SpecAugment: A Simple Data Augmentation Method for Automatic Speech Recognition
Written for the Research library (claude-opus-5-5) from the paper text, in our own words: the license "arXiv — non-exclusive license to distribute" does not allow redistributing the text itself. Original — arxiv.org/pdf/1904.08779
Summary
End-to-end speech recognizers overfit easily and need large amounts of labelled audio. Earlier augmentation methods mostly changed the raw waveform, for example through speed perturbation, added noise or room simulation. The authors propose SpecAugment, which instead alters the log mel spectrogram directly, treating it like an image. It needs no extra data and is cheap enough to run online during training. It combines three operations: time warping, masking runs of consecutive mel frequency channels, and masking runs of consecutive time steps. The masking is inspired by Cutout from computer vision. They apply it to Listen, Attend and Spell (LAS) models on LibriSpeech 960h and Switchboard 300h. They pair it with longer learning-rate schedules, since augmentation benefits more from extended training. The augmented LAS models beat the previous best hybrid systems even without a language model. Shallow fusion with an RNN LM improves results further. The work shows that a very simple, domain-agnostic input perturbation can let plain end-to-end models outperform heavily engineered hybrid pipelines.
Key points
- Results on LibriSpeech 960h:
- Without an LM, WER is 2.8% on test-clean and 6.8% on test-other.
- With shallow fusion, WER is 2.5% and 5.8%.
- The previous best hybrid result on test-other was 7.5%, so this is roughly a 22% relative gain.
- Results on Switchboard 300h (Hub5'00, Switchboard/CallHome portions):
- Without an LM, WER is 7.2%/14.6%.
- With fusion, WER is 6.8%/14.1%.
- The prior hybrid state of the art was 8.3%/17.3%.
- Augmentation policies: four hand-designed policies, LB, LD, SM and SS, each setting the warp parameter W, mask widths F and T, mask counts, and a cap p on time-mask width.
- Models and tokenization: LASd-w models (e.g., LAS-4-1024, LAS-6-1280) use a CNN front end, bi-LSTM encoder and 2-layer RNN decoder, with Word Piece Models of 16k (LibriSpeech) and 1k (Switchboard).
- Training schedules: three learning-rate schedules, B, D and the longest L (up to 320k steps). Longer schedules help more when augmentation is used. LibriSpeech runs took 7 days on 32 Cloud TPU chips.
Cited by
- wav2vec 2.0: A Framework for Self-Supervised Learning of Speech Representations
- Qwen-Audio: Advancing Universal Audio Understanding via Unified Large-Scale Audio-Language Models
- Seamless: Multilingual Expressive and Streaming Speech Translation
- FLEURS: Few-shot Learning Evaluation of Universal Representations of Speech
- Conformer: Convolution-augmented Transformer for Speech Recognition
- Perceiver: General Perception with Iterative Attention
- MLS: A Large-Scale Multilingual Dataset for Speech Research
- Robust Speech Recognition via Large-Scale Weak Supervision
- Unsupervised Data Augmentation for Consistency Training
Cite
@misc{park2019specaugment,
title = {{SpecAugment: A Simple Data Augmentation Method for Automatic Speech Recognition}},
author = {Park, Daniel S. and Chan, William and Zhang, Yu and Chiu, Chung-Cheng and Zoph, Barret and Cubuk, Ekin D. and Le, Quoc V.},
year = {2019},
doi = {10.21437/interspeech.2019-2680},
eprint = {1904.08779},
archivePrefix = {arXiv},
url = {https://doi.org/10.21437/interspeech.2019-2680}
}References
- G. Hinton, L. Deng, D. Yu, G. Dahl, A.-r. Mohamed, N. Jaitly, A. Senior, V. Vanhoucke, P. Nguyen, B. Kingsbury et al., “Deep neural networks for acoustic modeling in speech recognition,” IEEE Signal processing magazine, vol. 29, 2012. find
