Skip to main content

Building Real-Time Speech Recognition Without CMVN

  • Conference paper
  • First Online:
Book cover Speech and Computer (SPECOM 2018)

Part of the book series: Lecture Notes in Computer Science ((LNAI,volume 11096))

Included in the following conference series:

Abstract

Estimating cepstral mean and variance normalization (CMVN) in run-on and real-time settings poses several challenges. Using a moving average for variance and mean estimation requires a comparatively long history of data from a speaker which is not appropriate for short utterances or conversations. Using a pre-estimated global CMVN for speakers instead reduces the recognition performance due to potential mismatch between training and testing data. This paper investigates how to build a real-time run-on speech recognition system using acoustic features without applying CMVN. We propose a feature extraction architecture which can transform unnormalized log mel features to normalized bottleneck features without using historical data. We empirically show that mean and variance normalization is not critical for training neural networks on speech data. Using the proposed feature extraction, we achieved 4.1% word error rate reduction compared to global CMVN on the Skype conversations test set. We also reveal many cases when features without zero-mean can be learnt well by neural networks which stands in contrast to prior work.

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

References

  1. Alam, M.J., Ouellet, P., Kenny, P., O’Shaughnessy, D.: Comparative evaluation of feature normalization techniques for speaker verification. In: Travieso-González, C.M., Alonso-Hernández, J.B. (eds.) NOLISP 2011. LNCS (LNAI), vol. 7015, pp. 246–253. Springer, Heidelberg (2011). https://doi.org/10.1007/978-3-642-25020-0_32

    Chapter  Google Scholar 

  2. Cettolo, M., Niehues, J., Stüker, S., Bentivogli, L., Frederico, M.: Report on the 10th IWSLT evaluation campaign. In: The International Workshop on Spoken Language Translation (IWSLT) 2013 (2013)

    Google Scholar 

  3. Dehak, N., Kenny, P.J., Dehak, R., Dumouchel, P., Ouellet, P.: Front-end factor analysis for speaker verification. IEEE Trans. Audio Speech Lang. Process. 19(4), 788–798 (2011)

    Article  Google Scholar 

  4. Duda, R.O., Hart, P.E., Stork, D.G.: Pattern Classification. Wiley-Interscience (2000)

    Google Scholar 

  5. Federmann, C., Lewis, W.D.: Microsoft speech language translation (MSLT) corpus: the IWSLT 2016 release for English, French and German. In: The International Workshop on Spoken Language Translation (IWSLT) 2016 (2016)

    Google Scholar 

  6. Finke, M., Geutner, P., Hild, H., Kemp, T., Ries, K.R., Westphal, M.: The karlsruhe VERBMOBIL speech recognition engine. In: Proceedings of ICASSP (1997)

    Google Scholar 

  7. Furui, S.: Cepstral analysis technique for automatic speaker verification. IEEE Trans. Acoust. Speech Signal Process. 29(2), 254–272 (1981)

    Article  Google Scholar 

  8. Gehring, J., Miao, Y., Metze, F., Waibel, A.: Extracting deep bottleneck features using stacked auto-encoders. In: 2013 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pp. 3377–3381. IEEE (2013)

    Google Scholar 

  9. Graff, D.: The 1996 broadcast news speech and language-model corpus. In: Proceedings of the DARPA Workshop on Spoken Language Technology (1997)

    Google Scholar 

  10. Hannun, A., et al.: Deep speech: Scaling up end-to-end speech recognition. arXiv preprint arXiv:1412.5567 (2014)

  11. Jaitly, N., Hinton, G.E.: Vocal tract length perturbation (VTLP) improves speech recognition. In: Proceedings of ICML Workshop on Deep Learning for Audio, Speech and Language (2013)

    Google Scholar 

  12. Ko, T., Peddinti, V., Povey, D., Khudanpur, S.: Audio augmentation for speech recognition. In: INTERSPEECH, pp. 3586–3589 (2015)

    Google Scholar 

  13. LeCun, Y.A., Bottou, L., Orr, G.B., Müller, K.-R.: Efficient backprop. In: Montavon, G., Orr, G.B., Müller, K.-R. (eds.) Neural Networks: Tricks of the Trade. LNCS, vol. 7700, pp. 9–48. Springer, Heidelberg (2012). https://doi.org/10.1007/978-3-642-35289-8_3

