486 lines
15 KiB
Python
486 lines
15 KiB
Python
from io import BytesIO
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from typing import Tuple
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import librosa
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import numpy as np
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import scipy
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import soundfile as sf
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from librosa import magphase, pyin
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# For using kwargs
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# pylint: disable=unused-argument
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def build_mel_basis(
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*,
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sample_rate: int = None,
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fft_size: int = None,
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num_mels: int = None,
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mel_fmax: int = None,
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mel_fmin: int = None,
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**kwargs,
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) -> np.ndarray:
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"""Build melspectrogram basis.
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Returns:
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np.ndarray: melspectrogram basis.
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"""
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if mel_fmax is not None:
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assert mel_fmax <= sample_rate // 2
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assert mel_fmax - mel_fmin > 0
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return librosa.filters.mel(sr=sample_rate, n_fft=fft_size, n_mels=num_mels, fmin=mel_fmin, fmax=mel_fmax)
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def millisec_to_length(
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*, frame_length_ms: int = None, frame_shift_ms: int = None, sample_rate: int = None, **kwargs
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) -> Tuple[int, int]:
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"""Compute hop and window length from milliseconds.
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Returns:
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Tuple[int, int]: hop length and window length for STFT.
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"""
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factor = frame_length_ms / frame_shift_ms
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assert (factor).is_integer(), " [!] frame_shift_ms should divide frame_length_ms"
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win_length = int(frame_length_ms / 1000.0 * sample_rate)
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hop_length = int(win_length / float(factor))
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return win_length, hop_length
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def _log(x, base):
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if base == 10:
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return np.log10(x)
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return np.log(x)
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def _exp(x, base):
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if base == 10:
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return np.power(10, x)
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return np.exp(x)
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def amp_to_db(*, x: np.ndarray = None, gain: float = 1, base: int = 10, **kwargs) -> np.ndarray:
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"""Convert amplitude values to decibels.
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Args:
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x (np.ndarray): Amplitude spectrogram.
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gain (float): Gain factor. Defaults to 1.
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base (int): Logarithm base. Defaults to 10.
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Returns:
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np.ndarray: Decibels spectrogram.
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"""
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assert (x < 0).sum() == 0, " [!] Input values must be non-negative."
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return gain * _log(np.maximum(1e-8, x), base)
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# pylint: disable=no-self-use
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def db_to_amp(*, x: np.ndarray = None, gain: float = 1, base: int = 10, **kwargs) -> np.ndarray:
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"""Convert decibels spectrogram to amplitude spectrogram.
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Args:
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x (np.ndarray): Decibels spectrogram.
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gain (float): Gain factor. Defaults to 1.
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base (int): Logarithm base. Defaults to 10.
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Returns:
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np.ndarray: Amplitude spectrogram.
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"""
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return _exp(x / gain, base)
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def preemphasis(*, x: np.ndarray, coef: float = 0.97, **kwargs) -> np.ndarray:
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"""Apply pre-emphasis to the audio signal. Useful to reduce the correlation between neighbouring signal values.
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Args:
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x (np.ndarray): Audio signal.
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Raises:
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RuntimeError: Preemphasis coeff is set to 0.
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Returns:
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np.ndarray: Decorrelated audio signal.
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"""
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if coef == 0:
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raise RuntimeError(" [!] Preemphasis is set 0.0.")
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return scipy.signal.lfilter([1, -coef], [1], x)
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def deemphasis(*, x: np.ndarray = None, coef: float = 0.97, **kwargs) -> np.ndarray:
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"""Reverse pre-emphasis."""
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if coef == 0:
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raise RuntimeError(" [!] Preemphasis is set 0.0.")
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return scipy.signal.lfilter([1], [1, -coef], x)
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def spec_to_mel(*, spec: np.ndarray, mel_basis: np.ndarray = None, **kwargs) -> np.ndarray:
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"""Convert a full scale linear spectrogram output of a network to a melspectrogram.
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Args:
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spec (np.ndarray): Normalized full scale linear spectrogram.
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Shapes:
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- spec: :math:`[C, T]`
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Returns:
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np.ndarray: Normalized melspectrogram.
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"""
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return np.dot(mel_basis, spec)
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def mel_to_spec(*, mel: np.ndarray = None, mel_basis: np.ndarray = None, **kwargs) -> np.ndarray:
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"""Convert a melspectrogram to full scale spectrogram."""
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assert (mel < 0).sum() == 0, " [!] Input values must be non-negative."
