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hparams.py
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hparams.py
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import tensorflow as tf
import numpy as np
# NOTE: If you want full control for model architecture. please take a look
# at the code and change whatever you want. Some hyper parameters are hardcoded.
# Default hyperparameters:
hparams = tf.contrib.training.HParams(
name="wavenet",
# Convenient model builder
builder="wavenet",
# Input type:
# 1. raw [-1, 1]
# 2. mulaw [-1, 1]
# 3. mulaw-quantize [0, mu]
# If input_type is raw or mulaw, network assumes scalar input and
# discretized mixture of logistic distributions output, otherwise one-hot
# input and softmax output are assumed.
# **NOTE**: if you change the one of the two parameters below, you need to
# re-run preprocessing before training.
input_type="raw",
quantize_channels=65536, # 65536 or 256
# Audio:
sample_rate=125,
# this is only valid for mulaw is True
silence_threshold=2,
num_mels=74,
fmin=125,
fmax=7600,
fft_size=1024,
# shift can be specified by either hop_size or frame_shift_ms
hop_size=256,
frame_shift_ms=None,
min_level_db=-100,
ref_level_db=20,
# whether to rescale waveform or not.
# Let x is an input waveform, rescaled waveform y is given by:
# y = x / np.abs(x).max() * rescaling_max
rescaling=False,
rescaling_max=0.999,
# mel-spectrogram is normalized to [0, 1] for each utterance and clipping may
# happen depends on min_level_db and ref_level_db, causing clipping noise.
# If False, assertion is added to ensure no clipping happens.o0
allow_clipping_in_normalization=False,
# Mixture of logistic distributions:
log_scale_min=float(np.log(1e-14)),
# Model:
# This should equal to `quantize_channels` if mu-law quantize enabled
# otherwise num_mixture * 3 (pi, mean, log_scale)
out_channels=80,
layers=15,
stacks=3,
residual_channels=512,
gate_channels=512, # split into 2 gropus internally for gated activation
skip_out_channels=256,
dropout=1 - 0.95,
kernel_size=3,
# If True, apply weight normalization as same as DeepVoice3
weight_normalization=True,
# Use legacy code or not. Default is True since we already provided a model
# based on the legacy code that can generate high-quality audio.
# Ref: https://github.com/r9y9/wavenet_vocoder/pull/73
legacy=True,
# Local conditioning (set negative value to disable))
cin_channels=215,
# If True, use transposed convolutions to upsample conditional features,
# otherwise repeat features to adjust time resolution
upsample_conditional_features=False,
# should np.prod(upsample_scales) == hop_size
upsample_scales=[4, 4, 4, 4],
# Freq axis kernel size for upsampling network
freq_axis_kernel_size=3,
# Global conditioning (set negative value to disable)
# currently limited for speaker embedding
# this should only be enabled for multi-speaker dataset
gin_channels=-1, # i.e., speaker embedding dim
n_speakers=7, # 7 for CMU ARCTIC
# Data loader
pin_memory=True,
num_workers=2,
# train/test
# test size can be specified as portion or num samples
test_size=0.05, # 50 for CMU ARCTIC single speaker
test_num_samples=None,
random_state=1234,
# Loss
# Training:
batch_size=5,
adam_beta1=0.9,
adam_beta2=0.999,
adam_eps=1e-8,
amsgrad=False,
initial_learning_rate=2e-3,
# see lrschedule.py for available lr_schedule
lr_schedule="noam_learning_rate_decay",
lr_schedule_kwargs={}, # {"anneal_rate": 0.5, "anneal_interval": 50000},
# lr_schedule_kwargs={"T": 600000, "M": 10}, # {"anneal_rate": 0.5, "anneal_interval": 50000},
nepochs=2000,
weight_decay=0.0,
clip_thresh=-1,
# max time steps can either be specified as sec or steps
# if both are None, then full audio samples are used in a batch
max_time_sec=None,
max_time_steps=8000,
# Hold moving averaged parameters and use them for evaluation
exponential_moving_average=True,
# averaged = decay * averaged + (1 - decay) * x
ema_decay=0.9999,
# Save
# per-step intervals
checkpoint_interval=5000,
train_eval_interval=5000,
# per-epoch interval
test_eval_epoch_interval=1,
save_optimizer_state=True,
# Eval:
)
def hparams_debug_string():
values = hparams.values()
hp = [' %s: %s' % (name, values[name]) for name in sorted(values)]
return 'Hyperparameters:\n' + '\n'.join(hp)