During the last two decades a slow but steady change of external factors has set-up new conditions affecting the way in how software producing firms create and leverage innovations. Firms now need to look outside of their boundaries and start interacting with the open environment that encompasses them in order to stay innovative and keep a competitive advantage. To facilitate this shift Requiremen

Request PermissionsThis paper analyzes subband beamforming schemes mainly aimed at speech enhancement and acoustic echo suppression applications such as hands-free telephony for both mobile and office environments, Internet telephony and video conferencing. Analytical descriptions of both causal finite-length and noncausal infinite-length subband microphone array structures are given. More specifi

Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. However, subband processing causes signal degradations due to aliasing effects and amplitude distortions. This problem is unavoidable due to further filtering operations in subbands. In this letter, the problems of aliasing effect an

Subband adaptive filters have been proposed to speed up the convergence and to lower the computational complexity of time domain adaptive filters. However, subband processing causes signal degradations due to aliasing effects and amplitude distortions. This problem is unavoidable due to further filtering operations in subbands. In this paper, the problem of aliasing effect and amplitude distortion

A new robust microphone array method to enhance speech signals generated by a moving person in a noisy environment is presented. This blind approach is based on a two-stage scheme. First, a subband time-delay estimation method is used to localize the dominant speech source. The second stage involves speech enhancement, based on the acquired spatial information, by means of a soft-constrained subba

This paper presents a theoretical analysis of acertain criterion for complex-valued independent componentanalysis (ICA) with a focus on blind speech extraction (BSE) of aspatio–temporally nonstationary speech source. In the paper, theproposed criteria denoted KSICA is related to the well-known FastICAmethod with the Kurtosis contrast function. The proposedmethod is shown to share the important fix

This article presents an analysis of the sensitivity of geometrical sensor errors in acoustic source localization using the well-established SRP-PHAT method. The array in this analysis is a uniform linear array and the intended source is human speech in the far field. Two major results are presented: inner-sensor geometrical errors in the linear array produce smaller localization errors than corre

Julia Brinck (1854–1926) är den första svenska kvinnan med europeisk läkarutbildning i modern tid. Hon föddes i Helsingborg och tog sjukgymnastiskexamen i Stockholm vid 21 års ålder. Därefter arbetade hon på flera gymnastiska institut i Europa innan hon blev läkare 1886, efter fyra års studier i London. Julia Brinck var också en av de första svenska kvinnorna som ägnade sig åt naturvetenskaplig fo

This article presents a new method for localization of multiple concurrent speech sources that relies on simultaneous blind signal separation and direction of arrival (DOA) estimation, as well as a method to solve the intersection point selection problem that arises when locating multiple speech sources using multiple sensor arrays. The proposed method is based on a low complexity non-parametric b

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