Skip to main content

Model-Driven Speech Enhancement for Multisource Reverberant Environment (Signal Separation Evaluation Campaign (SiSEC) 2011)

  • Conference paper
Latent Variable Analysis and Signal Separation (LVA/ICA 2012)

Part of the book series: Lecture Notes in Computer Science ((LNTCS,volume 7191))

Abstract

We present a low complexity speech enhancement technique for real-life multi-source environments. Assuming that the speaker identity is known a priori, we present the idea of incorporating speaker model to enhance a target signal corrupted in non-stationary noise in a reverberant scenario. Based on experiments, this helps to improve the limited performance of noise-tracking based speech enhancement methods under unpredictable and non-stationary noise scenarios. Using pre-trained speaker model captures a constrained subspace for target speech and is capable to provide enhanced speech estimate by rejecting the non-stationary noise sources. Experimental results on Signal Separation Evaluation Campaign (SiSEC) showed that the proposed approach is successful in canceling the interference signal in the noisy input and providing an enhanced output signal.

The work of Pejman Mowlaee was funded by the European Commission within the Marie Curie ITN AUDIS, grant PITNGA-2008-214699. The work of Rahim Saeidi was funded by the European Community’s Seventh Framework Programme (FP7 2007-2013) under grant agreement no. 238803.

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 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

Preview

Unable to display preview. Download preview PDF.

Unable to display preview. Download preview PDF.

References

  1. Ephraim, Y., Malah, D.: Speech enhancement using a minimum-mean square error short-time spectral amplitude estimator. IEEE Trans. Audio, Speech, and Language Process. 32(6), 1109–1121 (1984)

    Article  Google Scholar 

  2. Ephraim, Y., Malah, D.: Speech enhancement using a minimum mean-square error log-spectral amplitude estimator. IEEE Transactions on Acoustics, Speech and Signal Processing 33(2), 443–445 (1985)

    Article  Google Scholar 

  3. Hendriks, R.C., Heusdens, R., Jensen, J.: MMSE based noise PSD tracking with low complexity. In: Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 4266–4269 (2010)

    Google Scholar 

  4. Christensen, H., Barker, J., Ma, N., Green, P.: The CHiME corpus: a resource and a challenge for computational hearing in multisource environments. In: Proc. Interspeech, pp. 1918–1921 (2010)

    Google Scholar 

  5. Mowlaee, P.: New Stategies for Single-channel Speech Separation, Ph.D. thesis, Institut for Elektroniske Systemer, Aalborg Universitet (2010)

    Google Scholar 

  6. Mowlaee, P., Christensen, M., Jensen, S.: New results on single-channel speech separation using sinusoidal modeling. IEEE Trans. Audio, Speech, and Language Process. 19(5), 1265–1277 (2011)

    Article  Google Scholar 

  7. Rangachari, S., Loizou, P.C.: A noise-estimation algorithm for highly non-stationary environments. Speech Communication 48(2), 220–231 (2006)

    Article  Google Scholar 

  8. Cohen, I., Berdugo, B.: Speech enhancement for non-stationary noise environments. Signal Processing 81(11), 2403–2418 (2001)

    Article  MATH  Google Scholar 

  9. Wang, D.: On ideal binary mask as the computational goal of auditory scene analysis. In: Speech Separation by Humans and Machines, pp. 181–197. Kluwer (2005)

    Google Scholar 

  10. Erkelens, J., Hendriks, R., Heusdens, R., Jensen, J.: Minimum mean-square error estimation of discrete Fourier coefficients with generalized gamma priors. IEEE Transactions on Audio, Speech, and Language Processing 15(6), 1741–1752 (2007)

    Article  Google Scholar 

  11. Vincent, E., Gribonval, R., Fevotte, C.: Performance measurement in blind audio source separation. IEEE Transactions on Audio, Speech, and Language Processing 14(4), 1462–1469 (2006)

    Article  Google Scholar 

  12. The third community-based Signal Separation Evaluation Campaign (SiSEC 2011), http://sisec.wiki.irisa.fr/tiki-index.php

  13. Emiya, V., Vincent, E., Harlander, N., Hohmann, V.: Subjective and objective quality assessment of audio source separation. IEEE Transactions on Audio, Speech, and Language Processing (99), 1 (2011)

    Google Scholar 

Download references

Author information

Authors and Affiliations

Authors

Editor information

Fabian Theis Andrzej Cichocki Arie Yeredor Michael Zibulevsky

Rights and permissions

Reprints and permissions

Copyright information

© 2012 Springer-Verlag Berlin Heidelberg

About this paper

Cite this paper

Mowlaee, P., Saeidi, R., Martin, R. (2012). Model-Driven Speech Enhancement for Multisource Reverberant Environment (Signal Separation Evaluation Campaign (SiSEC) 2011). In: Theis, F., Cichocki, A., Yeredor, A., Zibulevsky, M. (eds) Latent Variable Analysis and Signal Separation. LVA/ICA 2012. Lecture Notes in Computer Science, vol 7191. Springer, Berlin, Heidelberg. https://doi.org/10.1007/978-3-642-28551-6_56

Download citation

  • DOI: https://doi.org/10.1007/978-3-642-28551-6_56

  • Publisher Name: Springer, Berlin, Heidelberg

  • Print ISBN: 978-3-642-28550-9

  • Online ISBN: 978-3-642-28551-6

  • eBook Packages: Computer ScienceComputer Science (R0)

Publish with us

Policies and ethics