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from io import BytesIO |
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from typing import Dict, Tuple |
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import librosa |
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import numpy as np |
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import scipy.io.wavfile |
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import scipy.signal |
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from TTS.tts.utils.helpers import StandardScaler |
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from TTS.utils.audio.numpy_transforms import ( |
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amp_to_db, |
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build_mel_basis, |
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compute_f0, |
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db_to_amp, |
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deemphasis, |
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find_endpoint, |
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griffin_lim, |
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load_wav, |
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mel_to_spec, |
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millisec_to_length, |
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preemphasis, |
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rms_volume_norm, |
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spec_to_mel, |
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stft, |
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trim_silence, |
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volume_norm, |
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) |
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class AudioProcessor(object): |
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"""Audio Processor for TTS. |
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Note: |
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All the class arguments are set to default values to enable a flexible initialization |
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of the class with the model config. They are not meaningful for all the arguments. |
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Args: |
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sample_rate (int, optional): |
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target audio sampling rate. Defaults to None. |
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resample (bool, optional): |
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enable/disable resampling of the audio clips when the target sampling rate does not match the original sampling rate. Defaults to False. |
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num_mels (int, optional): |
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number of melspectrogram dimensions. Defaults to None. |
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log_func (int, optional): |
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log exponent used for converting spectrogram aplitude to DB. |
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min_level_db (int, optional): |
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minimum db threshold for the computed melspectrograms. Defaults to None. |
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frame_shift_ms (int, optional): |
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milliseconds of frames between STFT columns. Defaults to None. |
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frame_length_ms (int, optional): |
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milliseconds of STFT window length. Defaults to None. |
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hop_length (int, optional): |
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number of frames between STFT columns. Used if ```frame_shift_ms``` is None. Defaults to None. |
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win_length (int, optional): |
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STFT window length. Used if ```frame_length_ms``` is None. Defaults to None. |
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ref_level_db (int, optional): |
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reference DB level to avoid background noise. In general <20DB corresponds to the air noise. Defaults to None. |
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fft_size (int, optional): |
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FFT window size for STFT. Defaults to 1024. |
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power (int, optional): |
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Exponent value applied to the spectrogram before GriffinLim. Defaults to None. |
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preemphasis (float, optional): |
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Preemphasis coefficient. Preemphasis is disabled if == 0.0. Defaults to 0.0. |
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signal_norm (bool, optional): |
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enable/disable signal normalization. Defaults to None. |
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symmetric_norm (bool, optional): |
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enable/disable symmetric normalization. If set True normalization is performed in the range [-k, k] else [0, k], Defaults to None. |
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max_norm (float, optional): |
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```k``` defining the normalization range. Defaults to None. |
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mel_fmin (int, optional): |
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minimum filter frequency for computing melspectrograms. Defaults to None. |
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mel_fmax (int, optional): |
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maximum filter frequency for computing melspectrograms. Defaults to None. |
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pitch_fmin (int, optional): |
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minimum filter frequency for computing pitch. Defaults to None. |
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pitch_fmax (int, optional): |
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maximum filter frequency for computing pitch. Defaults to None. |
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spec_gain (int, optional): |
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gain applied when converting amplitude to DB. Defaults to 20. |
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stft_pad_mode (str, optional): |
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Padding mode for STFT. Defaults to 'reflect'. |
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clip_norm (bool, optional): |
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enable/disable clipping the our of range values in the normalized audio signal. Defaults to True. |
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griffin_lim_iters (int, optional): |
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Number of GriffinLim iterations. Defaults to None. |
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do_trim_silence (bool, optional): |
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enable/disable silence trimming when loading the audio signal. Defaults to False. |
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trim_db (int, optional): |
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DB threshold used for silence trimming. Defaults to 60. |