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inv_mel_basis = np.linalg.pinv(mel_basis)
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return np.maximum(1e-10, np.dot(inv_mel_basis, mel))
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def wav_to_spec(*, wav: np.ndarray = None, **kwargs) -> np.ndarray:
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"""Compute a spectrogram from a waveform.
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Args:
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wav (np.ndarray): Waveform. Shape :math:`[T_wav,]`
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Returns:
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np.ndarray: Spectrogram. Shape :math:`[C, T_spec]`. :math:`T_spec == T_wav / hop_length`
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"""
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D = stft(y=wav, **kwargs)
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S = np.abs(D)
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return S.astype(np.float32)
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def wav_to_mel(*, wav: np.ndarray = None, mel_basis=None, **kwargs) -> np.ndarray:
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"""Compute a melspectrogram from a waveform."""
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D = stft(y=wav, **kwargs)
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S = spec_to_mel(spec=np.abs(D), mel_basis=mel_basis, **kwargs)
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return S.astype(np.float32)
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def spec_to_wav(*, spec: np.ndarray, power: float = 1.5, **kwargs) -> np.ndarray:
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"""Convert a spectrogram to a waveform using Griffi-Lim vocoder."""
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S = spec.copy()
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return griffin_lim(spec=S**power, **kwargs)
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def mel_to_wav(*, mel: np.ndarray = None, power: float = 1.5, **kwargs) -> np.ndarray:
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"""Convert a melspectrogram to a waveform using Griffi-Lim vocoder."""
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S = mel.copy()
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S = mel_to_spec(mel=S, mel_basis=kwargs["mel_basis"]) # Convert back to linear
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return griffin_lim(spec=S**power, **kwargs)
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### STFT and ISTFT ###
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def stft(
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*,
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y: np.ndarray = None,
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fft_size: int = None,
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hop_length: int = None,
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win_length: int = None,
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pad_mode: str = "reflect",
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window: str = "hann",
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center: bool = True,
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**kwargs,
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) -> np.ndarray:
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"""Librosa STFT wrapper.
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Check http://librosa.org/doc/main/generated/librosa.stft.html argument details.
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Returns:
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np.ndarray: Complex number array.
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"""
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return librosa.stft(
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y=y,
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n_fft=fft_size,
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hop_length=hop_length,
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win_length=win_length,
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pad_mode=pad_mode,
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window=window,
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center=center,
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)
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def istft(
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*,
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y: np.ndarray = None,
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hop_length: int = None,
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win_length: int = None,
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window: str = "hann",
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center: bool = True,
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**kwargs,
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) -> np.ndarray:
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"""Librosa iSTFT wrapper.
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Check http://librosa.org/doc/main/generated/librosa.istft.html argument details.
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Returns:
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np.ndarray: Complex number array.
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"""
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return librosa.istft(y, hop_length=hop_length, win_length=win_length, center=center, window=window)
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def griffin_lim(*, spec: np.ndarray = None, num_iter=60, **kwargs) -> np.ndarray:
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angles = np.exp(2j * np.pi * np.random.rand(*spec.shape))
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S_complex = np.abs(spec).astype(complex)
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y = istft(y=S_complex * angles, **kwargs)
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if not np.isfinite(y).all():
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print(" [!] Waveform is not finite everywhere. Skipping the GL.")
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return np.array([0.0])
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for _ in range(num_iter):
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angles = np.exp(1j * np.angle(stft(y=y, **kwargs)))
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y = istft(y=S_complex * angles, **kwargs)
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return y
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def compute_stft_paddings(
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*, x: np.ndarray = None, hop_length: int = None, pad_two_sides: bool = False, **kwargs
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) -> Tuple[int, int]:
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"""Compute paddings used by Librosa's STFT. Compute right padding (final frame) or both sides padding
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(first and final frames)"""
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pad = (x.shape[0] // hop_length + 1) * hop_length - x.shape[0]
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if not pad_two_sides:
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return 0, pad
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return pad // 2, pad // 2 + pad % 2
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def compute_f0(
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*,
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x: np.ndarray = None,
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pitch_fmax: float = None,
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pitch_fmin: float = None,
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hop_length: int = None,
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win_length: int = None,
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sample_rate: int = None,
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stft_pad_mode: str = "reflect",
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center: bool = True,
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**kwargs,
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) -> np.ndarray:
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"""Compute pitch (f0) of a waveform using the same parameters used for computing melspectrogram.
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Args:
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x (np.ndarray): Waveform. Shape :math:`[T_wav,]`
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pitch_fmax (float): Pitch max value.
