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TTS/tts/datasets/output/ljspeech-ddc-June-14-2023_02+02PM-d0d1618/config.json
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{ | ||
"github_branch":"* master", | ||
"model": "Tacotron2", | ||
"run_name": "ljspeech-ddc", | ||
"run_description": "tacotron2 with DDC and differential spectral loss.", | ||
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// 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. | ||
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// Audio processing parameters | ||
"sample_rate": 22050, // DATASET-RELATED: wav sample-rate. | ||
"preemphasis": 0.0, // pre-emphasis to reduce spec noise and make it more structured. If 0.0, no -pre-emphasis. | ||
"ref_level_db": 20, // reference level db, theoretically 20db is the sound of air. | ||
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// Silence trimming | ||
"do_trim_silence": true,// enable trimming of slience of audio as you load it. LJspeech (true), TWEB (false), Nancy (true) | ||
"trim_db": 60, // threshold for timming silence. Set this according to your dataset. | ||
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// 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. | ||
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// 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, | ||
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// 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": "TTS/tts/datasets/output/scale_stats.npy" // 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 | ||
}, | ||
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// 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" | ||
// }, | ||
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// DISTRIBUTED TRAINING | ||
"distributed":{ | ||
"backend": "nccl", | ||
"url": "tcp:\/\/localhost:54321" | ||
}, | ||
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"reinit_layers": [], // give a list of layer names to restore from the given checkpoint. If not defined, it reloads all heuristically matching layers. | ||
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// TRAINING | ||
"batch_size": 100, // 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. | ||
"mixed_precision": true, // 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. | ||
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// LOSS SETTINGS | ||
"loss_masking": true, // enable / disable loss masking against the sequence padding. | ||
"decoder_loss_alpha": 0.5, // original decoder loss weight. If > 0, it is enabled | ||
"postnet_loss_alpha": 0.25, // original postnet loss weight. If > 0, it is enabled | ||
"postnet_diff_spec_alpha": 0.25, // differential spectral loss weight. If > 0, it is enabled | ||
"decoder_diff_spec_alpha": 0.25, // differential spectral loss weight. If > 0, it is enabled | ||
"decoder_ssim_alpha": 0.5, // decoder ssim loss weight. If > 0, it is enabled | ||
"postnet_ssim_alpha": 0.25, // postnet ssim loss weight. If > 0, it is enabled | ||
"ga_alpha": 5.0, // weight for guided attention loss. If > 0, guided attention is enabled. | ||
"stopnet_pos_weight": 15.0, // pos class weight for stopnet loss since there are way more negative samples than positive samples. | ||
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// 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. | ||
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// 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": 0.0001, // 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. | ||
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// 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": false, // enable/disable dropout at prenet. | ||
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// TACOTRON ATTENTION | ||
"attention_type": "original", // 'original' , 'graves', 'dynamic_convolution' | ||
"attention_heads": 4, // number of attention heads (only for 'graves') | ||
"attention_norm": "sigmoid", // softmax or sigmoid. | ||
"windowing": false, // Enables attention windowing. Used only in eval mode. | ||
"use_forward_attn": false, // 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": true, // use DDC explained here https://erogol.com/solving-attention-problems-of-tts-models-with-double-decoder-consistency-draft/ | ||
"ddc_r": 7, // reduction rate for coarse decoder. | ||
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// 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. | ||
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// 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. | ||
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// 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 | ||
"compute_input_seq_cache": false, // if true, text sequences are computed before starting training. If phonemes are enabled, they are also computed at this stage. | ||
"use_noise_augment": true, | ||
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// PATHS | ||
"output_path": "TTS/tts/datasets/output/", | ||
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// PHONEMES | ||
"phoneme_cache_path": "TTS/tts/phoneme_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 | ||
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// MULTI-SPEAKER and GST | ||
"use_speaker_embedding": false, // use speaker embedding to enable multi-speaker learning. | ||
"use_gst": false, // use global style tokens | ||
"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 | ||
"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, | ||
"gst_use_speaker_embedding": false | ||
}, | ||
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// DATASETS | ||
"datasets": // List of datasets. They all merged and they get different speaker_ids. | ||
[ | ||
{ | ||
"name": "ljspeech", | ||
"path": "TTS/tts/datasets/", | ||
// "data_path":"TTS/tts/datasets/wavs", | ||
"meta_file_train": "Amharic.txt", // for vtck if list, ignore speakers id in list for train, its useful for test cloning with new speakers | ||
"meta_file_val": "Amharic.txt" | ||
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} | ||
] | ||
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} | ||
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