Source code for TTS.utils.audio

from typing import Dict, Tuple

import librosa
import numpy as np
import pyworld as pw
import scipy.io.wavfile
import scipy.signal
import soundfile as sf
import torch
from torch import nn

from TTS.tts.utils.helpers import StandardScaler


class TorchSTFT(nn.Module):  # pylint: disable=abstract-method
    """Some of the audio processing funtions using Torch for faster batch processing.

    TODO: Merge this with audio.py

    Args:

        n_fft (int):
            FFT window size for STFT.

        hop_length (int):
            number of frames between STFT columns.

        win_length (int, optional):
            STFT window length.

        pad_wav (bool, optional):
            If True pad the audio with (n_fft - hop_length) / 2). Defaults to False.

        window (str, optional):
            The name of a function to create a window tensor that is applied/multiplied to each frame/window. Defaults to "hann_window"

        sample_rate (int, optional):
            target audio sampling rate. Defaults to None.

        mel_fmin (int, optional):
            minimum filter frequency for computing melspectrograms. Defaults to None.

        mel_fmax (int, optional):
            maximum filter frequency for computing melspectrograms. Defaults to None.

        n_mels (int, optional):
            number of melspectrogram dimensions. Defaults to None.

        use_mel (bool, optional):
            If True compute the melspectrograms otherwise. Defaults to False.

        do_amp_to_db_linear (bool, optional):
            enable/disable amplitude to dB conversion of linear spectrograms. Defaults to False.

        spec_gain (float, optional):
            gain applied when converting amplitude to DB. Defaults to 1.0.

        power (float, optional):
            Exponent for the magnitude spectrogram, e.g., 1 for energy, 2 for power, etc.  Defaults to None.

        use_htk (bool, optional):
            Use HTK formula in mel filter instead of Slaney.

        mel_norm (None, 'slaney', or number, optional):
            If 'slaney', divide the triangular mel weights by the width of the mel band
            (area normalization).

            If numeric, use `librosa.util.normalize` to normalize each filter by to unit l_p norm.
            See `librosa.util.normalize` for a full description of supported norm values
            (including `+-np.inf`).

            Otherwise, leave all the triangles aiming for a peak value of 1.0. Defaults to "slaney".
    """

    def __init__(
        self,
        n_fft,
        hop_length,
        win_length,
        pad_wav=False,
        window="hann_window",
        sample_rate=None,
        mel_fmin=0,
        mel_fmax=None,
        n_mels=80,
        use_mel=False,
        do_amp_to_db=False,
        spec_gain=1.0,
        power=None,
        use_htk=False,
        mel_norm="slaney",
    ):
        super().__init__()
        self.n_fft = n_fft
        self.hop_length = hop_length
        self.win_length = win_length
        self.pad_wav = pad_wav
        self.sample_rate = sample_rate
        self.mel_fmin = mel_fmin
        self.mel_fmax = mel_fmax
        self.n_mels = n_mels
        self.use_mel = use_mel
        self.do_amp_to_db = do_amp_to_db
        self.spec_gain = spec_gain
        self.power = power
        self.use_htk = use_htk
        self.mel_norm = mel_norm
        self.window = nn.Parameter(getattr(torch, window)(win_length), requires_grad=False)
        self.mel_basis = None
        if use_mel:
            self._build_mel_basis()

    def __call__(self, x):
        """Compute spectrogram frames by torch based stft.

        Args:
            x (Tensor): input waveform

        Returns:
            Tensor: spectrogram frames.

