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This paper discusses signal processing methods for speech extraction in use with voice communication applications such as personal digital assistants (PDA), mobile telephone terminals and personal computers. The user is distant from the device and thus the speech signal entering the device may be subject to reverberation and may be disturbed by background noise. The proposed structure consists of
Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters for acoustic echo cancellation. Subband processing introduces transmission delays caused by the filter bank and signal degradations due to aliasing effects. One efficient way to reduce the aliasing effects is to allow a higher sample
This paper introduces a new constrained adaptive subband beamformer algorithm for speech enhancement in acoustic telecommunication systems. The solution relies on a pre-calculated source covariance matrix and recursive estimates of background noise- and handsfree signal covariance matrices. The constraint acts as an eye-opening in a vicinity of the near-field location of the source and degradation
This paper presents a method for the design of nonuniform DFT filter banks for subband beamforming. Filter banks designed with the method are evaluated in subband beamforming in a real-world microphone array application. Different source positions in array applications give rise to different signal delays, which means that adaptive beamformers in the subbands alter the phase information of the sub
This paper presents a new robust microphone array to enhance speech signal under the influence of noise and jammer(s). The proposed structure comprises of a soft constrained subband beamformer, a blocking system and a non-coherent processing technique. The soft constrained beamformer enhances the desired speech signal in a specified region by suppressing all side-lobes. This enhanced signal is the
The paper proposes a new calibrated adaptive frequency domain beamformer for speech enhancement. The beamformer is based on the principle of a soft constraint formed from calibration data, rather than precalculated from free-field assumptions. The benefit is that the real room acoustical properties are taken into account. The proposed algorithm continuously estimates the spatial information for ea
This paper presents a new robust microphone array processing technique to enhance speech signals under the influence of noise and jammer(s). The new structure comprises two soft constrained subband beamformers and a non-coherent processing technique. Essentially, the first beamformer enhances the desired speech signal in a specified constrained region. The residual interference in the beamformer's
This paper analyses optimal subband beamforming performance mainly aimed at speech enhancement and acoustic echo suppression for personal communication devices, personal computers and wireless cellular telephones. The focus is on theoretical limits of finite impulse response (FIR) beamformers for spatially spread sources in the array near-field. Performance of the Wiener solution is compared to th
The paper proposes an efficient microphone array processing technique for speech enhancement. The scheme incorporates the source and interference predefined regions into a subband generalized sidelobe canceller (GSC). The lower path consists of two space constrained beamformers to extract the interference information outside the source region. This information then serves as a reference for an imp
A novel adaptive beamformer employing recursively updated soft constraints for acoustic speech enhancement is proposed. The beamformer operates in a subband structure to allow a time-frequency operation for each channel. Consequently, the processing performed can be viewed as a combined weighted spatial,frequency and temporal filter. The major benefit of the new recursive soft constrained beamformer
A novel adaptive beamformer employing recursively updated soft constraints for acoustic speech enhancement is proposed. The beamformer operates in a subband structure to allow time-frequency operation for each channel. As such, the processing can be viewed as a combination of weighted spatial and temporal filters. The major benefit of this recursive soft constrained beamformer is that it allows th
This paper discusses speech enhancement in an enclosed environment such as communication in a motorcycle helmet. A new constrained subband adaptive beamformer is proposed, which uses the concept of an earlier proposed calibrated beamformer mainly developed for a hands-free in-car environment. The highly non-stationary nature of the disturbing sound field encountered in an motorcycle helmet and the
In this paper, a new spatial filter bank design method for speech enhancement beamforming applications is presented. The aim of this design is to construct a set of different filter banks that would include the constraint of signal passage at one position (and closing in other positions corresponding to known disturbing sources). By performing the directional opening towards the desired location i
We investigate the potential of different pre-fetchingand/or caching strategies for different user behaviour withrespect to surfing or browsing in a catch-up-TV network. To thisend we identify accounts and channels associated with strong orweak surfing or browsing respectively and study the distributionsof hold times for the different types of behaviour. Finally wepresent results from a request prWe investigate the potential of different pre-fetching and/or caching strategies for different user behaviour with respect to surfing or browsing in a catch-up-TV network. To this end we identify accounts and channels associated with strong or weak surfing or browsing respectively and study the distributions of hold times for the different types of behaviour. Finally we present results from a requ
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
