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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
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