        Shapes:
            x: [B x T] or [:math:`[B, 1, T]`]
        """
        if x.ndim == 2:
            x = x.unsqueeze(1)
        if self.pad_wav:
            padding = int((self.n_fft - self.hop_length) / 2)
            x = torch.nn.functional.pad(x, (padding, padding), mode="reflect")
        # B x D x T x 2
        o = torch.stft(
            x.squeeze(1),
            self.n_fft,
            self.hop_length,
            self.win_length,
            self.window,
            center=True,
            pad_mode="reflect",  # compatible with audio.py
            normalized=False,
            onesided=True,
            return_complex=False,
        )
        M = o[:, :, :, 0]
        P = o[:, :, :, 1]
        S = torch.sqrt(torch.clamp(M**2 + P**2, min=1e-8))

        if self.power is not None:
            S = S**self.power

        if self.use_mel:
            S = torch.matmul(self.mel_basis.to(x), S)
        if self.do_amp_to_db:
            S = self._amp_to_db(S, spec_gain=self.spec_gain)
        return S

    def _build_mel_basis(self):
        mel_basis = librosa.filters.mel(
            self.sample_rate,
            self.n_fft,
            n_mels=self.n_mels,
            fmin=self.mel_fmin,
            fmax=self.mel_fmax,
            htk=self.use_htk,
            norm=self.mel_norm,
        )
        self.mel_basis = torch.from_numpy(mel_basis).float()

    @staticmethod
    def _amp_to_db(x, spec_gain=1.0):
        return torch.log(torch.clamp(x, min=1e-5) * spec_gain)

    @staticmethod
    def _db_to_amp(x, spec_gain=1.0):
        return torch.exp(x) / spec_gain


