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A very simple and extremely computationally efficient algorithm for blind separation of two speech sources from two mixtures is presented in this paper. The algorithm exploits the approximate W-disjoint orthogonality of speech signals and assumes specific sensors (microphones) setting that allows the sources to possess a feature we call cross high-low diversity. Two sources are said to be cross hi

A real-time digital signal processor (DSP) based implementation of a subband beamforming algorithm and its evaluation for dual microphone speech enhancement is presented. The algorithm, a calibrated constrained beamformer, is described theoretically and a real-time structure is proposed, including an efficient approach for multichannel data transformation. Measurements show that the battery driven

This paper presents a method for blind beamforming with application in realtime speech extraction in a non-stationary environment. The blind beamforming is carried out using an online kurtosis maximization approach where the optimization is based on Newton's method. The main novelty of the paper lies in the formulation of the subband kurtosis approximation, where a locally quadratic criterion is s

This paper presents a new microphone array method to enhance speech signals in a noisy reverberant environment. A time-delay estimation method is used for the speech source localization. The robustness of the localization method in high noise levels is provided by a subband Kurtosis-weighted structure. The estimated inter-sensor time-delays are directly used in an adaptive soft-constrained subband

-This paper presents a method for automatically detecting vehicle mounted auditory reverse alarms, or other similar warning signals, based on hidden Markov model and pattern matching techniques. The method is designed for embedded realtime platforms. The purpose of the method is to embed it with active hearing protection devices, aiding the user in detecting warning signals in low SNR environments

This paper focuses on realtime speech extraction using blind adaptive beamforming. The speech extraction is carried out using an approximation of the kurtosis measure in a subband domain. The introduced kurtosis approximation is an improvement of a recently proposed approximation technique where a locally quadratic criterion was solved at each iteration. The improvement introduced in this paper re

In many applications where speech separation and enhancement is of interest, e.g. conferencing systems, mobile phones and hearing aids, accurate speaker localization is important. This paper presents an alternative criteria for the well known steered response power with phase transform (SRP-PHAT) algorithm, in which the steered response relates to peaks in the fourth order cross cumulant, rather t

The performance of single-channel temporal noise reduction methods generally deteriorate in high noise environments, whereas spatial beamformers can maintain some level of speech enhancement. This paper presents a solution where a low complexity single-channel noise reduction method is integrated into the feedback control loop of an adaptive blind beamformer with the purpose of robust blind speech

A permutation problem arises in the case of locating multiple speech sources using several sensor arrays in the far field. The intersection of different direction of arrival (DOA) estimates between sensor arrays leads to a set of real source locations as well as a set of false intersections. This paper presents a novel method for pairing DOA estimates from different sensor arrays, resulting in the

Blind people may detect objects from the information in reflected sounds, echolocation. Detection as a function of the number of clicks compared to a continuous noise was tested by presenting clicks of 5 ms with rates from 1 to 64 clicks during a 500 ms period and a 500-ms continuous white noise. The sounds were recorded in an ordinary room through an artificial binaural head. The reflecting objec

In this chapter, speech enhancement and echo cancellation for hands-free mobile telephony are discussed. A number of microphone array methods have been tested and results from car measurements are given. Traditional methods such as linearly constrained beamforming and generalized sidelobe cancelers are discussed as well as array gain optimization methods. An in situ calibrated method which gives a

The convenience and the ease of use provided by hands-free operation of speech communication devices mean that speech enhancement schemes are becoming indispensable. In this chapter, two subband adaptive microphone array schemes are presented, which aim to provide good speech enhancement capability in poor signal to noise ratio situations. The basic commonality of the adaptive microphone array sch

The introduction of smart phones has redefined the usage of mobile phones in the communications world. Smart phones are now equipped with various sophisticated features such as Wi-Fi, Global Positioning System (GPS) navigation capability, high-resolution camera, touch screen, and broadband access which helps the mobile phone users to stay connected and updated. Many of these features are primarily

The invention relates to an adaptive method of extracting at least of des ired electro magnetic wave signals, sound wave signals (40, 42), and any oth er signals from a mixture of signals (40, 42, 44, 46) and suppressing noise and interfering signals to produce enhanced signals (50) corresponding to de sired (10) signals, and an apparatus (70) therefore. It relies on the concep t of at least one o

This study analyzes the intergenerational effects of late child bearing on offspring’s adult longevity in a population in Utah (USA) that does not display evidence of parity-specific birth control – a so-called natural fertility population. Studies have found that for women who experience late menopause and prolonged reproduction, aging is postponed and longevity is increased. This is believed to

Theoretical models and experiments suggest that the transport of suspended particles in microfluidics-based sorting devices can be modeled by a two-dimensional effective advection-diffusion process characterized by constant average velocity, $$\mathbf {W}$$W, and a typically anisotropic dispersion tensor, $$\mathbb {D}$$D, whose principal axes are slanted with respect to the direction of the effec

A survey of techniques to solve ATS + SA + Q = 0 is presented, and nine algorithms are coded and tested on a batch of examples. Which algorithm to be recommended depends mainly on the order of the system.

Inversion of a linear dynamical system is shown to be an operator equivalence to the well-known matrix lemma: (D + CLB)−1 = [D−1 − D−1C(L−1 + BD−1C)−1BD−1]

The linear filtering, prediction and smoothing problems as well as the linear quadratic control problems can very generally be formulated as operator equations using basic linear algebra.The equations are of Fredholm type II, and they are difficult to solve directly.It is shown how the operator can be factorized into two Volterra operators using a matrix Riccati equation. Recursive solution of the