Skip to main content

Automatic Speech Recognition

  • Chapter
  • First Online:
Speech-to-Speech Translation

Part of the book series: SpringerBriefs in Computer Science ((BRIEFSCOMPUTER))

Abstract

The main task of automatic speech recognition (ASR) is to convert voice signals to text transcriptions. It is one of the most important research fields in natural language processing (NLP). With more than a half century of endeavor, the word error rate (WER), which is a metric unit for transcription performance, has significantly been reduced. Particularly in recent years, due to the increase of computational power, large quantity of collected data, and efficient neural learning algorithms, the dominant power of deep learning technology further enhanced the performance of ASR systems to a practical level. However, there are still many issues that need to be further investigated for these systems to be adapted to a wide range of applications. In this chapter, we will introduce the main stream and pipeline of ASR frameworks, particularly the two dominant frameworks, i.e., Hidden Markov Model (HMM) with Gaussian Mixture model (GMM)-based ASR which dominated the field in the early decades, and deep learning model-based ASR which dominates the techniques used now. In addition, noisy robustness, which is one of the most important challenges for ASR in real applications, will also be introduced.

This is a preview of subscription content, log in via an institution to check access.

Access this chapter

Chapter
USD 29.95
Price excludes VAT (USA)
  • Available as PDF
  • Read on any device
  • Instant download
  • Own it forever
eBook
USD 39.99
Price excludes VAT (USA)
  • Available as EPUB and PDF
  • Read on any device
  • Instant download
  • Own it forever
Softcover Book
USD 54.99
Price excludes VAT (USA)
  • Compact, lightweight edition
  • Dispatched in 3 to 5 business days
  • Free shipping worldwide - see info

Tax calculation will be finalised at checkout

Purchases are for personal use only

Institutional subscriptions

Notes

  1. 1.

    http://voicetra.nict.go.jp/en/.

  2. 2.

    SPINE: Speech in noisy environments, http://www.speech.sri.com/projects/spine/.

  3. 3.

    Aurora speech recognition experimental framework, http://aurora.hsnr.de/index-2.html.

  4. 4.

    Computational hearing in multisource environments (CHiME) challenge, http://spandh.dcs.shef.ac.uk/projects/chime/.

  5. 5.

    Reverberant voice enhancement and recognition benchmark (REVERB) challenge, https://reverb2014.dereverberation.com/.

References

  1. Xiong, W., Wu, L., Alleva, F., Droppo, J., Huang, X., Stolcke, A.: The Microsoft 2016 conversational speech recognition system. Microsoft Technical Report MSR-TR-2017-39. http://arxiv.org/pdf/1708.06073.pdf

  2. Dixon, P.R., Hori, C., Kashioka, H.: Development of the SprinTra WFST speech decoder. NICT Res. J., 15–20 (2012)

    Google Scholar 

  3. Hinton, G., Deng, L., Yu, D., Dahl, G., Mohamed, A., Jaitly, N., Senior, A., Vanhoucke, V., Nguyen, P., Sainath, T., Kingsbury, B.: Deep neural networks for acoustic modeling in speech recognition: the shared views of four research groups. IEEE Signal Process. Mag. 29, 82–97 (2012)

    Article  Google Scholar 

  4. Hochreiter, S., Schmidhuber, J.: Long short-term memory. Neural Comput. 9(8), 1735–1780 (1997)

    Article  Google Scholar 

  5. Cho, K., Merrienboer, B., Bahdanau, D., Bengio, Y.: On the properties of neural machine translation: encoder-decoder approaches. In: The 8th Workshop on Syntax, Semantics and Structure in Statistical Translation, SSST-8 (2014)

    Google Scholar 

  6. Sainath, T.N., Vinyals, O., Senior, A., Sak, H.: Convolutional, long short-term memory, fully connected deep neural networks. In: IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) (2015). https://doi.org/10.1109/icassp.2015.7178838

  7. Csáji, B.C.: Approximation with artificial neural networks. Faculty of Sciences; Eötvös Loránd University, Hungary (2001)

    Google Scholar 

  8. Krizhevsky, A., Sutskever, I., Hinton, G.: Imagenet classification with deep convolutional neural networks. In: Proceedings Advances in Neural Information Processing Systems (NIPS) (2012)

