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* Init App * init server.py (#93) * init server.py * Update requirements.txt Add requirement Co-authored-by: auau <auau@test.com> Co-authored-by: babysor00 <babysor00@gmail.com> * Run web.py! Run web.py! Co-authored-by: balala <Ozgay@users.noreply.github.com> Co-authored-by: auau <auau@test.com>
90 lines
4.6 KiB
Python
90 lines
4.6 KiB
Python
import ast
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import pprint
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class HParams(object):
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def __init__(self, **kwargs): self.__dict__.update(kwargs)
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def __setitem__(self, key, value): setattr(self, key, value)
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def __getitem__(self, key): return getattr(self, key)
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def __repr__(self): return pprint.pformat(self.__dict__)
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def parse(self, string):
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# Overrides hparams from a comma-separated string of name=value pairs
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if len(string) > 0:
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overrides = [s.split("=") for s in string.split(",")]
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keys, values = zip(*overrides)
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keys = list(map(str.strip, keys))
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values = list(map(str.strip, values))
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for k in keys:
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self.__dict__[k] = ast.literal_eval(values[keys.index(k)])
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return self
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hparams = HParams(
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### Signal Processing (used in both synthesizer and vocoder)
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sample_rate = 16000,
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n_fft = 800,
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num_mels = 80,
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hop_size = 200, # Tacotron uses 12.5 ms frame shift (set to sample_rate * 0.0125)
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win_size = 800, # Tacotron uses 50 ms frame length (set to sample_rate * 0.050)
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fmin = 55,
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min_level_db = -100,
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ref_level_db = 20,
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max_abs_value = 4., # Gradient explodes if too big, premature convergence if too small.
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preemphasis = 0.97, # Filter coefficient to use if preemphasize is True
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preemphasize = True,
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### Tacotron Text-to-Speech (TTS)
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tts_embed_dims = 512, # Embedding dimension for the graphemes/phoneme inputs
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tts_encoder_dims = 256,
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tts_decoder_dims = 128,
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tts_postnet_dims = 512,
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tts_encoder_K = 5,
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tts_lstm_dims = 1024,
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tts_postnet_K = 5,
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tts_num_highways = 4,
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tts_dropout = 0.5,
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tts_cleaner_names = ["basic_cleaners"],
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tts_stop_threshold = -3.4, # Value below which audio generation ends.
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# For example, for a range of [-4, 4], this
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# will terminate the sequence at the first
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# frame that has all values < -3.4
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### Tacotron Training
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tts_schedule = [(2, 1e-3, 20_000, 24), # Progressive training schedule
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(2, 5e-4, 40_000, 24), # (r, lr, step, batch_size)
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(2, 2e-4, 80_000, 24), #
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(2, 1e-4, 160_000, 24), # r = reduction factor (# of mel frames
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(2, 3e-5, 320_000, 24), # synthesized for each decoder iteration)
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(2, 1e-5, 640_000, 24)], # lr = learning rate
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tts_clip_grad_norm = 1.0, # clips the gradient norm to prevent explosion - set to None if not needed
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tts_eval_interval = 500, # Number of steps between model evaluation (sample generation)
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# Set to -1 to generate after completing epoch, or 0 to disable
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tts_eval_num_samples = 1, # Makes this number of samples
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### Data Preprocessing
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max_mel_frames = 900,
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rescale = True,
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rescaling_max = 0.9,
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synthesis_batch_size = 16, # For vocoder preprocessing and inference.
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### Mel Visualization and Griffin-Lim
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signal_normalization = True,
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power = 1.5,
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griffin_lim_iters = 60,
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### Audio processing options
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fmax = 7600, # Should not exceed (sample_rate // 2)
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allow_clipping_in_normalization = True, # Used when signal_normalization = True
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clip_mels_length = True, # If true, discards samples exceeding max_mel_frames
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use_lws = False, # "Fast spectrogram phase recovery using local weighted sums"
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symmetric_mels = True, # Sets mel range to [-max_abs_value, max_abs_value] if True,
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# and [0, max_abs_value] if False
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trim_silence = True, # Use with sample_rate of 16000 for best results
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### SV2TTS
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speaker_embedding_size = 256, # Dimension for the speaker embedding
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silence_min_duration_split = 0.4, # Duration in seconds of a silence for an utterance to be split
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utterance_min_duration = 1.6, # Duration in seconds below which utterances are discarded
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)
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