#!/usr/bin/env python
# -*- coding: utf-8 -*-
"""
Effects
=======
Harmonic-percussive source separation
-------------------------------------
.. autosummary::
:toctree: generated/
hpss
harmonic
percussive
Time and frequency
------------------
.. autosummary::
:toctree: generated/
time_stretch
pitch_shift
Miscellaneous
-------------
.. autosummary::
:toctree: generated/
remix
trim
split
"""
import numpy as np
from . import core
from . import decompose
from . import feature
from . import util
from .util.exceptions import ParameterError
__all__ = ['hpss', 'harmonic', 'percussive',
'time_stretch', 'pitch_shift',
'remix', 'trim', 'split']
[docs]def hpss(y, **kwargs):
'''Decompose an audio time series into harmonic and percussive components.
This function automates the STFT->HPSS->ISTFT pipeline, and ensures that
the output waveforms have equal length to the input waveform `y`.
Parameters
----------
y : np.ndarray [shape=(n,)]
audio time series
kwargs : additional keyword arguments.
See `librosa.decompose.hpss` for details.
Returns
-------
y_harmonic : np.ndarray [shape=(n,)]
audio time series of the harmonic elements
y_percussive : np.ndarray [shape=(n,)]
audio time series of the percussive elements
See Also
--------
harmonic : Extract only the harmonic component
percussive : Extract only the percussive component
librosa.decompose.hpss : HPSS on spectrograms
Examples
--------
>>> # Extract harmonic and percussive components
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> y_harmonic, y_percussive = librosa.effects.hpss(y)
>>> # Get a more isolated percussive component by widening its margin
>>> y_harmonic, y_percussive = librosa.effects.hpss(y, margin=(1.0,5.0))
'''
# Compute the STFT matrix
stft = core.stft(y)
# Decompose into harmonic and percussives
stft_harm, stft_perc = decompose.hpss(stft, **kwargs)
# Invert the STFTs. Adjust length to match the input.
y_harm = util.fix_length(core.istft(stft_harm, dtype=y.dtype), len(y))
y_perc = util.fix_length(core.istft(stft_perc, dtype=y.dtype), len(y))
return y_harm, y_perc
[docs]def harmonic(y, **kwargs):
'''Extract harmonic elements from an audio time-series.
Parameters
----------
y : np.ndarray [shape=(n,)]
audio time series
kwargs : additional keyword arguments.
See `librosa.decompose.hpss` for details.
Returns
-------
y_harmonic : np.ndarray [shape=(n,)]
audio time series of just the harmonic portion
See Also
--------
hpss : Separate harmonic and percussive components
percussive : Extract only the percussive component
librosa.decompose.hpss : HPSS for spectrograms
Examples
--------
>>> # Extract harmonic component
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> y_harmonic = librosa.effects.harmonic(y)
>>> # Use a margin > 1.0 for greater harmonic separation
>>> y_harmonic = librosa.effects.harmonic(y, margin=3.0)
'''
# Compute the STFT matrix
stft = core.stft(y)
# Remove percussives
stft_harm = decompose.hpss(stft, **kwargs)[0]
# Invert the STFTs
y_harm = util.fix_length(core.istft(stft_harm, dtype=y.dtype), len(y))
return y_harm
[docs]def percussive(y, **kwargs):
'''Extract percussive elements from an audio time-series.
Parameters
----------
y : np.ndarray [shape=(n,)]
audio time series
kwargs : additional keyword arguments.
See `librosa.decompose.hpss` for details.
Returns
-------
y_percussive : np.ndarray [shape=(n,)]
audio time series of just the percussive portion
See Also
--------
hpss : Separate harmonic and percussive components
harmonic : Extract only the harmonic component
librosa.decompose.hpss : HPSS for spectrograms
Examples
--------
>>> # Extract percussive component
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> y_percussive = librosa.effects.percussive(y)
>>> # Use a margin > 1.0 for greater percussive separation
>>> y_percussive = librosa.effects.percussive(y, margin=3.0)
'''
# Compute the STFT matrix
stft = core.stft(y)
# Remove harmonics
stft_perc = decompose.hpss(stft, **kwargs)[1]
# Invert the STFT
y_perc = util.fix_length(core.istft(stft_perc, dtype=y.dtype), len(y))
return y_perc
[docs]def time_stretch(y, rate):
'''Time-stretch an audio series by a fixed rate.
