Improving robustness of adaptive beamforming for hearing devices

  • Alastair Moore Department of Electrical and Electronic Engineering, Imperial College, London, UK
  • Patrick Naylor Department of Electrical and Electronic Engineering, Imperial College, London, UK
  • Mike Brookes Department of Electrical and Electronic Engineering, Imperial College, London, UK

Abstract

Fixed beamforming for hearing aids is suboptimal due to mismatches in real-world situations between the assumed and encountered sound fields. Adaptive beamforming potentially provides better performance but may degrade it if the characteristics of the signal required by the design procedure are inaccurately estimated. This paper proposes a straightforward but sufficiently rich model for the sound field that can be used to increase the robustness of adaptive beamformer design. A method for estimating the model parameters is also presented. In reverberant acoustic conditions, the proposed method improves performance by > 1 dB even at −16 dB SNR, the lowest signal to noise ratio (SNR) tested. Furthermore, it is shown to be robust in a variety of acoustic conditions which do not conform to the sound field model, and to inaccurate steering of the array.

References

Avargel, Y., and Cohen, I. (2007), “On multiplicative transfer function approximation in the short-time Fourier transform domain,” IEEE Signal Process. Lett., 14(5), 337-340.

Bitzer, J., and Simmer, K.U. (2001), “Superdirective microphone arrays,” in Microphone Arrays: Signal Processing Techniques and Applications, M. S. Brandstein and D. B. Ward, Eds. Berlin, Germany: Springer-Verlag, 2001, 19-38.

Braun, S., and Habets, E.A.P (2015), “A multichannel diffuse power estimator for dereverberation in the presence of multiple sources,” EURASIP J. Audio Speech Music Process., vol. 2015, no. 1, p. 34.

Braun, S., Kuklasi´nski, A., Schwartz, O., Thiergart, O., Habets, E.A.P., Gannot, S., Doclo, S., and Jensen, J. (2018), “Evaluation and comparison of late reverberation power spectral density estimators,” IEEE/ACM Trans. Audio Speech Lang. Process., 26(6), 1056-1071.

Brookes, D.M. (1997), “VOICEBOX: A speech processing toolbox for MATLAB,” 1997–2016. [Online]. Available: http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/voicebox.html

Capon, J. (1969), “High resolution frequency-wavenumber spectrum analysis,” Proc. IEEE, 57, 1408-1418.

Chakrabarty, S., and Habets, E.A.P. (2018), “A Bayesian approach to informed spatial filtering with robustness against DOA estimation errors,” IEEE/ACM Trans. Audio Speech Lang. Process., 26(1), 145-160.

Cox, H., Zeskind, R.M., and Owen, M.M. (1987), “Robust adaptive beamforming,”

IEEE Trans. Acoust. Speech Signal Process., 35(10), 1365-1376.

Ehrenberg, L., Gannot, S., Leshem, A., and Zehavi, E. (2010), “Sensitivity analysis of MVDR and MPDR beamformers,” Proc. IEEE Conv. Electrical and Electronics Engineers, 416-420.

Gannot, S., Burshtein, D., and Weinstein, E. (2001), “Signal enhancement using

beamforming and nonstationarity with applications to speech,” IEEE Trans. Signal Process., 49(8), 1614-1626.

ITU-T (1993), “Objective measurement of active speech level,” Intl. Telecommunications Union (ITU-T), Recommendation P.56, Mar. 1993.

ITU-T (2003), “Perceptual evaluation of speech quality (PESQ), an objective method for end-to-end speech quality assessment of narrowband telephone networks and speech codecs,” Intl. Telecommunications Union (ITU-T), Recommendation P.862, Nov. 2003.

Jarrett, D.P., Habets, E.A.P., and Naylor, P.A. (2017), Theory and Applications of Spherical Microphone Array Processing, ser. Springer Topics in Signal Processing. Springer International Publishing, 2017.

Klasen, T.J., Bogaert, T.V. den, Moonen, M., and Wouters, J. (2007), “Binaural noise reduction algorithms for hearing aids that preserve interaural time delay cues,” IEEE Trans. Signal Process., 55(4), 1579-1585.

Li, J., Stoica, P., and Wang, Z. (2003), “On robust Capon beamforming and diagonal loading,” IEEE Trans. Signal Process., 51(7), 1702-1715.

Löllmann, H.W., Moore, A.H., Naylor, P.A., Rafaely, B., Horaud, R., Mazel, A., and Kellermann, W. (2017), “Microphone array signal processing for robot audition,” Proc. HSCMA, 51–55.

Markovich, S., Gannot, S., and Cohen, I. (2009), “Multichannel eigenspace beamforming in a reverberant noisy environment with multiple interfering speech signals,” IEEE Trans. Audio, Speech, Lang. Process., 17(6), 1071-1086.

Markovich-Golan, S. and Gannot, S. (2015), “Performance analysis of the covariance subtraction method for relative transfer function estimation and comparison to the covariance whitening method,” Proc. ICASSP, 544–548.

Moore, A.H., Lightburn, L., Xue, W., Naylor, P.A., and Brookes, M. (2018), “Binaural mask-informed speech enhancement for hearing aids with head tracking,” Proc. IWAENC.

Moore, A.H., Xue, W., Naylor, P.A., and Brookes, M. (2019), “Noise covariance matrix estimation for rotating microphone arrays,” IEEE/ACM Trans. Audio Speech Lang. Process., 27(3), 519-530.

Moore, A.H., de Haan, J.M., Pedersen, M.S., Naylor, P.A., Brookes, M., and Jensen, J. (2019), “Personalized signal-independent beamforming for binaural hearing aids,” J. Acoust. Soc. Am., 145, 971–2981.

Rafaely, B. (2015), Fundamentals of Spherical Array Processing, ser. Springer Topics in Signal Processing. Berlin Heidelberg: Springer-Verlag, 2015.

Schwartz, O., Gannot, S., and Habets, E.A.P (2016), “Joint estimation of late reverberant and speech power spectral densities in noisy environments using frobenius norm,” Proc. EUSIPCO, 1123–1127.

Schwarz, A., and Kellermann, W. (2015), “Coherent-to-diffuse power ratio estimation for dereverberation,” IEEE/ACM Trans. Audio Speech Lang. Process., 23(6), 1006–1018.

Taal, C.H., Hendriks, R.C., Heusdens, R., and Jensen, J. (2011), “An algorithm for intelligibility prediction of time-frequency weighted noisy speech,” IEEE Trans. Audio Speech Lang. Process., 19(7), 2125-2136.

Tamai, Y., Kagami, S., Amemiya, Y., Sasaki, Y., Mizoguchi, H., and Takano, T. (2004), “Circular microphone array for robot’s audition,” Proc. IEEE Sensors, 565–570.

Thiergart, O., and Habets, E.A.P. (2013), “An informed LCMV filter based on multiple instantaneous direction-of-arrival estimates,” Proc. ICASSP, 659–663.

van Trees, H.L. (2002), Optimum Array Processing, ser. Detection, Estimation and Modulation Theory. John Wiley & Sons, Inc., 2002.

Yilmaz, O., and Rickard, S. (2004), “Blind separation of speech mixtures via time- frequency masking,” IEEE Trans. Signal Process., 52(7), 1830-1847.

Published
2020-05-01
How to Cite
Moore, A., Naylor, P., & Brookes, M. (2020). Improving robustness of adaptive beamforming for hearing devices. Proceedings of the International Symposium on Auditory and Audiological Research, 7, 305-316. Retrieved from https://proceedings.isaar.eu/index.php/isaarproc/article/view/2019-35
Section
2019/4. Novel directions in hearing-instrument technology