# pylint: disable=too-many-public-methods
[docs]class AudioProcessor(object): """Audio Processor for TTS used by all the data pipelines. TODO: Make this a dataclass to replace `BaseAudioConfig`. Note: All the class arguments are set to default values to enable a flexible initialization of the class with the model config. They are not meaningful for all the arguments. Args: sample_rate (int, optional): target audio sampling rate. Defaults to None. resample (bool, optional): enable/disable resampling of the audio clips when the target sampling rate does not match the original sampling rate. Defaults to False. num_mels (int, optional): number of melspectrogram dimensions. Defaults to None. log_func (int, optional): log exponent used for converting spectrogram aplitude to DB. min_level_db (int, optional): minimum db threshold for the computed melspectrograms. Defaults to None. frame_shift_ms (int, optional): milliseconds of frames between STFT columns. Defaults to None. frame_length_ms (int, optional): milliseconds of STFT window length. Defaults to None. hop_length (int, optional): number of frames between STFT columns. Used if ```frame_shift_ms``` is None. Defaults to None. win_length (int, optional): STFT window length. Used if ```frame_length_ms``` is None. Defaults to None. ref_level_db (int, optional): reference DB level to avoid background noise. In general <20DB corresponds to the air noise. Defaults to None. fft_size (int, optional): FFT window size for STFT. Defaults to 1024. power (int, optional): Exponent value applied to the spectrogram before GriffinLim. Defaults to None. preemphasis (float, optional): Preemphasis coefficient. Preemphasis is disabled if == 0.0. Defaults to 0.0. signal_norm (bool, optional): enable/disable signal normalization. Defaults to None. symmetric_norm (bool, optional): enable/disable symmetric normalization. If set True normalization is performed in the range [-k, k] else [0, k], Defaults to None. max_norm (float, optional): ```k``` defining the normalization range. Defaults to None. mel_fmin (int, optional): minimum filter frequency for computing melspectrograms. Defaults to None. mel_fmax (int, optional): maximum filter frequency for computing melspectrograms. Defaults to None. pitch_fmin (int, optional): minimum filter frequency for computing pitch. Defaults to None. pitch_fmax (int, optional): maximum filter frequency for computing pitch. Defaults to None. spec_gain (int, optional): gain applied when converting amplitude to DB. Defaults to 20. stft_pad_mode (str, optional): Padding mode for STFT. Defaults to 'reflect'. clip_norm (bool, optional): enable/disable clipping the our of range values in the normalized audio signal. Defaults to True. griffin_lim_iters (int, optional): Number of GriffinLim iterations. Defaults to None. do_trim_silence (bool, optional): enable/disable silence trimming when loading the audio signal. Defaults to False. trim_db (int, optional): DB threshold used for silence trimming. Defaults to 60. do_sound_norm (bool, optional): enable/disable signal normalization. Defaults to False. do_amp_to_db_linear (bool, optional): enable/disable amplitude to dB conversion of linear spectrograms. Defaults to True. do_amp_to_db_mel (bool, optional): enable/disable amplitude to dB conversion of mel spectrograms. Defaults to True. do_rms_norm (bool, optional): enable/disable RMS volume normalization when loading an audio file. Defaults to False. db_level (int, optional): dB level used for rms normalization. The range is -99 to 0. Defaults to None. stats_path (str, optional): Path to the computed stats file. Defaults to None. verbose (bool, optional): enable/disable logging. Defaults to True. """ def __init__( self, sample_rate=None, resample=False, num_mels=None, log_func="np.log10", min_level_db=None, frame_shift_ms=None, frame_length_ms=None, hop_length=None, win_length=None, ref_level_db=None, fft_size=1024, power=None, preemphasis=0.0, signal_norm=None, symmetric_norm=None, max_norm=None, mel_fmin=None, mel_fmax=None, pitch_fmax=None, pitch_fmin=None, spec_gain=20, stft_pad_mode="reflect", clip_norm=True, griffin_lim_iters=None, do_trim_silence=False, trim_db=60, do_sound_norm=False, do_amp_to_db_linear=True, do_amp_to_db_mel=True, do_rms_norm=False, db_level=None, stats_path=None, verbose=True, **_, ): # setup