Restore checkpoint from pretrained model

[JRMeyer]

>>> CrazyJoeDevola
[September 15, 2020, 3:03pm]

Hi

Not sure if and what i am doing wrong, but i downloaded one of the LJ
trained models of 130k. I restored training after adding some missing
parameters. I would expect the model to produce audible voice after 10
epochs or so, due to the pretraining (i suppose this is the whole
purpose). but Even now, at step 165725, there is only echo in the audio
previews.

Is this due to the fact that i cloned the latest DEV repository, and
should have trained based on a different, specific branch?

I was looking for a shortcut to do a custom voice, with the basics of
the language already learnt, and thereby improving the results of the
custom voice.

[This is an archived TTS discussion thread from discourse.mozilla.org/t/restore-checkpoint-from-pretrained-model]

[JRMeyer]

>>> erogol
[September 15, 2020, 3:08pm]

it is better if you use the same commit as the checkpoint version.

If it is something about your training, we need more info to help.
Tensorboard output, examples audios etc.

[Archived Post]

[JRMeyer]

>>> CrazyJoeDevola
[September 15, 2020, 4:24pm]

Test audios:
https://drive.google.com/drive/folders/1D2vIqx2oQzKXPSVa1dyLnKgtAN2rjiJu?usp=sharing

config

> { slash
> 'model': 'Tacotron2', slash
> 'run_name': 'ljspeech-tac2', slash
> 'run_description': 'tacotron2 mean-scale normalization with gradual
> training',
>
> // AUDIO PARAMETERS
> 'audio':{
> // stft parameters
> 'fft_size': 1024, // number of stft frequency levels. Size of the linear spectogram frame.
> 'win_length': 1024, // stft window length in ms.
> 'hop_length': 256, // stft window hop-lengh in ms.
> 'frame_length_ms': null, // stft window length in ms.If null, 'win_length' is used.
> 'frame_shift_ms': null, // stft window hop-lengh in ms. If null, 'hop_length' is used.
>
> // Audio processing parameters
> 'sample_rate': 22050, // DATASET-RELATED: wav sample-rate. If different than the original data, it is resampled.
> 'preemphasis': 0.0, // pre-emphasis to reduce spec noise and make it more structured. If 0.0, no -pre-emphasis.
> 'ref_level_db': 0, // reference level db, theoretically 20db is the sound of air.
>
> // Silence trimming
> 'do_trim_silence': true,// enable trimming of slience of audio as you load it. LJspeech (false), TWEB (false), Nancy (true)
> 'trim_db': 60, // threshold for timming silence. Set this according to your dataset.
>
> // Griffin-Lim
> 'power': 1.5, // value to sharpen wav signals after GL algorithm.
> 'griffin_lim_iters': 60,// #griffin-lim iterations. 30-60 is a good range. Larger the value, slower the generation.
>
> // MelSpectrogram parameters
> 'num_mels': 80, // size of the mel spec frame.
> 'mel_fmin': 50.0, // minimum freq level for mel-spec. ~50 for male and 95 for female voices. Tune for dataset!!
> 'mel_fmax': 7600.0, // maximum freq level for mel-spec. Tune for dataset!!
> 'spec_gain': 1.0, // scaler value appplied after log transform of spectrogram.
>
> // Normalization parameters
> 'signal_norm': true, // normalize spec values. Mean-Var normalization if 'stats_path' is defined otherwise range normalization defined by the other params.
> 'min_level_db': -100, // lower bound for normalization
> 'symmetric_norm': true, // move normalization to range [-1, 1]
> 'max_norm': 4.0, // scale normalization to range [-max_norm, max_norm] or [0, max_norm]
> 'clip_norm': true, // clip normalized values into the range.
> 'stats_path': null // DO NOT USE WITH MULTI_SPEAKER MODEL. scaler stats file computed by 'compute_statistics.py'. If it is defined, mean-std based notmalization is used and other normalization params are ignored
> },
>
> // VOCABULARY PARAMETERS
> // if custom character set is not defined,
> // default set in symbols.py is used
> // 'characters':{
> // 'pad': '_',
> // 'eos': '',
> // 'bos': '^',
