klampropoulos / end-to-end-asr-pytorch

This is an open source project (formerly named Listen, Attend and Spell - PyTorch Implementation) for end-to-end ASR implemented with Pytorch, the well known deep learning toolkit.

End-to-end Automatic Speech Recognition Systems - PyTorch Implementation

This is an open source project (formerly named Listen, Attend and Spell - PyTorch Implementation) for end-to-end ASR implemented with Pytorch, the well known deep learning toolkit.

The end-to-end ASR was based on Listen, Attend and Spell¹. Multiple techniques proposed recently were also implemented, serving as additional plug-ins for better performance. For the list of techniques implemented, please refer to the highlights, configuration and references.

Feel free to use/modify them, any bug report or improvement suggestion will be appreciated. If you have any questions, please contact r07922013[AT]ntu.edu.tw. If you find this project helpful for your research, please do consider to cite my paper, thanks!

Highlights

• Acoustic feature extraction

  • Purepython extraction using librosa as the backend
  • One-click execution scripts (currently supporting TIMT & LibriSpeech)
  • Phoneme/character/subword²/word embedding for text encoding

• End-to-end ASR

  • Seq2seq ASR with different types of encoder/attention³
  • CTC-based ASR⁴, which can also be hybrid⁵ with the former
  • yaml-styled model construction and hyper parameters setting
  • Training process visualization with TensorBoard, including attention alignment

• Speech recognition (decoding)

  • Beam search decoding
  • RNN language model training and joint decoding for ASR⁶
  • Joint CTC-attention based decoding⁶

You may checkout some example log files with TensorBoard by downloading them from log/log_url.txt

Requirements

• Python 3
• Computing power (high-end GPU) and memory space (both RAM/GPU's RAM) is extremely important if you'd like to train your own model.
• Required packages and their use are listed here.

Instructions

Before you start, make sure all the packages required were installed correctly

Step 0. Preprocessing - Acoustic Feature Extraction & Text Encoding

Preprocessing scripts are placed under data/, you may execute them directly. The extracted data, which is ready for training, will be stored in the same place by default. For example,

cd data/
python3 preprocess_libri.py --data_path <path to LibriSpeech on your computer> 

The parameters available for these scripts are as follow,

Options	Description
data_path	Path to the raw dataset (can be obtained by download & unzip)
feature_type	Which type of acoustic feature to be extracted, fbank or mfcc
feature_dim	Feature dimension, usually depends on the feature type (e.g. 13 for mfcc)
apply_delta	Append delta of the acoustic feature to itself
apply_delta_delta	Append delta of delta
apply_cmvn	Normalize acoustic feature
output_path	Output path for extracted data (by default, it's data/)
target	Text encoding target, one of phoneme/char/subword/word
n_tokens	Vocabulary size, only applies for subword/word

You may check the parameter type and default value by using the option --help for each script.

Step 1. Configuring - Model Design & Hyperparameter Setup

All the parameters related to training/decoding will be stored in a yaml file. Hyperparameter tuning and massive experiment and can be managed easily this way. See documentation and examples for the exact format. Note that the example configs provided were not fine-tuned, you may want to write your own config for best performance.

Step 2. Training - End-to-end ASR (or RNN-LM) Learning

Once the config file is ready, run the following command to train end-to-end ASR (or language model)

python3 main.py --config <path of config file> 

All settings will be parsed from the config file automatically to start training, the log file can be accessed through TensorBoard. Please notice that the error rate reported on the TensorBoard is biased (see issue #10), you should run the testing phase in order to get the true performance of model. For example, train an ASR on LibriSpeech and watch the log with

python3 main.py --config config/libri_example.yaml
# open TensorBoard to see log
tensorboard --logdir log/
# Train an external language model
python3 main.py --config config/libri_example_rnnlm.yaml --rnnlm

There are also some options, which do not influence the performance (except seed), are available in this phase including the followings

Options	Description
config	Path of config file
seed	Random seed, note this is an option that affects the result
name	Experiments for logging and saving model, by default it's _
logdir	Path to store training logs (log files for tensorboard), default log/
ckpdir	Path to store results, default result/<name>
njobs	Number of workers for Pytorch DataLoader
cpu	CPU-only mode, not recommended
no-msg	Hide all message from stdout
rnnlm	Switch to rnnlm training mode
test	Switch to decoding mode (do not use during training phase)

Step 3. Testing - Speech Recognition & Performance Evaluation

Once a model was trained, run the following command to test it

python3 main.py --config <path of config file> --test

Recognition result will be stored at result/<name>/ as a txt file with auto-naming according to the decoding parameters specified in config file. The result file may be evaluated with eval.py. For example, test the ASR trained on LibriSpeech and check performance with

python3 main.py --config config/libri_example.yaml --test
# Check WER/CER
python3 eval.py --file result/libri_example_sd0/decode_*.txt

Notice that the meaning of options for main.py in this phase will change

Options	Description
test	Must be enabled
config	Path of config file
name	Must be identical to the models training phase
ckpdir	Must be identical to the models training phase
njobs	Number of threads used for decoding, very important in terms of efficiency. Large value equals fast decoding yet RAM/GPU RAM expensive.
cpu	CPU-only mode, not recommended
no-msg	Hide all message from stdout

Rest of the options are ineffective in the testing phase.

ToDo

• Plot attention map during testing
• Pure CTC training
• Resume model training

Acknowledgements

• Parts of the implementation refer to ESPnet, a great end-to-end speech processing toolkit by Watanabe et al.
• Special thanks to William Chan, the first author of LAS, for answering my questions during implementation.
• Thanks xiaoming, Odie Ko, b-etienne, Jinserk Baik and Zhong-Yi Li for identifying several issues in our implementation.

Reference

1. Listen, Attend and Spell, W Chan et al.
2. Neural Machine Translation of Rare Words with Subword Units, R Sennrich et al.
3. Attention-Based Models for Speech Recognition, J Chorowski et al.
4. Connectionist Temporal Classification: Labelling Unsegmented Sequence Data with Recurrent Neural Networks, A Graves et al.
5. Joint CTC-Attention based End-to-End Speech Recognition using Multi-task Learning, S Kim et al.
6. Advances in Joint CTC-Attention based End-to-End Speech Recognition with a Deep CNN Encoder and RNN-LM, T Hori et al.

Citation

@inproceedings{liu2019adversarial,
  title={Adversarial Training of End-to-end Speech Recognition Using a Criticizing Language Model},
  author={Liu, Alexander and Lee, Hung-yi and Lee, Lin-shan},
  booktitle={Acoustics, Speech and Signal Processing (ICASSP)},
  year={2019},
  organization={IEEE}
}