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hparams.py
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hparams.py
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# CONFIG -----------------------------------------------------------------------------------------------------------#
# Here are the input and output data paths (Note: you can override wav_path in preprocess.py)
wav_path = '/path/to/wav_files/'
data_path = 'data/'
# model ids are separate - that way you can use a new tts with an old wavernn and vice versa
# NB: expect undefined behaviour if models were trained on different DSP settings
voc_model_id = 'ljspeech_mol'
tts_model_id = 'ljspeech_lsa_smooth_attention'
# set this to True if you are only interested in WaveRNN
ignore_tts = True
# DSP --------------------------------------------------------------------------------------------------------------#
# Settings for all models
sample_rate = 22050 # 采样点数量
n_fft = 2048
fft_bins = n_fft // 2 + 1
num_mels = 80 # bin/infer.py parser.add_argument("--n_mels", default=80, type=int)
hop_length = 275 # 12.5ms - in line with Tacotron 2 paper 22050*0.03
win_length = 1100 # 50ms - same reason as above
fmin = 40 # check
min_level_db = -100 # check
ref_level_db = 20 # check
bits = 9 # check bit depth of signal
mu_law = True # check Recommended to suppress noise if using raw bits in hp.voc_mode below
peak_norm = False # check Normalise to the peak of each wav file
# WAVERNN / VOCODER ------------------------------------------------------------------------------------------------#
# Model Hparams
voc_mode = 'MOL' # either 'RAW' (softmax on raw bits) or 'MOL' (sample from mixture of logistics)
voc_upsample_factors = (5, 5, 11) # NB - this needs to correctly factorise hop_length
voc_rnn_dims = 512
voc_fc_dims = 512
voc_compute_dims = 128
voc_res_out_dims = 128
voc_res_blocks = 10
# Training
voc_batch_size = 32
voc_lr = 1e-4
voc_checkpoint_every = 25_000
voc_gen_at_checkpoint = 5 # number of samples to generate at each checkpoint
voc_total_steps = 1_000_000 # Total number of training steps
voc_test_samples = 50 # How many unseen samples to put aside for testing
voc_pad = 2 # this will pad the input so that the resnet can 'see' wider than input length
voc_seq_len = hop_length * 5 # must be a multiple of hop_length
voc_clip_grad_norm = 4 # set to None if no gradient clipping needed
# Generating / Synthesizing
voc_gen_batched = True # very fast (realtime+) single utterance batched generation
voc_target = 11_000 # target number of samples to be generated in each batch entry
voc_overlap = 550 # number of samples for crossfading between batches
# TACOTRON/TTS -----------------------------------------------------------------------------------------------------#
# Model Hparams
tts_embed_dims = 256 # embedding dimension for the graphemes/phoneme inputs
tts_encoder_dims = 128
tts_decoder_dims = 256
tts_postnet_dims = 128
tts_encoder_K = 16
tts_lstm_dims = 512
tts_postnet_K = 8
tts_num_highways = 4
tts_dropout = 0.5
tts_cleaner_names = ['english_cleaners']
tts_stop_threshold = -3.4 # Value below which audio generation ends.
# For example, for a range of [-4, 4], this
# will terminate the sequence at the first
# frame that has all values < -3.4
# Training
tts_schedule = [(7, 1e-3, 10_000, 32), # progressive training schedule
(5, 1e-4, 100_000, 32), # (r, lr, step, batch_size)
(2, 1e-4, 180_000, 16),
(2, 1e-4, 350_000, 8)]
tts_max_mel_len = 1250 # if you have a couple of extremely long spectrograms you might want to use this
tts_bin_lengths = True # bins the spectrogram lengths before sampling in data loader - speeds up training
tts_clip_grad_norm = 1.0 # clips the gradient norm to prevent explosion - set to None if not needed
tts_checkpoint_every = 2_000 # checkpoints the model every X steps
# TODO: tts_phoneme_prob = 0.0 # [0 <-> 1] probability for feeding model phonemes vrs graphemes
# ------------------------------------------------------------------------------------------------------------------#