Closed a-froghyar closed 3 years ago
@WeberJulian has an idea as he is also using GST
"r": 4,
set this to "r": 7
I think since you are using gradual training it loads the first batch of data with r=4 but tries to train with r=7
Edit: Just checked and got the same error
Traceback (most recent call last):
File "TTS/bin/train_tacotron.py", line 721, in <module>
main(args)
File "TTS/bin/train_tacotron.py", line 623, in main
scaler_st)
File "TTS/bin/train_tacotron.py", line 166, in train
text_input, text_lengths, mel_input, mel_lengths, speaker_ids=speaker_ids, speaker_embeddings=speaker_embeddings)
File "/media/alexander/LinuxFS/Projects/TTS-dev/.venv/lib/python3.6/site-packages/torch/nn/modules/module.py", line 727, in _call_impl
result = self.forward(*input, **kwargs)
File "/media/alexander/LinuxFS/Projects/TTS-dev/TTS/tts/models/tacotron.py", line 173, in forward
decoder_outputs = decoder_outputs * output_mask.unsqueeze(1).expand_as(decoder_outputs)
RuntimeError: The expanded size of the tensor (216) must match the existing size (378) at non-singleton dimension 2. Target sizes: [64, 80, 216]. Tensor sizes: [64, 1, 378]
The error is not related to GST. Just set r=7
in your config and it should work.
Hum the error I had was with speaker_id initialisation in eval, but here it's concerning gst and it's during training. Here my best advice would be to compare tacotron and tacotron 2 in the debugger.
Hey @WeberJulian thanks for your answer. What should I be looking for when I compare the two?
@a-froghyar I would look for differences in shapes of the different tensors (like encoder_outputs before and after compute_gst)
Have you checked yet if the same config in tacotron 2 works for you ?
@WeberJulian not yet, I'll be on it later today and I'll report back, thanks again!
@WeberJulian @erogol I'm trying to set up my debuging config, could you shed some light on debugging best practices please? :)
@a-froghyar have you checked @SanjaESC answer?
@erogol yes I did, same error with T1, T2 threw a librosa error, my Dataset seems to be stereo apparently, will check back. librosa.util.exceptions.ParameterError: Invalid shape for monophonic audio: ndim=2, shape=(19680, 2)
- I've checked and weirdly SOME of the wav files in my corpus are stereo, going to convert to mono now and try again.
for debugging you can use the small dataset set under tests
folder. There are also some sample configs for model testing that you can copy and paste for debugging.
Thanks, what I meant was actually how to launch the debugger in the IDE using a specific config.json for the training run? If I launch the debugger on train_tacotron.py
, how do I feed it the config that I'm using?
Solved my problem, sorry for the noob question, just needed to add the argument into my launch.json file like so:
"args": ["--config_path", "TTS/tts/configs/gst_blizzard.json"]
,
where gst_blizzard.json
is my custom config file
@WeberJulian I've realised that I had a higher numpy version before, I downgraded to the one in the requirements and T2 is running, T1 now throws this error:
loss = functional.l1_loss(x * mask, target * mask, reduction='sum') RuntimeError: The size of tensor a (80) must match the size of tensor b (513) at non-singleton dimension 2
After a few iterations, target - [64,111,80]
actually has a different size to mask - [64,111,513]
t1 needs linear spectrograms. I guess you set the dataloader to return melspectrograms or there is something wrong in your settings. Pls pot tyour config here. @a-froghyar
For debugging I don't use the IDE. I just use breakpoint()
or import pdb; pdb.set_trace()
depending on th pythonversion and co through the code on termina l
Thanks, doesn't t1 need Mel and Linear? Here's my config:
{ "model": "Tacotron", "run_name": "blizzard-gts", "run_description": "tacotron with DDC and differential spectral loss.",
// 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": 24000, // 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": 95.0, // minimum freq level for mel-spec. ~50 for male and ~95 for female voices. Tune for dataset!!
"mel_fmax": 12000.0, // 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": "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": 64, // 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.
// LOSS SETTINGS
"loss_masking": false, // 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.
// 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": 300000, // 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.
// TACOTRON ATTENTION
"attention_type": "graves", // '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": false, // 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.
// 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": 5000, // 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": 8, // number of training data loader processes. Don't set it too big. 4-8 are good values.
"num_val_loader_workers": 8, // 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,
// PATHS
"output_path": "/home/big-boy/Models/Blizzard/",
// PHONEMES
"phoneme_cache_path": "/home/big-boy/Models/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
// MULTI-SPEAKER and GST
"use_speaker_embedding": false, // use speaker embedding to enable multi-speaker learning.
"use_gst": true, // 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
},
// DATASETS
"datasets": // List of datasets. They all merged and they get different speaker_ids.
[{
"name": "ljspeech",
"path": "/home/big-boy/Data/blizzard2013/segmented/",
"meta_file_train": "metadata.csv", // for vtck if list, ignore speakers id in list for train, its useful for test cloning with new speakers
"meta_file_val": null
}]
}
can you post the full error here? I need the line number/
/home/big-boy/anaconda3/envs/PyCapacitron/lib/python3.8/site-packages/torch/nn/modules/loss.py:94: UserWarning: Using a target size (torch.Size([64, 90, 80])) that is different to the input size (torch.Size([64, 90, 513])). This will likely lead to incorrect results due to broadcasting. Please ensure they have the same size.
return F.l1_loss(input, target, reduction=self.reduction)
! Run is removed from /home/big-boy/Models/Blizzard/blizzard-gts-March-11-2021_05+38PM-45068a9
Traceback (most recent call last):
File "/home/big-boy/projects/TTS/TTS/bin/train_tacotron.py", line 721, in
Ok I figured. It's a :bug: :)
If I understand correctly it's the linear spectrogram that gets passed down instead of the mel? The linear is only needed for the Griffin Lin conversion, right?