class attributed self.sample_rate = sample_rate self.resample = resample self.num_mels = num_mels self.log_func = log_func self.min_level_db = min_level_db or 0 self.frame_shift_ms = frame_shift_ms self.frame_length_ms = frame_length_ms self.ref_level_db = ref_level_db self.fft_size = fft_size self.power = power self.preemphasis = preemphasis self.griffin_lim_iters = griffin_lim_iters self.signal_norm = signal_norm self.symmetric_norm = symmetric_norm self.mel_fmin = mel_fmin or 0 self.mel_fmax = mel_fmax self.pitch_fmin = pitch_fmin self.pitch_fmax = pitch_fmax self.spec_gain = float(spec_gain) self.stft_pad_mode = stft_pad_mode self.max_norm = 1.0 if max_norm is None else float(max_norm) self.clip_norm = clip_norm self.do_trim_silence = do_trim_silence self.trim_db = trim_db self.do_sound_norm = do_sound_norm self.do_amp_to_db_linear = do_amp_to_db_linear self.do_amp_to_db_mel = do_amp_to_db_mel self.do_rms_norm = do_rms_norm self.db_level = db_level self.stats_path = stats_path # setup exp_func for db to amp conversion if log_func == "np.log": self.base = np.e elif log_func == "np.log10": self.base = 10 else: raise ValueError(" [!] unknown `log_func` value.") # setup stft parameters if hop_length is None: # compute stft parameters from given time values self.hop_length, self.win_length = self._stft_parameters() else: # use stft parameters from config file self.hop_length = hop_length self.win_length = win_length assert min_level_db != 0.0, " [!] min_level_db is 0" assert ( self.win_length <= self.fft_size ), f" [!] win_length cannot be larger than fft_size - {self.win_length} vs {self.fft_size}" members = vars(self) if verbose: print(" > Setting up Audio Processor...") for key, value in members.items(): print(" | > {}:{}".format(key, value)) # create spectrogram utils self.mel_basis = self._build_mel_basis() self.inv_mel_basis = np.linalg.pinv(self._build_mel_basis()) # setup scaler if stats_path and signal_norm: mel_mean, mel_std, linear_mean, linear_std, _ = self.load_stats(stats_path) self.setup_scaler(mel_mean, mel_std, linear_mean, linear_std) self.signal_norm = True self.max_norm = None self.clip_norm = None self.symmetric_norm = None @staticmethod def init_from_config(config: "Coqpit", verbose=True): if "audio" in config: return AudioProcessor(verbose=verbose, **config.audio) return AudioProcessor(verbose=verbose, **config) ### setting up the parameters ### def _build_mel_basis( self, ) -> np.ndarray: """Build melspectrogram basis. Returns: np.ndarray: melspectrogram basis. """ if self.mel_fmax is not None: assert self.mel_fmax <= self.sample_rate // 2 return librosa.filters.mel( self.sample_rate, self.fft_size, n_mels=self.num_mels, fmin=self.mel_fmin, fmax=self.mel_fmax ) def _stft_parameters( self, ) -> Tuple[int, int]: """Compute the real STFT parameters from the time values. Returns: Tuple[int, int]: hop length and window length for STFT. """ factor = self.frame_length_ms / self.frame_shift_ms assert (factor).is_integer(), " [!] frame_shift_ms should divide frame_length_ms" hop_length = int(self.frame_shift_ms / 1000.0 * self.sample_rate) win_length = int(hop_length * factor) return hop_length, win_length ### normalization ###
[docs] def normalize(self, S: np.ndarray) -> np.ndarray: """Normalize values into `[0, self.max_norm]` or `[-self.max_norm, self.max_norm]` Args: S (np.ndarray): Spectrogram to normalize. Raises: RuntimeError: Mean and variance is computed from incompatible parameters. Returns: np.ndarray: Normalized spectrogram. """ # pylint: disable=no-else-return S = S.copy() if self.signal_norm: # mean-var scaling if hasattr(self, "mel_scaler"): if S.shape[0] == self.num_mels: return self.mel_scaler.transform(S.T).T elif S.shape[0] == self.fft_size / 2: return self.linear_scaler.transform(S.T).T else: raise RuntimeError(" [!] Mean-Var stats does not match the given feature dimensions.") # range normalization S -= self.ref_level_db # discard certain range of DB assuming it is air noise S_norm = (S - self.min_level_db) / (-self.min_level_db) if self.symmetric_norm: S_norm = ((2 * self.max_norm) * S_norm) - self.max_norm if self.clip_norm: S_norm = np.clip( S_norm, -self.max_norm, self.max_norm # pylint: disable=invalid-unary-operand-type ) return S_norm else: S_norm = self.max_norm * S_norm if self.clip_norm: S_norm = np.clip(S_norm, 0, self.max_norm) return S_norm else: return S