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do_sound_norm (bool, optional): |
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enable/disable signal normalization. Defaults to False. |
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do_amp_to_db_linear (bool, optional): |
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enable/disable amplitude to dB conversion of linear spectrograms. Defaults to True. |
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do_amp_to_db_mel (bool, optional): |
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enable/disable amplitude to dB conversion of mel spectrograms. Defaults to True. |
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do_rms_norm (bool, optional): |
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enable/disable RMS volume normalization when loading an audio file. Defaults to False. |
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db_level (int, optional): |
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dB level used for rms normalization. The range is -99 to 0. Defaults to None. |
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stats_path (str, optional): |
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Path to the computed stats file. Defaults to None. |
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verbose (bool, optional): |
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enable/disable logging. Defaults to True. |
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""" |
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def __init__( |
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self, |
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sample_rate=None, |
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resample=False, |
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num_mels=None, |
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log_func="np.log10", |
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min_level_db=None, |
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frame_shift_ms=None, |
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frame_length_ms=None, |
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hop_length=None, |
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win_length=None, |
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ref_level_db=None, |
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fft_size=1024, |
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power=None, |
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preemphasis=0.0, |
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signal_norm=None, |
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symmetric_norm=None, |
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max_norm=None, |
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mel_fmin=None, |
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mel_fmax=None, |
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pitch_fmax=None, |
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pitch_fmin=None, |
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spec_gain=20, |
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stft_pad_mode="reflect", |
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clip_norm=True, |
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griffin_lim_iters=None, |
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do_trim_silence=False, |
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trim_db=60, |
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do_sound_norm=False, |
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do_amp_to_db_linear=True, |
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do_amp_to_db_mel=True, |
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do_rms_norm=False, |
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db_level=None, |
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stats_path=None, |
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verbose=True, |
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**_, |
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): |
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self.sample_rate = sample_rate |
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self.resample = resample |
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self.num_mels = num_mels |
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self.log_func = log_func |
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self.min_level_db = min_level_db or 0 |
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self.frame_shift_ms = frame_shift_ms |
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self.frame_length_ms = frame_length_ms |
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self.ref_level_db = ref_level_db |
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self.fft_size = fft_size |
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self.power = power |
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self.preemphasis = preemphasis |
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self.griffin_lim_iters = griffin_lim_iters |
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self.signal_norm = signal_norm |
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self.symmetric_norm = symmetric_norm |
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self.mel_fmin = mel_fmin or 0 |
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self.mel_fmax = mel_fmax |
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self.pitch_fmin = pitch_fmin |
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self.pitch_fmax = pitch_fmax |
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self.spec_gain = float(spec_gain) |
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self.stft_pad_mode = stft_pad_mode |
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self.max_norm = 1.0 if max_norm is None else float(max_norm) |
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self.clip_norm = clip_norm |
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self.do_trim_silence = do_trim_silence |
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self.trim_db = trim_db |
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self.do_sound_norm = do_sound_norm |
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self.do_amp_to_db_linear = do_amp_to_db_linear |
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self.do_amp_to_db_mel = do_amp_to_db_mel |
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self.do_rms_norm = do_rms_norm |
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self.db_level = db_level |
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self.stats_path = stats_path |
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if log_func == "np.log": |
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self.base = np.e |
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elif log_func == "np.log10": |
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self.base = 10 |
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else: |
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raise ValueError(" [!] unknown `log_func` value.") |
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if hop_length is None: |
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self.win_length, self.hop_length = millisec_to_length( |
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frame_length_ms=self.frame_length_ms, frame_shift_ms=self.frame_shift_ms, sample_rate=self.sample_rate |
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) |
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else: |
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self.hop_length = hop_length |
