Spaces:
Runtime error
Runtime error
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)) | |