[docs] def denormalize(self, S: np.ndarray) -> np.ndarray: """Denormalize spectrogram values. Args: S (np.ndarray): Spectrogram to denormalize. Raises: RuntimeError: Mean and variance are incompatible. Returns: np.ndarray: Denormalized spectrogram. """ # pylint: disable=no-else-return S_denorm = S.copy() if self.signal_norm: # mean-var scaling if hasattr(self, "mel_scaler"): if S_denorm.shape[0] == self.num_mels: return self.mel_scaler.inverse_transform(S_denorm.T).T elif S_denorm.shape[0] == self.fft_size / 2: return self.linear_scaler.inverse_transform(S_denorm.T).T else: raise RuntimeError(" [!] Mean-Var stats does not match the given feature dimensions.") if self.symmetric_norm: if self.clip_norm: S_denorm = np.clip( S_denorm, -self.max_norm, self.max_norm # pylint: disable=invalid-unary-operand-type ) S_denorm = ((S_denorm + self.max_norm) * -self.min_level_db / (2 * self.max_norm)) + self.min_level_db return S_denorm + self.ref_level_db else: if self.clip_norm: S_denorm = np.clip(S_denorm, 0, self.max_norm) S_denorm = (S_denorm * -self.min_level_db / self.max_norm) + self.min_level_db return S_denorm + self.ref_level_db else: return S_denorm
### Mean-STD scaling ###
[docs] def load_stats(self, stats_path: str) -> Tuple[np.array, np.array, np.array, np.array, Dict]: """Loading mean and variance statistics from a `npy` file. Args: stats_path (str): Path to the `npy` file containing Returns: Tuple[np.array, np.array, np.array, np.array, Dict]: loaded statistics and the config used to compute them. """ stats = np.load(stats_path, allow_pickle=True).item() # pylint: disable=unexpected-keyword-arg mel_mean = stats["mel_mean"] mel_std = stats["mel_std"] linear_mean = stats["linear_mean"] linear_std = stats["linear_std"] stats_config = stats["audio_config"] # check all audio parameters used for computing stats skip_parameters = ["griffin_lim_iters", "stats_path", "do_trim_silence", "ref_level_db", "power"] for key in stats_config.keys(): if key in skip_parameters: continue if key not in ["sample_rate", "trim_db"]: assert ( stats_config[key] == self.__dict__[key] ), f" [!] Audio param {key} does not match the value used for computing mean-var stats. {stats_config[key]} vs {self.__dict__[key]}" return mel_mean, mel_std, linear_mean, linear_std, stats_config
# pylint: disable=attribute-defined-outside-init
[docs] def setup_scaler( self, mel_mean: np.ndarray, mel_std: np.ndarray, linear_mean: np.ndarray, linear_std: np.ndarray ) -> None: """Initialize scaler objects used in mean-std normalization. Args: mel_mean (np.ndarray): Mean for melspectrograms. mel_std (np.ndarray): STD for melspectrograms. linear_mean (np.ndarray): Mean for full scale spectrograms. linear_std (np.ndarray): STD for full scale spectrograms. """ self.mel_scaler = StandardScaler() self.mel_scaler.set_stats(mel_mean, mel_std) self.linear_scaler = StandardScaler() self.linear_scaler.set_stats(linear_mean, linear_std)
### DB and AMP conversion ### # pylint: disable=no-self-use def _amp_to_db(self, x: np.ndarray) -> np.ndarray: """Convert amplitude values to decibels. Args: x (np.ndarray): Amplitude spectrogram. Returns: np.ndarray: Decibels spectrogram. """ return self.spec_gain * _log(np.maximum(1e-5, x), self.base) # pylint: disable=no-self-use def _db_to_amp(self, x: np.ndarray) -> np.ndarray: """Convert decibels spectrogram to amplitude spectrogram. Args: x (np.ndarray): Decibels spectrogram. Returns: np.ndarray: Amplitude spectrogram. """ return _exp(x / self.spec_gain, self.base) ### Preemphasis ###
[docs] def apply_preemphasis(self, x: np.ndarray) -> 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 self.preemphasis == 0: raise RuntimeError(" [!] Preemphasis is set 0.0.") return scipy.signal.lfilter([1, -self.preemphasis], [1], x)
[docs] def apply_inv_preemphasis(self, x: np.ndarray) -> np.ndarray: """Reverse pre-emphasis.""" if self.preemphasis == 0: raise RuntimeError(" [!] Preemphasis is set 0.0.") return scipy.signal.lfilter([1], [1, -self.preemphasis], x)
