Independent Component Analysis.

Binary time-frequency masks are powerful tools for the separation of sources from a single mixture. Perfect demixing via binary time-frequency masks is possible provided the time-frequency representations of the sources do not overlap: a condition we call W-disjoint orthogonality. We introduce here the concept of approximate W-disjoint orthogonality and present experimental results demonstrating the level of approximate W-disjoint orthogonality of speech in mixtures of various orders. The results demonstrate that there exist ideal binary time-frequency masks that can separate several speech signals from one mixture. While determining these masks blindly from just one mixture is an open problem, we show that we can approximate the ideal masks in the case where two anechoic mixtures are provided. Motivated by the maximum likelihood mixing parameter estimators, we define a power weighted two-dimensional (2-D) histogram constructed from the ratio of the time-frequency representations of the mixtures that is shown to have one peak for each source with peak location corresponding to the relative attenuation and delay mixing parameters. The histogram is used to create time-frequency masks that partition one of the mixtures into the original sources. Experimental results on speech mixtures verify the technique. Example demixing results can be found online at http://alum.mit.edu/www/rickard/bss.html.

Blind separation of speech mixtures via time-frequency masking

Active noise control exploits the long wavelengths associated with low frequency sound. It works on the principle of destructive interference between the sound fields generated by the original primary sound source and that due to other secondary sources, acoustic outputs of which can be controlled. The acoustic objectives of different active noise control systems and the electrical control methodologies that are used to achieve these objectives are examined. The importance of having a clear understanding of the principles behind both the acoustics and the electrical control in order to appreciate the advantages and limitations of active noise control is emphasized. A brief discussion of the physical basis of active sound control that concentrates on three-dimensional sound fields is presented. >

Active noise control

On typical echo paths, the proportionate normalized least-mean-squares (PNLMS) adaptation algorithm converges significantly faster than the normalized least-mean-squares (NLMS) algorithm generally used in echo cancelers to date. In PNLMS adaptation, the adaptation gain at each tap position varies from position to position and is roughly proportional at each tap position to the absolute value of the current tap weight estimate. The total adaptation gain being distributed over the taps is carefully monitored and controlled so as to hold the adaptation quality (misadjustment noise) constant. PNLMS adaptation only entails a modest increase in computational complexity.

Proportionate normalized least-mean-squares adaptation in echo cancelers

There has been a great deal of material in the area of discrete-time transforms that has been published in recent years. This book does an excellent job of presenting important aspects of such material in a clear manner. The book has 11 chapters and a very useful appendix. Seven of these chapters are essentially devoted to the Fourier series/transform, discrete Fourier transform, fast Fourier transform (FFT), and applications of the FFT in the area of spectral estimation. Chapters 8 through 10 deal with many other discrete-time transforms and algorithms to compute them. Of these transforms, the KarhunenLoeve, the discrete cosine, and the Walsh-Hadamard transform are perhaps the most well-known. A lucid discussion of number theoretic transforms i5 presented in Chapter 11. This reviewer feels that the authors have done a fine job of compiling the pertinent material and presenting it in a concise and clear manner. There are a number of problems at the end of each chapter, an appreciable number of which are challenging. The authors have included a comprehensive set of references at the end of the book. In brief, this book is a valuable contribution to the general areas of signal processing and communications. It can be used for a graduate level course in perhaps two ways. One would be to cover the first seven chapters in great detail. The other would be to cover the whole book by focussing on different topics in a selective manner. This book  by  Elliott  and Rao  is extremely  useful to researchers/engineers who are working  in the areas of signal processing and communications. It i s also an excellent reference book, and hence a valuable addition to one’s library

http://ieeexplore.ieee.org/iel5/5/31347/01457444.pdf

Multirate digital signal processing

PURPOSE: To make a final estimation error constant regardless of the degree of a projection. CONSTITUTION: A transfer function of the unknown system 11 is estimated from a projection algorithm by performing a convolution operation on an input x(k) with a transfer function HΛ(k), obtaining a difference e(k) between the result yΛ(k) and an output of an unknown system 11, obtaining G(k) from the solution of simultaneous linear equations with p-unknowns G(k)=Cp(k) E(k) consisting of the vector E(k) and a covariance matrix Cp(k) of x(k), obtaining a corrected vector δHΛ(k) by computing G(k) Xp(k), adding the product of this δHΛ(k) and step size μ(k) to the estimation transfer function HΛ(k) and repeating an updating of the estimation transfer function. μ(k) is controlled at the time of updating, by inputting σx , a variance of x(k), F(k) , a sum of linear square predictive errors of x(k) selecting one of the diagonal terms of Cp(k), and a value μ1 deciding the final estimation error, constants c and e, and obtaining μ(k) having a positive correlation between μ(k) and F(k).

Adaptive transfer function estimation method and estimation device using the same

http://www.asp.c.dendai.ac.jp/thesis/aclt_0909_suzuki.pdf

Improving the robustness of multiple signal classification (MUSIC) method to reflected sounds by sub-band peak-hold processing

PROBLEM TO BE SOLVED: To provide a speaker form each point with a sound image in a voice conference. SOLUTION: Voice signals Sa-Sd from spots A-D are transmitted through communication liens to a center 24, and converted into first signals Sa1 -Sd1 and second signals Sa2 -Sd2 (Sa1 >Sb1 >Sc1 >Sd1 , Sa2 <Sb2 <Sc2 <Sd2 ) in two sound pressure levels by sound pressure modulating parts 12a-12d, and the mutual first and second signals except the voices to other spot from own spot are added by signal adding parts 13a-13d, for example, the first signals Sb1 , Sc1 , and Sd1 are added, and the second signals Sb2 , Sc2 and Sd2 are added by the adding part 13a in the center 24. The added first and second signals of the adding parts 13a-13d are transmitted to the spots A-D, and stereo reproduced by left speakers 27a1-27d1 and right speakers 27a2-27d2 at each spot, and a sound image is localized at a position different at each uttered voice spot.

Multi-point voice communication equipment

In many telecommunication systems, hands‐free operation is becoming popular. For this to be comfortable and easy, high echo cancellation performance and a simple setup are required. This paper presents a new hands‐free teleconferencing unit that consists of a main loudspeaker, an auxiliary loudspeaker, a microphone, and an echo canceller DSP board. The auxiliary loudspeaker is used to reduce the acoustic coupling between the main loudspeaker and the microphone by reproducing the anti‐phase sound of the main loudspeaker. Since the auxiliary loudspeaker is set very close to the microphone, its sound level is low enough for users not to hear. Experimental results showed a reduction in acoustic coupling of about 6 dB from 200 Hz to 1 kHz. This unit also incorporates a newly developed acoustic echo canceller (AEC) DSP. The AEC algorithm, which is based on a duo‐filter control system [Y. Haneda, S. Makino, J. Kojima, and S. Shimauchi, ‘‘Implementation and Evaluation of an Acoustic Echo Canceller Using Duo‐Filte...

Hands‐free teleconferencing unit using auxiliary loudspeaker and duo‐filter echo canceller DSP

/pdf/study-of-acoustical-impulse-response-measurement-with-small-40p3mzqi63.pdf

Yutaka Kaneda

Papers

Adaptive transfer function estimation method and estimation device using the same

Improving the robustness of multiple signal classification (MUSIC) method to reflected sounds by sub-band peak-hold processing

Multi-point voice communication equipment

Hands‐free teleconferencing unit using auxiliary loudspeaker and duo‐filter echo canceller DSP

Study of acoustical impulse response measurement with small error caused by harmonic distortion