Discrete-time variance tracking with application to speech processing

Two new discrete-time algorithms are presented for tracking variance and reciprocal variance. The closed loop nature of the solutions to these problems makes this approach highly accurate and can be used recursively in real time. Since the Least-Mean Squares (LMS) method of parameter estimation requires an estimate of variance to compute the step size, this technique is well suited to applications such as speech processing and adaptive filtering

Automatic variance control and variance estimation loops

A closed loop servo approach is applied to the problem of controlling and estimating variance in nonstationary signals. The new circuit closely resembles but is not the same as, automatic gain control (AGC) which is common in radio and other circuits. The closed loop nature of the solution to this problem makes this approach highly accurate and can be used recursively in real time

Analysis of a non-minimum phase acoustic beamformer

The two input Griffiths-Jim acoustic beamformer is analysed in the frequency domain using a Wiener type formulation. Unlike previous solutions the approach here is to look at the problem of non-minimum phase acoustic transfer functions which are encountered in many real filtering problems. The polynomial transfer function approach gives an elegant way of obtaining the frequency response of the beamformer and gives new insight to the problem in general

On feed-through terms in the lms algorithm

The well known least mean squares (LMS) algorithm is studied as a control system. When applied in a noise canceller a block diagram approach is used to show that the step size has two upper limits. One is the conventional limit beyond which instability results. The second limit shows that if the step size is chosen to be too large then feed-through terms consisting of signal times noise will result in an additive term at the noise canceller output. This second limit is smaller than the first and will cause distortion at the noise canceller output

Kepstrum approach to real-time speech-enhancement methods using two microphones

The objective of this paper is to provide improved real-time noise canceling performance by using kepstrum analysis. The method is applied to typically existing two-microphone approaches using modified adaptive noise canceling and speech beamforming methods. It will be shown that the kepstrum approach gives an improved effect for optimally enhancing a speech signal in the primary input when it is applied to the front-end of a beamformer or speech directivity system. As a result, enhanced performance in the form of an improved noise reduction ratio with highly reduced adaptive filter size can be achieved. Experiments according to 20cm broadside microphone configuration are implemented in real-time in a real environment, which is a typical indoor office with a moderate reverberation condition

Automotive three-microphone voice activity detector and noise-canceller

This paper addresses issues in improving hands-free speech recognition performance in car environments. A three-microphone array has been used to form a beamformer with leastmean squares (LMS) to improve Signal to Noise Ratio (SNR). A three-microphone array has been paralleled to a Voice Activity Detection (VAD). The VAD uses time-delay estimation together with magnitude-squared coherence (MSC)

A kepstrum approach to filtering, smoothing and prediction

The kepstrum (or complex cepstrum) method is revisited and applied to the problem of spectral factorization where the spectrum is directly estimated from observations. The solution to this problem in turn leads to a new approach to optimal filtering, smoothing and prediction using the Wiener theory. Unlike previous approaches to adaptive and self-tuning filtering, the technique, when implemented, does not require a priori information on the type or order of the signal generating model. And unlike other approaches - with the exception of spectral subtraction - no state-space or polynomial model is necessary. In this first paper results are restricted to stationary signal and additive white noise

Design of an electrostatic end-plugged plasma-confinement device

A laboratory-scale experimental device having an outside diameter of 1.2 m has been designed to test the idea of electrostatic end plugging of an open-ended magnetic-field configuration. The configuration is a toroidal quadrupole having four very thin (less than 1-mm-thick) line cusps produced by four circular copper coils. Iron is used to concentrate the magnetic flux density to 2.0 T; without the use of iron, the power consumption, which is about 1 MW, would be about 25 times higher. The use of iron also produces a precisely known magnetic field and allows good access for diagnostics and pumping. Iron is also used for both the flux return path and the vacuum chamber. A hollow anode with an adjustable (nominally 1-mm-wide) gap is biased from 10 to 20 kV. Plasma densities of about 10/sup 13/ cm/sup -3/ and temperatures of about 1 keV might be produced by an electron beam and by electron cyclotron resonance heating. Higher-order multipoles (hexapoles and octopoles) also are described