### SPECTROGRAMs ### def _linear_to_mel(self, spectrogram: np.ndarray) -> np.ndarray: """Project a full scale spectrogram to a melspectrogram. Args: spectrogram (np.ndarray): Full scale spectrogram. Returns: np.ndarray: Melspectrogram """ return np.dot(self.mel_basis, spectrogram) def _mel_to_linear(self, mel_spec: np.ndarray) -> np.ndarray: """Convert a melspectrogram to full scale spectrogram.""" return np.maximum(1e-10, np.dot(self.inv_mel_basis, mel_spec))
[docs] def spectrogram(self, y: np.ndarray) -> np.ndarray: """Compute a spectrogram from a waveform. Args: y (np.ndarray): Waveform. Returns: np.ndarray: Spectrogram. """ if self.preemphasis != 0: D = self._stft(self.apply_preemphasis(y)) else: D = self._stft(y) if self.do_amp_to_db_linear: S = self._amp_to_db(np.abs(D)) else: S = np.abs(D) return self.normalize(S).astype(np.float32)
[docs] def melspectrogram(self, y: np.ndarray) -> np.ndarray: """Compute a melspectrogram from a waveform.""" if self.preemphasis != 0: D = self._stft(self.apply_preemphasis(y)) else: D = self._stft(y) if self.do_amp_to_db_mel: S = self._amp_to_db(self._linear_to_mel(np.abs(D))) else: S = self._linear_to_mel(np.abs(D)) return self.normalize(S).astype(np.float32)
[docs] def inv_spectrogram(self, spectrogram: np.ndarray) -> np.ndarray: """Convert a spectrogram to a waveform using Griffi-Lim vocoder.""" S = self.denormalize(spectrogram) S = self._db_to_amp(S) # Reconstruct phase if self.preemphasis != 0: return self.apply_inv_preemphasis(self._griffin_lim(S**self.power)) return self._griffin_lim(S**self.power)
[docs] def inv_melspectrogram(self, mel_spectrogram: np.ndarray) -> np.ndarray: """Convert a melspectrogram to a waveform using Griffi-Lim vocoder.""" D = self.denormalize(mel_spectrogram) S = self._db_to_amp(D) S = self._mel_to_linear(S) # Convert back to linear if self.preemphasis != 0: return self.apply_inv_preemphasis(self._griffin_lim(S**self.power)) return self._griffin_lim(S**self.power)
[docs] def out_linear_to_mel(self, linear_spec: np.ndarray) -> np.ndarray: """Convert a full scale linear spectrogram output of a network to a melspectrogram. Args: linear_spec (np.ndarray): Normalized full scale linear spectrogram. Returns: np.ndarray: Normalized melspectrogram. """ S = self.denormalize(linear_spec) S = self._db_to_amp(S) S = self._linear_to_mel(np.abs(S)) S = self._amp_to_db(S) mel = self.normalize(S) return mel
### STFT and ISTFT ### def _stft(self, y: np.ndarray) -> np.ndarray: """Librosa STFT wrapper. Args: y (np.ndarray): Audio signal. Returns: np.ndarray: Complex number array. """ return librosa.stft( y=y, n_fft=self.fft_size, hop_length=self.hop_length, win_length=self.win_length, pad_mode=self.stft_pad_mode, window="hann", center=True, ) def _istft(self, y: np.ndarray) -> np.ndarray: """Librosa iSTFT wrapper.""" return librosa.istft(y, hop_length=self.hop_length, win_length=self.win_length) def _griffin_lim(self, S): angles = np.exp(2j * np.pi * np.random.rand(*S.shape)) S_complex = np.abs(S).astype(np.complex) y = self._istft(S_complex * angles) if not np.isfinite(y).all(): print(" [!] Waveform is not finite everywhere. Skipping the GL.") return np.array([0.0]) for _ in range(self.griffin_lim_iters): angles = np.exp(1j * np.angle(self._stft(y))) y = self._istft(S_complex * angles) return y
[docs] def compute_stft_paddings(self, x, pad_sides=1): """Compute paddings used by Librosa's STFT. Compute right padding (final frame) or both sides padding (first and final frames)""" assert pad_sides in (1, 2) pad = (x.shape[0] // self.hop_length + 1) * self.hop_length - x.shape[0] if pad_sides == 1: return 0, pad return pad // 2, pad // 2 + pad % 2
[docs] def compute_f0(self, x: np.ndarray) -> np.ndarray: """Compute pitch (f0) of a waveform using the same parameters used for computing melspectrogram. Args: x (np.ndarray): Waveform. Returns: np.ndarray: Pitch. Examples: >>> WAV_FILE = filename = librosa.util.example_audio_file() >>> from TTS.config import BaseAudioConfig >>> from TTS.utils.audio import AudioProcessor >>> conf = BaseAudioConfig(pitch_fmax=8000) >>> ap = AudioProcessor(**conf) >>> wav = ap.load_wav(WAV_FILE, sr=22050)[:5 * 22050] >>> pitch = ap.compute_f0(wav) """ assert self.pitch_fmax is not None, " [!] Set `pitch_fmax` before caling `compute_f0`." # align F0 length to the spectrogram length if len(x) % self.hop_length == 0: x = np.pad(x, (0, self.hop_length // 2), mode="reflect") f0, t = pw.dio( x.astype(np.double), fs=self.sample_rate, f0_ceil=self.pitch_fmax, frame_period=1000 * self.hop_length / self.sample_rate, ) f0 = pw.stonemask(x.astype(np.double), f0, t, self.sample_rate) return f0
