Hi there , im training a Polish speech model , spectrograms looks quite good, even the sound in /Log category is not so bad like for 35k steps. Even thoo I can't synthesize any audio , from text file or in live mode. Here is the error ps.
and here is the screen from the training begining .
Maybe it's becouse there's something wrong in my hparams.py ? Im running on M-AILABS model.
In Windows system I can't set parameters to synthesize.py from command promt, only thing I could do to make it works, is to set default values inside script.
`
Audio
num_mels = 80, #Number of mel-spectrogram channels and local conditioning dimensionality
num_freq = 513 , # (= n_fft / 2 + 1) only used when adding linear spectrograms post processing network
rescale = True, #Whether to rescale audio prior to preprocessing
rescaling_max = 0.999, #Rescaling value
trim_silence = True, #Whether to clip silence in Audio (at beginning and end of audio only, not the middle)
clip_mels_length = True, #For cases of OOM (Not really recommended, working on a workaround)
max_mel_frames = 1300, #Only relevant when clip_mels_length = True
# Use LWS (https://github.com/Jonathan-LeRoux/lws) for STFT and phase reconstruction
# It's preferred to set True to use with https://github.com/r9y9/wavenet_vocoder
# Does not work if n_ffit is not multiple of hop_size!!
use_lws=False,
silence_threshold=2, #silence threshold used for sound trimming for wavenet preprocessing
#Mel spectrogram
n_fft = 1024, #Extra window size is filled with 0 paddings to match this parameter
hop_size = 256, #For 22050Hz, 275 ~= 12.5 ms
win_size = 1024, #For 22050Hz, 1100 ~= 50 ms (If None, win_size = n_fft)
sample_rate = 16000, #22050 Hz (corresponding to ljspeech dataset)
frame_shift_ms = None,
#M-AILABS (and other datasets) trim params
trim_fft_size = 512,
trim_hop_size = 128,
trim_top_db = 23,
#Mel and Linear spectrograms normalization/scaling and clipping
signal_normalization = True,
allow_clipping_in_normalization = True, #Only relevant if mel_normalization = True
symmetric_mels = False, #Whether to scale the data to be symmetric around 0
max_abs_value = 4., #max absolute value of data. If symmetric, data will be [-max, max] else [0, max]
normalize_for_wavenet = True, #whether to rescale to [0, 1] for wavenet.
#Limits
min_level_db = -100,
ref_level_db = 20,
fmin = 0, #Set this to 75 if your speaker is male! if female, 125 should help taking off noise. (To test depending on dataset)
fmax = 7600,
#Griffin Lim
power = 1.5,
griffin_lim_iters = 60,
###########################################################################################################################################
#Tacotron
outputs_per_step = 3, #number of frames to generate at each decoding step (speeds up computation and allows for higher batch size)
stop_at_any = True, #Determines whether the decoder should stop when predicting <stop> to any frame or to all of them
embedding_dim = 512, #dimension of embedding space
enc_conv_num_layers = 3, #number of encoder convolutional layers
enc_conv_kernel_size = (5, ), #size of encoder convolution filters for each layer
enc_conv_channels = 512, #number of encoder convolutions filters for each layer
encoder_lstm_units = 256, #number of lstm units for each direction (forward and backward)
smoothing = False, #Whether to smooth the attention normalization function
attention_dim = 128, #dimension of attention space
attention_filters = 32, #number of attention convolution filters
attention_kernel = (31, ), #kernel size of attention convolution
cumulative_weights = True, #Whether to cumulate (sum) all previous attention weights or simply feed previous weights (Recommended: True)
prenet_layers = [256, 256], #number of layers and number of units of prenet
decoder_layers = 2, #number of decoder lstm layers
decoder_lstm_units = 1024, #number of decoder lstm units on each layer
max_iters = 1000, #Max decoder steps during inference (Just for safety from infinite loop cases)
postnet_num_layers = 5, #number of postnet convolutional layers
postnet_kernel_size = (5, ), #size of postnet convolution filters for each layer
postnet_channels = 512, #number of postnet convolution filters for each layer
mask_encoder = False, #whether to mask encoder padding while computing attention
mask_decoder = False, #Whether to use loss mask for padded sequences (if False, <stop_token> loss function will not be weighted, else recommended pos_weight = 20)
cross_entropy_pos_weight = 1, #Use class weights to reduce the stop token classes imbalance (by adding more penalty on False Negatives (FN)) (1 = disabled)
predict_linear = True, #Whether to add a post-processing network to the Tacotron to predict linear spectrograms (True mode Not tested!!)