Parameters
----------
y : np.ndarray [shape=(n,)]
audio time series
rate : float > 0 [scalar]
Stretch factor. If `rate > 1`, then the signal is sped up.
If `rate < 1`, then the signal is slowed down.
Returns
-------
y_stretch : np.ndarray [shape=(rate * n,)]
audio time series stretched by the specified rate
See Also
--------
pitch_shift : pitch shifting
librosa.core.phase_vocoder : spectrogram phase vocoder
Examples
--------
Compress to be twice as fast
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> y_fast = librosa.effects.time_stretch(y, 2.0)
Or half the original speed
>>> y_slow = librosa.effects.time_stretch(y, 0.5)
'''
if rate <= 0:
raise ParameterError('rate must be a positive number')
# Construct the stft
stft = core.stft(y)
# Stretch by phase vocoding
stft_stretch = core.phase_vocoder(stft, rate)
# Invert the stft
y_stretch = core.istft(stft_stretch, dtype=y.dtype)
return y_stretch
[docs]def pitch_shift(y, sr, n_steps, bins_per_octave=12):
'''Pitch-shift the waveform by `n_steps` half-steps.
Parameters
----------
y : np.ndarray [shape=(n,)]
audio time-series
sr : number > 0 [scalar]
audio sampling rate of `y`
n_steps : float [scalar]
how many (fractional) half-steps to shift `y`
bins_per_octave : float > 0 [scalar]
how many steps per octave
Returns
-------
y_shift : np.ndarray [shape=(n,)]
The pitch-shifted audio time-series
See Also
--------
time_stretch : time stretching
librosa.core.phase_vocoder : spectrogram phase vocoder
Examples
--------
Shift up by a major third (four half-steps)
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> y_third = librosa.effects.pitch_shift(y, sr, n_steps=4)
Shift down by a tritone (six half-steps)
>>> y_tritone = librosa.effects.pitch_shift(y, sr, n_steps=-6)
Shift up by 3 quarter-tones
>>> y_three_qt = librosa.effects.pitch_shift(y, sr, n_steps=3,
... bins_per_octave=24)
'''
if bins_per_octave < 1 or not np.issubdtype(type(bins_per_octave), np.integer):
raise ParameterError('bins_per_octave must be a positive integer.')
rate = 2.0 ** (-float(n_steps) / bins_per_octave)
# Stretch in time, then resample
y_shift = core.resample(time_stretch(y, rate), float(sr) / rate, sr)
# Crop to the same dimension as the input
return util.fix_length(y_shift, len(y))
[docs]def remix(y, intervals, align_zeros=True):
'''Remix an audio signal by re-ordering time intervals.
Parameters
----------
y : np.ndarray [shape=(t,) or (2, t)]
Audio time series
intervals : iterable of tuples (start, end)
An iterable (list-like or generator) where the `i`th item
`intervals[i]` indicates the start and end (in samples)
of a slice of `y`.
align_zeros : boolean
If `True`, interval boundaries are mapped to the closest
zero-crossing in `y`. If `y` is stereo, zero-crossings
are computed after converting to mono.
Returns
-------
y_remix : np.ndarray [shape=(d,) or (2, d)]
`y` remixed in the order specified by `intervals`
Examples
--------
Load in the example track and reverse the beats
>>> y, sr = librosa.load(librosa.util.example_audio_file())
Compute beats
>>> _, beat_frames = librosa.beat.beat_track(y=y, sr=sr,
... hop_length=512)
Convert from frames to sample indices
>>> beat_samples = librosa.frames_to_samples(beat_frames)
Generate intervals from consecutive events
>>> intervals = librosa.util.frame(beat_samples, frame_length=2,
... hop_length=1).T
Reverse the beat intervals
>>> y_out = librosa.effects.remix(y, intervals[::-1])
'''
# Validate the audio buffer
util.valid_audio(y, mono=False)
y_out = []
if align_zeros:
y_mono = core.to_mono(y)
zeros = np.nonzero(core.zero_crossings(y_mono))[-1]
# Force end-of-signal onto zeros
zeros = np.append(zeros, [len(y_mono)])
clip = [slice(None)] * y.ndim
for interval in intervals:
if align_zeros:
interval = zeros[util.match_events(interval, zeros)]
clip[-1] = slice(interval[0], interval[1])
y_out.append(y[clip])
return np.concatenate(y_out, axis=-1)
def _signal_to_frame_nonsilent(y, frame_length=2048, hop_length=512, top_db=60,
ref=np.max):
'''Frame-wise non-silent indicator for audio input.