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self.win_length = win_length |
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assert min_level_db != 0.0, " [!] min_level_db is 0" |
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assert ( |
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self.win_length <= self.fft_size |
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), f" [!] win_length cannot be larger than fft_size - {self.win_length} vs {self.fft_size}" |
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members = vars(self) |
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if verbose: |
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print(" > Setting up Audio Processor...") |
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for key, value in members.items(): |
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print(" | > {}:{}".format(key, value)) |
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self.mel_basis = build_mel_basis( |
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sample_rate=self.sample_rate, |
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fft_size=self.fft_size, |
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num_mels=self.num_mels, |
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mel_fmax=self.mel_fmax, |
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mel_fmin=self.mel_fmin, |
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) |
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if stats_path and signal_norm: |
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mel_mean, mel_std, linear_mean, linear_std, _ = self.load_stats(stats_path) |
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self.setup_scaler(mel_mean, mel_std, linear_mean, linear_std) |
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self.signal_norm = True |
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self.max_norm = None |
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self.clip_norm = None |
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self.symmetric_norm = None |
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@staticmethod |
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def init_from_config(config: "Coqpit", verbose=True): |
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if "audio" in config: |
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return AudioProcessor(verbose=verbose, **config.audio) |
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return AudioProcessor(verbose=verbose, **config) |
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def normalize(self, S: np.ndarray) -> np.ndarray: |
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"""Normalize values into `[0, self.max_norm]` or `[-self.max_norm, self.max_norm]` |
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Args: |
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S (np.ndarray): Spectrogram to normalize. |
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Raises: |
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RuntimeError: Mean and variance is computed from incompatible parameters. |
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Returns: |
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np.ndarray: Normalized spectrogram. |
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""" |
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S = S.copy() |
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if self.signal_norm: |
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if hasattr(self, "mel_scaler"): |
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if S.shape[0] == self.num_mels: |
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return self.mel_scaler.transform(S.T).T |
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elif S.shape[0] == self.fft_size / 2: |
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return self.linear_scaler.transform(S.T).T |
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else: |
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raise RuntimeError(" [!] Mean-Var stats does not match the given feature dimensions.") |
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S -= self.ref_level_db |
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S_norm = (S - self.min_level_db) / (-self.min_level_db) |
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if self.symmetric_norm: |
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S_norm = ((2 * self.max_norm) * S_norm) - self.max_norm |
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if self.clip_norm: |
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S_norm = np.clip( |
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S_norm, -self.max_norm, self.max_norm |
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) |
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return S_norm |
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else: |
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S_norm = self.max_norm * S_norm |
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if self.clip_norm: |
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S_norm = np.clip(S_norm, 0, self.max_norm) |
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return S_norm |
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else: |
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return S |
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def denormalize(self, S: np.ndarray) -> np.ndarray: |
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"""Denormalize spectrogram values. |
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Args: |
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S (np.ndarray): Spectrogram to denormalize. |
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Raises: |
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RuntimeError: Mean and variance are incompatible. |
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Returns: |
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np.ndarray: Denormalized spectrogram. |
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""" |
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S_denorm = S.copy() |
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if self.signal_norm: |
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if hasattr(self, "mel_scaler"): |
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if S_denorm.shape[0] == self.num_mels: |
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return self.mel_scaler.inverse_transform(S_denorm.T).T |
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elif S_denorm.shape[0] == self.fft_size / 2: |
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return self.linear_scaler.inverse_transform(S_denorm.T).T |
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else: |
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raise RuntimeError(" [!] Mean-Var stats does not match the given feature dimensions.") |
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if self.symmetric_norm: |
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if self.clip_norm: |
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S_denorm = np.clip( |
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S_denorm, -self.max_norm, self.max_norm |
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) |
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S_denorm = ((S_denorm + self.max_norm) * -self.min_level_db / (2 * self.max_norm)) + self.min_level_db |
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return S_denorm + self.ref_level_db |
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else: |
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if self.clip_norm: |
