| from typing import Dict, Tuple |
|
|
| import librosa |
| import numpy as np |
| import scipy.io.wavfile |
| import scipy.signal |
| import soundfile as sf |
|
|
| from TTS.tts.utils.helpers import StandardScaler |
| from TTS.utils.audio.numpy_transforms import compute_f0 |
|
|
| |
|
|
|
|
| class AudioProcessor(object): |
| """Audio Processor for TTS. |
| |
| 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, |
| **_, |
| ): |
| |
| 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 |
| |
| if log_func == "np.log": |
| self.base = np.e |
| elif log_func == "np.log10": |
| self.base = 10 |
| else: |
| raise ValueError(" [!] unknown `log_func` value.") |
| |
| if hop_length is None: |
| |
| self.hop_length, self.win_length = self._stft_parameters() |
| else: |
| |
| 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)) |
| |
| self.mel_basis = self._build_mel_basis() |
| self.inv_mel_basis = np.linalg.pinv(self._build_mel_basis()) |
| |
| 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) |
|
|
| |
| 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( |
| sr=self.sample_rate, n_fft=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 |
|
|
| |
| 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. |
| """ |
| |
| S = S.copy() |
| if self.signal_norm: |
| |
| 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.") |
| |
| S -= self.ref_level_db |
| 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 |
| ) |
| 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 |
|
|
| 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. |
| """ |
| |
| S_denorm = S.copy() |
| if self.signal_norm: |
| |
| 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 |
| ) |
| 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 |
|
|
| |
| 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() |
| 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"] |
| |
| 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 |
|
|
| |
| 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) |
|
|
| |
| |
| 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) |
|
|
| |
| 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) |
|
|
| |
| 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) |
|
|
| 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) |
|
|
| |
| 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)) |
|
|
| 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) |
|
|
| 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) |
|
|
| 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) |
| |
| if self.preemphasis != 0: |
| return self.apply_inv_preemphasis(self._griffin_lim(S**self.power)) |
| return self._griffin_lim(S**self.power) |
|
|
| 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) |
| if self.preemphasis != 0: |
| return self.apply_inv_preemphasis(self._griffin_lim(S**self.power)) |
| return self._griffin_lim(S**self.power) |
|
|
| 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 |
|
|
| |
| 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 |
|
|
| 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 |
|
|
| 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.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 self.pitch_fmax is not None, " [!] Set `pitch_fmax` before caling `compute_f0`." |
| assert self.pitch_fmin is not None, " [!] Set `pitch_fmin` before caling `compute_f0`." |
| |
| if len(x) % self.hop_length == 0: |
| x = np.pad(x, (0, self.hop_length // 2), mode=self.stft_pad_mode) |
|
|
| f0 = compute_f0( |
| x=x, |
| pitch_fmax=self.pitch_fmax, |
| pitch_fmin=self.pitch_fmin, |
| hop_length=self.hop_length, |
| win_length=self.win_length, |
| sample_rate=self.sample_rate, |
| stft_pad_mode=self.stft_pad_mode, |
| center=True, |
| ) |
|
|
| return f0 |
|
|
| |
| 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) |
|
|
| 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 |
| ] |
|
|
| @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 |
|
|
| 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 |
|
|
| |
| 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: |
| |
| x, sr = librosa.load(filename, sr=self.sample_rate) |
| elif sr is None: |
| |
| 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 |
|
|
| 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. |
| """ |
| if self.do_rms_norm: |
| wav_norm = self.rms_volume_norm(wav, self.db_level) * 32767 |
| else: |
| 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)) |
|
|
| 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(path=filename) |
|
|
| @staticmethod |
| def mulaw_encode(wav: np.ndarray, qc: int) -> np.ndarray: |
| mu = 2**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, |
| ) |
|
|
| @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) |
|
|
| @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 |
|
|
| @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) |
|
|