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pitch_fmin (float): Pitch min value.
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hop_length (int): Number of frames between STFT columns.
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win_length (int): STFT window length.
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sample_rate (int): Audio sampling rate.
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stft_pad_mode (str): Padding mode for STFT.
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center (bool): Centered padding.
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Returns:
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np.ndarray: Pitch. Shape :math:`[T_pitch,]`. :math:`T_pitch == T_wav / hop_length`
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Examples:
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>>> WAV_FILE = filename = librosa.example('vibeace')
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>>> from TTS.config import BaseAudioConfig
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>>> from TTS.utils.audio import AudioProcessor
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>>> conf = BaseAudioConfig(pitch_fmax=640, pitch_fmin=1)
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>>> ap = AudioProcessor(**conf)
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>>> wav = ap.load_wav(WAV_FILE, sr=ap.sample_rate)[:5 * ap.sample_rate]
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>>> pitch = ap.compute_f0(wav)
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"""
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assert pitch_fmax is not None, " [!] Set `pitch_fmax` before caling `compute_f0`."
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assert pitch_fmin is not None, " [!] Set `pitch_fmin` before caling `compute_f0`."
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f0, voiced_mask, _ = pyin(
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y=x.astype(np.double),
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fmin=pitch_fmin,
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fmax=pitch_fmax,
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sr=sample_rate,
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frame_length=win_length,
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win_length=win_length // 2,
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hop_length=hop_length,
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pad_mode=stft_pad_mode,
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center=center,
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n_thresholds=100,
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beta_parameters=(2, 18),
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boltzmann_parameter=2,
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resolution=0.1,
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max_transition_rate=35.92,
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switch_prob=0.01,
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no_trough_prob=0.01,
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)
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f0[~voiced_mask] = 0.0
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return f0
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def compute_energy(y: np.ndarray, **kwargs) -> np.ndarray:
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"""Compute energy of a waveform using the same parameters used for computing melspectrogram.
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Args:
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x (np.ndarray): Waveform. Shape :math:`[T_wav,]`
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Returns:
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np.ndarray: energy. Shape :math:`[T_energy,]`. :math:`T_energy == T_wav / hop_length`
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Examples:
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>>> WAV_FILE = filename = librosa.example('vibeace')
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>>> from TTS.config import BaseAudioConfig
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>>> from TTS.utils.audio import AudioProcessor
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>>> conf = BaseAudioConfig()
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>>> ap = AudioProcessor(**conf)
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>>> wav = ap.load_wav(WAV_FILE, sr=ap.sample_rate)[:5 * ap.sample_rate]
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>>> energy = ap.compute_energy(wav)
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"""
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x = stft(y=y, **kwargs)
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mag, _ = magphase(x)
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energy = np.sqrt(np.sum(mag**2, axis=0))
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return energy
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### Audio Processing ###
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def find_endpoint(
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*,
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wav: np.ndarray = None,
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trim_db: float = -40,
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sample_rate: int = None,
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min_silence_sec=0.8,
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gain: float = None,
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base: int = None,
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**kwargs,
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) -> int:
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"""Find the last point without silence at the end of a audio signal.
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Args:
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wav (np.ndarray): Audio signal.
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threshold_db (int, optional): Silence threshold in decibels. Defaults to -40.
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min_silence_sec (float, optional): Ignore silences that are shorter then this in secs. Defaults to 0.8.
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gian (float, optional): Gain to be used to convert trim_db to trim_amp. Defaults to None.
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base (int, optional): Base of the logarithm used to convert trim_db to trim_amp. Defaults to 10.
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Returns:
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int: Last point without silence.
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"""
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window_length = int(sample_rate * min_silence_sec)
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hop_length = int(window_length / 4)
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threshold = db_to_amp(x=-trim_db, gain=gain, base=base)
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for x in range(hop_length, len(wav) - window_length, hop_length):
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if np.max(wav[x : x + window_length]) < threshold:
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return x + hop_length
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return len(wav)
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def trim_silence(
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*,
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wav: np.ndarray = None,
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sample_rate: int = None,
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trim_db: float = None,
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win_length: int = None,
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hop_length: int = None,
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**kwargs,
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) -> np.ndarray:
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"""Trim silent parts with a threshold and 0.01 sec margin"""
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margin = int(sample_rate * 0.01)
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wav = wav[margin:-margin]
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return librosa.effects.trim(wav, top_db=trim_db, frame_length=win_length, hop_length=hop_length)[0]
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def volume_norm(*, x: np.ndarray = None, coef: float = 0.95, **kwargs) -> np.ndarray:
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"""Normalize the volume of an audio signal.