###########################################################################################################################################
#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.
input_type="raw",
quantize_channels=2 ** 16, # 65536 (16-bit) (raw) or 256 (8-bit) (mulaw or mulaw-quantize) // number of classes = 256 <=> mu = 255
log_scale_min=float(np.log(1e-14)), #Mixture of logistic distributions minimal log scale
log_scale_min_gauss = float(np.log(1e-7)), #Gaussian distribution minimal allowed log scale
#To use Gaussian distribution as output distribution instead of mixture of logistics, set "out_channels = 2" instead of "out_channels = 10 * 3". (UNDER TEST)
out_channels = 2, #This should be equal to quantize channels when input type is 'mulaw-quantize' else: num_distributions * 3 (prob, mean, log_scale).
layers = 30, #Number of dilated convolutions (Default: Simplified Wavenet of Tacotron-2 paper)
stacks = 3, #Number of dilated convolution stacks (Default: Simplified Wavenet of Tacotron-2 paper)
residual_channels = 512,
gate_channels = 512, #split in 2 in gated convolutions
skip_out_channels = 256,
kernel_size = 3,
cin_channels = 80, #Set this to -1 to disable local conditioning, else it must be equal to num_mels!!
upsample_conditional_features = True, #Whether to repeat conditional features or upsample them (The latter is recommended)
upsample_scales = [15, 20], #prod(upsample_scales) should be equal to hop_size
freq_axis_kernel_size = 3,
leaky_alpha = 0.4,
gin_channels = -1, #Set this to -1 to disable global conditioning, Only used for multi speaker dataset. It defines the depth of the embeddings (Recommended: 16)
use_speaker_embedding = True, #whether to make a speaker embedding
n_speakers = 5, #number of speakers (rows of the embedding)
use_bias = True, #Whether to use bias in convolutional layers of the Wavenet
max_time_sec = None,
max_time_steps = 13000, #Max time steps in audio used to train wavenet (decrease to save memory) (Recommend: 8000 on modest GPUs, 13000 on stronger ones)
###########################################################################################################################################
#Tacotron Training
tacotron_random_seed = 5339, #Determines initial graph and operations (i.e: model) random state for reproducibility
tacotron_swap_with_cpu = False, #Whether to use cpu as support to gpu for decoder computation (Not recommended: may cause major slowdowns! Only use when critical!)
tacotron_batch_size = 6, #number of training samples on each training steps
tacotron_reg_weight = 1e-6, #regularization weight (for L2 regularization)
tacotron_scale_regularization = True, #Whether to rescale regularization weight to adapt for outputs range (used when reg_weight is high and biasing the model)
tacotron_test_size = None, #% of data to keep as test data, if None, tacotron_test_batches must be not None
tacotron_test_batches = 48, #number of test batches (For Ljspeech: 10% ~= 41 batches of 32 samples)
tacotron_data_random_state=1234, #random state for train test split repeatability
#Usually your GPU can handle 16x tacotron_batch_size during synthesis for the same memory amount during training (because no gradients to keep and ops to register for backprop)
tacotron_synthesis_batch_size = 32 * 16, #This ensures GTA synthesis goes up to 40x faster than one sample at a time and uses 100% of your GPU computation power.