### Audio Processing ###
[docs] def find_endpoint(self, wav: np.ndarray, min_silence_sec=0.8) -> 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. Returns: int: Last point without silence. """ window_length = int(self.sample_rate * min_silence_sec) hop_length = int(window_length / 4) threshold = self._db_to_amp(-self.trim_db) 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)
[docs] def trim_silence(self, wav): """Trim silent parts with a threshold and 0.01 sec margin""" margin = int(self.sample_rate * 0.01) wav = wav[margin:-margin] return librosa.effects.trim(wav, top_db=self.trim_db, frame_length=self.win_length, hop_length=self.hop_length)[ 0 ]
[docs] @staticmethod def sound_norm(x: np.ndarray) -> np.ndarray: """Normalize the volume of an audio signal. Args: x (np.ndarray): Raw waveform. Returns: np.ndarray: Volume normalized waveform. """ return x / abs(x).max() * 0.95
@staticmethod def _rms_norm(wav, db_level=-27): r = 10 ** (db_level / 20) a = np.sqrt((len(wav) * (r**2)) / np.sum(wav**2)) return wav * a
[docs] def rms_volume_norm(self, x: np.ndarray, db_level: float = None) -> np.ndarray: """Normalize the volume based on RMS of the signal. Args: x (np.ndarray): Raw waveform. Returns: np.ndarray: RMS normalized waveform. """ if db_level is None: db_level = self.db_level assert -99 <= db_level <= 0, " [!] db_level should be between -99 and 0" wav = self._rms_norm(x, db_level) return wav
### save and load ###
[docs] def load_wav(self, filename: str, sr: int = None) -> 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. Returns: np.ndarray: Loaded waveform. """ if self.resample: # loading with resampling. It is significantly slower. x, sr = librosa.load(filename, sr=self.sample_rate) elif sr is None: # SF is faster than librosa for loading files x, sr = sf.read(filename) assert self.sample_rate == sr, "%s vs %s" % (self.sample_rate, sr) else: x, sr = librosa.load(filename, sr=sr) if self.do_trim_silence: try: x = self.trim_silence(x) except ValueError: print(f" [!] File cannot be trimmed for silence - {filename}") if self.do_sound_norm: x = self.sound_norm(x) if self.do_rms_norm: x = self.rms_volume_norm(x, self.db_level) return x
[docs] def save_wav(self, wav: np.ndarray, path: str, sr: int = None) -> None: """Save a waveform to a file using Scipy. Args: wav (np.ndarray): Waveform to save. path (str): Path to a output file. sr (int, optional): Sampling rate used for saving to the file. Defaults to None. """ wav_norm = wav * (32767 / max(0.01, np.max(np.abs(wav)))) scipy.io.wavfile.write(path, sr if sr else self.sample_rate, wav_norm.astype(np.int16))
[docs] def get_duration(self, filename: str) -> float: """Get the duration of a wav file using Librosa. Args: filename (str): Path to the wav file. """ return librosa.get_duration(filename)
@staticmethod def mulaw_encode(wav: np.ndarray, qc: int) -> np.ndarray: mu = 2**qc - 1 # wav_abs = np.minimum(np.abs(wav), 1.0) signal = np.sign(wav) * np.log(1 + mu * np.abs(wav)) / np.log(1.0 + mu) # Quantize signal to the specified number of levels. signal = (signal + 1) / 2 * mu + 0.5 return np.floor( signal, )
[docs] @staticmethod def mulaw_decode(wav, qc): """Recovers waveform from quantized values.""" mu = 2**qc - 1 x = np.sign(wav) / mu * ((1 + mu) ** np.abs(wav) - 1) return x
@staticmethod def encode_16bits(x): return np.clip(x * 2**15, -(2**15), 2**15 - 1).astype(np.int16)
[docs] @staticmethod def quantize(x: np.ndarray, bits: int) -> 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]`. bits (int): Number of quantization bits. Returns: np.ndarray: Quantized waveform. """ return (x + 1.0) * (2**bits - 1) / 2
[docs] @staticmethod def dequantize(x, bits): """Dequantize a waveform from the given number of bits.""" return 2 * x / (2**bits - 1) - 1
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)