In this paper, a binaural beamforming algorithm for hearing aid applications is introduced. The beamforming algorithm is designed to be robust to some error in the estimate of the target speaker direction. The algorithm has two main components: a robust target linearly constrained minimum variance (TLCMV) algorithm based on imposing two constraints around the estimated direction of the target signal, and a post-processor to help with the preservation of binaural cues. The robust TLCMV provides a good level of noise reduction and low level of target distortion under realistic conditions. The post-processor enhances the beamformer abilities to preserve the binaural cues for both diffuse-like background noise and directional interferers (competing speakers), while keeping a good level of noise reduction. The introduced algorithm does not require knowledge or estimation of the directional interferers' directions nor the second-order statistics of noise-only components. The introduced algorithm requires an estimate of the target speaker direction, but it is designed to be robust to some deviation from the estimated direction. Compared with recently proposed state-of-the-art methods, comprehensive evaluations are performed under complex realistic acoustic scenarios generated in both anechoic and mildly reverberant environments, considering a mismatch between estimated and true sources direction of arrival. Mismatch between the anechoic propagation models used for the design of the beamformers and the mildly reverberant propagation models used to generate the simulated directional signals is also considered. The results illustrate the robustness of the proposed algorithm to such mismatches.

The use of a nearby external microphone for addressing front-back ambiguity in single-microphone hearing aid devices is investigated. Strategic placement of the external microphone is able to provide benefits from the body shielding back-directional noise and therefore information for discriminating between the frontal and back hemispheres. The scattering effects of the body are first analyzed to yield a placement strategy of the external microphone for maximizing the shielding effect of the body against back-directional noise sources while also optimizing for speech intelligibility. Assuming optimal placement, a frontal target source presence probability (FTSPP) estimator is derived. Using the FTSPP estimator, a more comprehensive noise estimator is proposed, which considers both stationary and nonstationary interferers. The performance of the proposed noise estimator is evaluated in its application for postfiltering the output of a binaural beamformer. The effect of postfiltering using the proposed noise estimator is to provide further reduction of directional noise from the lateral and back direction while preserving the frontal target signal. The resulting enhancement with the proposed noise estimator provides a better signal-to-noise ratio and improved objective speech quality and intelligibility of a frontal target speaker compared to the state of the art.

This paper presents a strategy for using an external microphone for enhancing noisy speech in single-microphone completely-in-canal (CIC) hearing aids. The external microphone is placed such that it benefits from the body shielding noise from the back hemisphere. The presented algorithm first enhances the external microphone signal without assuming an exact known location of the external microphone. The proposed algorithm then automatically incorporates the enhanced external microphone signal for post-processing enhancement of a conventional dual-channel binaural beamformer whenever the external microphone has a significant SNR advantage. The overall enhancement scheme avoids error-prone estimations of target voice activity detection and relative transfer functions between the microphones. Unlike single-channel post-processing filters which are limited to reducing stationary or diffuse noise, the proposed postprocessing filter is able to reduce highly non-stationary directional noise from the back hemisphere. The resulting system provides enhancement even for noise arriving from the backward direction, which conventionally is difficult for CIC hearing aids where there exists a front-back ambiguity.

The current generation of digital hearing aids allows the implementation of advanced noise reduction schemes. However, most current noise reduction algorithms are monaural and are therefore intended for only bilateral hearing aids. Recently, binaural in contrast to monaural noise reduction schemes have been proposed, targeting future high-end binaural hearing aids. Those new types of hearing aids would allow the sharing of information/signals received from both left and right hearing aid microphones (via a wireless link) to generate an output for the left and right ear. This paper presents a novel noise power spectral density estimator for binaural hearing aids operating in a diffuse noise field environment, by taking advantage of the left and right reference signals that will be accessible, as opposed to the single reference signal currently available in bilateral hearing aids. In contrast with some previously published noise estimation methods for hearing aids or speech enhancement, the proposed noise estimator does not assume stationary noise, it can work for colored noise in a diffuse noise field, it does not require a voice activity detection, the noise power spectrum can be estimated during speech activity or not, it does not experience noise tracking latency and most importantly, it is not essential for the target speaker to be in front of the binaural hearing aid user to estimate the noise power spectrum, i.e., the direction of arrival of the source speech signal can be arbitrary. Finally, the proposed noise estimator can be combined with any hearing aid noise reduction technique, where the accuracy of the noise estimation can be critical to achieve a satisfactory de-noising performance.

This letter uses the twisted-pair common-mode signal as the input of an adaptive canceller to remove some wideband crosstalk in a digital subscriber line (xDSL) differential signal. Simulations using simple crosstalk and line balance models show the potential benefits of the canceller to improve the signal-to-noise ratio of an xDSL channel.

Crosstalk originating from multiple high-speed data services in the same telephone bundle is a limiting factor for the maximum bit rate, the loop length and the number of data services that a bundle can support. Due to the nature of twisted-pairs, external interferences (including crosstalk) mostly couple to the twisted-pair line in common mode, and then leaks to differential mode due to line imperfect balance. The result is a degradation of the received differential signal quality. This paper uses the common mode signal as a reference to an adaptive wideband crosstalk canceller, as an attempt to remove the effect of crosstalk on the differential signal. Simulation results show the potential benefits of using this technique to reduce the crosstalk levels.