- G. Dahl, D. Yu, L. Deng, and A. Acero, “Context-Dependent Pre-Trained Deep Neural Networks for Large-Vocabulary Speech Recognition,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 20, Jan 2012. find
- T. Sainath, A. rahman Mohamed, B. Kingsbury, and B. Ramabhadran, “Deep Convolutional Neural Networks for LVCSR,” in ICASSP, 2013. find
- A. Graves, A. rahman Mohamed, and G. Hinton, “Speech Recognition with Deep Recurrent Neural Networks,” in ICASSP, 2013. find
- A. Graves and N. Jaitly, “Towards End-to-End Speech Recognition with Recurrent Neural Networks,” in ICML, 2014. find
- W. Chan, N. Jaitly, Q. V. Le, and O. Vinyals, “Listen, Attend and Spell: A Neural Network for Large Vocabulary Conversational Speech Recognition,” in ICASSP, 2016. find
- D. Bahdanau, J. Chorowski, D. Serdyuk, P. Brakel, and Y. Bengio, “End-to-End Attention-based Large Vocabulary Speech Recognition,” in ICASSP, 2016. find
- C.-C. Chiu, T. N. Sainath, Y. Wu, R. Prabhavalkar, P. Nguyen, Z. Chen, A. Kannan, R. J. Weiss, K. Rao, E. Gonina, N. Jaitly, B. Li, J. Chorowski, and M. Bacchiani, “State-of-the-art Speech Recognition With Sequence-to-Sequence Models,” in ICASSP, 2018. find
- N. Kanda, R. Takeda, and Y. Obuchi, “Elastic spectral distortion for low resource speech recognition with deep neural networks,” in ASRU, 2013. find
- A. Ragni, K. M. Knill, S. P. Rath, and M. J. F. Gales, “Data augmentation for low resource languages,” in INTERSPEECH, 2014. find
- N. Jaitly and G. Hinton, “Vocal Tract Length Perturbation (VTLP) improves speech recognition,” in ICML Workshop on Deep Learning for Audio, Speech and Language Processing, 2013. find
- A. Hannun, C. Case, J. Casper, B. Catanzaro, G. Diamos, E. Elsen, R. Prenger, S. Satheesh, S. Sengupta, A. Coates, and A. Ng, “Deep Speech: Scaling up end-to-end speech recognition,” in arXiv, 2014. find our card
- T. Ko, V. Peddinti, D. Povey, and S. Khudanpur, “Audio Augmentation for Speech Recognition,” in INTERSPEECH, 2015. find
- C. Kim, A. Misra, K. Chin, T. Hughes, A. Narayanan, T. Sainath, and M. Bacchiani, “Generation of large-scale simulated utterances in virtual rooms to train deep-neural networks for far-field speech recognition in Google Home,” in INTERSPEECH, 2017. find
- R. Prabhavalkar, R. Alvarez, C. Parada, P. Nakkiran, and T. N. Sainath, “Automatic gain control and multi-style training for robust small-footprint keyword spotting with deep neural networks,” in ICASSP, 2015. find
- A. Raju, S. Panchapagesan, X. Liu, A. Mandal, and N. Strom, “Data Augmentation for Robust Keyword Spotting under Playback Interference,” in arXiv, 2018. find
- S. H. Mallidi and H. Hermansky, “Novel neural network based fusion for Multistream ASR,” in ICASSP, 2016. find
- E. D. Cubuk, B. Zoph, D. Mané, V. Vasudevan, and Q. V. Le, “Autoaugment: Learning augmentation policies from data,” in CVPR, 2019. find
- T. DeVries and G. Taylor, “Improved Regularization of Convolutional Neural Networks with Cutout,” in arXiv, 2017. find
- V. Panayotov, G. Chen, D. Povey, and S. Khudanpur, “Librispeech: An ASR corpus based on public domain audio books,” in ICASSP, 2015. find
- Ç. Gülçehre, O. Firat, K. Xu, K. Cho, L. Barrault, H. Lin, F. Bougares, H. Schwenk, and Y. Bengio, “On using monolingual corpora in neural machine translation,” in arxiv, 2015. find our card
- J. Godfrey, E. Holliman, and J. McDaniel, “SWITCHBOARD: telephone speech corpus for research and development,” in ICASSP, 1992. find
- C. Cieri, D. Miller, and K. Walker, “The fisher corpus: a resource for the next generations of speech-to-text,” in LREC, 2004. find
- A. Zeyer, K. Irie, R. Schlüter, and H. Ney, “Improved training of end-to-end attention models for speech recognition,” in INTER-SPEECH, 2018. find
- K. Irie, R. Prabhavalkar, A. Kannan, A. Bruguier, D. Rybach, and P. Nguyen, “Model Unit Exploration for Sequence-to-Sequence Speech Recognition,” in arXiv, 2019. find
