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audio.py
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audio.py
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# -*- coding: utf-8 -*-
#!/usr/bin/env python
from scipy import signal
from pydub import AudioSegment
import os
import librosa
import soundfile as sf
import numpy as np
def read_wav(path, sr, duration=None, mono=True):
wav, _ = librosa.load(path, mono=mono, sr=sr, duration=duration)
return wav
def write_wav(wav, sr, path, format='wav', subtype='PCM_16'):
sf.write(path, wav, sr, format=format, subtype=subtype)
def read_mfcc(prefix):
filename = '{}.mfcc.npy'.format(prefix)
mfcc = np.load(filename)
return mfcc
def write_mfcc(prefix, mfcc):
filename = '{}.mfcc'.format(prefix)
np.save(filename, mfcc)
def read_spectrogram(prefix):
filename = '{}.spec.npy'.format(prefix)
spec = np.load(filename)
return spec
def write_spectrogram(prefix, spec):
filename = '{}.spec'.format(prefix)
np.save(filename, spec)
def split_wav(wav, top_db):
intervals = librosa.effects.split(wav, top_db=top_db)
wavs = map(lambda i: wav[i[0]: i[1]], intervals)
return wavs
def trim_wav(wav):
wav, _ = librosa.effects.trim(wav)
return wav
def fix_length(wav, length, mode='constant'):
if len(wav) != length:
wav = librosa.util.fix_length(wav, length, mode=mode)
return wav
def crop_random_wav(wav, length):
"""
Randomly cropped a part in a wav file.
:param wav: a waveform
:param length: length to be randomly cropped.
:return: a randomly cropped part of wav.
"""
assert (wav.ndim <= 2)
assert (type(length) == int)
wav_len = wav.shape[-1]
start = np.random.choice(range(np.maximum(1, wav_len - length)), 1)[0]
end = start + length
if wav.ndim == 1:
wav = wav[start:end]
else:
wav = wav[:, start:end]
return wav
def mp3_to_wav(src_path, tar_path):
"""
Read mp3 file from source path, convert it to wav and write it to target path.
Necessary libraries: ffmpeg, libav.
:param src_path: source mp3 file path
:param tar_path: target wav file path
"""
basepath, filename = os.path.split(src_path)
os.chdir(basepath)
AudioSegment.from_mp3(src_path).export(tar_path, format='wav')
def prepro_audio(source_path, target_path, format=None, sr=None, db=None):
"""
Read a wav, change sample rate, format, and average decibel and write to target path.
:param source_path: source wav file path
:param target_path: target wav file path
:param sr: sample rate.
:param format: output audio format.
:param db: decibel.
"""
sound = AudioSegment.from_file(source_path, format)
if sr:
sound = sound.set_frame_rate(sr)
if db:
change_dBFS = db - sound.dBFS
sound = sound.apply_gain(change_dBFS)
sound.export(target_path, 'wav')
def _split_path(path):
"""
Split path to basename, filename and extension. For example, 'a/b/c.wav' => ('a/b', 'c', 'wav')
:param path: file path
:return: basename, filename, and extension
"""
basepath, filename = os.path.split(path)
filename, extension = os.path.splitext(filename)
return basepath, filename, extension
def wav2spec(wav, n_fft, win_length, hop_length, time_first=True):
"""
Get magnitude and phase spectrogram from waveforms.
Parameters
----------
wav : np.ndarray [shape=(n,)]
The real-valued waveform.
n_fft : int > 0 [scalar]
FFT window size.
win_length : int <= n_fft [scalar]
The window will be of length `win_length` and then padded
with zeros to match `n_fft`.
hop_length : int > 0 [scalar]
Number audio of frames between STFT columns.
time_first : boolean. optional.
if True, time axis is followed by bin axis. In this case, shape of returns is (t, 1 + n_fft/2)
Returns
-------
mag : np.ndarray [shape=(t, 1 + n_fft/2) or (1 + n_fft/2, t)]
Magnitude spectrogram.
phase : np.ndarray [shape=(t, 1 + n_fft/2) or (1 + n_fft/2, t)]
Phase spectrogram.
"""
stft = librosa.stft(y=wav, n_fft=n_fft, hop_length=hop_length, win_length=win_length)
mag = np.abs(stft)
phase = np.angle(stft)
if time_first:
mag = mag.T
phase = phase.T
return mag, phase
def spec2wav(mag, n_fft, win_length, hop_length, num_iters=30, phase=None):
"""
Get a waveform from the magnitude spectrogram by Griffin-Lim Algorithm.
Parameters
----------
mag : np.ndarray [shape=(1 + n_fft/2, t)]
Magnitude spectrogram.
n_fft : int > 0 [scalar]
FFT window size.
win_length : int <= n_fft [scalar]
The window will be of length `win_length` and then padded
with zeros to match `n_fft`.
hop_length : int > 0 [scalar]
Number audio of frames between STFT columns.
num_iters: int > 0 [scalar]
Number of iterations of Griffin-Lim Algorithm.
phase : np.ndarray [shape=(1 + n_fft/2, t)]
Initial phase spectrogram.
Returns
-------
wav : np.ndarray [shape=(n,)]
The real-valued waveform.
