 + n_fmt//2)
# Truncate and length-normalize
return result[idx] * np.sqrt(n) / n_fmt
def _spectrogram(y=None, S=None, n_fft=2048, hop_length=512, power=1):
'''Helper function to retrieve a magnitude spectrogram.
This is primarily used in feature extraction functions that can operate on
either audio time-series or spectrogram input.
Parameters
----------
y : None or np.ndarray [ndim=1]
If provided, an audio time series
S : None or np.ndarray
Spectrogram input, optional
n_fft : int > 0
STFT window size
hop_length : int > 0
STFT hop length
power : float > 0
Exponent for the magnitude spectrogram,
e.g., 1 for energy, 2 for power, etc.
Returns
-------
S_out : np.ndarray [dtype=np.float32]
- If `S` is provided as input, then `S_out == S`
- Else, `S_out = |stft(y, n_fft=n_fft, hop_length=hop_length)|**power`
n_fft : int > 0
- If `S` is provided, then `n_fft` is inferred from `S`
- Else, copied from input
'''
if S is not None:
# Infer n_fft from spectrogram shape
n_fft = 2 * (S.shape[0] - 1)
else:
# Otherwise, compute a magnitude spectrogram from input
S = np.abs(stft(y, n_fft=n_fft, hop_length=hop_length))**power
return S, n_fft
附上参考文献参考博客
2984
被折叠的 条评论
为什么被折叠?
到【灌水乐园】发言
