import os import librosa import pickle import numpy as np from scipy import signal _mel_basis = None class AudioProcessor(object): def __init__(self, sample_rate, num_mels, min_level_db, frame_shift_ms, frame_length_ms, preemphasis, ref_level_db, num_freq, power, griffin_lim_iters=None): self.sample_rate = sample_rate self.num_mels = num_mels self.min_level_db = min_level_db self.frame_shift_ms = frame_shift_ms self.frame_length_ms = frame_length_ms self.preemphasis = preemphasis self.ref_level_db = ref_level_db self.num_freq = num_freq self.power = power self.griffin_lim_iters = griffin_lim_iters def save_wav(self, wav, path): wav *= 32767 / max(0.01, np.max(np.abs(wav))) librosa.output.write_wav(path, wav.astype(np.int16), self.sample_rate) def _linear_to_mel(self, spectrogram): global _mel_basis if _mel_basis is None: _mel_basis = self._build_mel_basis() return np.dot(_mel_basis, spectrogram) def _build_mel_basis(self, ): n_fft = (self.num_freq - 1) * 2 return librosa.filters.mel(self.sample_rate, n_fft, n_mels=self.num_mels) def _normalize(self, S): return np.clip((S - self.min_level_db) / -self.min_level_db, 0, 1) def _denormalize(self, S): return (np.clip(S, 0, 1) * -self.min_level_db) + self.min_level_db def _stft_parameters(self, ): n_fft = (self.num_freq - 1) * 2 hop_length = int(self.frame_shift_ms / 1000 * self.sample_rate) win_length = int(self.frame_length_ms / 1000 * self.sample_rate) return n_fft, hop_length, win_length def _amp_to_db(self, x): return 20 * np.log10(np.maximum(1e-5, x)) def _db_to_amp(self, x): return np.power(10.0, x * 0.05) def apply_preemphasis(self, x): return signal.lfilter([1, -self.preemphasis], [1], x) def apply_inv_preemphasis(self, x): return signal.lfilter([1], [1, -self.preemphasis], x) def spectrogram(self, y): D = self._stft(self.apply_preemphasis(y)) S = self._amp_to_db(np.abs(D)) - self.ref_level_db return self._normalize(S) def inv_spectrogram(self, spectrogram): '''Converts spectrogram to waveform using librosa''' S = self._denormalize(spectrogram) S = self._db_to_amp(S + self.ref_level_db) # Convert back to linear # Reconstruct phase return self.apply_inv_preemphasis(self._griffin_lim(S ** self.power)) def _griffin_lim(self, S): '''librosa implementation of Griffin-Lim Based on https://github.com/librosa/librosa/issues/434 ''' 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) for i in range(self.griffin_lim_iters): angles = np.exp(1j * np.angle(self._stft(y))) y = self._istft(S_complex * angles) return y def melspectrogram(self, y): D = self._stft(self.apply_preemphasis(y)) S = self._amp_to_db(self._linear_to_mel(np.abs(D))) - self.ref_level_db return self._normalize(S) def _stft(self, y): n_fft, hop_length, win_length = self._stft_parameters() return librosa.stft(y=y, n_fft=n_fft, hop_length=hop_length, win_length=win_length) def _istft(self, y): _, hop_length, win_length = self._stft_parameters() return librosa.istft(y, hop_length=hop_length, win_length=win_length) def find_endpoint(self, wav, threshold_db=-40, min_silence_sec=0.8): window_length = int(self.sample_rate * min_silence_sec) hop_length = int(window_length / 4) threshold = self._db_to_amp(threshold_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)