mirror of https://github.com/coqui-ai/TTS.git
testing mean-var scalingand updating test config
parent
40cb4a53a6
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52b0dc39a6
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@ -140,3 +140,24 @@ class TestAudio(unittest.TestCase):
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assert x_norm.min() < 0, x_norm.min()
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assert x_norm.min() < 0, x_norm.min()
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x_ = self.ap._denormalize(x_norm)
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x_ = self.ap._denormalize(x_norm)
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assert (x - x_).sum() < 1e-3
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assert (x - x_).sum() < 1e-3
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def test_scaler(self):
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scaler_stats_path = os.path.join(get_tests_input_path(), 'scale_stats.npy')
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conf.audio['stats_path'] = scaler_stats_path
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conf.audio['preemphasis'] = 0.0
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conf.audio['do_trim_silence'] = True
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conf.audio['signal_norm'] = True
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ap = AudioProcessor(**conf.audio)
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mel_mean, mel_std, linear_mean, linear_std, _ = ap.load_stats(scaler_stats_path)
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ap.setup_scaler(mel_mean, mel_std, linear_mean, linear_std)
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self.ap.signal_norm = False
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self.ap.preemphasis = 0.0
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# test scaler forward and backward transforms
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wav = self.ap.load_wav(WAV_FILE)
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mel_reference = self.ap.melspectrogram(wav)
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mel_norm = ap.melspectrogram(wav)
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mel_denorm = ap._denormalize(mel_norm)
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assert abs(mel_reference - mel_denorm).max() < 1e-4
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@ -2,10 +2,12 @@
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"audio":{
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"audio":{
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"audio_processor": "audio", // to use dictate different audio processors, if available.
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"audio_processor": "audio", // to use dictate different audio processors, if available.
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"num_mels": 80, // size of the mel spec frame.
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"num_mels": 80, // size of the mel spec frame.
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"num_freq": 1025, // number of stft frequency levels. Size of the linear spectogram frame.
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"num_freq": 513, // number of stft frequency levels. Size of the linear spectogram frame.
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"sample_rate": 22050, // wav sample-rate. If different than the original data, it is resampled.
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"sample_rate": 22050, // wav sample-rate. If different than the original data, it is resampled.
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"frame_length_ms": 50, // stft window length in ms.
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"frame_length_ms": null, // stft window length in ms.
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"frame_shift_ms": 12.5, // stft window hop-lengh in ms.
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"frame_shift_ms": null, // stft window hop-lengh in ms.
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"hop_length": 256,
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"win_length": 1024,
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"preemphasis": 0.97, // pre-emphasis to reduce spec noise and make it more structured. If 0.0, no -pre-emphasis.
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"preemphasis": 0.97, // pre-emphasis to reduce spec noise and make it more structured. If 0.0, no -pre-emphasis.
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"min_level_db": -100, // normalization range
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"min_level_db": -100, // normalization range
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"ref_level_db": 20, // reference level db, theoretically 20db is the sound of air.
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"ref_level_db": 20, // reference level db, theoretically 20db is the sound of air.
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@ -15,8 +17,8 @@
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"symmetric_norm": true, // move normalization to range [-1, 1]
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"symmetric_norm": true, // move normalization to range [-1, 1]
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"clip_norm": true, // clip normalized values into the range.
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"clip_norm": true, // clip normalized values into the range.
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"max_norm": 4, // scale normalization to range [-max_norm, max_norm] or [0, max_norm]
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"max_norm": 4, // scale normalization to range [-max_norm, max_norm] or [0, max_norm]
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"mel_fmin": 95, // minimum freq level for mel-spec. ~50 for male and ~95 for female voices. Tune for dataset!!
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"mel_fmin": 0, // minimum freq level for mel-spec. ~50 for male and ~95 for female voices. Tune for dataset!!
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"mel_fmax": 7600, // maximum freq level for mel-spec. Tune for dataset!!
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"mel_fmax": 8000, // maximum freq level for mel-spec. Tune for dataset!!
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"do_trim_silence": false
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"do_trim_silence": false
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},
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},
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