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S_denorm = np.clip(S_denorm, 0, self.max_norm) |
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S_denorm = (S_denorm * -self.min_level_db / self.max_norm) + self.min_level_db |
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return S_denorm + self.ref_level_db |
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else: |
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return S_denorm |
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def load_stats(self, stats_path: str) -> Tuple[np.array, np.array, np.array, np.array, Dict]: |
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"""Loading mean and variance statistics from a `npy` file. |
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Args: |
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stats_path (str): Path to the `npy` file containing |
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Returns: |
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Tuple[np.array, np.array, np.array, np.array, Dict]: loaded statistics and the config used to |
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compute them. |
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""" |
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stats = np.load(stats_path, allow_pickle=True).item() |
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mel_mean = stats["mel_mean"] |
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mel_std = stats["mel_std"] |
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linear_mean = stats["linear_mean"] |
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linear_std = stats["linear_std"] |
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stats_config = stats["audio_config"] |
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skip_parameters = ["griffin_lim_iters", "stats_path", "do_trim_silence", "ref_level_db", "power"] |
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for key in stats_config.keys(): |
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if key in skip_parameters: |
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continue |
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if key not in ["sample_rate", "trim_db"]: |
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assert ( |
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stats_config[key] == self.__dict__[key] |
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), f" [!] Audio param {key} does not match the value used for computing mean-var stats. {stats_config[key]} vs {self.__dict__[key]}" |
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return mel_mean, mel_std, linear_mean, linear_std, stats_config |
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def setup_scaler( |
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self, mel_mean: np.ndarray, mel_std: np.ndarray, linear_mean: np.ndarray, linear_std: np.ndarray |
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) -> None: |
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"""Initialize scaler objects used in mean-std normalization. |
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Args: |
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mel_mean (np.ndarray): Mean for melspectrograms. |
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mel_std (np.ndarray): STD for melspectrograms. |
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linear_mean (np.ndarray): Mean for full scale spectrograms. |
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linear_std (np.ndarray): STD for full scale spectrograms. |
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""" |
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self.mel_scaler = StandardScaler() |
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self.mel_scaler.set_stats(mel_mean, mel_std) |
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self.linear_scaler = StandardScaler() |
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self.linear_scaler.set_stats(linear_mean, linear_std) |
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def apply_preemphasis(self, x: np.ndarray) -> np.ndarray: |
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"""Apply pre-emphasis to the audio signal. Useful to reduce the correlation between neighbouring signal values. |
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Args: |
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x (np.ndarray): Audio signal. |
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Raises: |
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RuntimeError: Preemphasis coeff is set to 0. |
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Returns: |
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np.ndarray: Decorrelated audio signal. |
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""" |
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return preemphasis(x=x, coef=self.preemphasis) |
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def apply_inv_preemphasis(self, x: np.ndarray) -> np.ndarray: |
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"""Reverse pre-emphasis.""" |
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return deemphasis(x=x, coef=self.preemphasis) |
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def spectrogram(self, y: np.ndarray) -> np.ndarray: |
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"""Compute a spectrogram from a waveform. |
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Args: |
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y (np.ndarray): Waveform. |
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Returns: |
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np.ndarray: Spectrogram. |
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""" |
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if self.preemphasis != 0: |
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y = self.apply_preemphasis(y) |
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D = stft( |
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y=y, |
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fft_size=self.fft_size, |
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hop_length=self.hop_length, |
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win_length=self.win_length, |
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pad_mode=self.stft_pad_mode, |
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) |
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if self.do_amp_to_db_linear: |
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S = amp_to_db(x=np.abs(D), gain=self.spec_gain, base=self.base) |
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else: |
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S = np.abs(D) |
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return self.normalize(S).astype(np.float32) |
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def melspectrogram(self, y: np.ndarray) -> np.ndarray: |
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"""Compute a melspectrogram from a waveform.""" |
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if self.preemphasis != 0: |
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y = self.apply_preemphasis(y) |
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D = stft( |
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y=y, |
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fft_size=self.fft_size, |
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hop_length=self.hop_length, |
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win_length=self.win_length, |
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pad_mode=self.stft_pad_mode, |
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) |