    Chapter  Google Scholar 

  14. Peddinti, V., Povey, D., Khudanpur, S.: A time delay neural network architecture for efficient modeling of long temporal contexts. In: INTERSPEECH, pp. 3214–3218 (2015)

    Google Scholar 

  15. Povey, D., et al.: The Kaldi speech recognition toolkit. In: IEEE 2011 Workshop on Automatic Speech Recognition and Understanding. IEEE Signal Processing Society, December 2011

    Google Scholar 

  16. Prisyach, T., Mendelev, V., Ubskiy, D.: Data augmentation for training of noise robust acoustic models. In: Ignatov, D.I., et al. (eds.) AIST 2016. CCIS, vol. 661, pp. 17–25. Springer, Cham (2017). https://doi.org/10.1007/978-3-319-52920-2_2

    Chapter  Google Scholar 

  17. Pujol, P., Macho, D., Nadeu, C.: On real-time mean-and-variance normalization of speech recognition features. In: 2006 IEEE International Conference on Acoustics, Speech and Signal Processing, 2006. ICASSP 2006 Proceedings, vol. 1, p. I. IEEE (2006)

    Google Scholar 

  18. Rousseau, A., Deléglise, P., Estève, Y.: Enhancing the TED-LIUM corpus with selected data for language modeling and more TED talks. In: Proceedings of LREC (2014)

    Google Scholar 

  19. Sainath, T.N., Kingsbury, B., Mohamed, A.R., Ramabhadran, B.: Learning filter banks within a deep neural network framework. In: 2013 IEEE Workshop on Automatic Speech Recognition and Understanding (ASRU), pp. 297–302. IEEE (2013)

    Google Scholar 

  20. Sainath, T.N., Mohamed, A.R., Kingsbury, B., Ramabhadran, B.: Deep convolutional neural networks for LVCSR. In: 2013 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pp. 8614–8618. IEEE (2013)

    Google Scholar 

  21. Stüker, S., Kilgour, K., Kraft, F.: Quaero 2010 speech-to-text evaluation systems. In: Nagel, W., Kröner, D., Resch, M. (eds.) High Performance Computing in Science and Engineering ’11. Springer, Heidelberg (2012). https://doi.org/10.1007/978-3-642-23869-7_44

    Chapter  Google Scholar 

  22. Viikki, O., Laurila, K.: Cepstral domain segmental feature vector normalization for noise robust speech recognition. Speech Commun. 25(1), 133–147 (1998)

    Article  Google Scholar 

  23. Vincent, P., Larochelle, H., Bengio, Y., Manzagol, P.A.: Extracting and composing robust features with denoising autoencoders. In: The 25th International Conference on Machine Learning, pp. 1096–1103. ACM (2008)

    Google Scholar 

  24. Yu, D., Seltzer, M.L.: Improved bottleneck features using pretrained deep neural networks. In: Interspeech, vol. 237, p. 240 (2011)

    Google Scholar 

  25. Zeiler, M.D., et al.: On rectified linear units for speech processing. In: 2013 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pp. 3517–3521. IEEE (2013)

    Google Scholar 

  26. Zeyer, A., Schlüter, R., Ney, H.: Towards online-recognition with deep bidirectional LSTM acoustic models. In: Interspeech 2016, pp. 3424–3428 (2016)

    Google Scholar 

Download references

Author information

Authors and Affiliations

Authors

Corresponding author

Correspondence to Thai Son Nguyen .

Editor information

Editors and Affiliations

Rights and permissions

Reprints and permissions

Copyright information

© 2018 Springer Nature Switzerland AG

About this paper

Check for updates. Verify currency and authenticity via CrossMark

Cite this paper

Nguyen, T.S., Sperber, M., Stüker, S., Waibel, A. (2018). Building Real-Time Speech Recognition Without CMVN. In: Karpov, A., Jokisch, O., Potapova, R. (eds) Speech and Computer. SPECOM 2018. Lecture Notes in Computer Science(), vol 11096. Springer, Cham. https://doi.org/10.1007/978-3-319-99579-3_47

Download citation

  • DOI: https://doi.org/10.1007/978-3-319-99579-3_47

  • Published:

  • Publisher Name: Springer, Cham

  • Print ISBN: 978-3-319-99578-6

  • Online ISBN: 978-3-319-99579-3

  • eBook Packages: Computer ScienceComputer Science (R0)

Publish with us

Policies and ethics