> // 'characters': 'ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz!'(),-.:;? ',
> // 'punctuations':'!'(),-.:;? ',
> // 'phonemes':'iyɨʉɯuɪʏʊeøɘəɵɤoɛœɜɞʌɔæɐaɶɑɒᵻʘɓǀɗǃʄǂɠǁʛpbtdʈɖcɟkɡqɢʔɴŋɲɳnɱmʙrʀⱱɾɽɸβfvθðszʃʒʂʐçʝxɣχʁħʕhɦɬɮʋɹɻjɰlɭʎʟˈˌːˑʍwɥʜʢʡɕʑɺɧɚ˞ɫ'
> // },
>
> // DISTRIBUTED TRAINING
> 'distributed':{
> 'backend': 'nccl',
> 'url': 'tcp: slash / slash /localhost:54321'
> },
>
> 'reinit_layers': [], // give a list of layer names to restore from the given checkpoint. If not defined, it reloads all heuristically matching layers.
>
> // TRAINING
> 'batch_size': 32, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
> 'eval_batch_size':16,
> 'r': 7, // Number of decoder frames to predict per iteration. Set the initial values if gradual training is enabled.
> 'gradual_training': [[0, 7, 64], [1, 5, 64], [50000, 3, 32], [130000, 2, 32], [290000, 1, 32]], //set gradual training steps [first_step, r, batch_size]. If it is null, gradual training is disabled. For Tacotron, you might need to reduce the 'batch_size' as you proceeed.
> 'loss_masking': true, // enable / disable loss masking against the sequence padding.
> 'ga_alpha': 1.0, // weight for guided attention loss. If > 0, guided attention is enabled.
> 'apex_amp_level': null, // level of optimization with NVIDIA's apex feature for automatic mixed FP16/FP32 precision (AMP), NOTE: currently only O1 is supported, and use 'O1' to activate.
>
>
> // VALIDATION
> 'run_eval': true,
> 'test_delay_epochs': 10, //Until attention is aligned, testing only wastes computation time.
> 'test_sentences_file': null, // set a file to load sentences to be used for testing. If it is null then we use default english sentences.
>
> // OPTIMIZER
> 'noam_schedule': false, // use noam warmup and lr schedule.
> 'grad_clip': 1.0, // upper limit for gradients for clipping.
> 'epochs': 1000, // total number of epochs to train.
> 'lr': 0.0003, // Initial learning rate. If Noam decay is active, maximum learning rate.
> 'wd': 0.000001, // Weight decay weight.
> 'warmup_steps': 4000, // Noam decay steps to increase the learning rate from 0 to 'lr'
> 'seq_len_norm': false, // Normalize eash sample loss with its length to alleviate imbalanced datasets. Use it if your dataset is small or has skewed distribution of sequence lengths.
>
> // TACOTRON PRENET
> 'memory_size': -1, // ONLY TACOTRON - size of the memory queue used fro storing last decoder predictions for auto-regression. If < 0, memory queue is disabled and decoder only uses the last prediction frame.
> 'prenet_type': 'original', // 'original' or 'bn'.
> 'prenet_dropout': true, // enable/disable dropout at prenet.
>
> // ATTENTION
> 'attention_type': 'original', // 'original' or 'graves'
> 'attention_heads': 4, // number of attention heads (only for 'graves')
> 'attention_norm': 'sigmoid', // softmax or sigmoid. Suggested to use softmax for Tacotron2 and sigmoid for Tacotron.
> 'windowing': false, // Enables attention windowing. Used only in eval mode.
> 'use_forward_attn': true, // if it uses forward attention. In general, it aligns faster.
> 'forward_attn_mask': false, // Additional masking forcing monotonicity only in eval mode.
> 'transition_agent': false, // enable/disable transition agent of forward attention.
> 'location_attn': true, // enable_disable location sensitive attention. It is enabled for TACOTRON by default.
> 'bidirectional_decoder': false, // use https://arxiv.org/abs/1907.09006. Use it, if attention does not work well with your dataset.
> 'double_decoder_consistency': false, // use DDC explained here https://erogol.com/solving-attention-problems-of-tts-models-with-double-decoder-consistency-draft/
> 'ddc_r': 1, // reduction rate for coarse decoder.
>
> // STOPNET
> 'stopnet': true, // Train stopnet predicting the end of synthesis.