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S = spec_to_mel(spec=np.abs(D), mel_basis=self.mel_basis) |
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if self.do_amp_to_db_mel: |
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S = amp_to_db(x=S, gain=self.spec_gain, base=self.base) |
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return self.normalize(S).astype(np.float32) |
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def inv_spectrogram(self, spectrogram: np.ndarray) -> np.ndarray: |
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"""Convert a spectrogram to a waveform using Griffi-Lim vocoder.""" |
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S = self.denormalize(spectrogram) |
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S = db_to_amp(x=S, gain=self.spec_gain, base=self.base) |
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W = self._griffin_lim(S**self.power) |
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return self.apply_inv_preemphasis(W) if self.preemphasis != 0 else W |
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def inv_melspectrogram(self, mel_spectrogram: np.ndarray) -> np.ndarray: |
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"""Convert a melspectrogram to a waveform using Griffi-Lim vocoder.""" |
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D = self.denormalize(mel_spectrogram) |
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S = db_to_amp(x=D, gain=self.spec_gain, base=self.base) |
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S = mel_to_spec(mel=S, mel_basis=self.mel_basis) |
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W = self._griffin_lim(S**self.power) |
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return self.apply_inv_preemphasis(W) if self.preemphasis != 0 else W |
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def out_linear_to_mel(self, linear_spec: np.ndarray) -> np.ndarray: |
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"""Convert a full scale linear spectrogram output of a network to a melspectrogram. |
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Args: |
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linear_spec (np.ndarray): Normalized full scale linear spectrogram. |
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Returns: |
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np.ndarray: Normalized melspectrogram. |
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""" |
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S = self.denormalize(linear_spec) |
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S = db_to_amp(x=S, gain=self.spec_gain, base=self.base) |
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S = spec_to_mel(spec=np.abs(S), mel_basis=self.mel_basis) |
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S = amp_to_db(x=S, gain=self.spec_gain, base=self.base) |
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mel = self.normalize(S) |
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return mel |
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|
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def _griffin_lim(self, S): |
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return griffin_lim( |
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spec=S, |
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num_iter=self.griffin_lim_iters, |
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hop_length=self.hop_length, |
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win_length=self.win_length, |
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fft_size=self.fft_size, |
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pad_mode=self.stft_pad_mode, |
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) |
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def compute_f0(self, x: np.ndarray) -> np.ndarray: |
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"""Compute pitch (f0) of a waveform using the same parameters used for computing melspectrogram. |
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Args: |
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x (np.ndarray): Waveform. |
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Returns: |
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np.ndarray: Pitch. |
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|
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Examples: |
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>>> WAV_FILE = filename = librosa.example('vibeace') |
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>>> from TTS.config import BaseAudioConfig |
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>>> from TTS.utils.audio import AudioProcessor |
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>>> conf = BaseAudioConfig(pitch_fmax=640, pitch_fmin=1) |
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>>> ap = AudioProcessor(**conf) |
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>>> wav = ap.load_wav(WAV_FILE, sr=ap.sample_rate)[:5 * ap.sample_rate] |
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>>> pitch = ap.compute_f0(wav) |
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""" |
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|
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if len(x) % self.hop_length == 0: |
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x = np.pad(x, (0, self.hop_length // 2), mode=self.stft_pad_mode) |
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f0 = compute_f0( |
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x=x, |
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pitch_fmax=self.pitch_fmax, |
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pitch_fmin=self.pitch_fmin, |
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hop_length=self.hop_length, |
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win_length=self.win_length, |
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sample_rate=self.sample_rate, |
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stft_pad_mode=self.stft_pad_mode, |
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center=True, |
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) |
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return f0 |
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|
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def find_endpoint(self, wav: np.ndarray, min_silence_sec=0.8) -> int: |
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"""Find the last point without silence at the end of a audio signal. |
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|
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Args: |
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wav (np.ndarray): Audio signal. |
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threshold_db (int, optional): Silence threshold in decibels. Defaults to -40. |
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min_silence_sec (float, optional): Ignore silences that are shorter then this in secs. Defaults to 0.8. |
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|
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Returns: |
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int: Last point without silence. |
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""" |
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return find_endpoint( |
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wav=wav, |
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trim_db=self.trim_db, |
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sample_rate=self.sample_rate, |
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min_silence_sec=min_silence_sec, |
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gain=self.spec_gain, |
