TTS/tests/inputs/test_align_tts.json

{
    "model": "align_tts",
    "run_name": "test_sample_dataset_run",
    "run_description": "sample dataset test run",

    // 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.
         "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.

         // 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.

         // 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,

         // 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": "ABCDEFGHIJKLMNOPQRSTUVWXYZÇÃÀÁÂÊÉÍÓÔÕÚÛabcdefghijklmnopqrstuvwxyzçãàáâêéíóôõúû!(),-.:;? ",
    //     "punctuations":"!'(),-.:;? ",
    //     "phonemes":"iyɨʉɯuɪʏʊeøɘəɵɤoɛœɜɞʌɔæɐaɶɑɒᵻʘɓǀɗǃʄǂɠǁʛpbtdʈɖcɟkɡqɢʔɴŋɲɳnɱmʙrʀⱱɾɽɸβfvθðszʃʒʂʐçʝxɣχʁħʕhɦɬɮʋɹɻjɰlɭʎʟˈˌːˑʍwɥʜʢʡɕʑɺɧɚ˞ɫ'̃' "
    // },

    "add_blank": false, // if true add a new token after each token of the sentence. This increases the size of the input sequence, but has considerably improved the prosody of the GlowTTS model.

    // DISTRIBUTED TRAINING
    "distributed":{
        "backend": "nccl",
        "url": "tcp:\/\/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.

    // MODEL PARAMETERS
    "positional_encoding": true,
    "hidden_channels": 256,
    "hidden_channels_dp": 256,
    "encoder_type": "fftransformer",
    "encoder_params":{
        "hidden_channels_ffn": 1024 ,
        "num_heads": 2,
        "num_layers": 6,
        "dropout_p": 0.1
    },
    "decoder_type": "fftransformer",
    "decoder_params":{
        "hidden_channels_ffn": 1024 ,
        "num_heads": 2,
        "num_layers": 6,
        "dropout_p": 0.1
    },


    // TRAINING
    "batch_size":2,       // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
    "eval_batch_size":1,
    "r": 1,                 // Number of decoder frames to predict per iteration. Set the initial values if gradual training is enabled.
    "loss_masking": true,   // enable / disable loss masking against the sequence padding.
    "phase_start_steps": null,


    // LOSS PARAMETERS
    "ssim_alpha": 1,
    "spec_loss_alpha": 1,
    "dur_loss_alpha": 1,
    "mdn_alpha": 1,

    // VALIDATION
    "run_eval": true,
    "test_delay_epochs": -1,       //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": true,         // use noam warmup and lr schedule.
    "grad_clip": 1.0,              // upper limit for gradients for clipping.
    "epochs": 1,               // total number of epochs to train.
    "lr": 0.002,                    // Initial learning rate. If Noam decay is active, maximum learning rate.
    "warmup_steps": 4000,          // Noam decay steps to increase the learning rate from 0 to "lr"

   // TENSORBOARD and LOGGING
   "print_step": 1,       // 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": 5000,      // Number of training steps expected to save traninpg stats and checkpoints.
   "checkpoint": true,     // If true, it saves checkpoints per "save_step"
   "keep_all_best": true,  // If true, keeps all best_models after keep_after steps
   "keep_after": 10000,    // Global step after which to keep best models if keep_all_best is true
   "tb_model_param_stats": false,     // true, plots param stats per layer on tensorboard. Might be memory consuming, but good for debugging.:set n
   "mixed_precision": false,

   // DATA LOADING
   "text_cleaner": "english_cleaners",
   "enable_eos_bos_chars": false, // enable/disable beginning of sentence and end of sentence chars.
   "num_loader_workers": 0,        // number of training data loader processes. Don't set it too big. 4-8 are good values.
   "num_eval_loader_workers": 0,    // number of evaluation data loader processes.
   "batch_group_size": 0,  //Number of batches to shuffle after bucketing.
   "min_seq_len": 2,       // DATASET-RELATED: minimum text length to use in training
   "max_seq_len": 300,     // DATASET-RELATED: maximum text length
   "compute_f0": false,     // compute f0 values in data-loader
   "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.

   // PATHS
   "output_path": "tests/train_outputs/",

   // PHONEMES
   "phoneme_cache_path": "tests/train_outputs/phoneme_cache/",  // phoneme computation is slow, therefore, it caches results in the given folder.
   "use_phonemes": false,           // use phonemes instead of raw characters. It is suggested for better pronoun[ciation.
   "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_d_vector_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
   "d_vector_file": "/home/erogol/Data/libritts/speakers.json", // if not null and use_d_vector_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


    // DATASETS
    "datasets":   // List of datasets. They all merged and they get different speaker_ids.
    [
        {
            "formatter": "ljspeech",
            "path": "tests/data/ljspeech/",
            "meta_file_train": "metadata.csv",
            "meta_file_val": "metadata.csv",
            "meta_file_attn_mask": null
        }
    ]
}