tacotron_decay_learning_rate = True, #boolean, determines if the learning rate will follow an exponential decay
tacotron_start_decay = 50000, #Step at which learning decay starts
tacotron_decay_steps = 50000, #Determines the learning rate decay slope (UNDER TEST)
tacotron_decay_rate = 0.4, #learning rate decay rate (UNDER TEST)
tacotron_initial_learning_rate = 1e-3, #starting learning rate
tacotron_final_learning_rate = 1e-5, #minimal learning rate
tacotron_adam_beta1 = 0.9, #AdamOptimizer beta1 parameter
tacotron_adam_beta2 = 0.999, #AdamOptimizer beta2 parameter
tacotron_adam_epsilon = 1e-6, #AdamOptimizer beta3 parameter
tacotron_zoneout_rate = 0.1, #zoneout rate for all LSTM cells in the network
tacotron_dropout_rate = 0.5, #dropout rate for all convolutional layers + prenet
tacotron_clip_gradients = False, #whether to clip gradients
natural_eval = False, #Whether to use 100% natural eval (to evaluate Curriculum Learning performance) or with same teacher-forcing ratio as in training (just for overfit)
#Decoder RNN learning can take be done in one of two ways:
# Teacher Forcing: vanilla teacher forcing (usually with ratio = 1). mode='constant'
# Curriculum Learning Scheme: From Teacher-Forcing to sampling from previous outputs is function of global step. (teacher forcing ratio decay) mode='scheduled'
#The second approach is inspired by:
#Bengio et al. 2015: Scheduled Sampling for Sequence Prediction with Recurrent Neural Networks.
#Can be found under: https://arxiv.org/pdf/1506.03099.pdf
tacotron_teacher_forcing_mode = 'constant', #Can be ('constant' or 'scheduled'). 'scheduled' mode applies a cosine teacher forcing ratio decay. (Preference: scheduled)
tacotron_teacher_forcing_ratio = 1., #Value from [0., 1.], 0.=0%, 1.=100%, determines the % of times we force next decoder inputs, Only relevant if mode='constant'
tacotron_teacher_forcing_init_ratio = 1., #initial teacher forcing ratio. Relevant if mode='scheduled'
tacotron_teacher_forcing_final_ratio = 0., #final teacher forcing ratio. Relevant if mode='scheduled'
tacotron_teacher_forcing_start_decay = 10000, #starting point of teacher forcing ratio decay. Relevant if mode='scheduled'
tacotron_teacher_forcing_decay_steps = 280000, #Determines the teacher forcing ratio decay slope. Relevant if mode='scheduled'
tacotron_teacher_forcing_decay_alpha = 0., #teacher forcing ratio decay rate. Relevant if mode='scheduled'
###########################################################################################################################################
#Wavenet Training
wavenet_random_seed = 5339, # S=5, E=3, D=9 :)
wavenet_swap_with_cpu = False, #Whether to use cpu as support to gpu for decoder computation (Not recommended: may cause major slowdowns! Only use when critical!)
wavenet_batch_size = 4, #batch size used to train wavenet.
wavenet_test_size = 0.0441, #% of data to keep as test data, if None, wavenet_test_batches must be not None
wavenet_test_batches = None, #number of test batches.
wavenet_data_random_state = 1234, #random state for train test split repeatability
#During synthesis, there is no max_time_steps limitation so the model can sample much longer audio than 8k(or 13k) steps. (Audio can go up to 500k steps, equivalent to ~21sec on 24kHz)
#Usually your GPU can handle 1x~2x wavenet_batch_size during synthesis for the same memory amount during training (because no gradients to keep and ops to register for backprop)
wavenet_synthesis_batch_size = 4 * 2, #This ensure that wavenet synthesis goes up to 4x~8x faster when synthesizing multiple sentences. Watch out for OOM with long audios.
wavenet_learning_rate = 1e-3,
wavenet_adam_beta1 = 0.9,
wavenet_adam_beta2 = 0.999,
wavenet_adam_epsilon = 1e-8,
wavenet_ema_decay = 0.9999, #decay rate of exponential moving average
wavenet_dropout = 0.05, #drop rate of wavenet layers
train_with_GTA = True, #Whether to use GTA mels to train WaveNet instead of ground truth mels.
###########################################################################################################################################`
Hi there , im training a Polish speech model , spectrograms looks quite good, even the sound in /Log category is not so bad like for 35k steps. Even thoo I can't synthesize any audio , from text file or in live mode. Here is the error ps.
and here is the screen from the training begining .
Maybe it's becouse there's something wrong in my hparams.py ? Im running on M-AILABS model. In Windows system I can't set parameters to synthesize.py from command promt, only thing I could do to make it works, is to set default values inside script.
`
Audio
Please , take a look on my issue, Thank you.