> 'separate_stopnet': true, // Train stopnet seperately if 'stopnet==true'. It prevents stopnet loss to influence the rest of the model. It causes a better model, but it trains SLOWER.
>
> // TENSORBOARD and LOGGING
> 'print_step': 25, // Number of steps to log training on console.
> 'tb_plot_step': 100, // Number of steps to plot TB training figures.
> 'print_eval': false, // If True, it prints intermediate loss values in evalulation.
> 'save_step': 10000, // Number of training steps expected to save traninpg stats and checkpoints.
> 'checkpoint': true, // If true, it saves checkpoints per 'save_step'
> 'tb_model_param_stats': false, // true, plots param stats per layer on tensorboard. Might be memory consuming, but good for debugging.
>
> // DATA LOADING
> 'text_cleaner': 'phoneme_cleaners',
> 'enable_eos_bos_chars': false, // enable/disable beginning of sentence and end of sentence chars.
> 'num_loader_workers': 4, // number of training data loader processes. Don't set it too big. 4-8 are good values.
> 'num_val_loader_workers': 4, // number of evaluation data loader processes.
> 'batch_group_size': 4, //Number of batches to shuffle after bucketing.
> 'min_seq_len': 6, // DATASET-RELATED: minimum text length to use in training
> 'max_seq_len': 153, // DATASET-RELATED: maximum text length
>
> // PATHS
> 'output_path': '/content/drive/My Drive/Pretrained2.0',
>
> // PHONEMES
> 'phoneme_cache_path': '/content/drive/My Drive/Pretrained2.0/cache', // phoneme computation is slow, therefore, it caches results in the given folder.
> 'use_phonemes': true, // use phonemes instead of raw characters. It is suggested for better pronounciation.
> 'phoneme_language': 'en-us', // depending on your target language, pick one from https://github.com/bootphon/phonemizer#languages
>
> // MULTI-SPEAKER and GST
> 'use_speaker_embedding': false, // use speaker embedding to enable multi-speaker learning.
> 'use_external_speaker_embedding_file': false, // if true, forces the model to use external embedding per sample instead of nn.embeddings, that is, it supports external embeddings such as those used at: https://arxiv.org/abs /1806.04558
> 'external_speaker_embedding_file': '../../speakers-vctk-en.json', // if not null and use_external_speaker_embedding_file is true, it is used to load a specific embedding file and thus uses these embeddings instead of nn.embeddings, that is, it supports external embeddings such as those used at: https://arxiv.org/abs /1806.04558
> 'use_gst': false, // use global style tokens
> 'gst': { // gst parameter if gst is enabled
> 'gst_style_input': null, // Condition the style input either on a
> // -> wave file [path to wave] or
> // -> dictionary using the style tokens {'token1': 'value', 'token2': 'value'} example {'0': 0.15, '1': 0.15, '5': -0.15}
> // with the dictionary being len(dict) <= len(gst_style_tokens).
> 'gst_embedding_dim': 512,
> 'gst_num_heads': 4,
> 'gst_style_tokens': 10
>
> }, slash
> // DATASETS slash
> 'datasets': // List of datasets. They all merged and they get
> different speaker_ids. slash
> slash [ slash
> { slash
> 'name': 'ljspeech', slash
> 'path': '/content/LJSpeech-1.1', slash
> 'meta_file_train': 'metadata.csv', slash
> 'meta_file_val': 'metadata_val.csv' slash
> } slash
> slash ]
>
> }

> EVALUATION

slash | slash > Synthesizing test sentences slash
slash | slash > Decoder stopped with slash 'max_decoder_steps slash
slash | slash > Decoder stopped with slash 'max_decoder_steps slash
slash | slash > Decoder stopped with slash 'max_decoder_steps slash
warning: audio amplitude out of range, auto clipped. slash
slash | slash > avg_decoder_loss: 0.00337 (-0.00013) slash
slash | slash > avg_postnet_loss: 0.00334 (-0.00008) slash
slash | slash > avg_stopnet_loss: 0.15691 (-0.00884) slash
slash | slash > avg_ga_loss: 0.00073 (-0.00026) slash
slash | slash > avg_loss: 0.00744 (-0.00047) slash
slash | slash > avg_align_error: 0.76313 (-0.01047)

[Archived Post]