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For pt.I see ibid., vol.50, p.2644-2655 (2002). This paper proposes a novel algorithm to estimate the time-varying spectral centroid of the narrowband Doppler volume backscattering signal. It is constructed in a semi-parametric way, that is, modeling parametrically the local narrowband evolutionary spectrum using an AR(2) and nonparametrically adapting its time-varying coefficients using the wavel
A new design method for broadband microphone arrays is presented. Using sequences of calibration signals, the method is able to design finite-impulse response (FIR) filters with specific performance. The method can control and adjust the speech distortion, noise suppression, and echo cancellation directly. It turns out that a significantly shorter filter length can be applied to achieve better ove
Effective utilization of the spatial domain enhances the capacity of a mobile radio network. A common technique is to use sector antennas, where the sectors are formed by weighting the outputs from the antenna elements. This results in spatial domain selectivity, which significantly improves the signal-to-(noise and interference) ratio in the received signals. However, the operation of the sector
The letter provides an analysis of optimal finite-impulse response subband beamforming for speech enhancement in multipath environments. A modification of the direct-path standard Wiener formulation is shown to give nearly optimal performance when it comes to signal-to-noise-plus-interference ratio, by including coherent multipath propagation in the criterion.
[Context & motivation] This research preview paper presents ongoing work on the metamodel of a free software requirements modeling tool called reqT that is developed in an educational context. The work aims to make an initial validation of a survey instrument that elicits views on the metamodel of the reqT tool, which aims to engage computer science students in Requirements Engineering (RE) th
This paper presents a new method for the design of oversampled uniform DFT-filter banks for the special application of subband adaptive beamforming with microphone arrays. Since array applications rely on the fact that different source positions give rise to different signal delays, a beamformer alters the phase information of the signals. This in turn leads to signal degradations when perfect rec
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
