mirror of https://github.com/coqui-ai/TTS.git
refactor(audio.processor): remove duplicate stft_parameters
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@ -15,6 +15,7 @@ from TTS.utils.audio.numpy_transforms import (
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db_to_amp,
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db_to_amp,
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griffin_lim,
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griffin_lim,
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mel_to_spec,
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mel_to_spec,
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millisec_to_length,
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spec_to_mel,
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spec_to_mel,
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stft,
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stft,
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)
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)
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@ -209,7 +210,9 @@ class AudioProcessor(object):
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# setup stft parameters
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# setup stft parameters
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if hop_length is None:
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if hop_length is None:
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# compute stft parameters from given time values
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# compute stft parameters from given time values
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self.hop_length, self.win_length = self._stft_parameters()
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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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else:
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# use stft parameters from config file
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# use stft parameters from config file
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self.hop_length = hop_length
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self.hop_length = hop_length
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@ -246,21 +249,6 @@ class AudioProcessor(object):
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return AudioProcessor(verbose=verbose, **config.audio)
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return AudioProcessor(verbose=verbose, **config.audio)
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return AudioProcessor(verbose=verbose, **config)
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return AudioProcessor(verbose=verbose, **config)
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### setting up the parameters ###
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def _stft_parameters(
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self,
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) -> Tuple[int, int]:
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"""Compute the real STFT parameters from the time values.
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Returns:
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Tuple[int, int]: hop length and window length for STFT.
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"""
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factor = self.frame_length_ms / self.frame_shift_ms
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assert (factor).is_integer(), " [!] frame_shift_ms should divide frame_length_ms"
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hop_length = int(self.frame_shift_ms / 1000.0 * self.sample_rate)
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win_length = int(hop_length * factor)
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return hop_length, win_length
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### normalization ###
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### normalization ###
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def normalize(self, S: np.ndarray) -> np.ndarray:
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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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"""Normalize values into `[0, self.max_norm]` or `[-self.max_norm, self.max_norm]`
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