This paper introduces a new subband adaptive space constrained beamforming structure for use in hands-free speech enhancement applications. The scheme incorporates a space constrained source model and voice activity information through the integration of a voice activity detector (VAD). The VAD information is used to estimate noise covariance information during non-speech periods and to optimally

This paper presents a new constrained subband beamforming algorithm to enhance speech signals generated by a moving source in a noisy environment. The beamformer is based on the principle of a soft constraint defined for a specified region corresponding to a known source location. The soft constraint secures the spatial-temporal passage of the desired source signal in the adaptive update of the be

A modern hearing aid should be aesthetically appealing as well as offer sufficient and adequate signal amplification. Due to the small physical size of these devices, acoustical feedback (howling) is a major problem. Apart from the annoyance and potential hearing damaging effects that howling implies, it also reduces the supplied maximum Real Ear Aided Gain (REAG). This paper proposes a novel method

This paper presents a new approach for multiple speaker DOA estimation using an array of microphones. The method relies on the fact that multiple independent speakers have a small overlap in the time-frequency domain, i.e. the individual signals are almost W-disjoint orthogonal. By introducing a time-frequency mask and by continuously tracking the set of time-frequency points corresponding to each

This paper presents an idea to extend a certain class of single channel speech enhancement algorithms to include the spatial domain. The resulting blind beamformer does not rely on a-priori knowledge of source and sensor positions and it enhances one or several speech sources based only on received data. The underlying principle in this approach is the fact that speech signals are short time stati

This paper presents a real time implementation of a blind beamformer for subband speech enhancement. The beamformer adaptivelymaximizes the statistical kurtosis measure of the beamformer’soutput signal. Speech carries high kurtosis and noiseoften exhibit lower kurtosis. Hence, maximization of the outputsignal’s kurtosis enhances speech, in general. The implementationis carried out on a novel frame

A very simple and extremely computationally efficient algorithm for blind separation of two speech sources from two mixtures is presented in this paper. The algorithm exploits the approximate W-disjoint orthogonality of speech signals and assumes specific sensors (microphones) setting that allows the sources to possess a feature we call cross high-low diversity. Two sources are said to be cross hi

A real-time digital signal processor (DSP) based implementation of a subband beamforming algorithm and its evaluation for dual microphone speech enhancement is presented. The algorithm, a calibrated constrained beamformer, is described theoretically and a real-time structure is proposed, including an efficient approach for multichannel data transformation. Measurements show that the battery driven

This paper presents a method for blind beamforming with application in realtime speech extraction in a non-stationary environment. The blind beamforming is carried out using an online kurtosis maximization approach where the optimization is based on Newton's method. The main novelty of the paper lies in the formulation of the subband kurtosis approximation, where a locally quadratic criterion is s

This paper presents a new microphone array method to enhance speech signals in a noisy reverberant environment. A time-delay estimation method is used for the speech source localization. The robustness of the localization method in high noise levels is provided by a subband Kurtosis-weighted structure. The estimated inter-sensor time-delays are directly used in an adaptive soft-constrained subband

-This paper presents a method for automatically detecting vehicle mounted auditory reverse alarms, or other similar warning signals, based on hidden Markov model and pattern matching techniques. The method is designed for embedded realtime platforms. The purpose of the method is to embed it with active hearing protection devices, aiding the user in detecting warning signals in low SNR environments

This paper focuses on realtime speech extraction using blind adaptive beamforming. The speech extraction is carried out using an approximation of the kurtosis measure in a subband domain. The introduced kurtosis approximation is an improvement of a recently proposed approximation technique where a locally quadratic criterion was solved at each iteration. The improvement introduced in this paper re

In many applications where speech separation and enhancement is of interest, e.g. conferencing systems, mobile phones and hearing aids, accurate speaker localization is important. This paper presents an alternative criteria for the well known steered response power with phase transform (SRP-PHAT) algorithm, in which the steered response relates to peaks in the fourth order cross cumulant, rather t

The performance of single-channel temporal noise reduction methods generally deteriorate in high noise environments, whereas spatial beamformers can maintain some level of speech enhancement. This paper presents a solution where a low complexity single-channel noise reduction method is integrated into the feedback control loop of an adaptive blind beamformer with the purpose of robust blind speech

A permutation problem arises in the case of locating multiple speech sources using several sensor arrays in the far field. The intersection of different direction of arrival (DOA) estimates between sensor arrays leads to a set of real source locations as well as a set of false intersections. This paper presents a novel method for pairing DOA estimates from different sensor arrays, resulting in the

Blind people may detect objects from the information in reflected sounds, echolocation. Detection as a function of the number of clicks compared to a continuous noise was tested by presenting clicks of 5 ms with rates from 1 to 64 clicks during a 500 ms period and a 500-ms continuous white noise. The sounds were recorded in an ordinary room through an artificial binaural head. The reflecting objec

In this chapter, speech enhancement and echo cancellation for hands-free mobile telephony are discussed. A number of microphone array methods have been tested and results from car measurements are given. Traditional methods such as linearly constrained beamforming and generalized sidelobe cancelers are discussed as well as array gain optimization methods. An in situ calibrated method which gives a

The convenience and the ease of use provided by hands-free operation of speech communication devices mean that speech enhancement schemes are becoming indispensable. In this chapter, two subband adaptive microphone array schemes are presented, which aim to provide good speech enhancement capability in poor signal to noise ratio situations. The basic commonality of the adaptive microphone array sch