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The "Knowledger" - a Case Study of a Small Knowledge Focused Consulting Firm
Complex-valued independent component analysis for online blind speech extraction
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
Statistical Analysis of a Local Quadratic Criterion for Blind Speech Extraction
Calibration Errors of Uniform Linear Sensor Arrays for DOA Estimation: an Analysis with SRP-PHAT
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
Första svenskan med en riktig läkarlicens. : Julia Brinck var sjukgymnast, cellbiolog och brevvän med Florence nightingale.
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
Source Localization for Multiple Speech Sources Using Low Complexity Non-Parametric Source Separation and Clustering
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
A hybrid design of beamformers for voice control devices
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
Blind Source Separation Using Time-Frequency Masking
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
Slopa produktivitetsavdraget i högre utbildning
The measurement of selection when detection is imperfect: How good are naïve methods?
Governance of higher education – the role of proximity in teaching quality
Adaptive Microphone Array System for Speech Enhancement
Neural Network Based Adaptive Microphone Array System for Speech Enhancement
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
Skolkulturer - i skolan, i ämneslärarutbildningen och i blivande lärare
Performance Improvement of Multiple Array 3d Sonic Digitizer System via Calibration
Performance Analysis and Bias Correction in Synchronized Acoustic 3D Measuring System
Optimal Beamforming for Voice Input to Personal Communication Devices
A New Pilot-Signal based Space-Time Adaptive Algorithm
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