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