Presents a microphone array system for use in handsfree mobile telephone equipment. The array is based on a fast and efficient “on-site” and “self-calibration” scheme. The performance in suppressing the interior car cabin noise and the far-end speech is approximately 17 dB, respectively, while maintaining the near-end speaker level. The near-end signal is almost undistorted. The performance of two

In the application of adaptive antenna arrays to wireless communications, a known pilot signal sequence may be used for estimating the array response at the beginning of each data frame. This pilot sequence is usually very short and conventional training methods which estimate the array response, based solely on this training sequence, may incur large estimation errors. In this paper, we propose a

In order to increase the capacity of a mobile radio network, it is desirable to exploit the spatial domain in an efficient way. A common technique is to use a sector antennas. Sectors can be formed by using antenna arrays which add spatial domain selectivity in order to improve the Signal-to-Noise and Interference Ratio (SNIR). In this paper an interference cancellation scheme is presented that co

Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. 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 rate than critically needed in

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

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

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

This study focuses on Seved, a segregated and socioeconomically "poor" neighborhood in the city of Malmö in Sweden. It has attracted wide media coverage, a possible consequence of which is its increased stigmatization. The wide disparity between perceived or imagined fear and the actual incidence of, or exposure to, violence attests to the important role of the media in shaping mental maps and pla