xTTS-fr-cpu / TTS /utils /audio /processor.py
Shadhil's picture
voice-clone with single audio sample input
9b2107c
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
@staticmethod
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,
)
@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 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)