    Google Scholar 

  9. Srivastava, R.K., Greff, K., Schmidhuber, J.: Training very deep networks. In: Proceedings of NIPS (2015)

    Google Scholar 

  10. He, K., Zhang, X., Ren, S., Sun, J.: Deep residual learning for image recognition. In: Proceedings of the IEEE Conference on Computer Vision and Pattern Recognition (CVPR), pp. 770–778 (2016)

    Google Scholar 

  11. Graves, A., Fernandez, S., Gomez, F., Shmidhuber, J.: Connectionist temporal classification: labelling unsegmented sequence data with recurrent neural networks. In: Proceedings of the International Conference on Machine Learning (ICML) (2006)

    Google Scholar 

  12. Graves, A., Jaitly, N.: Towards end-to-end speech recognition with recurrent neural networks. In: Proceedings of the International Conference on Machine Learning (ICML) (2014)

    Google Scholar 

  13. Chorowski, J., Bahdanau, D., Cho, K., Bengio, Y.: End-to-end continuous speech recognition using attention-based recurrent NN: First results. arXiv preprint arXiv:14121602 (2014)

  14. Miao, Y., Gowayyed, M., Metze, F.: EESEN: end-to-end speech recognition using deep RNN models and WFST-based decoding. In: Proceedings of IEEE-ASRU (2015)

    Google Scholar 

  15. Kanda, N., Lu, X., Kawai, H.: Maximum a posteriori based decoding for CTC acoustic models. In: Proceedings of INTERSPEECH, pp. 1868–1872 (2016)

    Google Scholar 

  16. Boll, S.F.: Suppression of acoustic noise in speech using spectral subtraction. IEEE Trans. Audio Speech Signal Process. 27(2), 113–120 (1979)

    Article  Google Scholar 

  17. Ephraim, Y., Malah, D.: Speech enhancement using a minimum mean-square error short-time spectral amplitude estimator. IEEE Trans. Audio Speech Signal Process. 32, 1109–1121 (1984)

    Article  Google Scholar 

  18. Lu, X., Tsao, Y., Matsuda, S, Hori, C.: Speech enhancement based on deep denoising autoencoder. In: Proceedings of Interspeech ’13, pp. 436–440, August 2013

    Google Scholar 

  19. Yoshioka, T., Nakatani, T.: Generalization of multi-channel linear prediction methods for blind MIMO impulse response shortening. IEEE Trans. Audio Speech Lang. Process. 20(10), 2707–2720 (2012)

    Article  Google Scholar 

  20. Wölfel, M., McDonough, M.: Minimum variance distortionless response spectral estimation. IEEE Signal Process. Mag. 22(5) (2005)

    Google Scholar 

  21. Liao, H.: Speaker adaptation of context dependent deep neural networks. In: Proceedings of ICASSP ’13, pp. 7947–7951, May 2013

    Google Scholar 

  22. Seltzer, M., Yu, D., Wang, Y.: An investigation of deep neural networks for noise robust speech recognition. In: Proceedings of ICASSP ’13, pp. 7398–7402, May 2013

    Google Scholar 

  23. Wang, Z., Wang, D.: A joint training framework for robust automatic speech recognition. IEEE/ACM Transa. Audio Speech Lang. Process. (2016)

    Google Scholar 

  24. Li, L., Sim, K.C.: Improving robustness of deep neural networks via spectral masking for automatic speech recognition. In: Proceedings of ASRU ’13, pp. 279–284, December 2013

    Google Scholar 

Download references

Author information

Authors and Affiliations

Authors

Corresponding author

Correspondence to Xugang Lu .

Editor information

Editors and Affiliations

Rights and permissions

Reprints and permissions

Copyright information

© 2020 The Author(s), under exclusive license to Springer Nature Singapore Pte Ltd.

About this chapter

Check for updates. Verify currency and authenticity via CrossMark

Cite this chapter

Lu, X., Li, S., Fujimoto, M. (2020). Automatic Speech Recognition. In: Kidawara, Y., Sumita, E., Kawai, H. (eds) Speech-to-Speech Translation. SpringerBriefs in Computer Science. Springer, Singapore. https://doi.org/10.1007/978-981-15-0595-9_2

Download citation

  • DOI: https://doi.org/10.1007/978-981-15-0595-9_2

  • Published:

  • Publisher Name: Springer, Singapore

  • Print ISBN: 978-981-15-0594-2

  • Online ISBN: 978-981-15-0595-9

  • eBook Packages: Computer ScienceComputer Science (R0)

Publish with us

Policies and ethics