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from io import BytesIO | |
from typing import Dict, Tuple | |
import librosa | |
import numpy as np | |
import scipy.io.wavfile | |
import scipy.signal | |
from TTS.tts.utils.helpers import StandardScaler | |
from TTS.utils.audio.numpy_transforms import ( | |
amp_to_db, | |
build_mel_basis, | |
compute_f0, | |
db_to_amp, | |
deemphasis, | |
find_endpoint, | |
griffin_lim, | |
load_wav, | |
mel_to_spec, | |
millisec_to_length, | |
preemphasis, | |
rms_volume_norm, | |
spec_to_mel, | |
stft, | |
trim_silence, | |
volume_norm, | |
) | |
# pylint: disable=too-many-public-methods | |
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, | |
**_, | |
): | |
# 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.win_length, self.hop_length = millisec_to_length( | |
frame_length_ms=self.frame_length_ms, frame_shift_ms=self.frame_shift_ms, sample_rate=self.sample_rate | |
) | |
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 = build_mel_basis( | |
sample_rate=self.sample_rate, | |
fft_size=self.fft_size, | |
num_mels=self.num_mels, | |
mel_fmax=self.mel_fmax, | |
mel_fmin=self.mel_fmin, | |
) | |
# 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 | |
def init_from_config(config: "Coqpit", verbose=True): | |
if "audio" in config: | |
return AudioProcessor(verbose=verbose, **config.audio) | |
return AudioProcessor(verbose=verbose, **config) | |
### normalization ### | |
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 | |
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 ### | |
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 | |
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) | |
### Preemphasis ### | |
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. | |
""" | |
return preemphasis(x=x, coef=self.preemphasis) | |
def apply_inv_preemphasis(self, x: np.ndarray) -> np.ndarray: | |
"""Reverse pre-emphasis.""" | |
return deemphasis(x=x, coef=self.preemphasis) | |
### SPECTROGRAMs ### | |
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: | |
y = self.apply_preemphasis(y) | |
D = stft( | |
y=y, | |
fft_size=self.fft_size, | |
hop_length=self.hop_length, | |
win_length=self.win_length, | |
pad_mode=self.stft_pad_mode, | |
) | |
if self.do_amp_to_db_linear: | |
S = amp_to_db(x=np.abs(D), gain=self.spec_gain, base=self.base) | |
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: | |
y = self.apply_preemphasis(y) | |
D = stft( | |
y=y, | |
fft_size=self.fft_size, | |
hop_length=self.hop_length, | |
win_length=self.win_length, | |
pad_mode=self.stft_pad_mode, | |
) | |
S = spec_to_mel(spec=np.abs(D), mel_basis=self.mel_basis) | |
if self.do_amp_to_db_mel: | |
S = amp_to_db(x=S, gain=self.spec_gain, base=self.base) | |
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 = db_to_amp(x=S, gain=self.spec_gain, base=self.base) | |
# Reconstruct phase | |
W = self._griffin_lim(S**self.power) | |
return self.apply_inv_preemphasis(W) if self.preemphasis != 0 else W | |
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 = db_to_amp(x=D, gain=self.spec_gain, base=self.base) | |
S = mel_to_spec(mel=S, mel_basis=self.mel_basis) # Convert back to linear | |
W = self._griffin_lim(S**self.power) | |
return self.apply_inv_preemphasis(W) if self.preemphasis != 0 else W | |
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 = db_to_amp(x=S, gain=self.spec_gain, base=self.base) | |
S = spec_to_mel(spec=np.abs(S), mel_basis=self.mel_basis) | |
S = amp_to_db(x=S, gain=self.spec_gain, base=self.base) | |
mel = self.normalize(S) | |
return mel | |
def _griffin_lim(self, S): | |
return griffin_lim( | |
spec=S, | |
num_iter=self.griffin_lim_iters, | |
hop_length=self.hop_length, | |
win_length=self.win_length, | |
fft_size=self.fft_size, | |
pad_mode=self.stft_pad_mode, | |
) | |
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) | |
""" | |
# align F0 length to the spectrogram length | |
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 | |
### Audio Processing ### | |
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. | |
""" | |
return find_endpoint( | |
wav=wav, | |
trim_db=self.trim_db, | |
sample_rate=self.sample_rate, | |
min_silence_sec=min_silence_sec, | |
gain=self.spec_gain, | |
base=self.base, | |
) | |
def trim_silence(self, wav): | |
"""Trim silent parts with a threshold and 0.01 sec margin""" | |
return trim_silence( | |
wav=wav, | |
sample_rate=self.sample_rate, | |
trim_db=self.trim_db, | |
win_length=self.win_length, | |
hop_length=self.hop_length, | |
) | |
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 volume_norm(x=x) | |
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 | |
return rms_volume_norm(x=x, db_level=db_level) | |
### save and load ### | |
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 sr is not None: | |
x = load_wav(filename=filename, sample_rate=sr, resample=True) | |
else: | |
x = load_wav(filename=filename, sample_rate=self.sample_rate, resample=self.resample) | |
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, pipe_out=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. | |
pipe_out (BytesIO, optional): Flag to stdout the generated TTS wav file for shell pipe. | |
""" | |
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)))) | |
wav_norm = wav_norm.astype(np.int16) | |
if pipe_out: | |
wav_buffer = BytesIO() | |
scipy.io.wavfile.write(wav_buffer, sr if sr else self.sample_rate, wav_norm) | |
wav_buffer.seek(0) | |
pipe_out.buffer.write(wav_buffer.read()) | |
scipy.io.wavfile.write(path, sr if sr else self.sample_rate, wav_norm) | |
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=filename) | |