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In this paper, a new approach to designing beamformers for voice control device is proposed. It is well-known that under a strong near-field noise with low signal-to-noise ratios (SNR), the performance of speech recognition is deteriorated significantly. However, designing the beamformer for enhancing speech recognition is a slow process and might not be adapted. easily to the changing noise envir
In blind source separation (BSS), multiple mixtures acquired by an array of sensors are processed in order to recover the initial multiple source signals. While a variety of Independent Component Analysis (ICA)-based techniques are being used, in this paper we used a newly proposed method: The Degenerate Unmixing and Estimation Technique (DUET). The method applies when sources are W-disjoint ortho
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
