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audio.py
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audio.py
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import librosa
import librosa.filters
import numpy as np
from hparams import hparams
from scipy.io import wavfile
from nnmnkwii import preprocessing as P
def low_cut_filter(x, fs, cutoff=70):
"""APPLY LOW CUT FILTER.
https://github.com/kan-bayashi/PytorchWaveNetVocoder
Args:
x (ndarray): Waveform sequence.
fs (int): Sampling frequency.
cutoff (float): Cutoff frequency of low cut filter.
Return:
ndarray: Low cut filtered waveform sequence.
"""
nyquist = fs // 2
norm_cutoff = cutoff / nyquist
from scipy.signal import firwin, lfilter
# low cut filter
fil = firwin(255, norm_cutoff, pass_zero=False)
lcf_x = lfilter(fil, 1, x)
return lcf_x
def load_wav(path):
sr, x = wavfile.read(path)
signed_int16_max = 2**15
if x.dtype == np.int16:
x = x.astype(np.float32) / signed_int16_max
if sr != hparams.sample_rate:
x = librosa.resample(x, sr, hparams.sample_rate)
x = np.clip(x, -1.0, 1.0)
return x
def save_wav(wav, path):
wav *= 32767 / max(0.01, np.max(np.abs(wav)))
wavfile.write(path, hparams.sample_rate, wav.astype(np.int16))
def trim(quantized):
start, end = start_and_end_indices(quantized, hparams.silence_threshold)
return quantized[start:end]
def preemphasis(x, coef=0.85):
return P.preemphasis(x, coef)
def inv_preemphasis(x, coef=0.85):
return P.inv_preemphasis(x, coef)
def adjust_time_resolution(quantized, mel):
"""Adjust time resolution by repeating features
Args:
quantized (ndarray): (T,)
mel (ndarray): (N, D)
Returns:
tuple: Tuple of (T,) and (T, D)
"""
assert len(quantized.shape) == 1
assert len(mel.shape) == 2
upsample_factor = quantized.size // mel.shape[0]
mel = np.repeat(mel, upsample_factor, axis=0)
n_pad = quantized.size - mel.shape[0]
if n_pad != 0:
assert n_pad > 0
mel = np.pad(mel, [(0, n_pad), (0, 0)], mode="constant", constant_values=0)
# trim
start, end = start_and_end_indices(quantized, hparams.silence_threshold)
return quantized[start:end], mel[start:end, :]
def start_and_end_indices(quantized, silence_threshold=2):
for start in range(quantized.size):
if abs(quantized[start] - 127) > silence_threshold:
break
for end in range(quantized.size - 1, 1, -1):
if abs(quantized[end] - 127) > silence_threshold:
break
assert abs(quantized[start] - 127) > silence_threshold
assert abs(quantized[end] - 127) > silence_threshold
return start, end
def logmelspectrogram(y, pad_mode="reflect"):
"""Same log-melspectrogram computation as espnet
https://github.com/espnet/espnet
from espnet.transform.spectrogram import logmelspectrogram
"""
D = _stft(y, pad_mode=pad_mode)
S = _linear_to_mel(np.abs(D))
S = np.log10(np.maximum(S, 1e-10))
return S
def get_hop_size():
hop_size = hparams.hop_size
if hop_size is None:
assert hparams.frame_shift_ms is not None
hop_size = int(hparams.frame_shift_ms / 1000 * hparams.sample_rate)
return hop_size
def get_win_length():
win_length = hparams.win_length
if win_length < 0:
assert hparams.win_length_ms > 0
win_length = int(hparams.win_length_ms / 1000 * hparams.sample_rate)
return win_length
def _stft(y, pad_mode="constant"):
# use constant padding (defaults to zeros) instead of reflection padding
return librosa.stft(y=y, n_fft=hparams.fft_size, hop_length=get_hop_size(),
win_length=get_win_length(), window=hparams.window,
pad_mode=pad_mode)
def pad_lr(x, fsize, fshift):
return (0, fsize)
# Conversions:
_mel_basis = None
def _linear_to_mel(spectrogram):
global _mel_basis
if _mel_basis is None:
_mel_basis = _build_mel_basis()
return np.dot(_mel_basis, spectrogram)
def _build_mel_basis():
if hparams.fmax is not None:
assert hparams.fmax <= hparams.sample_rate // 2
return librosa.filters.mel(hparams.sample_rate, hparams.fft_size,
fmin=hparams.fmin, fmax=hparams.fmax,
n_mels=hparams.num_mels)
def _amp_to_db(x):
min_level = np.exp(hparams.min_level_db / 20 * np.log(10))
return 20 * np.log10(np.maximum(min_level, x))
def _db_to_amp(x):
return np.power(10.0, x * 0.05)
def _normalize(S):
return np.clip((S - hparams.min_level_db) / -hparams.min_level_db, 0, 1)
def _denormalize(S):
return (np.clip(S, 0, 1) * -hparams.min_level_db) + hparams.min_level_db