From cedc22847beff4a3f8f0b088463b37355717b36d Mon Sep 17 00:00:00 2001 From: erogol Date: Wed, 20 May 2020 18:16:24 +0200 Subject: [PATCH] fix synthesize.py --- config_template.json | 134 ------------------------------------------- synthesize.py | 3 +- 2 files changed, 2 insertions(+), 135 deletions(-) delete mode 100644 config_template.json diff --git a/config_template.json b/config_template.json deleted file mode 100644 index e525ec31..00000000 --- a/config_template.json +++ /dev/null @@ -1,134 +0,0 @@ -{ - "model": "Tacotron2", // one of the model in models/ - "run_name": "ljspeech-stft_params", - "run_description": "tacotron2 cosntant stf parameters", - - // AUDIO PARAMETERS - "audio":{ - // Audio processing parameters - "num_mels": 80, // size of the mel spec frame. - "num_freq": 513, // number of stft frequency levels. Size of the linear spectogram frame. - "sample_rate": 22050, // DATASET-RELATED: wav sample-rate. If different than the original data, it is resampled. - "win_length": 1024, // stft window length in ms. - "hop_length": 256, // stft window hop-lengh in ms. - "preemphasis": 0.0, // pre-emphasis to reduce spec noise and make it more structured. If 0.0, no -pre-emphasis. - "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. - "min_level_db": -100, // normalization range - "ref_level_db": 20, // reference level db, theoretically 20db is the sound of air. - "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. - // Normalization parameters - "signal_norm": true, // normalize the spec values in range [0, 1] - "symmetric_norm": true, // move normalization to range [-1, 1] - "max_norm": 1.0, // scale normalization to range [-max_norm, max_norm] or [0, max_norm] - "clip_norm": true, // clip normalized values into the range. - "mel_fmin": 0.0, // minimum freq level for mel-spec. ~50 for male and ~95 for female voices. Tune for dataset!! - "mel_fmax": 8000.0, // maximum freq level for mel-spec. Tune for dataset!! - "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. - }, - - // 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:\/\/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": 10.0, // weight for guided attention loss. If > 0, guided attention is enabled. - - // 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": 0.0001, // 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. - - // 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": 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. - - // 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 traning on console. - "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": 0, //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": "/data4/rw/home/Trainings/", - - // PHONEMES - "phoneme_cache_path": "mozilla_us_phonemes_3", // 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. - "style_wav_for_test": null, // path to style wav file to be used in TacotronGST inference. - "use_gst": false, // TACOTRON ONLY: use global style tokens - - // DATASETS - "datasets": // List of datasets. They all merged and they get different speaker_ids. - [ - { - "name": "ljspeech", - "path": "/root/LJSpeech-1.1/", - "meta_file_train": "metadata.csv", - "meta_file_val": null - } - ] - -} - diff --git a/synthesize.py b/synthesize.py index 1a760268..18048c2f 100644 --- a/synthesize.py +++ b/synthesize.py @@ -7,7 +7,8 @@ import json import string from TTS.utils.synthesis import synthesis -from TTS.utils.generic_utils import load_config, setup_model +from TTS.utils.generic_utils import setup_model +from TTS.utils.io import load_config from TTS.utils.text.symbols import make_symbols, symbols, phonemes from TTS.utils.audio import AudioProcessor