30stomercury/Automatic-Speech-Recognition

End-to-End Speech Recognition Using Tensorflow

A TensorFlow Implementation of Listen Attention and Spell

This is a tensorflow implementation of end-to-end ASR. Though there are several fantastic github repos in tensorflow, I tried to implement it without using tf.contrib.seq2seq API in this repo. In addition, the performance on LibriSpeech dev/test sets is evaluated and the evaluation results are reasonable.

Overview

• Components:
  • Char/Subword text encoding.
  • MFCC/fbank acoustic features with per utterance CMVN.
  • LAS training (visualized with tensorboard: loss, sample text outputs, features, alignments).
  • TFrecord dataset pipline.
  • Batch testing using greedy decoder.
  • Beam search decoder.
  • RNNLM.

Remarks

Note that this project is still in progress.

• Notes

  • Currently, I only test this model on MFCC 39 (13+delta+accelerate) features.
  • CTC related parts are not yet fully tested.
  • Volume augmentation is currently commented out because it shows little improvements.

• Improvements

  • CNN-based Listener.
  • Location-aware attention.
  • Augmentation include speed perturbation.
  • Label smoothing. (IMPORANT)
  • Scheduled learning rate.
  • Scheduled sampling.
  • Bucketing.

• Some advice

  • Generally, LibriSpeech-100 is not large enough to train a well-perform LAS.
  • In my experience, adding more data is the best policy.
  • A better way to check if your model is learning in a right way is to monitor the speech-text aligments in tensorboard or set verbosity=1.

Requirements

pip3 install virtualenv
virtualenv --python=python3 venv
source venv/bin/activate
pip3 install -r requirements.txt

Arguments

The definitions of the args are described in las/arguments.py. You can modify all args there before preprocessing, training, testing and decoding.

Usage

0) Prepare data

• Libirspeech train/dev/test data

bash prepare_libri_data.sh 

1) Preprocess Audios & Texts. / Training pipeline.

I include dataset pipeline and training pipeline in run.sh.

bash run.sh

Train RNNLM: (NOT READY)

2) Testing

Testing with gready decoder.

python3 test.py --split SPLIT \           # test or dev
                --unit UNIT \ 
                --feat_dim FEAT_DIM \ 
                --feat_dir FEAT_DIR \
                --save_dir SAVE_DIR 

3) Decode

Beam search decoder.

python3 decode.py --split SPLIT \         # test or dev
                  --unit UNIT \ 
                  --beam_size BEAM_SIZE \
                  --convert_rate 0.24 \   # 0.24 is large enough.
                  --apply_lm APPLY_LM \         
                  --lm_weight LM_WEIGHT \
                  --feat_dim FEAT_DIM \ 
                  --feat_dir FEAT_DIR \
                  --save_dir SAVE_DIR 

Tensorboard

tensorboard --logdir ./summary

Results

Results trained on LibriSpeech-360 (WER)

Model	dev-clean	test-clean
Char LAS	0.249	0.262

Reference

• Listen, Attend and Spell
• Attention-Based Models for Speech Recognition
• On the Choice of Modeling Unit for Sequence-to-Sequence Speech Recognition
• Char RNNLM: TensorFlow-Char-RNN
• A nice Pytorch version: End-to-end-ASR-Pytorch.

TODO

• Add scheduled sampling.
• Add location-aware attention.
• Evaluate the performance with subword unit: Subword las training.
• Decoding with subword-based RNNLM.
• Evaluate the performance on joint CTC training, decoding.