This is a helper function for `trim` and `split`.
Parameters
----------
y : np.ndarray, shape=(n,) or (2,n)
Audio signal, mono or stereo
frame_length : int > 0
The number of samples per frame
hop_length : int > 0
The number of samples between frames
top_db : number > 0
The threshold (in decibels) below reference to consider as
silence
ref : callable or float
The reference power
Returns
-------
non_silent : np.ndarray, shape=(m,), dtype=bool
Indicator of non-silent frames
'''
# Convert to mono
y_mono = core.to_mono(y)
# Compute the MSE for the signal
mse = feature.rmse(y=y_mono,
frame_length=frame_length,
hop_length=hop_length)**2
return (core.power_to_db(mse.squeeze(),
ref=ref,
top_db=None) > - top_db)
[docs]def trim(y, top_db=60, ref=np.max, frame_length=2048, hop_length=512):
'''Trim leading and trailing silence from an audio signal.
Parameters
----------
y : np.ndarray, shape=(n,) or (2,n)
Audio signal, can be mono or stereo
top_db : number > 0
The threshold (in decibels) below reference to consider as
silence
ref : number or callable
The reference power. By default, it uses `np.max` and compares
to the peak power in the signal.
frame_length : int > 0
The number of samples per analysis frame
hop_length : int > 0
The number of samples between analysis frames
Returns
-------
y_trimmed : np.ndarray, shape=(m,) or (2, m)
The trimmed signal
index : np.ndarray, shape=(2,)
the interval of `y` corresponding to the non-silent region:
`y_trimmed = y[index[0]:index[1]]` (for mono) or
`y_trimmed = y[:, index[0]:index[1]]` (for stereo).
Examples
--------
>>> # Load some audio
>>> y, sr = librosa.load(librosa.util.example_audio_file())
>>> # Trim the beginning and ending silence
>>> yt, index = librosa.effects.trim(y)
>>> # Print the durations
>>> print(librosa.get_duration(y), librosa.get_duration(yt))
61.45886621315193 60.58086167800454
'''
non_silent = _signal_to_frame_nonsilent(y,
frame_length=frame_length,
hop_length=hop_length,
ref=ref,
top_db=top_db)
nonzero = np.flatnonzero(non_silent)
if nonzero.size > 0:
# Compute the start and end positions
# End position goes one frame past the last non-zero
start = int(core.frames_to_samples(nonzero[0], hop_length))
end = min(y.shape[-1],
int(core.frames_to_samples(nonzero[-1] + 1, hop_length)))
else:
# The signal only contains zeros
start, end = 0, 0
# Build the mono/stereo index
full_index = [slice(None)] * y.ndim
full_index[-1] = slice(start, end)
return y[full_index], np.asarray([start, end])
[docs]def split(y, top_db=60, ref=np.max, frame_length=2048, hop_length=512):
'''Split an audio signal into non-silent intervals.
Parameters
----------
y : np.ndarray, shape=(n,) or (2, n)
An audio signal
top_db : number > 0
The threshold (in decibels) below reference to consider as
silence
ref : number or callable
The reference power. By default, it uses `np.max` and compares
to the peak power in the signal.
frame_length : int > 0
The number of samples per analysis frame
hop_length : int > 0
The number of samples between analysis frames
Returns
-------
intervals : np.ndarray, shape=(m, 2)
`intervals[i] == (start_i, end_i)` are the start and end time
(in samples) of non-silent interval `i`.
'''
non_silent = _signal_to_frame_nonsilent(y,
frame_length=frame_length,
hop_length=hop_length,
ref=ref,
top_db=top_db)
# Interval slicing, adapted from
# https://stackoverflow.com/questions/2619413/efficiently-finding-the-interval-with-non-zeros-in-scipy-numpy-in-python
# Find points where the sign flips
edges = np.flatnonzero(np.diff(non_silent.astype(int)))
# Pad back the sample lost in the diff
edges = [edges + 1]
# If the first frame had high energy, count it
if non_silent[0]:
edges.insert(0, [0])
# Likewise for the last frame
if non_silent[-1]:
edges.append([len(non_silent)])
# Convert from frames to samples
edges = core.frames_to_samples(np.concatenate(edges),
hop_length=hop_length)
# Clip to the signal duration
edges = np.minimum(edges, y.shape[-1])
# Stack the results back as an ndarray
return edges.reshape((-1, 2))