There is no "need" but it just works better with GL.
@a-froghyar dev branch should work now
Thanks so much!
@erogol I was actually working in the main branch, I got the following error in the dev branch, not sure how it got to try to phonemize chinese:
Traceback (most recent call last):
File "TTS/bin/train_tacotron.py", line 13, in
did you pip install -e . again after the checkout ?
Training is running, thank you guys so much!
I had to downgrade to librosa==0.6.3, though because of this:
librosa.util.exceptions.ParameterError: Audio buffer is not Fortran-contiguous. Use numpy.asfortranarray to ensure Fortran contiguity.
I have the same problem with v0.0.12:
CUDA_VISIBLE_DEVICES="0" python ../../TTS/bin/train_tacotron.py --config_path model_config.json
> Using CUDA: True
> Number of GPUs: 1
> Git Hash: 59ab268
> Experiment folder: /home/lpierron/Mozilla_TTS/CORPUS_LP/Models/maiLabs-fr-dca/mailabs-fr-ddc-April-23-2021_03+38PM-59ab268
> Setting up Audio Processor...
| > sample_rate:22050
| > resample:False
| > num_mels:80
| > min_level_db:-100
| > frame_shift_ms:None
| > frame_length_ms:None
| > ref_level_db:20
| > fft_size:1024
| > power:1.5
| > preemphasis:0.0
| > griffin_lim_iters:60
| > signal_norm:True
| > symmetric_norm:True
| > mel_fmin:0
| > mel_fmax:8000.0
| > spec_gain:1.0
| > stft_pad_mode:reflect
| > max_norm:4.0
| > clip_norm:True
| > do_trim_silence:True
| > trim_db:60
| > do_sound_norm:False
| > stats_path:./scale_stats.npy
| > log_func:<ufunc 'log10'>
| > exp_func:<function AudioProcessor.__init__.<locals>.<lambda> at 0x7f1ef7ac6c10>
| > hop_length:256
| > win_length:1024
| > /tmp/tts/by_book/female/ezwa/monsieur_lecoq/metadata.csv
| > Found 14211 files in /tmp/tts
> Using model: Tacotron2
> Model has 28183506 parameters
> Starting with inf best loss.
> DataLoader initialization
| > Use phonemes: True
| > phoneme language: fr-fr
| > Number of instances : 14069
| > Max length sequence: 281
| > Min length sequence: 3
| > Avg length sequence: 105.0826640130784
| > Num. instances discarded by max-min (max=153, min=6) seq limits: 2420
| > Batch group size: 128.
> EPOCH: 0/1000
> Number of output frames: 7
> TRAINING (2021-04-23 15:38:18)
! Run is removed from /home/lpierron/Mozilla_TTS/CORPUS_LP/Models/maiLabs-fr-dca/mailabs-fr-ddc-April-23-2021_03+38PM-59ab268
Traceback (most recent call last):
File "../../TTS/bin/train_tacotron.py", line 744, in <module>
main(args)
File "../../TTS/bin/train_tacotron.py", line 704, in main
train_avg_loss_dict, global_step = train(
File "../../TTS/bin/train_tacotron.py", line 198, in train
decoder_output, postnet_output, alignments, stop_tokens = model(
File "/home/lpierron/miniconda3/envs/tts/lib/python3.8/site-packages/torch/nn/modules/module.py", line 889, in _call_impl
result = self.forward(*input, **kwargs)
File "/home/lpierron/Mozilla_TTS/COQUI-TTS/TTS/TTS/tts/models/tacotron2.py", line 226, in forward
decoder_outputs = decoder_outputs * output_mask.unsqueeze(1).expand_as(decoder_outputs)
RuntimeError: The expanded size of the tensor (12) must match the existing size (84) at non-singleton dimension 2. Target sizes: [64, 80, 12]. Tensor sizes: [64, 1, 84]
I have downgraded librosa==0.6.3
but it doesn't work.
See my configuration next:
Hey,
I'm trying to run a training with Tacotron 1 using GST. I get the error on the first batch already.
Pytorch version: 1.8 and 1.7.1 (both yielded the same error) Python version: 3.8.0
Traceback (most recent call last): File "TTS/bin/train_tacotron.py", line 721, in <module> main(args) File "TTS/bin/train_tacotron.py", line 619, in main train_avg_loss_dict, global_step = train(train_loader, model, File "TTS/bin/train_tacotron.py", line 168, in train decoder_output, postnet_output, alignments, stop_tokens = model( File "/home/big-boy/anaconda3/envs/PyCapacitron/lib/python3.8/site-packages/torch/nn/modules/module.py", line 889, in _call_impl result = self.forward(*input, **kwargs) File "/home/big-boy/projects/TTS/TTS/tts/models/tacotron.py", line 173, in forward decoder_outputs = decoder_outputs * output_mask.unsqueeze(1).expand_as(decoder_outputs) RuntimeError: The expanded size of the tensor (64) must match the existing size (112) at non-singleton dimension 2. Target sizes: [64, 80, 64]. Tensor sizes: [64, 1, 112]
My hyperparams: // TRAINING "batch_size": 64, "eval_batch_size": 16, "r": 4, "gradual_training": [ [0, 7, 64], [1, 5, 64], [50000, 3, 32], [130000, 2, 32], [290000, 1, 32] ], "mixed_precision": true,
// MULTI-SPEAKER and GST "use_speaker_embedding": false, // use speaker embedding to enable multi-speaker learning. "use_gst": true, "use_external_speaker_embedding_file": false, "external_speaker_embedding_file": "../../speakers-vctk-en.json", "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 },