"""
assert (num_iters > 0)
if phase is None:
phase = np.pi * np.random.rand(*mag.shape)
stft = mag * np.exp(1.j * phase)
wav = None
for i in range(num_iters):
wav = librosa.istft(stft, win_length=win_length, hop_length=hop_length)
if i != num_iters - 1:
stft = librosa.stft(wav, n_fft=n_fft, win_length=win_length, hop_length=hop_length)
_, phase = librosa.magphase(stft)
phase = np.angle(phase)
stft = mag * np.exp(1.j * phase)
return wav
def preemphasis(wav, coeff=0.97):
"""
Emphasize high frequency range of the waveform by increasing power(squared amplitude).
Parameters
----------
wav : np.ndarray [shape=(n,)]
Real-valued the waveform.
coeff: float <= 1 [scalar]
Coefficient of pre-emphasis.
Returns
-------
preem_wav : np.ndarray [shape=(n,)]
The pre-emphasized waveform.
"""
preem_wav = signal.lfilter([1, -coeff], [1], wav)
return preem_wav
def inv_preemphasis(preem_wav, coeff=0.97):
"""
Invert the pre-emphasized waveform to the original waveform.
Parameters
----------
preem_wav : np.ndarray [shape=(n,)]
The pre-emphasized waveform.
coeff: float <= 1 [scalar]
Coefficient of pre-emphasis.
Returns
-------
wav : np.ndarray [shape=(n,)]
Real-valued the waveform.
"""
wav = signal.lfilter([1], [1, -coeff], preem_wav)
return wav
def linear_to_mel(linear, sr, n_fft, n_mels, **kwargs):
"""
Convert a linear-spectrogram to mel-spectrogram.
:param linear: Linear-spectrogram.
:param sr: Sample rate.
:param n_fft: FFT window size.
:param n_mels: Number of mel filters.
:return: Mel-spectrogram.
"""
mel_basis = librosa.filters.mel(sr, n_fft, n_mels, **kwargs) # (n_mels, 1+n_fft//2)
mel = np.dot(mel_basis, linear) # (n_mels, t) # mel spectrogram
return mel
def amp2db(amp):
return librosa.amplitude_to_db(amp)
def db2amp(db):
return librosa.db_to_amplitude(db)
def normalize_db(db, max_db, min_db):
"""
Normalize dB-scaled spectrogram values to be in range of 0~1.
:param db: Decibel-scaled spectrogram.
:param max_db: Maximum dB.
:param min_db: Minimum dB.
:return: Normalized spectrogram.
"""
norm_db = np.clip((db - min_db) / (max_db - min_db), 0, 1)
return norm_db
def denormalize_db(norm_db, max_db, min_db):
"""
Denormalize the normalized values to be original dB-scaled value.
:param norm_db: Normalized spectrogram.
:param max_db: Maximum dB.
:param min_db: Minimum dB.
:return: Decibel-scaled spectrogram.
"""
db = np.clip(norm_db, 0, 1) * (max_db - min_db) + min_db
return db
def dynamic_range_compression(db, threshold, ratio, method='downward'):
"""
Execute dynamic range compression(https://en.wikipedia.org/wiki/Dynamic_range_compression) to dB.
:param db: Decibel-scaled magnitudes
:param threshold: Threshold dB
:param ratio: Compression ratio.
:param method: Downward or upward.
:return: Range compressed dB-scaled magnitudes
"""
if method is 'downward':
db[db > threshold] = (db[db > threshold] - threshold) / ratio + threshold
elif method is 'upward':
db[db < threshold] = threshold - ((threshold - db[db < threshold]) / ratio)
return db
def emphasize_magnitude(mag, power=1.2):
"""
Emphasize a magnitude spectrogram by applying power function. This is used for removing noise.
:param mag: magnitude spectrogram.
:param power: exponent.
:return: emphasized magnitude spectrogram.
"""
emphasized_mag = np.power(mag, power)
return emphasized_mag
def wav2melspec(wav, sr, n_fft, win_length, hop_length, n_mels, time_first=True, **kwargs):
# Linear spectrogram
mag_spec, phase_spec = wav2spec(wav, n_fft, win_length, hop_length, time_first=False)
# Mel-spectrogram
mel_spec = linear_to_mel(mag_spec, sr, n_fft, n_mels, **kwargs)
# Time-axis first
if time_first:
mel_spec = mel_spec.T # (t, n_mels)
return mel_spec
def wav2melspec_db(wav, sr, n_fft, win_length, hop_length, n_mels, normalize=False, max_db=None, min_db=None, time_first=True, **kwargs):
# Mel-spectrogram
mel_spec = wav2melspec(wav, sr, n_fft, win_length, hop_length, n_mels, time_first=False, **kwargs)
# Decibel
mel_db = librosa.amplitude_to_db(mel_spec)
# Normalization
mel_db = normalize_db(mel_db, max_db, min_db) if normalize else mel_db
# Time-axis first
if time_first:
mel_db = mel_db.T # (t, n_mels)
return mel_db
def wav2mfcc(wav, sr, n_fft, win_length, hop_length, n_mels, n_mfccs, preemphasis_coeff=0.97, time_first=True, **kwargs):
# Pre-emphasis
wav_preem = preemphasis(wav, coeff=preemphasis_coeff)
# Decibel-scaled mel-spectrogram
mel_db = wav2melspec_db(wav_preem, sr, n_fft, win_length, hop_length, n_mels, time_first=False, **kwargs)
# MFCCs
mfccs = np.dot(librosa.filters.dct(n_mfccs, mel_db.shape[0]), mel_db)
# Time-axis first
if time_first:
mfccs = mfccs.T # (t, n_mfccs)
return mfccs