- M. Schuster and K. Nakajima, “Japanese and korean voice search,” in ICASSP, 2012. find
- A. Graves, “Practical Variational Inference for Neural Networks,” in NIPS, 2011. find
- C. Szegedy, V. Vanhoucke, S. Ioffe, J. Shlens, and Z. Wojna, “Rethinking the inception architecture for computer vision,” in CVPR, 2016. find
- J. Chorowski and N. Jaitly, “Towards better decoding and language model integration in sequence to sequence models,” in IN-TERSPEECH, 2017. find
- D. Povey, V. Peddinti, D. Galvez, P. Ghahrmani, V. Manohar, X. Na, Y. Wang, and S. Khudanpur, “Purely sequence-trained neural networks for ASR based on lattice-free MMI,” in INTER-SPEECH, 2016. find
- K. J. Han, A. Chandrashekaran, J. Kim, and I. Lane, “The CA-PIO 2017 Conversational Speech Recognition System,” in arXiv, 2017. find
- X. Yang, J. Li, and X. Zhou, “A novel pyramidal-FSMN architecture with lattice-free MMI for speech recognition,” in arXiv, 2018. find
- R. Collobert, C. Puhrsch, and G. Synnaeve, “Wav2Letter: an End-to-End ConvNet-based Speech Recognition System,” in arXiv, 2016. find
- V. Liptchinsky, G. Synnaeve, and R. Collobert, “Letter-Based Speech Recognition with Gated ConvNets,” in arXiv, 2017. find
- Y. Zhou, C. Xiong, and R. Socher, “Improving End-to-End Speech Recognition with Policy Learning,” in ICASSP, 2018. find
- N. Zeghidour, Q. Xu, V. Liptchinsky, N. Usunier, G. Synnaeve, and R. Collobert, “Fully Convolutional Speech Recognition,” in arXiv, 2018. find
- J. Li, V. Lavrukhin, B. Ginsburg, R. Leary, O. Kuchaiev, J. M. Cohen, H. Nguyen, and R. T. Gadde, “Jasper: An End-to-End Convolutional Neural Acoustic Model,” in arXiv, 2019. find
- A. Zeyer, A. Merboldt, R. Schlüter, and H. Ney, “A comprehensive analysis on attention models,” in NIPS: Workshop IRASL, 2018. find
- S. Sabour, W. Chan, and M. Norouzi, “Optimal Completion Distillation for Sequence Learning,” in ICLR, 2019. find
- D. Povey, A. Ghoshal, G. Boulianne, L. Burget, O. Glembek, N. Goel, M. Hannemann, P. Motlicek, Y. Qian, P. Schwarz, J. Silovsky, G. Stemmer, and K. Vesely, “The Kaldi Speech Recognition Toolkit,” in ASRU, 2011. find
- K. Vesely, A. Ghoshal, L. Burger, and D. Povey, “Sequence-discriminative training of deep neural networks,” in INTER-SPEECH, 2013. find
- H. Hadian, H. Sameti, D. Povey, and S. Khudanpur, “End-to-end speech recognition using lattice-free MMI,” in INTERSPEECH, 2018. find
- G. Zweig, C. Yu, J. Droppo, and A. Stolcke, “Advances in All-Neural Speech Recognition,” in ICASSP, 2017. find
- K. Audhkhasi, B. Ramabhadran, G. Saon, M. Picheny, and D. Nahamoo, “Direct Acoustics-to-Word Models for English Conversational Speech Recognition,” in INTERSPEECH, 2018. find
- K. Audhkhasi, B. Kingsbury, B. Ramabhadran, G. Saon, and M. Picheny, “Building competitive direct acoustics-to-word models for english conversational speech recognition,” in ICASSP, 2018. find
- L. Lu, X. Zhang, and S. Renals, “On training the recurrent neural network encoder-decoder for large vocabulary end-to-end speech recognition,” in ICASSP, 2016. find
- S. Toshniwal, H. Tang, L. Lu, and K. Livescu, “Multitask Learning with Low-Level Auxiliary Tasks for Encoder-Decoder Based Speech Recognition,” in INTERSPEECH, 2017. find
- C. Weng, J. Cui, G. Wang, J. Wang, C. Yu, D. Su, and D. Yu, “Improving Attention Based Sequence-to-Sequence Models for End-to-End English Conversational Speech Recognition,” in INTER-SPEECH, 2018. find
- G. Kovács, L. Tóth, D. Van Compernolle, and S. Ganapathy, “Increasing the robustness of cnn acoustic models using autoregressive moving average spectrogram features and channel dropout,” Pattern Recognition Letters, vol. 100, pp. 44–50, 2017. find
- L. Tóth, G. Kovács, and D. Van Compernolle, “A perceptually inspired data augmentation method for noise robust cnn acoustic models,” in SPECOM, 2018. find