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Args:
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x (np.ndarray): Raw waveform.
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coef (float): Coefficient to rescale the maximum value. Defaults to 0.95.
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Returns:
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np.ndarray: Volume normalized waveform.
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"""
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return x / abs(x).max() * coef
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def rms_norm(*, wav: np.ndarray = None, db_level: float = -27.0, **kwargs) -> np.ndarray:
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r = 10 ** (db_level / 20)
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a = np.sqrt((len(wav) * (r**2)) / np.sum(wav**2))
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return wav * a
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def rms_volume_norm(*, x: np.ndarray, db_level: float = -27.0, **kwargs) -> np.ndarray:
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"""Normalize the volume based on RMS of the signal.
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Args:
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x (np.ndarray): Raw waveform.
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db_level (float): Target dB level in RMS. Defaults to -27.0.
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Returns:
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np.ndarray: RMS normalized waveform.
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"""
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assert -99 <= db_level <= 0, " [!] db_level should be between -99 and 0"
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wav = rms_norm(wav=x, db_level=db_level)
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return wav
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def load_wav(*, filename: str, sample_rate: int = None, resample: bool = False, **kwargs) -> np.ndarray:
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"""Read a wav file using Librosa and optionally resample, silence trim, volume normalize.
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Resampling slows down loading the file significantly. Therefore it is recommended to resample the file before.
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Args:
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filename (str): Path to the wav file.
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sr (int, optional): Sampling rate for resampling. Defaults to None.
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resample (bool, optional): Resample the audio file when loading. Slows down the I/O time. Defaults to False.
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Returns:
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np.ndarray: Loaded waveform.
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"""
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if resample:
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# loading with resampling. It is significantly slower.
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x, _ = librosa.load(filename, sr=sample_rate)
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else:
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# SF is faster than librosa for loading files
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x, _ = sf.read(filename)
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return x
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def save_wav(*, wav: np.ndarray, path: str, sample_rate: int = None, pipe_out=None, **kwargs) -> None:
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"""Save float waveform to a file using Scipy.
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Args:
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wav (np.ndarray): Waveform with float values in range [-1, 1] to save.
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path (str): Path to a output file.
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sr (int, optional): Sampling rate used for saving to the file. Defaults to None.
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pipe_out (BytesIO, optional): Flag to stdout the generated TTS wav file for shell pipe.
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"""
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wav_norm = wav * (32767 / max(0.01, np.max(np.abs(wav))))
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wav_norm = wav_norm.astype(np.int16)
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if pipe_out:
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wav_buffer = BytesIO()
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scipy.io.wavfile.write(wav_buffer, sample_rate, wav_norm)
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wav_buffer.seek(0)
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pipe_out.buffer.write(wav_buffer.read())
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scipy.io.wavfile.write(path, sample_rate, wav_norm)
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def mulaw_encode(*, wav: np.ndarray, mulaw_qc: int, **kwargs) -> np.ndarray:
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mu = 2**mulaw_qc - 1
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signal = np.sign(wav) * np.log(1 + mu * np.abs(wav)) / np.log(1.0 + mu)
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signal = (signal + 1) / 2 * mu + 0.5
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return np.floor(
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signal,
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)
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def mulaw_decode(*, wav, mulaw_qc: int, **kwargs) -> np.ndarray:
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"""Recovers waveform from quantized values."""
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mu = 2**mulaw_qc - 1
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x = np.sign(wav) / mu * ((1 + mu) ** np.abs(wav) - 1)
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return x
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def encode_16bits(*, x: np.ndarray, **kwargs) -> np.ndarray:
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return np.clip(x * 2**15, -(2**15), 2**15 - 1).astype(np.int16)
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def quantize(*, x: np.ndarray, quantize_bits: int, **kwargs) -> np.ndarray:
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"""Quantize a waveform to a given number of bits.
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Args:
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x (np.ndarray): Waveform to quantize. Must be normalized into the range `[-1, 1]`.
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quantize_bits (int): Number of quantization bits.
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Returns:
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np.ndarray: Quantized waveform.
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"""
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return (x + 1.0) * (2**quantize_bits - 1) / 2
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def dequantize(*, x, quantize_bits, **kwargs) -> np.ndarray:
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"""Dequantize a waveform from the given number of bits."""
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return 2 * x / (2**quantize_bits - 1) - 1
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