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base=self.base, |
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) |
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|
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def trim_silence(self, wav): |
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"""Trim silent parts with a threshold and 0.01 sec margin""" |
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return trim_silence( |
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wav=wav, |
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sample_rate=self.sample_rate, |
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trim_db=self.trim_db, |
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win_length=self.win_length, |
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hop_length=self.hop_length, |
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) |
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|
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@staticmethod |
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def sound_norm(x: np.ndarray) -> np.ndarray: |
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"""Normalize the volume of an audio signal. |
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|
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Args: |
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x (np.ndarray): Raw waveform. |
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|
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Returns: |
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np.ndarray: Volume normalized waveform. |
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""" |
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return volume_norm(x=x) |
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|
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def rms_volume_norm(self, x: np.ndarray, db_level: float = None) -> np.ndarray: |
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"""Normalize the volume based on RMS of the signal. |
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|
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Args: |
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x (np.ndarray): Raw waveform. |
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|
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Returns: |
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np.ndarray: RMS normalized waveform. |
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""" |
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if db_level is None: |
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db_level = self.db_level |
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return rms_volume_norm(x=x, db_level=db_level) |
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|
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def load_wav(self, filename: str, sr: int = None) -> np.ndarray: |
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"""Read a wav file using Librosa and optionally resample, silence trim, volume normalize. |
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|
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Resampling slows down loading the file significantly. Therefore it is recommended to resample the file before. |
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|
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Args: |
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filename (str): Path to the wav file. |
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sr (int, optional): Sampling rate for resampling. Defaults to None. |
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|
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Returns: |
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np.ndarray: Loaded waveform. |
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""" |
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if sr is not None: |
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x = load_wav(filename=filename, sample_rate=sr, resample=True) |
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else: |
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x = load_wav(filename=filename, sample_rate=self.sample_rate, resample=self.resample) |
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if self.do_trim_silence: |
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try: |
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x = self.trim_silence(x) |
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except ValueError: |
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print(f" [!] File cannot be trimmed for silence - {filename}") |
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if self.do_sound_norm: |
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x = self.sound_norm(x) |
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if self.do_rms_norm: |
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x = self.rms_volume_norm(x, self.db_level) |
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return x |
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|
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def save_wav(self, wav: np.ndarray, path: str, sr: int = None, pipe_out=None) -> None: |
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"""Save a waveform to a file using Scipy. |
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|
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Args: |
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wav (np.ndarray): Waveform to save. |
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path (str): Path to a output file. |
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sr (int, optional): Sampling rate used for saving to the file. Defaults to None. |
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pipe_out (BytesIO, optional): Flag to stdout the generated TTS wav file for shell pipe. |
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""" |
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if self.do_rms_norm: |
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wav_norm = self.rms_volume_norm(wav, self.db_level) * 32767 |
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else: |
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wav_norm = wav * (32767 / max(0.01, np.max(np.abs(wav)))) |
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|
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wav_norm = wav_norm.astype(np.int16) |
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if pipe_out: |
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wav_buffer = BytesIO() |
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scipy.io.wavfile.write(wav_buffer, sr if sr else self.sample_rate, wav_norm) |
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wav_buffer.seek(0) |
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pipe_out.buffer.write(wav_buffer.read()) |
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scipy.io.wavfile.write(path, sr if sr else self.sample_rate, wav_norm) |
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|
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def get_duration(self, filename: str) -> float: |
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"""Get the duration of a wav file using Librosa. |
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|
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Args: |
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filename (str): Path to the wav file. |
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""" |
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return librosa.get_duration(filename=filename) |
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