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from scipy.ndimage.morphology import binary_dilation
from speaker_encoder.params_data import *
from pathlib import Path
from typing import Optional, Union
import numpy as np
import webrtcvad
import librosa
import struct
int16_max = (2 ** 15) - 1
def preprocess_wav(fpath_or_wav: Union[str, Path, np.ndarray],
source_sr: Optional[int] = None):
"""
Applies the preprocessing operations used in training the Speaker Encoder to a waveform
either on disk or in memory. The waveform will be resampled to match the data hyperparameters.
:param fpath_or_wav: either a filepath to an audio file (many extensions are supported, not
just .wav), either the waveform as a numpy array of floats.
:param source_sr: if passing an audio waveform, the sampling rate of the waveform before
preprocessing. After preprocessing, the waveform's sampling rate will match the data
hyperparameters. If passing a filepath, the sampling rate will be automatically detected and
this argument will be ignored.
"""
# Load the wav from disk if needed
if isinstance(fpath_or_wav, str) or isinstance(fpath_or_wav, Path):
wav, source_sr = librosa.load(fpath_or_wav, sr=None)
else:
wav = fpath_or_wav
# Resample the wav if needed
if source_sr is not None and source_sr != sampling_rate:
wav = librosa.resample(wav, source_sr, sampling_rate)
# Apply the preprocessing: normalize volume and shorten long silences
wav = normalize_volume(wav, audio_norm_target_dBFS, increase_only=True)
wav = trim_long_silences(wav)
return wav
def wav_to_mel_spectrogram(wav):
"""
Derives a mel spectrogram ready to be used by the encoder from a preprocessed audio waveform.
Note: this not a log-mel spectrogram.
"""
frames = librosa.feature.melspectrogram(
y=wav,
sr=sampling_rate,
n_fft=int(sampling_rate * mel_window_length / 1000),
hop_length=int(sampling_rate * mel_window_step / 1000),
n_mels=mel_n_channels
)
return frames.astype(np.float32).T
def trim_long_silences(wav):
"""
Ensures that segments without voice in the waveform remain no longer than a
threshold determined by the VAD parameters in params.py.
:param wav: the raw waveform as a numpy array of floats
:return: the same waveform with silences trimmed away (length <= original wav length)
"""
# Compute the voice detection window size
samples_per_window = (vad_window_length * sampling_rate) // 1000
# Trim the end of the audio to have a multiple of the window size
wav = wav[:len(wav) - (len(wav) % samples_per_window)]
# Convert the float waveform to 16-bit mono PCM
pcm_wave = struct.pack("%dh" % len(wav), *(np.round(wav * int16_max)).astype(np.int16))
# Perform voice activation detection
voice_flags = []
vad = webrtcvad.Vad(mode=3)
for window_start in range(0, len(wav), samples_per_window):
window_end = window_start + samples_per_window
voice_flags.append(vad.is_speech(pcm_wave[window_start * 2:window_end * 2],
sample_rate=sampling_rate))
voice_flags = np.array(voice_flags)
# Smooth the voice detection with a moving average
def moving_average(array, width):
array_padded = np.concatenate((np.zeros((width - 1) // 2), array, np.zeros(width // 2)))
ret = np.cumsum(array_padded, dtype=float)
ret[width:] = ret[width:] - ret[:-width]
return ret[width - 1:] / width
audio_mask = moving_average(voice_flags, vad_moving_average_width)
audio_mask = np.round(audio_mask).astype(np.bool)
# Dilate the voiced regions
audio_mask = binary_dilation(audio_mask, np.ones(vad_max_silence_length + 1))
audio_mask = np.repeat(audio_mask, samples_per_window)
return wav[audio_mask == True]
def normalize_volume(wav, target_dBFS, increase_only=False, decrease_only=False):
if increase_only and decrease_only:
raise ValueError("Both increase only and decrease only are set")
dBFS_change = target_dBFS - 10 * np.log10(np.mean(wav ** 2))
if (dBFS_change < 0 and increase_only) or (dBFS_change > 0 and decrease_only):
return wav
return wav * (10 ** (dBFS_change / 20))
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